add vad code.

VocieProcess/CMakeLists.txt

@@ -1,5 +1,3 @@
-cmake_minimum_required(VERSION 3.29)
-
 project(VocieProcess)
 
 set(CMAKE_CXX_STANDARD 17)
@@ -15,6 +13,10 @@ FetchContent_MakeAvailable(absl)
 
 
 add_library(VocieProcess
+    api/rtp_headers.h api/rtp_headers.cc
+    api/rtp_packet_info.h api/rtp_packet_info.cc
+
+    api/audio/audio_frame.h api/audio/audio_frame.cc
     api/audio/audio_processing_statistics.h api/audio/audio_processing_statistics.cc
     api/audio/audio_processing.h api/audio/audio_processing.cc
     api/audio/channel_layout.h api/audio/channel_layout.cc
@@ -26,6 +28,12 @@ add_library(VocieProcess
     api/units/time_delta.h api/units/time_delta.cc
     api/units/timestamp.h api/units/timestamp.cc
 
+    api/video/color_space.h api/video/color_space.cc
+    api/video/hdr_metadata.h api/video/hdr_metadata.cc
+    api/video/video_content_type.h api/video/video_content_type.cc
+    api/video/video_timing.h api/video/video_timing.cc
+
+    common_audio/audio_converter.h common_audio/audio_converter.cc
     common_audio/audio_util.cc
     common_audio/channel_buffer.h common_audio/channel_buffer.cc
     common_audio/fir_filter_neon.h common_audio/fir_filter_neon.cc
@@ -59,8 +67,11 @@ add_library(VocieProcess
     common_audio/third_party/spl_sqrt_floor/spl_sqrt_floor.h common_audio/third_party/spl_sqrt_floor/spl_sqrt_floor.c
 
     rtc_base/checks.h rtc_base/checks.cc
+    rtc_base/event_tracer.h rtc_base/event_tracer.cc
+    rtc_base/event.h rtc_base/event.cc
     rtc_base/logging.h rtc_base/logging.cc
     rtc_base/platform_thread_types.h rtc_base/platform_thread_types.cc
+    rtc_base/platform_thread.h rtc_base/platform_thread.cc
     rtc_base/race_checker.h rtc_base/race_checker.cc
     rtc_base/string_encode.h rtc_base/string_encode.cc
     rtc_base/string_to_number.h rtc_base/string_to_number.cc
@@ -77,13 +88,27 @@ add_library(VocieProcess
 
     rtc_base/strings/string_builder.h rtc_base/strings/string_builder.cc
 
+    rtc_base/synchronization/sequence_checker_internal.h rtc_base/synchronization/sequence_checker_internal.cc
+    rtc_base/synchronization/yield_policy.h rtc_base/synchronization/yield_policy.cc
+
     rtc_base/system/file_wrapper.h rtc_base/system/file_wrapper.cc
+    rtc_base/system/warn_current_thread_is_deadlocked.h rtc_base/system/warn_current_thread_is_deadlocked.cc
+
+    modules/audio_coding/codecs/isac/main/source/filter_functions.h modules/audio_coding/codecs/isac/main/source/filter_functions.c
+    modules/audio_coding/codecs/isac/main/source/isac_vad.h modules/audio_coding/codecs/isac/main/source/isac_vad.c
+    modules/audio_coding/codecs/isac/main/source/pitch_estimator.h modules/audio_coding/codecs/isac/main/source/pitch_estimator.c
+    modules/audio_coding/codecs/isac/main/source/pitch_filter.h modules/audio_coding/codecs/isac/main/source/pitch_filter.c
 
     modules/audio_processing/audio_buffer.h modules/audio_processing/audio_buffer.cc
+    modules/audio_processing/echo_control_mobile_impl.h modules/audio_processing/echo_control_mobile_impl.cc
     modules/audio_processing/high_pass_filter.h modules/audio_processing/high_pass_filter.cc
+    modules/audio_processing/rms_level.h modules/audio_processing/rms_level.cc
     modules/audio_processing/splitting_filter.h modules/audio_processing/splitting_filter.cc
     modules/audio_processing/three_band_filter_bank.h modules/audio_processing/three_band_filter_bank.cc
 
+    modules/audio_processing/include/aec_dump.h modules/audio_processing/include/aec_dump.cc
+    modules/audio_processing/include/audio_frame_proxies.h modules/audio_processing/include/audio_frame_proxies.cc
+
     modules/audio_processing/aec3/adaptive_fir_filter_erl.h modules/audio_processing/aec3/adaptive_fir_filter_erl.cc
     modules/audio_processing/aec3/adaptive_fir_filter.h modules/audio_processing/aec3/adaptive_fir_filter.cc
     modules/audio_processing/aec3/aec_state.h modules/audio_processing/aec3/aec_state.cc
@@ -146,6 +171,9 @@ add_library(VocieProcess
     modules/audio_processing/aecm/aecm_core_neon.cc
     modules/audio_processing/aecm/echo_control_mobile.h  modules/audio_processing/aecm/echo_control_mobile.cc
 
+    modules/audio_processing/capture_levels_adjuster/audio_samples_scaler.h modules/audio_processing/capture_levels_adjuster/audio_samples_scaler.cc
+    modules/audio_processing/capture_levels_adjuster/capture_levels_adjuster.h modules/audio_processing/capture_levels_adjuster/capture_levels_adjuster.cc
+
     modules/audio_processing/logging/apm_data_dumper.h modules/audio_processing/logging/apm_data_dumper.cc
 
     modules/audio_processing/ns/fast_math.h modules/audio_processing/ns/fast_math.cc
@@ -166,6 +194,17 @@ add_library(VocieProcess
     modules/audio_processing/utility/delay_estimator_wrapper.h  modules/audio_processing/utility/delay_estimator_wrapper.cc
     modules/audio_processing/utility/delay_estimator.h modules/audio_processing/utility/delay_estimator.cc
 
+    modules/audio_processing/vad/gmm.h modules/audio_processing/vad/gmm.cc
+    modules/audio_processing/vad/pitch_based_vad.h modules/audio_processing/vad/pitch_based_vad.cc
+    modules/audio_processing/vad/pitch_internal.h modules/audio_processing/vad/pitch_internal.cc
+    modules/audio_processing/vad/pole_zero_filter.h modules/audio_processing/vad/pole_zero_filter.cc
+    modules/audio_processing/vad/standalone_vad.h modules/audio_processing/vad/standalone_vad.cc
+    modules/audio_processing/vad/vad_audio_proc.h modules/audio_processing/vad/vad_audio_proc.cc
+    modules/audio_processing/vad/vad_circular_buffer.h modules/audio_processing/vad/vad_circular_buffer.cc
+    modules/audio_processing/vad/voice_activity_detector.h modules/audio_processing/vad/voice_activity_detector.cc
+
+    modules/third_party/fft/fft.h modules/third_party/fft/fft.c
+
     system_wrappers/source/field_trial.cc
     system_wrappers/source/metrics.cc
 )

VocieProcess/api/audio/audio_frame.cc

@@ -0,0 +1,235 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/audio/audio_frame.h"
+
+#include <string.h>
+
+#include <cstdint>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/audio/audio_view.h"
+#include "api/audio/channel_layout.h"
+#include "api/rtp_packet_infos.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/time_utils.h"
+
+namespace webrtc {
+
+AudioFrame::AudioFrame() {
+  // Visual Studio doesn't like this in the class definition.
+  static_assert(sizeof(data_) == kMaxDataSizeBytes, "kMaxDataSizeBytes");
+}
+
+AudioFrame::AudioFrame(int sample_rate_hz,
+                       size_t num_channels,
+                       ChannelLayout layout /*= CHANNEL_LAYOUT_UNSUPPORTED*/)
+    : samples_per_channel_(SampleRateToDefaultChannelSize(sample_rate_hz)),
+      sample_rate_hz_(sample_rate_hz),
+      num_channels_(num_channels),
+      channel_layout_(layout == CHANNEL_LAYOUT_UNSUPPORTED
+                          ? GuessChannelLayout(num_channels)
+                          : layout) {
+  RTC_DCHECK_LE(num_channels_, kMaxConcurrentChannels);
+  RTC_DCHECK_GT(sample_rate_hz_, 0);
+  RTC_DCHECK_GT(samples_per_channel_, 0u);
+}
+
+void AudioFrame::Reset() {
+  ResetWithoutMuting();
+  muted_ = true;
+}
+
+void AudioFrame::ResetWithoutMuting() {
+  // TODO(wu): Zero is a valid value for `timestamp_`. We should initialize
+  // to an invalid value, or add a new member to indicate invalidity.
+  timestamp_ = 0;
+  elapsed_time_ms_ = -1;
+  ntp_time_ms_ = -1;
+  samples_per_channel_ = 0;
+  sample_rate_hz_ = 0;
+  num_channels_ = 0;
+  channel_layout_ = CHANNEL_LAYOUT_NONE;
+  speech_type_ = kUndefined;
+  vad_activity_ = kVadUnknown;
+  profile_timestamp_ms_ = 0;
+  packet_infos_ = RtpPacketInfos();
+  absolute_capture_timestamp_ms_ = absl::nullopt;
+}
+
+void AudioFrame::UpdateFrame(uint32_t timestamp,
+                             const int16_t* data,
+                             size_t samples_per_channel,
+                             int sample_rate_hz,
+                             SpeechType speech_type,
+                             VADActivity vad_activity,
+                             size_t num_channels) {
+  RTC_CHECK_LE(num_channels, kMaxConcurrentChannels);
+  timestamp_ = timestamp;
+  samples_per_channel_ = samples_per_channel;
+  sample_rate_hz_ = sample_rate_hz;
+  speech_type_ = speech_type;
+  vad_activity_ = vad_activity;
+  num_channels_ = num_channels;
+  channel_layout_ = GuessChannelLayout(num_channels);
+  if (channel_layout_ != CHANNEL_LAYOUT_UNSUPPORTED) {
+    RTC_DCHECK_EQ(num_channels, ChannelLayoutToChannelCount(channel_layout_));
+  }
+
+  const size_t length = samples_per_channel * num_channels;
+  RTC_CHECK_LE(length, data_.size());
+  if (data != nullptr) {
+    memcpy(data_.data(), data, sizeof(int16_t) * length);
+    muted_ = false;
+  } else {
+    muted_ = true;
+  }
+}
+
+void AudioFrame::CopyFrom(const AudioFrame& src) {
+  if (this == &src)
+    return;
+
+  if (muted_ && !src.muted()) {
+    // TODO: bugs.webrtc.org/5647 - Since the default value for `muted_` is
+    // false and `data_` may still be uninitialized (because we don't initialize
+    // data_ as part of construction), we clear the full buffer here before
+    // copying over new values. If we don't, msan might complain in some tests.
+    // Consider locking down construction, avoiding the default constructor and
+    // prefering construction that initializes all state.
+    ClearSamples(data_);
+  }
+
+  timestamp_ = src.timestamp_;
+  elapsed_time_ms_ = src.elapsed_time_ms_;
+  ntp_time_ms_ = src.ntp_time_ms_;
+  packet_infos_ = src.packet_infos_;
+  muted_ = src.muted();
+  samples_per_channel_ = src.samples_per_channel_;
+  sample_rate_hz_ = src.sample_rate_hz_;
+  speech_type_ = src.speech_type_;
+  vad_activity_ = src.vad_activity_;
+  num_channels_ = src.num_channels_;
+  channel_layout_ = src.channel_layout_;
+  absolute_capture_timestamp_ms_ = src.absolute_capture_timestamp_ms();
+
+  auto data = src.data_view();
+  RTC_CHECK_LE(data.size(), data_.size());
+  if (!muted_ && !data.empty()) {
+    memcpy(&data_[0], &data[0], sizeof(int16_t) * data.size());
+  }
+}
+
+void AudioFrame::UpdateProfileTimeStamp() {
+  profile_timestamp_ms_ = rtc::TimeMillis();
+}
+
+int64_t AudioFrame::ElapsedProfileTimeMs() const {
+  if (profile_timestamp_ms_ == 0) {
+    // Profiling has not been activated.
+    return -1;
+  }
+  return rtc::TimeSince(profile_timestamp_ms_);
+}
+
+const int16_t* AudioFrame::data() const {
+  return muted_ ? zeroed_data().begin() : data_.data();
+}
+
+InterleavedView<const int16_t> AudioFrame::data_view() const {
+  // If you get a nullptr from `data_view()`, it's likely because the
+  // samples_per_channel_ and/or num_channels_ members haven't been properly
+  // set. Since `data_view()` returns an InterleavedView<> (which internally
+  // uses rtc::ArrayView<>), we inherit the behavior in InterleavedView when the
+  // view size is 0 that ArrayView<>::data() returns nullptr. So, even when an
+  // AudioFrame is muted and we want to return `zeroed_data()`, if
+  // samples_per_channel_ or  num_channels_ is 0, the view will point to
+  // nullptr.
+  return InterleavedView<const int16_t>(muted_ ? &zeroed_data()[0] : &data_[0],
+                                        samples_per_channel_, num_channels_);
+}
+
+int16_t* AudioFrame::mutable_data() {
+  // TODO: bugs.webrtc.org/5647 - Can we skip zeroing the buffer?
+  // Consider instead if we should rather zero the buffer when `muted_` is set
+  // to `true`.
+  if (muted_) {
+    ClearSamples(data_);
+    muted_ = false;
+  }
+  return &data_[0];
+}
+
+InterleavedView<int16_t> AudioFrame::mutable_data(size_t samples_per_channel,
+                                                  size_t num_channels) {
+  const size_t total_samples = samples_per_channel * num_channels;
+  RTC_CHECK_LE(total_samples, data_.size());
+  RTC_CHECK_LE(num_channels, kMaxConcurrentChannels);
+  // Sanity check for valid argument values during development.
+  // If `samples_per_channel` is < `num_channels` but larger than 0,
+  // then chances are the order of arguments is incorrect.
+  RTC_DCHECK((samples_per_channel == 0 && num_channels == 0) ||
+             num_channels <= samples_per_channel)
+      << "samples_per_channel=" << samples_per_channel
+      << "num_channels=" << num_channels;
+
+  // TODO: bugs.webrtc.org/5647 - Can we skip zeroing the buffer?
+  // Consider instead if we should rather zero the whole buffer when `muted_` is
+  // set to `true`.
+  if (muted_) {
+    ClearSamples(data_, total_samples);
+    muted_ = false;
+  }
+  samples_per_channel_ = samples_per_channel;
+  num_channels_ = num_channels;
+  return InterleavedView<int16_t>(&data_[0], samples_per_channel, num_channels);
+}
+
+void AudioFrame::Mute() {
+  muted_ = true;
+}
+
+bool AudioFrame::muted() const {
+  return muted_;
+}
+
+void AudioFrame::SetLayoutAndNumChannels(ChannelLayout layout,
+                                         size_t num_channels) {
+  channel_layout_ = layout;
+  num_channels_ = num_channels;
+#if RTC_DCHECK_IS_ON
+  // Do a sanity check that the layout and num_channels match.
+  // If this lookup yield 0u, then the layout is likely CHANNEL_LAYOUT_DISCRETE.
+  auto expected_num_channels = ChannelLayoutToChannelCount(layout);
+  if (expected_num_channels) {  // If expected_num_channels is 0
+    RTC_DCHECK_EQ(expected_num_channels, num_channels_);
+  }
+#endif
+  RTC_CHECK_LE(samples_per_channel_ * num_channels_, data_.size());
+}
+
+void AudioFrame::SetSampleRateAndChannelSize(int sample_rate) {
+  sample_rate_hz_ = sample_rate;
+  // We could call `AudioProcessing::GetFrameSize()` here, but that requires
+  // adding a dependency on the ":audio_processing" build target, which can
+  // complicate the dependency tree. Some refactoring is probably in order to
+  // get some consistency around this since there are many places across the
+  // code that assume this default buffer size.
+  samples_per_channel_ = SampleRateToDefaultChannelSize(sample_rate_hz_);
+}
+
+// static
+rtc::ArrayView<const int16_t> AudioFrame::zeroed_data() {
+  static int16_t* null_data = new int16_t[kMaxDataSizeSamples]();
+  return rtc::ArrayView<const int16_t>(null_data, kMaxDataSizeSamples);
+}
+
+}  // namespace webrtc

VocieProcess/api/audio/audio_frame.h

@@ -0,0 +1,230 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_AUDIO_AUDIO_FRAME_H_
+#define API_AUDIO_AUDIO_FRAME_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <array>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/audio/audio_view.h"
+#include "api/audio/channel_layout.h"
+#include "api/rtp_packet_infos.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+// Default webrtc buffer size in milliseconds.
+constexpr size_t kDefaultAudioBufferLengthMs = 10u;
+
+// Default total number of audio buffers per second based on the default length.
+constexpr size_t kDefaultAudioBuffersPerSec =
+    1000u / kDefaultAudioBufferLengthMs;
+
+// Returns the number of samples a buffer needs to hold for ~10ms of a single
+// audio channel at a given sample rate.
+// See also `AudioProcessing::GetFrameSize()`.
+inline size_t SampleRateToDefaultChannelSize(size_t sample_rate) {
+  // Basic sanity check. 192kHz is the highest supported input sample rate.
+  RTC_DCHECK_LE(sample_rate, 192000);
+  return sample_rate / kDefaultAudioBuffersPerSec;
+}
+/////////////////////////////////////////////////////////////////////
+
+/* This class holds up to 120 ms of super-wideband (32 kHz) stereo audio. It
+ * allows for adding and subtracting frames while keeping track of the resulting
+ * states.
+ *
+ * Notes
+ * - This is a de-facto api, not designed for external use. The AudioFrame class
+ *   is in need of overhaul or even replacement, and anyone depending on it
+ *   should be prepared for that.
+ * - The total number of samples is samples_per_channel_ * num_channels_.
+ * - Stereo data is interleaved starting with the left channel.
+ */
+class AudioFrame {
+ public:
+  // Using constexpr here causes linker errors unless the variable also has an
+  // out-of-class definition, which is impractical in this header-only class.
+  // (This makes no sense because it compiles as an enum value, which we most
+  // certainly cannot take the address of, just fine.) C++17 introduces inline
+  // variables which should allow us to switch to constexpr and keep this a
+  // header-only class.
+  enum : size_t {
+    // Stereo, 32 kHz, 120 ms (2 * 32 * 120)
+    // Stereo, 192 kHz, 20 ms (2 * 192 * 20)
+    kMaxDataSizeSamples = 7680,
+    kMaxDataSizeBytes = kMaxDataSizeSamples * sizeof(int16_t),
+  };
+
+  enum VADActivity { kVadActive = 0, kVadPassive = 1, kVadUnknown = 2 };
+  enum SpeechType {
+    kNormalSpeech = 0,
+    kPLC = 1,
+    kCNG = 2,
+    kPLCCNG = 3,
+    kCodecPLC = 5,
+    kUndefined = 4
+  };
+
+  AudioFrame();
+
+  // Construct an audio frame with frame length properties and channel
+  // information. `samples_per_channel()` will be initialized to a 10ms buffer
+  // size and if `layout` is not specified (default value of
+  // CHANNEL_LAYOUT_UNSUPPORTED is set), then the channel layout is derived
+  // (guessed) from `num_channels`.
+  AudioFrame(int sample_rate_hz,
+             size_t num_channels,
+             ChannelLayout layout = CHANNEL_LAYOUT_UNSUPPORTED);
+
+  AudioFrame(const AudioFrame&) = delete;
+  AudioFrame& operator=(const AudioFrame&) = delete;
+
+  // Resets all members to their default state.
+  void Reset();
+  // Same as Reset(), but leaves mute state unchanged. Muting a frame requires
+  // the buffer to be zeroed on the next call to mutable_data(). Callers
+  // intending to write to the buffer immediately after Reset() can instead use
+  // ResetWithoutMuting() to skip this wasteful zeroing.
+  void ResetWithoutMuting();
+
+  // TODO: b/335805780 - Accept InterleavedView.
+  void UpdateFrame(uint32_t timestamp,
+                   const int16_t* data,
+                   size_t samples_per_channel,
+                   int sample_rate_hz,
+                   SpeechType speech_type,
+                   VADActivity vad_activity,
+                   size_t num_channels = 1);
+
+  void CopyFrom(const AudioFrame& src);
+
+  // Sets a wall-time clock timestamp in milliseconds to be used for profiling
+  // of time between two points in the audio chain.
+  // Example:
+  //   t0: UpdateProfileTimeStamp()
+  //   t1: ElapsedProfileTimeMs() => t1 - t0 [msec]
+  void UpdateProfileTimeStamp();
+  // Returns the time difference between now and when UpdateProfileTimeStamp()
+  // was last called. Returns -1 if UpdateProfileTimeStamp() has not yet been
+  // called.
+  int64_t ElapsedProfileTimeMs() const;
+
+  // data() returns a zeroed static buffer if the frame is muted.
+  // TODO: b/335805780 - Return InterleavedView.
+  const int16_t* data() const;
+
+  // Returns a read-only view of all the valid samples held by the AudioFrame.
+  // For a muted AudioFrame, the samples will all be 0.
+  InterleavedView<const int16_t> data_view() const;
+
+  // mutable_frame() always returns a non-static buffer; the first call to
+  // mutable_frame() zeros the buffer and marks the frame as unmuted.
+  // TODO: b/335805780 - Return an InterleavedView.
+  int16_t* mutable_data();
+
+  // Grants write access to the audio buffer. The size of the returned writable
+  // view is determined by the `samples_per_channel` and `num_channels`
+  // dimensions which the function checks for correctness and stores in the
+  // internal member variables; `samples_per_channel()` and `num_channels()`
+  // respectively.
+  // If the state is currently muted, the returned view will be zeroed out.
+  InterleavedView<int16_t> mutable_data(size_t samples_per_channel,
+                                        size_t num_channels);
+
+  // Prefer to mute frames using AudioFrameOperations::Mute.
+  void Mute();
+  // Frame is muted by default.
+  bool muted() const;
+
+  size_t max_16bit_samples() const { return data_.size(); }
+  size_t samples_per_channel() const { return samples_per_channel_; }
+  size_t num_channels() const { return num_channels_; }
+
+  ChannelLayout channel_layout() const { return channel_layout_; }
+  // Sets the `channel_layout` property as well as `num_channels`.
+  void SetLayoutAndNumChannels(ChannelLayout layout, size_t num_channels);
+
+  int sample_rate_hz() const { return sample_rate_hz_; }
+
+  void set_absolute_capture_timestamp_ms(
+      int64_t absolute_capture_time_stamp_ms) {
+    absolute_capture_timestamp_ms_ = absolute_capture_time_stamp_ms;
+  }
+
+  absl::optional<int64_t> absolute_capture_timestamp_ms() const {
+    return absolute_capture_timestamp_ms_;
+  }
+
+  // Sets the sample_rate_hz and samples_per_channel properties based on a
+  // given sample rate and calculates a default 10ms samples_per_channel value.
+  void SetSampleRateAndChannelSize(int sample_rate);
+
+  // RTP timestamp of the first sample in the AudioFrame.
+  uint32_t timestamp_ = 0;
+  // Time since the first frame in milliseconds.
+  // -1 represents an uninitialized value.
+  int64_t elapsed_time_ms_ = -1;
+  // NTP time of the estimated capture time in local timebase in milliseconds.
+  // -1 represents an uninitialized value.
+  int64_t ntp_time_ms_ = -1;
+  size_t samples_per_channel_ = 0;
+  int sample_rate_hz_ = 0;
+  size_t num_channels_ = 0;
+  SpeechType speech_type_ = kUndefined;
+  VADActivity vad_activity_ = kVadUnknown;
+  // Monotonically increasing timestamp intended for profiling of audio frames.
+  // Typically used for measuring elapsed time between two different points in
+  // the audio path. No lock is used to save resources and we are thread safe
+  // by design.
+  // TODO(nisse@webrtc.org): consider using absl::optional.
+  int64_t profile_timestamp_ms_ = 0;
+
+  // Information about packets used to assemble this audio frame. This is needed
+  // by `SourceTracker` when the frame is delivered to the RTCRtpReceiver's
+  // MediaStreamTrack, in order to implement getContributingSources(). See:
+  // https://w3c.github.io/webrtc-pc/#dom-rtcrtpreceiver-getcontributingsources
+  //
+  // TODO(bugs.webrtc.org/10757):
+  //   Note that this information might not be fully accurate since we currently
+  //   don't have a proper way to track it across the audio sync buffer. The
+  //   sync buffer is the small sample-holding buffer located after the audio
+  //   decoder and before where samples are assembled into output frames.
+  //
+  // `RtpPacketInfos` may also be empty if the audio samples did not come from
+  // RTP packets. E.g. if the audio were locally generated by packet loss
+  // concealment, comfort noise generation, etc.
+  RtpPacketInfos packet_infos_;
+
+ private:
+  // A permanently zeroed out buffer to represent muted frames. This is a
+  // header-only class, so the only way to avoid creating a separate zeroed
+  // buffer per translation unit is to wrap a static in an inline function.
+  static rtc::ArrayView<const int16_t> zeroed_data();
+
+  std::array<int16_t, kMaxDataSizeSamples> data_;
+  bool muted_ = true;
+  ChannelLayout channel_layout_ = CHANNEL_LAYOUT_NONE;
+
+  // Absolute capture timestamp when this audio frame was originally captured.
+  // This is only valid for audio frames captured on this machine. The absolute
+  // capture timestamp of a received frame is found in `packet_infos_`.
+  // This timestamp MUST be based on the same clock as rtc::TimeMillis().
+  absl::optional<int64_t> absolute_capture_timestamp_ms_;
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_AUDIO_FRAME_H_

VocieProcess/api/function_view.h

@@ -0,0 +1,131 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_FUNCTION_VIEW_H_
+#define API_FUNCTION_VIEW_H_
+
+#include <cstddef>
+#include <type_traits>
+#include <utility>
+
+#include "rtc_base/checks.h"
+
+// Just like std::function, FunctionView will wrap any callable and hide its
+// actual type, exposing only its signature. But unlike std::function,
+// FunctionView doesn't own its callable---it just points to it. Thus, it's a
+// good choice mainly as a function argument when the callable argument will
+// not be called again once the function has returned.
+//
+// Its constructors are implicit, so that callers won't have to convert lambdas
+// and other callables to FunctionView<Blah(Blah, Blah)> explicitly. This is
+// safe because FunctionView is only a reference to the real callable.
+//
+// Example use:
+//
+//   void SomeFunction(rtc::FunctionView<int(int)> index_transform);
+//   ...
+//   SomeFunction([](int i) { return 2 * i + 1; });
+//
+// Note: FunctionView is tiny (essentially just two pointers) and trivially
+// copyable, so it's probably cheaper to pass it by value than by const
+// reference.
+
+namespace rtc {
+
+template <typename T>
+class FunctionView;  // Undefined.
+
+template <typename RetT, typename... ArgT>
+class FunctionView<RetT(ArgT...)> final {
+ public:
+  // Constructor for lambdas and other callables; it accepts every type of
+  // argument except those noted in its enable_if call.
+  template <
+      typename F,
+      typename std::enable_if<
+          // Not for function pointers; we have another constructor for that
+          // below.
+          !std::is_function<typename std::remove_pointer<
+              typename std::remove_reference<F>::type>::type>::value &&
+
+          // Not for nullptr; we have another constructor for that below.
+          !std::is_same<std::nullptr_t,
+                        typename std::remove_cv<F>::type>::value &&
+
+          // Not for FunctionView objects; we have another constructor for that
+          // (the implicitly declared copy constructor).
+          !std::is_same<FunctionView,
+                        typename std::remove_cv<typename std::remove_reference<
+                            F>::type>::type>::value>::type* = nullptr>
+  FunctionView(F&& f)
+      : call_(CallVoidPtr<typename std::remove_reference<F>::type>) {
+    f_.void_ptr = &f;
+  }
+
+  // Constructor that accepts function pointers. If the argument is null, the
+  // result is an empty FunctionView.
+  template <
+      typename F,
+      typename std::enable_if<std::is_function<typename std::remove_pointer<
+          typename std::remove_reference<F>::type>::type>::value>::type* =
+          nullptr>
+  FunctionView(F&& f)
+      : call_(f ? CallFunPtr<typename std::remove_pointer<F>::type> : nullptr) {
+    f_.fun_ptr = reinterpret_cast<void (*)()>(f);
+  }
+
+  // Constructor that accepts nullptr. It creates an empty FunctionView.
+  template <typename F,
+            typename std::enable_if<std::is_same<
+                std::nullptr_t,
+                typename std::remove_cv<F>::type>::value>::type* = nullptr>
+  FunctionView(F&& f) : call_(nullptr) {}
+
+  // Default constructor. Creates an empty FunctionView.
+  FunctionView() : call_(nullptr) {}
+
+  RetT operator()(ArgT... args) const {
+    RTC_DCHECK(call_);
+    return call_(f_, std::forward<ArgT>(args)...);
+  }
+
+  // Returns true if we have a function, false if we don't (i.e., we're null).
+  explicit operator bool() const { return !!call_; }
+
+ private:
+  union VoidUnion {
+    void* void_ptr;
+    void (*fun_ptr)();
+  };
+
+  template <typename F>
+  static RetT CallVoidPtr(VoidUnion vu, ArgT... args) {
+    return (*static_cast<F*>(vu.void_ptr))(std::forward<ArgT>(args)...);
+  }
+  template <typename F>
+  static RetT CallFunPtr(VoidUnion vu, ArgT... args) {
+    return (reinterpret_cast<typename std::add_pointer<F>::type>(vu.fun_ptr))(
+        std::forward<ArgT>(args)...);
+  }
+
+  // A pointer to the callable thing, with type information erased. It's a
+  // union because we have to use separate types depending on if the callable
+  // thing is a function pointer or something else.
+  VoidUnion f_;
+
+  // Pointer to a dispatch function that knows the type of the callable thing
+  // that's stored in f_, and how to call it. A FunctionView object is empty
+  // (null) iff call_ is null.
+  RetT (*call_)(VoidUnion, ArgT...);
+};
+
+}  // namespace rtc
+
+#endif  // API_FUNCTION_VIEW_H_

VocieProcess/api/make_ref_counted.h

@@ -0,0 +1,170 @@
+/*
+ *  Copyright 2022 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef API_MAKE_REF_COUNTED_H_
+#define API_MAKE_REF_COUNTED_H_
+
+#include <type_traits>
+#include <utility>
+
+#include "rtc_base/ref_counted_object.h"
+
+namespace webrtc {
+
+namespace webrtc_make_ref_counted_internal {
+// Determines if the given class has AddRef and Release methods.
+template <typename T>
+class HasAddRefAndRelease {
+ private:
+  template <typename C,
+            decltype(std::declval<C>().AddRef())* = nullptr,
+            decltype(std::declval<C>().Release())* = nullptr>
+  static int Test(int);
+  template <typename>
+  static char Test(...);
+
+ public:
+  static constexpr bool value = std::is_same_v<decltype(Test<T>(0)), int>;
+};
+}  // namespace webrtc_make_ref_counted_internal
+
+// General utilities for constructing a reference counted class and the
+// appropriate reference count implementation for that class.
+//
+// These utilities select either the `RefCountedObject` implementation or
+// `FinalRefCountedObject` depending on whether the to-be-shared class is
+// derived from the RefCountInterface interface or not (respectively).
+
+// `make_ref_counted`:
+//
+// Use this when you want to construct a reference counted object of type T and
+// get a `scoped_refptr<>` back. Example:
+//
+//   auto p = make_ref_counted<Foo>("bar", 123);
+//
+// For a class that inherits from RefCountInterface, this is equivalent to:
+//
+//   auto p = scoped_refptr<Foo>(new RefCountedObject<Foo>("bar", 123));
+//
+// If the class does not inherit from RefCountInterface, but does have
+// AddRef/Release methods (so a T* is convertible to rtc::scoped_refptr), this
+// is equivalent to just
+//
+//   auto p = scoped_refptr<Foo>(new Foo("bar", 123));
+//
+// Otherwise, the example is equivalent to:
+//
+//   auto p = scoped_refptr<FinalRefCountedObject<Foo>>(
+//       new FinalRefCountedObject<Foo>("bar", 123));
+//
+// In these cases, `make_ref_counted` reduces the amount of boilerplate code but
+// also helps with the most commonly intended usage of RefCountedObject whereby
+// methods for reference counting, are virtual and designed to satisfy the need
+// of an interface. When such a need does not exist, it is more efficient to use
+// the `FinalRefCountedObject` template, which does not add the vtable overhead.
+//
+// Note that in some cases, using RefCountedObject directly may still be what's
+// needed.
+
+// `make_ref_counted` for abstract classes that are convertible to
+// RefCountInterface. The is_abstract requirement rejects classes that inherit
+// both RefCountInterface and RefCounted object, which is a a discouraged
+// pattern, and would result in double inheritance of RefCountedObject if this
+// template was applied.
+template <
+    typename T,
+    typename... Args,
+    typename std::enable_if<std::is_convertible_v<T*, RefCountInterface*> &&
+                                std::is_abstract_v<T>,
+                            T>::type* = nullptr>
+scoped_refptr<T> make_ref_counted(Args&&... args) {
+  return scoped_refptr<T>(new RefCountedObject<T>(std::forward<Args>(args)...));
+}
+
+// `make_ref_counted` for complete classes that are not convertible to
+// RefCountInterface and already carry a ref count.
+template <
+    typename T,
+    typename... Args,
+    typename std::enable_if<
+        !std::is_convertible_v<T*, RefCountInterface*> &&
+            webrtc_make_ref_counted_internal::HasAddRefAndRelease<T>::value,
+        T>::type* = nullptr>
+scoped_refptr<T> make_ref_counted(Args&&... args) {
+  return scoped_refptr<T>(new T(std::forward<Args>(args)...));
+}
+
+// `make_ref_counted` for complete classes that are not convertible to
+// RefCountInterface and have no ref count of their own.
+template <
+    typename T,
+    typename... Args,
+    typename std::enable_if<
+        !std::is_convertible_v<T*, RefCountInterface*> &&
+            !webrtc_make_ref_counted_internal::HasAddRefAndRelease<T>::value,
+
+        T>::type* = nullptr>
+scoped_refptr<FinalRefCountedObject<T>> make_ref_counted(Args&&... args) {
+  return scoped_refptr<FinalRefCountedObject<T>>(
+      new FinalRefCountedObject<T>(std::forward<Args>(args)...));
+}
+
+}  // namespace webrtc
+
+// Backwards compatibe aliases.
+// TODO: https://issues.webrtc.org/42225969 - deprecate and remove.
+namespace rtc {
+// This doesn't work:
+// template <typename T, typename... Args>
+//    using make_ref_counted(Args&&... args) =
+//    webrtc::make_ref_counted<T>(Args&&... args);
+// Instead, reproduce the templates.
+template <typename T,
+          typename... Args,
+          typename std::enable_if<
+              std::is_convertible_v<T*, webrtc::RefCountInterface*> &&
+                  std::is_abstract_v<T>,
+              T>::type* = nullptr>
+scoped_refptr<T> make_ref_counted(Args&&... args) {
+  return webrtc::scoped_refptr<T>(
+      new webrtc::RefCountedObject<T>(std::forward<Args>(args)...));
+}
+
+// `make_ref_counted` for complete classes that are not convertible to
+// RefCountInterface and already carry a ref count.
+template <typename T,
+          typename... Args,
+          typename std::enable_if<
+              !std::is_convertible_v<T*, webrtc::RefCountInterface*> &&
+                  webrtc::webrtc_make_ref_counted_internal::HasAddRefAndRelease<
+                      T>::value,
+              T>::type* = nullptr>
+scoped_refptr<T> make_ref_counted(Args&&... args) {
+  return webrtc::scoped_refptr<T>(new T(std::forward<Args>(args)...));
+}
+
+// `make_ref_counted` for complete classes that are not convertible to
+// RefCountInterface and have no ref count of their own.
+template <typename T,
+          typename... Args,
+          typename std::enable_if<
+              !std::is_convertible_v<T*, webrtc::RefCountInterface*> &&
+                  !webrtc::webrtc_make_ref_counted_internal::
+                      HasAddRefAndRelease<T>::value,
+
+              T>::type* = nullptr>
+scoped_refptr<webrtc::FinalRefCountedObject<T>> make_ref_counted(
+    Args&&... args) {
+  return webrtc::scoped_refptr<FinalRefCountedObject<T>>(
+      new webrtc::FinalRefCountedObject<T>(std::forward<Args>(args)...));
+}
+
+}  // namespace rtc
+
+#endif  // API_MAKE_REF_COUNTED_H_

VocieProcess/api/ref_counted_base.h

@@ -0,0 +1,106 @@
+/*
+ *  Copyright 2017 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef API_REF_COUNTED_BASE_H_
+#define API_REF_COUNTED_BASE_H_
+
+#include <type_traits>
+
+#include "rtc_base/ref_counter.h"
+
+namespace webrtc {
+
+class RefCountedBase {
+ public:
+  RefCountedBase() = default;
+
+  RefCountedBase(const RefCountedBase&) = delete;
+  RefCountedBase& operator=(const RefCountedBase&) = delete;
+
+  void AddRef() const { ref_count_.IncRef(); }
+  RefCountReleaseStatus Release() const {
+    const auto status = ref_count_.DecRef();
+    if (status == RefCountReleaseStatus::kDroppedLastRef) {
+      delete this;
+    }
+    return status;
+  }
+
+ protected:
+  // Provided for internal webrtc subclasses for corner cases where it's
+  // necessary to know whether or not a reference is exclusively held.
+  bool HasOneRef() const { return ref_count_.HasOneRef(); }
+
+  virtual ~RefCountedBase() = default;
+
+ private:
+  mutable webrtc::webrtc_impl::RefCounter ref_count_{0};
+};
+
+// Template based version of `RefCountedBase` for simple implementations that do
+// not need (or want) destruction via virtual destructor or the overhead of a
+// vtable.
+//
+// To use:
+//   struct MyInt : public rtc::RefCountedNonVirtual<MyInt>  {
+//     int foo_ = 0;
+//   };
+//
+//   rtc::scoped_refptr<MyInt> my_int(new MyInt());
+//
+// sizeof(MyInt) on a 32 bit system would then be 8, int + refcount and no
+// vtable generated.
+template <typename T>
+class RefCountedNonVirtual {
+ public:
+  RefCountedNonVirtual() = default;
+
+  RefCountedNonVirtual(const RefCountedNonVirtual&) = delete;
+  RefCountedNonVirtual& operator=(const RefCountedNonVirtual&) = delete;
+
+  void AddRef() const { ref_count_.IncRef(); }
+  RefCountReleaseStatus Release() const {
+    // If you run into this assert, T has virtual methods. There are two
+    // options:
+    // 1) The class doesn't actually need virtual methods, the type is complete
+    //    so the virtual attribute(s) can be removed.
+    // 2) The virtual methods are a part of the design of the class. In this
+    //    case you can consider using `RefCountedBase` instead or alternatively
+    //    use `rtc::RefCountedObject`.
+    static_assert(!std::is_polymorphic<T>::value,
+                  "T has virtual methods. RefCountedBase is a better fit.");
+    const auto status = ref_count_.DecRef();
+    if (status == RefCountReleaseStatus::kDroppedLastRef) {
+      delete static_cast<const T*>(this);
+    }
+    return status;
+  }
+
+ protected:
+  // Provided for internal webrtc subclasses for corner cases where it's
+  // necessary to know whether or not a reference is exclusively held.
+  bool HasOneRef() const { return ref_count_.HasOneRef(); }
+
+  ~RefCountedNonVirtual() = default;
+
+ private:
+  mutable webrtc::webrtc_impl::RefCounter ref_count_{0};
+};
+
+}  // namespace webrtc
+
+// Backwards compatibe aliases.
+// TODO: https://issues.webrtc.org/42225969 - deprecate and remove.
+namespace rtc {
+using RefCountedBase = webrtc::RefCountedBase;
+template <typename T>
+using RefCountedNonVirtual = webrtc::RefCountedNonVirtual<T>;
+}  // namespace rtc
+
+#endif  // API_REF_COUNTED_BASE_H_

VocieProcess/api/rtp_headers.cc

@@ -0,0 +1,58 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/rtp_headers.h"
+
+namespace webrtc {
+
+AudioLevel::AudioLevel() : voice_activity_(false), audio_level_(0) {}
+
+AudioLevel::AudioLevel(bool voice_activity, int audio_level)
+    : voice_activity_(voice_activity), audio_level_(audio_level) {
+  RTC_CHECK_GE(audio_level, 0);
+  RTC_CHECK_LE(audio_level, 127);
+}
+
+RTPHeaderExtension::RTPHeaderExtension()
+    : hasTransmissionTimeOffset(false),
+      transmissionTimeOffset(0),
+      hasAbsoluteSendTime(false),
+      absoluteSendTime(0),
+      hasTransportSequenceNumber(false),
+      transportSequenceNumber(0),
+      hasVideoRotation(false),
+      videoRotation(kVideoRotation_0),
+      hasVideoContentType(false),
+      videoContentType(VideoContentType::UNSPECIFIED),
+      has_video_timing(false) {}
+
+RTPHeaderExtension::RTPHeaderExtension(const RTPHeaderExtension& other) =
+    default;
+
+RTPHeaderExtension& RTPHeaderExtension::operator=(
+    const RTPHeaderExtension& other) = default;
+
+RTPHeader::RTPHeader()
+    : markerBit(false),
+      payloadType(0),
+      sequenceNumber(0),
+      timestamp(0),
+      ssrc(0),
+      numCSRCs(0),
+      arrOfCSRCs(),
+      paddingLength(0),
+      headerLength(0),
+      extension() {}
+
+RTPHeader::RTPHeader(const RTPHeader& other) = default;
+
+RTPHeader& RTPHeader::operator=(const RTPHeader& other) = default;
+
+}  // namespace webrtc

VocieProcess/api/rtp_headers.h

@@ -0,0 +1,208 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_RTP_HEADERS_H_
+#define API_RTP_HEADERS_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <string>
+
+#include "absl/types/optional.h"
+#include "api/units/timestamp.h"
+#include "api/video/color_space.h"
+#include "api/video/video_content_type.h"
+#include "api/video/video_rotation.h"
+#include "api/video/video_timing.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+struct FeedbackRequest {
+  // Determines whether the recv delta as specified in
+  // https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions-01
+  // should be included.
+  bool include_timestamps;
+  // Include feedback of received packets in the range [sequence_number -
+  // sequence_count + 1, sequence_number]. That is, no feedback will be sent if
+  // sequence_count is zero.
+  int sequence_count;
+};
+
+// The Absolute Capture Time extension is used to stamp RTP packets with a NTP
+// timestamp showing when the first audio or video frame in a packet was
+// originally captured. The intent of this extension is to provide a way to
+// accomplish audio-to-video synchronization when RTCP-terminating intermediate
+// systems (e.g. mixers) are involved. See:
+// http://www.webrtc.org/experiments/rtp-hdrext/abs-capture-time
+struct AbsoluteCaptureTime {
+  // Absolute capture timestamp is the NTP timestamp of when the first frame in
+  // a packet was originally captured. This timestamp MUST be based on the same
+  // clock as the clock used to generate NTP timestamps for RTCP sender reports
+  // on the capture system.
+  //
+  // It’s not always possible to do an NTP clock readout at the exact moment of
+  // when a media frame is captured. A capture system MAY postpone the readout
+  // until a more convenient time. A capture system SHOULD have known delays
+  // (e.g. from hardware buffers) subtracted from the readout to make the final
+  // timestamp as close to the actual capture time as possible.
+  //
+  // This field is encoded as a 64-bit unsigned fixed-point number with the high
+  // 32 bits for the timestamp in seconds and low 32 bits for the fractional
+  // part. This is also known as the UQ32.32 format and is what the RTP
+  // specification defines as the canonical format to represent NTP timestamps.
+  uint64_t absolute_capture_timestamp;
+
+  // Estimated capture clock offset is the sender’s estimate of the offset
+  // between its own NTP clock and the capture system’s NTP clock. The sender is
+  // here defined as the system that owns the NTP clock used to generate the NTP
+  // timestamps for the RTCP sender reports on this stream. The sender system is
+  // typically either the capture system or a mixer.
+  //
+  // This field is encoded as a 64-bit two’s complement signed fixed-point
+  // number with the high 32 bits for the seconds and low 32 bits for the
+  // fractional part. It’s intended to make it easy for a receiver, that knows
+  // how to estimate the sender system’s NTP clock, to also estimate the capture
+  // system’s NTP clock:
+  //
+  //   Capture NTP Clock = Sender NTP Clock + Capture Clock Offset
+  absl::optional<int64_t> estimated_capture_clock_offset;
+};
+
+// The audio level extension is used to indicate the voice activity and the
+// audio level of the payload in the RTP stream. See:
+// https://tools.ietf.org/html/rfc6464#section-3.
+class AudioLevel {
+ public:
+  AudioLevel();
+  AudioLevel(bool voice_activity, int audio_level);
+  AudioLevel(const AudioLevel& other) = default;
+  AudioLevel& operator=(const AudioLevel& other) = default;
+
+  // Flag indicating whether the encoder believes the audio packet contains
+  // voice activity.
+  bool voice_activity() const { return voice_activity_; }
+
+  // Audio level in -dBov. Values range from 0 to 127, representing 0 to -127
+  // dBov. 127 represents digital silence.
+  int level() const { return audio_level_; }
+
+ private:
+  bool voice_activity_;
+  int audio_level_;
+};
+
+inline bool operator==(const AbsoluteCaptureTime& lhs,
+                       const AbsoluteCaptureTime& rhs) {
+  return (lhs.absolute_capture_timestamp == rhs.absolute_capture_timestamp) &&
+         (lhs.estimated_capture_clock_offset ==
+          rhs.estimated_capture_clock_offset);
+}
+
+inline bool operator!=(const AbsoluteCaptureTime& lhs,
+                       const AbsoluteCaptureTime& rhs) {
+  return !(lhs == rhs);
+}
+
+struct RTPHeaderExtension {
+  RTPHeaderExtension();
+  RTPHeaderExtension(const RTPHeaderExtension& other);
+  RTPHeaderExtension& operator=(const RTPHeaderExtension& other);
+
+  static constexpr int kAbsSendTimeFraction = 18;
+
+  Timestamp GetAbsoluteSendTimestamp() const {
+    RTC_DCHECK(hasAbsoluteSendTime);
+    RTC_DCHECK(absoluteSendTime < (1ul << 24));
+    return Timestamp::Micros((absoluteSendTime * 1000000ll) /
+                             (1 << kAbsSendTimeFraction));
+  }
+
+  bool hasTransmissionTimeOffset;
+  int32_t transmissionTimeOffset;
+  bool hasAbsoluteSendTime;
+  uint32_t absoluteSendTime;
+  absl::optional<AbsoluteCaptureTime> absolute_capture_time;
+  bool hasTransportSequenceNumber;
+  uint16_t transportSequenceNumber;
+  absl::optional<FeedbackRequest> feedback_request;
+
+  // Audio Level includes both level in dBov and voiced/unvoiced bit. See:
+  // https://tools.ietf.org/html/rfc6464#section-3
+  absl::optional<AudioLevel> audio_level() const { return audio_level_; }
+
+  void set_audio_level(absl::optional<AudioLevel> audio_level) {
+    audio_level_ = audio_level;
+  }
+
+  // For Coordination of Video Orientation. See
+  // http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/
+  // ts_126114v120700p.pdf
+  bool hasVideoRotation;
+  VideoRotation videoRotation;
+
+  // TODO(ilnik): Refactor this and one above to be absl::optional() and remove
+  // a corresponding bool flag.
+  bool hasVideoContentType;
+  VideoContentType videoContentType;
+
+  bool has_video_timing;
+  VideoSendTiming video_timing;
+
+  VideoPlayoutDelay playout_delay;
+
+  // For identification of a stream when ssrc is not signaled. See
+  // https://tools.ietf.org/html/rfc8852
+  std::string stream_id;
+  std::string repaired_stream_id;
+
+  // For identifying the media section used to interpret this RTP packet. See
+  // https://tools.ietf.org/html/rfc8843
+  std::string mid;
+
+  absl::optional<ColorSpace> color_space;
+
+ private:
+  absl::optional<AudioLevel> audio_level_;
+};
+
+enum { kRtpCsrcSize = 15 };  // RFC 3550 page 13
+
+struct RTC_EXPORT RTPHeader {
+  RTPHeader();
+  RTPHeader(const RTPHeader& other);
+  RTPHeader& operator=(const RTPHeader& other);
+
+  bool markerBit;
+  uint8_t payloadType;
+  uint16_t sequenceNumber;
+  uint32_t timestamp;
+  uint32_t ssrc;
+  uint8_t numCSRCs;
+  uint32_t arrOfCSRCs[kRtpCsrcSize];
+  size_t paddingLength;
+  size_t headerLength;
+  RTPHeaderExtension extension;
+};
+
+// RTCP mode to use. Compound mode is described by RFC 4585 and reduced-size
+// RTCP mode is described by RFC 5506.
+enum class RtcpMode { kOff, kCompound, kReducedSize };
+
+enum NetworkState {
+  kNetworkUp,
+  kNetworkDown,
+};
+
+}  // namespace webrtc
+
+#endif  // API_RTP_HEADERS_H_

VocieProcess/api/rtp_packet_info.cc

@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/rtp_packet_info.h"
+
+#include <stddef.h>
+
+#include <algorithm>
+#include <cstdint>
+#include <utility>
+#include <vector>
+
+#include "api/rtp_headers.h"
+#include "api/units/timestamp.h"
+
+namespace webrtc {
+
+RtpPacketInfo::RtpPacketInfo()
+    : ssrc_(0), rtp_timestamp_(0), receive_time_(Timestamp::MinusInfinity()) {}
+
+RtpPacketInfo::RtpPacketInfo(uint32_t ssrc,
+                             std::vector<uint32_t> csrcs,
+                             uint32_t rtp_timestamp,
+                             Timestamp receive_time)
+    : ssrc_(ssrc),
+      csrcs_(std::move(csrcs)),
+      rtp_timestamp_(rtp_timestamp),
+      receive_time_(receive_time) {}
+
+RtpPacketInfo::RtpPacketInfo(const RTPHeader& rtp_header,
+                             Timestamp receive_time)
+    : ssrc_(rtp_header.ssrc),
+      rtp_timestamp_(rtp_header.timestamp),
+      receive_time_(receive_time) {
+  const auto& extension = rtp_header.extension;
+  const auto csrcs_count = std::min<size_t>(rtp_header.numCSRCs, kRtpCsrcSize);
+
+  csrcs_.assign(&rtp_header.arrOfCSRCs[0], &rtp_header.arrOfCSRCs[csrcs_count]);
+
+  if (extension.audio_level()) {
+    audio_level_ = extension.audio_level()->level();
+  }
+
+  absolute_capture_time_ = extension.absolute_capture_time;
+}
+
+bool operator==(const RtpPacketInfo& lhs, const RtpPacketInfo& rhs) {
+  return (lhs.ssrc() == rhs.ssrc()) && (lhs.csrcs() == rhs.csrcs()) &&
+         (lhs.rtp_timestamp() == rhs.rtp_timestamp()) &&
+         (lhs.receive_time() == rhs.receive_time()) &&
+         (lhs.audio_level() == rhs.audio_level()) &&
+         (lhs.absolute_capture_time() == rhs.absolute_capture_time()) &&
+         (lhs.local_capture_clock_offset() == rhs.local_capture_clock_offset());
+}
+
+}  // namespace webrtc

VocieProcess/api/rtp_packet_info.h

@@ -0,0 +1,117 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_RTP_PACKET_INFO_H_
+#define API_RTP_PACKET_INFO_H_
+
+#include <cstdint>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/rtp_headers.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+//
+// Structure to hold information about a received `RtpPacket`. It is primarily
+// used to carry per-packet information from when a packet is received until
+// the information is passed to `SourceTracker`.
+//
+class RTC_EXPORT RtpPacketInfo {
+ public:
+  RtpPacketInfo();
+
+  RtpPacketInfo(uint32_t ssrc,
+                std::vector<uint32_t> csrcs,
+                uint32_t rtp_timestamp,
+                Timestamp receive_time);
+
+  RtpPacketInfo(const RTPHeader& rtp_header, Timestamp receive_time);
+
+  RtpPacketInfo(const RtpPacketInfo& other) = default;
+  RtpPacketInfo(RtpPacketInfo&& other) = default;
+  RtpPacketInfo& operator=(const RtpPacketInfo& other) = default;
+  RtpPacketInfo& operator=(RtpPacketInfo&& other) = default;
+
+  uint32_t ssrc() const { return ssrc_; }
+  void set_ssrc(uint32_t value) { ssrc_ = value; }
+
+  const std::vector<uint32_t>& csrcs() const { return csrcs_; }
+  void set_csrcs(std::vector<uint32_t> value) { csrcs_ = std::move(value); }
+
+  uint32_t rtp_timestamp() const { return rtp_timestamp_; }
+  void set_rtp_timestamp(uint32_t value) { rtp_timestamp_ = value; }
+
+  Timestamp receive_time() const { return receive_time_; }
+  void set_receive_time(Timestamp value) { receive_time_ = value; }
+
+  absl::optional<uint8_t> audio_level() const { return audio_level_; }
+  RtpPacketInfo& set_audio_level(absl::optional<uint8_t> value) {
+    audio_level_ = value;
+    return *this;
+  }
+
+  const absl::optional<AbsoluteCaptureTime>& absolute_capture_time() const {
+    return absolute_capture_time_;
+  }
+  RtpPacketInfo& set_absolute_capture_time(
+      const absl::optional<AbsoluteCaptureTime>& value) {
+    absolute_capture_time_ = value;
+    return *this;
+  }
+
+  const absl::optional<TimeDelta>& local_capture_clock_offset() const {
+    return local_capture_clock_offset_;
+  }
+  RtpPacketInfo& set_local_capture_clock_offset(
+      absl::optional<TimeDelta> value) {
+    local_capture_clock_offset_ = value;
+    return *this;
+  }
+
+ private:
+  // Fields from the RTP header:
+  // https://tools.ietf.org/html/rfc3550#section-5.1
+  uint32_t ssrc_;
+  std::vector<uint32_t> csrcs_;
+  uint32_t rtp_timestamp_;
+
+  // Local `webrtc::Clock`-based timestamp of when the packet was received.
+  Timestamp receive_time_;
+
+  // Fields from the Audio Level header extension:
+  // https://tools.ietf.org/html/rfc6464#section-3
+  absl::optional<uint8_t> audio_level_;
+
+  // Fields from the Absolute Capture Time header extension:
+  // http://www.webrtc.org/experiments/rtp-hdrext/abs-capture-time
+  absl::optional<AbsoluteCaptureTime> absolute_capture_time_;
+
+  // Clock offset between the local clock and the capturer's clock.
+  // Do not confuse with `AbsoluteCaptureTime::estimated_capture_clock_offset`
+  // which instead represents the clock offset between a remote sender and the
+  // capturer. The following holds:
+  //   Capture's NTP Clock = Local NTP Clock + Local-Capture Clock Offset
+  absl::optional<TimeDelta> local_capture_clock_offset_;
+};
+
+bool operator==(const RtpPacketInfo& lhs, const RtpPacketInfo& rhs);
+
+inline bool operator!=(const RtpPacketInfo& lhs, const RtpPacketInfo& rhs) {
+  return !(lhs == rhs);
+}
+
+}  // namespace webrtc
+
+#endif  // API_RTP_PACKET_INFO_H_

VocieProcess/api/rtp_packet_infos.h

@@ -0,0 +1,130 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_RTP_PACKET_INFOS_H_
+#define API_RTP_PACKET_INFOS_H_
+
+#include <utility>
+#include <vector>
+
+#include "api/make_ref_counted.h"
+#include "api/ref_counted_base.h"
+#include "api/rtp_packet_info.h"
+#include "api/scoped_refptr.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// Semi-immutable structure to hold information about packets used to assemble
+// an audio or video frame. Uses internal reference counting to make it very
+// cheap to copy.
+//
+// We should ideally just use `std::vector<RtpPacketInfo>` and have it
+// `std::move()`-ed as the per-packet information is transferred from one object
+// to another. But moving the info, instead of copying it, is not easily done
+// for the current video code.
+class RTC_EXPORT RtpPacketInfos {
+ public:
+  using vector_type = std::vector<RtpPacketInfo>;
+
+  using value_type = vector_type::value_type;
+  using size_type = vector_type::size_type;
+  using difference_type = vector_type::difference_type;
+  using const_reference = vector_type::const_reference;
+  using const_pointer = vector_type::const_pointer;
+  using const_iterator = vector_type::const_iterator;
+  using const_reverse_iterator = vector_type::const_reverse_iterator;
+
+  using reference = const_reference;
+  using pointer = const_pointer;
+  using iterator = const_iterator;
+  using reverse_iterator = const_reverse_iterator;
+
+  RtpPacketInfos() {}
+  explicit RtpPacketInfos(const vector_type& entries)
+      : data_(Data::Create(entries)) {}
+
+  explicit RtpPacketInfos(vector_type&& entries)
+      : data_(Data::Create(std::move(entries))) {}
+
+  RtpPacketInfos(const RtpPacketInfos& other) = default;
+  RtpPacketInfos(RtpPacketInfos&& other) = default;
+  RtpPacketInfos& operator=(const RtpPacketInfos& other) = default;
+  RtpPacketInfos& operator=(RtpPacketInfos&& other) = default;
+
+  const_reference operator[](size_type pos) const { return entries()[pos]; }
+
+  const_reference at(size_type pos) const { return entries().at(pos); }
+  const_reference front() const { return entries().front(); }
+  const_reference back() const { return entries().back(); }
+
+  const_iterator begin() const { return entries().begin(); }
+  const_iterator end() const { return entries().end(); }
+  const_reverse_iterator rbegin() const { return entries().rbegin(); }
+  const_reverse_iterator rend() const { return entries().rend(); }
+
+  const_iterator cbegin() const { return entries().cbegin(); }
+  const_iterator cend() const { return entries().cend(); }
+  const_reverse_iterator crbegin() const { return entries().crbegin(); }
+  const_reverse_iterator crend() const { return entries().crend(); }
+
+  bool empty() const { return entries().empty(); }
+  size_type size() const { return entries().size(); }
+
+ private:
+  class Data final : public rtc::RefCountedNonVirtual<Data> {
+   public:
+    static rtc::scoped_refptr<Data> Create(const vector_type& entries) {
+      // Performance optimization for the empty case.
+      if (entries.empty()) {
+        return nullptr;
+      }
+
+      return rtc::make_ref_counted<Data>(entries);
+    }
+
+    static rtc::scoped_refptr<Data> Create(vector_type&& entries) {
+      // Performance optimization for the empty case.
+      if (entries.empty()) {
+        return nullptr;
+      }
+
+      return rtc::make_ref_counted<Data>(std::move(entries));
+    }
+
+    const vector_type& entries() const { return entries_; }
+
+    explicit Data(const vector_type& entries) : entries_(entries) {}
+    explicit Data(vector_type&& entries) : entries_(std::move(entries)) {}
+    ~Data() = default;
+
+   private:
+    const vector_type entries_;
+  };
+
+  static const vector_type& empty_entries() {
+    static const vector_type& value = *new vector_type();
+    return value;
+  }
+
+  const vector_type& entries() const {
+    if (data_ != nullptr) {
+      return data_->entries();
+    } else {
+      return empty_entries();
+    }
+  }
+
+  rtc::scoped_refptr<Data> data_;
+};
+
+}  // namespace webrtc
+
+#endif  // API_RTP_PACKET_INFOS_H_

VocieProcess/api/sequence_checker.h

@@ -0,0 +1,141 @@
+/*
+ *  Copyright 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef API_SEQUENCE_CHECKER_H_
+#define API_SEQUENCE_CHECKER_H_
+
+#include "api/task_queue/task_queue_base.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/synchronization/sequence_checker_internal.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+
+// SequenceChecker is a helper class used to help verify that some methods
+// of a class are called on the same task queue or thread. A
+// SequenceChecker is bound to a a task queue if the object is
+// created on a task queue, or a thread otherwise.
+//
+//
+// Example:
+// class MyClass {
+//  public:
+//   void Foo() {
+//     RTC_DCHECK_RUN_ON(&sequence_checker_);
+//     ... (do stuff) ...
+//   }
+//
+//  private:
+//   SequenceChecker sequence_checker_;
+// }
+//
+// In Release mode, IsCurrent will always return true.
+class RTC_LOCKABLE SequenceChecker
+#if RTC_DCHECK_IS_ON
+    : public webrtc_sequence_checker_internal::SequenceCheckerImpl {
+  using Impl = webrtc_sequence_checker_internal::SequenceCheckerImpl;
+#else
+    : public webrtc_sequence_checker_internal::SequenceCheckerDoNothing {
+  using Impl = webrtc_sequence_checker_internal::SequenceCheckerDoNothing;
+#endif
+ public:
+  enum InitialState : bool { kDetached = false, kAttached = true };
+
+  // TODO(tommi): We could maybe join these two ctors and have fewer factory
+  // functions. At the moment they're separate to minimize code changes when
+  // we added the second ctor as well as avoiding to have unnecessary code at
+  // the SequenceChecker which much only run for the SequenceCheckerImpl
+  // implementation.
+  // In theory we could have something like:
+  //
+  //  SequenceChecker(InitialState initial_state = kAttached,
+  //                  TaskQueueBase* attached_queue = TaskQueueBase::Current());
+  //
+  // But the problem with that is having the call to `Current()` exist for
+  // `SequenceCheckerDoNothing`.
+  explicit SequenceChecker(InitialState initial_state = kAttached)
+      : Impl(initial_state) {}
+  explicit SequenceChecker(TaskQueueBase* attached_queue)
+      : Impl(attached_queue) {}
+
+  // Returns true if sequence checker is attached to the current sequence.
+  bool IsCurrent() const { return Impl::IsCurrent(); }
+  // Detaches checker from sequence to which it is attached. Next attempt
+  // to do a check with this checker will result in attaching this checker
+  // to the sequence on which check was performed.
+  void Detach() { Impl::Detach(); }
+};
+
+}  // namespace webrtc
+
+// RTC_RUN_ON/RTC_GUARDED_BY/RTC_DCHECK_RUN_ON macros allows to annotate
+// variables are accessed from same thread/task queue.
+// Using tools designed to check mutexes, it checks at compile time everywhere
+// variable is access, there is a run-time dcheck thread/task queue is correct.
+//
+// class SequenceCheckerExample {
+//  public:
+//   int CalledFromPacer() RTC_RUN_ON(pacer_sequence_checker_) {
+//     return var2_;
+//   }
+//
+//   void CallMeFromPacer() {
+//     RTC_DCHECK_RUN_ON(&pacer_sequence_checker_)
+//        << "Should be called from pacer";
+//     CalledFromPacer();
+//   }
+//
+//  private:
+//   int pacer_var_ RTC_GUARDED_BY(pacer_sequence_checker_);
+//   SequenceChecker pacer_sequence_checker_;
+// };
+//
+// class TaskQueueExample {
+//  public:
+//   class Encoder {
+//    public:
+//     rtc::TaskQueueBase& Queue() { return encoder_queue_; }
+//     void Encode() {
+//       RTC_DCHECK_RUN_ON(&encoder_queue_);
+//       DoSomething(var_);
+//     }
+//
+//    private:
+//     rtc::TaskQueueBase& encoder_queue_;
+//     Frame var_ RTC_GUARDED_BY(encoder_queue_);
+//   };
+//
+//   void Encode() {
+//     // Will fail at runtime when DCHECK is enabled:
+//     // encoder_->Encode();
+//     // Will work:
+//     rtc::scoped_refptr<Encoder> encoder = encoder_;
+//     encoder_->Queue().PostTask([encoder] { encoder->Encode(); });
+//   }
+//
+//  private:
+//   rtc::scoped_refptr<Encoder> encoder_;
+// }
+
+// Document if a function expected to be called from same thread/task queue.
+#define RTC_RUN_ON(x) \
+  RTC_THREAD_ANNOTATION_ATTRIBUTE__(exclusive_locks_required(x))
+
+// Checks current code is running on the desired sequence.
+//
+// First statement validates it is running on the sequence `x`.
+// Second statement annotates for the thread safety analyzer the check was done.
+// Such annotation has to be attached to a function, and that function has to be
+// called. Thus current implementation creates a noop lambda and calls it.
+#define RTC_DCHECK_RUN_ON(x)                                               \
+  RTC_DCHECK((x)->IsCurrent())                                             \
+      << webrtc::webrtc_sequence_checker_internal::ExpectationToString(x); \
+  []() RTC_ASSERT_EXCLUSIVE_LOCK(x) {}()
+
+#endif  // API_SEQUENCE_CHECKER_H_

VocieProcess/api/video/color_space.cc

@@ -0,0 +1,269 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/video/color_space.h"
+
+#include <cstddef>
+#include <cstdint>
+#include <string>
+
+#include "absl/types/optional.h"
+#include "api/video/hdr_metadata.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace webrtc {
+namespace {
+// Try to convert `enum_value` into the enum class T. `enum_bitmask` is created
+// by the funciton below. Returns true if conversion was successful, false
+// otherwise.
+template <typename T>
+bool SetFromUint8(uint8_t enum_value, uint64_t enum_bitmask, T* out) {
+  if ((enum_value < 64) && ((enum_bitmask >> enum_value) & 1)) {
+    *out = static_cast<T>(enum_value);
+    return true;
+  }
+  return false;
+}
+
+// This function serves as an assert for the constexpr function below. It's on
+// purpose not declared as constexpr so that it causes a build problem if enum
+// values of 64 or above are used. The bitmask and the code generating it would
+// have to be extended if the standard is updated to include enum values >= 64.
+int EnumMustBeLessThan64() {
+  return -1;
+}
+
+template <typename T, size_t N>
+constexpr int MakeMask(const int index, const int length, T (&values)[N]) {
+  return length > 1
+             ? (MakeMask(index, 1, values) +
+                MakeMask(index + 1, length - 1, values))
+             : (static_cast<uint8_t>(values[index]) < 64
+                    ? (uint64_t{1} << static_cast<uint8_t>(values[index]))
+                    : EnumMustBeLessThan64());
+}
+
+// Create a bitmask where each bit corresponds to one potential enum value.
+// `values` should be an array listing all possible enum values. The bit is set
+// to one if the corresponding enum exists. Only works for enums with values
+// less than 64.
+template <typename T, size_t N>
+constexpr uint64_t CreateEnumBitmask(T (&values)[N]) {
+  return MakeMask(0, N, values);
+}
+
+bool SetChromaSitingFromUint8(uint8_t enum_value,
+                              ColorSpace::ChromaSiting* chroma_siting) {
+  constexpr ColorSpace::ChromaSiting kChromaSitings[] = {
+      ColorSpace::ChromaSiting::kUnspecified,
+      ColorSpace::ChromaSiting::kCollocated, ColorSpace::ChromaSiting::kHalf};
+  constexpr uint64_t enum_bitmask = CreateEnumBitmask(kChromaSitings);
+
+  return SetFromUint8(enum_value, enum_bitmask, chroma_siting);
+}
+
+}  // namespace
+
+ColorSpace::ColorSpace() = default;
+ColorSpace::ColorSpace(const ColorSpace& other) = default;
+ColorSpace::ColorSpace(ColorSpace&& other) = default;
+ColorSpace& ColorSpace::operator=(const ColorSpace& other) = default;
+
+ColorSpace::ColorSpace(PrimaryID primaries,
+                       TransferID transfer,
+                       MatrixID matrix,
+                       RangeID range)
+    : ColorSpace(primaries,
+                 transfer,
+                 matrix,
+                 range,
+                 ChromaSiting::kUnspecified,
+                 ChromaSiting::kUnspecified,
+                 nullptr) {}
+
+ColorSpace::ColorSpace(PrimaryID primaries,
+                       TransferID transfer,
+                       MatrixID matrix,
+                       RangeID range,
+                       ChromaSiting chroma_siting_horz,
+                       ChromaSiting chroma_siting_vert,
+                       const HdrMetadata* hdr_metadata)
+    : primaries_(primaries),
+      transfer_(transfer),
+      matrix_(matrix),
+      range_(range),
+      chroma_siting_horizontal_(chroma_siting_horz),
+      chroma_siting_vertical_(chroma_siting_vert),
+      hdr_metadata_(hdr_metadata ? absl::make_optional(*hdr_metadata)
+                                 : absl::nullopt) {}
+
+ColorSpace::PrimaryID ColorSpace::primaries() const {
+  return primaries_;
+}
+
+ColorSpace::TransferID ColorSpace::transfer() const {
+  return transfer_;
+}
+
+ColorSpace::MatrixID ColorSpace::matrix() const {
+  return matrix_;
+}
+
+ColorSpace::RangeID ColorSpace::range() const {
+  return range_;
+}
+
+ColorSpace::ChromaSiting ColorSpace::chroma_siting_horizontal() const {
+  return chroma_siting_horizontal_;
+}
+
+ColorSpace::ChromaSiting ColorSpace::chroma_siting_vertical() const {
+  return chroma_siting_vertical_;
+}
+
+const HdrMetadata* ColorSpace::hdr_metadata() const {
+  return hdr_metadata_ ? &*hdr_metadata_ : nullptr;
+}
+
+#define PRINT_ENUM_CASE(TYPE, NAME) \
+  case TYPE::NAME:                  \
+    ss << #NAME;                    \
+    break;
+
+std::string ColorSpace::AsString() const {
+  char buf[1024];
+  rtc::SimpleStringBuilder ss(buf);
+  ss << "{primaries:";
+  switch (primaries_) {
+    PRINT_ENUM_CASE(PrimaryID, kBT709)
+    PRINT_ENUM_CASE(PrimaryID, kUnspecified)
+    PRINT_ENUM_CASE(PrimaryID, kBT470M)
+    PRINT_ENUM_CASE(PrimaryID, kBT470BG)
+    PRINT_ENUM_CASE(PrimaryID, kSMPTE170M)
+    PRINT_ENUM_CASE(PrimaryID, kSMPTE240M)
+    PRINT_ENUM_CASE(PrimaryID, kFILM)
+    PRINT_ENUM_CASE(PrimaryID, kBT2020)
+    PRINT_ENUM_CASE(PrimaryID, kSMPTEST428)
+    PRINT_ENUM_CASE(PrimaryID, kSMPTEST431)
+    PRINT_ENUM_CASE(PrimaryID, kSMPTEST432)
+    PRINT_ENUM_CASE(PrimaryID, kJEDECP22)
+  }
+  ss << ", transfer:";
+  switch (transfer_) {
+    PRINT_ENUM_CASE(TransferID, kBT709)
+    PRINT_ENUM_CASE(TransferID, kUnspecified)
+    PRINT_ENUM_CASE(TransferID, kGAMMA22)
+    PRINT_ENUM_CASE(TransferID, kGAMMA28)
+    PRINT_ENUM_CASE(TransferID, kSMPTE170M)
+    PRINT_ENUM_CASE(TransferID, kSMPTE240M)
+    PRINT_ENUM_CASE(TransferID, kLINEAR)
+    PRINT_ENUM_CASE(TransferID, kLOG)
+    PRINT_ENUM_CASE(TransferID, kLOG_SQRT)
+    PRINT_ENUM_CASE(TransferID, kIEC61966_2_4)
+    PRINT_ENUM_CASE(TransferID, kBT1361_ECG)
+    PRINT_ENUM_CASE(TransferID, kIEC61966_2_1)
+    PRINT_ENUM_CASE(TransferID, kBT2020_10)
+    PRINT_ENUM_CASE(TransferID, kBT2020_12)
+    PRINT_ENUM_CASE(TransferID, kSMPTEST2084)
+    PRINT_ENUM_CASE(TransferID, kSMPTEST428)
+    PRINT_ENUM_CASE(TransferID, kARIB_STD_B67)
+  }
+  ss << ", matrix:";
+  switch (matrix_) {
+    PRINT_ENUM_CASE(MatrixID, kRGB)
+    PRINT_ENUM_CASE(MatrixID, kBT709)
+    PRINT_ENUM_CASE(MatrixID, kUnspecified)
+    PRINT_ENUM_CASE(MatrixID, kFCC)
+    PRINT_ENUM_CASE(MatrixID, kBT470BG)
+    PRINT_ENUM_CASE(MatrixID, kSMPTE170M)
+    PRINT_ENUM_CASE(MatrixID, kSMPTE240M)
+    PRINT_ENUM_CASE(MatrixID, kYCOCG)
+    PRINT_ENUM_CASE(MatrixID, kBT2020_NCL)
+    PRINT_ENUM_CASE(MatrixID, kBT2020_CL)
+    PRINT_ENUM_CASE(MatrixID, kSMPTE2085)
+    PRINT_ENUM_CASE(MatrixID, kCDNCLS)
+    PRINT_ENUM_CASE(MatrixID, kCDCLS)
+    PRINT_ENUM_CASE(MatrixID, kBT2100_ICTCP)
+  }
+
+  ss << ", range:";
+  switch (range_) {
+    PRINT_ENUM_CASE(RangeID, kInvalid)
+    PRINT_ENUM_CASE(RangeID, kLimited)
+    PRINT_ENUM_CASE(RangeID, kFull)
+    PRINT_ENUM_CASE(RangeID, kDerived)
+  }
+  ss << "}";
+  return ss.str();
+}
+
+#undef PRINT_ENUM_CASE
+
+bool ColorSpace::set_primaries_from_uint8(uint8_t enum_value) {
+  constexpr PrimaryID kPrimaryIds[] = {
+      PrimaryID::kBT709,      PrimaryID::kUnspecified, PrimaryID::kBT470M,
+      PrimaryID::kBT470BG,    PrimaryID::kSMPTE170M,   PrimaryID::kSMPTE240M,
+      PrimaryID::kFILM,       PrimaryID::kBT2020,      PrimaryID::kSMPTEST428,
+      PrimaryID::kSMPTEST431, PrimaryID::kSMPTEST432,  PrimaryID::kJEDECP22};
+  constexpr uint64_t enum_bitmask = CreateEnumBitmask(kPrimaryIds);
+
+  return SetFromUint8(enum_value, enum_bitmask, &primaries_);
+}
+
+bool ColorSpace::set_transfer_from_uint8(uint8_t enum_value) {
+  constexpr TransferID kTransferIds[] = {
+      TransferID::kBT709,       TransferID::kUnspecified,
+      TransferID::kGAMMA22,     TransferID::kGAMMA28,
+      TransferID::kSMPTE170M,   TransferID::kSMPTE240M,
+      TransferID::kLINEAR,      TransferID::kLOG,
+      TransferID::kLOG_SQRT,    TransferID::kIEC61966_2_4,
+      TransferID::kBT1361_ECG,  TransferID::kIEC61966_2_1,
+      TransferID::kBT2020_10,   TransferID::kBT2020_12,
+      TransferID::kSMPTEST2084, TransferID::kSMPTEST428,
+      TransferID::kARIB_STD_B67};
+  constexpr uint64_t enum_bitmask = CreateEnumBitmask(kTransferIds);
+
+  return SetFromUint8(enum_value, enum_bitmask, &transfer_);
+}
+
+bool ColorSpace::set_matrix_from_uint8(uint8_t enum_value) {
+  constexpr MatrixID kMatrixIds[] = {
+      MatrixID::kRGB,       MatrixID::kBT709,       MatrixID::kUnspecified,
+      MatrixID::kFCC,       MatrixID::kBT470BG,     MatrixID::kSMPTE170M,
+      MatrixID::kSMPTE240M, MatrixID::kYCOCG,       MatrixID::kBT2020_NCL,
+      MatrixID::kBT2020_CL, MatrixID::kSMPTE2085,   MatrixID::kCDNCLS,
+      MatrixID::kCDCLS,     MatrixID::kBT2100_ICTCP};
+  constexpr uint64_t enum_bitmask = CreateEnumBitmask(kMatrixIds);
+
+  return SetFromUint8(enum_value, enum_bitmask, &matrix_);
+}
+
+bool ColorSpace::set_range_from_uint8(uint8_t enum_value) {
+  constexpr RangeID kRangeIds[] = {RangeID::kInvalid, RangeID::kLimited,
+                                   RangeID::kFull, RangeID::kDerived};
+  constexpr uint64_t enum_bitmask = CreateEnumBitmask(kRangeIds);
+
+  return SetFromUint8(enum_value, enum_bitmask, &range_);
+}
+
+bool ColorSpace::set_chroma_siting_horizontal_from_uint8(uint8_t enum_value) {
+  return SetChromaSitingFromUint8(enum_value, &chroma_siting_horizontal_);
+}
+
+bool ColorSpace::set_chroma_siting_vertical_from_uint8(uint8_t enum_value) {
+  return SetChromaSitingFromUint8(enum_value, &chroma_siting_vertical_);
+}
+
+void ColorSpace::set_hdr_metadata(const HdrMetadata* hdr_metadata) {
+  hdr_metadata_ =
+      hdr_metadata ? absl::make_optional(*hdr_metadata) : absl::nullopt;
+}
+
+}  // namespace webrtc

VocieProcess/api/video/color_space.h

@@ -0,0 +1,181 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_VIDEO_COLOR_SPACE_H_
+#define API_VIDEO_COLOR_SPACE_H_
+
+#include <stdint.h>
+
+#include <string>
+
+#include "absl/types/optional.h"
+#include "api/video/hdr_metadata.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// This class represents color information as specified in T-REC H.273,
+// available from https://www.itu.int/rec/T-REC-H.273.
+//
+// WebRTC's supported codecs:
+// - VP9 supports color profiles, see VP9 Bitstream & Decoding Process
+// Specification Version 0.6 Section 7.2.2 "Color config semantics" available
+// from https://www.webmproject.org.
+// - VP8 only supports BT.601, see
+// https://tools.ietf.org/html/rfc6386#section-9.2
+// - H264 uses the exact same representation as T-REC H.273. See T-REC-H.264
+// E.2.1, "VUI parameters semantics", available from
+// https://www.itu.int/rec/T-REC-H.264.
+
+class RTC_EXPORT ColorSpace {
+ public:
+  enum class PrimaryID : uint8_t {
+    // The indices are equal to the values specified in T-REC H.273 Table 2.
+    kBT709 = 1,
+    kUnspecified = 2,
+    kBT470M = 4,
+    kBT470BG = 5,
+    kSMPTE170M = 6,  // Identical to BT601
+    kSMPTE240M = 7,
+    kFILM = 8,
+    kBT2020 = 9,
+    kSMPTEST428 = 10,
+    kSMPTEST431 = 11,
+    kSMPTEST432 = 12,
+    kJEDECP22 = 22,  // Identical to EBU3213-E
+    // When adding/removing entries here, please make sure to do the
+    // corresponding change to kPrimaryIds.
+  };
+
+  enum class TransferID : uint8_t {
+    // The indices are equal to the values specified in T-REC H.273 Table 3.
+    kBT709 = 1,
+    kUnspecified = 2,
+    kGAMMA22 = 4,
+    kGAMMA28 = 5,
+    kSMPTE170M = 6,
+    kSMPTE240M = 7,
+    kLINEAR = 8,
+    kLOG = 9,
+    kLOG_SQRT = 10,
+    kIEC61966_2_4 = 11,
+    kBT1361_ECG = 12,
+    kIEC61966_2_1 = 13,
+    kBT2020_10 = 14,
+    kBT2020_12 = 15,
+    kSMPTEST2084 = 16,
+    kSMPTEST428 = 17,
+    kARIB_STD_B67 = 18,
+    // When adding/removing entries here, please make sure to do the
+    // corresponding change to kTransferIds.
+  };
+
+  enum class MatrixID : uint8_t {
+    // The indices are equal to the values specified in T-REC H.273 Table 4.
+    kRGB = 0,
+    kBT709 = 1,
+    kUnspecified = 2,
+    kFCC = 4,
+    kBT470BG = 5,
+    kSMPTE170M = 6,
+    kSMPTE240M = 7,
+    kYCOCG = 8,
+    kBT2020_NCL = 9,
+    kBT2020_CL = 10,
+    kSMPTE2085 = 11,
+    kCDNCLS = 12,
+    kCDCLS = 13,
+    kBT2100_ICTCP = 14,
+    // When adding/removing entries here, please make sure to do the
+    // corresponding change to kMatrixIds.
+  };
+
+  enum class RangeID {
+    // The indices are equal to the values specified at
+    // https://www.webmproject.org/docs/container/#colour for the element Range.
+    kInvalid = 0,
+    // Limited Rec. 709 color range with RGB values ranging from 16 to 235.
+    kLimited = 1,
+    // Full RGB color range with RGB values from 0 to 255.
+    kFull = 2,
+    // Range is defined by MatrixCoefficients/TransferCharacteristics.
+    kDerived = 3,
+    // When adding/removing entries here, please make sure to do the
+    // corresponding change to kRangeIds.
+  };
+
+  enum class ChromaSiting {
+    // Chroma siting specifies how chroma is subsampled relative to the luma
+    // samples in a YUV video frame.
+    // The indices are equal to the values specified at
+    // https://www.webmproject.org/docs/container/#colour for the element
+    // ChromaSitingVert and ChromaSitingHorz.
+    kUnspecified = 0,
+    kCollocated = 1,
+    kHalf = 2,
+    // When adding/removing entries here, please make sure to do the
+    // corresponding change to kChromaSitings.
+  };
+
+  ColorSpace();
+  ColorSpace(const ColorSpace& other);
+  ColorSpace(ColorSpace&& other);
+  ColorSpace& operator=(const ColorSpace& other);
+  ColorSpace(PrimaryID primaries,
+             TransferID transfer,
+             MatrixID matrix,
+             RangeID range);
+  ColorSpace(PrimaryID primaries,
+             TransferID transfer,
+             MatrixID matrix,
+             RangeID range,
+             ChromaSiting chroma_siting_horizontal,
+             ChromaSiting chroma_siting_vertical,
+             const HdrMetadata* hdr_metadata);
+  friend bool operator==(const ColorSpace& lhs, const ColorSpace& rhs) {
+    return lhs.primaries_ == rhs.primaries_ && lhs.transfer_ == rhs.transfer_ &&
+           lhs.matrix_ == rhs.matrix_ && lhs.range_ == rhs.range_ &&
+           lhs.chroma_siting_horizontal_ == rhs.chroma_siting_horizontal_ &&
+           lhs.chroma_siting_vertical_ == rhs.chroma_siting_vertical_ &&
+           lhs.hdr_metadata_ == rhs.hdr_metadata_;
+  }
+  friend bool operator!=(const ColorSpace& lhs, const ColorSpace& rhs) {
+    return !(lhs == rhs);
+  }
+
+  PrimaryID primaries() const;
+  TransferID transfer() const;
+  MatrixID matrix() const;
+  RangeID range() const;
+  ChromaSiting chroma_siting_horizontal() const;
+  ChromaSiting chroma_siting_vertical() const;
+  const HdrMetadata* hdr_metadata() const;
+  std::string AsString() const;
+
+  bool set_primaries_from_uint8(uint8_t enum_value);
+  bool set_transfer_from_uint8(uint8_t enum_value);
+  bool set_matrix_from_uint8(uint8_t enum_value);
+  bool set_range_from_uint8(uint8_t enum_value);
+  bool set_chroma_siting_horizontal_from_uint8(uint8_t enum_value);
+  bool set_chroma_siting_vertical_from_uint8(uint8_t enum_value);
+  void set_hdr_metadata(const HdrMetadata* hdr_metadata);
+
+ private:
+  PrimaryID primaries_ = PrimaryID::kUnspecified;
+  TransferID transfer_ = TransferID::kUnspecified;
+  MatrixID matrix_ = MatrixID::kUnspecified;
+  RangeID range_ = RangeID::kInvalid;
+  ChromaSiting chroma_siting_horizontal_ = ChromaSiting::kUnspecified;
+  ChromaSiting chroma_siting_vertical_ = ChromaSiting::kUnspecified;
+  absl::optional<HdrMetadata> hdr_metadata_;
+};
+
+}  // namespace webrtc
+#endif  // API_VIDEO_COLOR_SPACE_H_

VocieProcess/api/video/hdr_metadata.cc

@@ -0,0 +1,21 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/video/hdr_metadata.h"
+
+namespace webrtc {
+
+HdrMasteringMetadata::Chromaticity::Chromaticity() = default;
+
+HdrMasteringMetadata::HdrMasteringMetadata() = default;
+
+HdrMetadata::HdrMetadata() = default;
+
+}  // namespace webrtc

VocieProcess/api/video/hdr_metadata.h

@@ -0,0 +1,105 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_VIDEO_HDR_METADATA_H_
+#define API_VIDEO_HDR_METADATA_H_
+
+namespace webrtc {
+
+// SMPTE ST 2086 mastering metadata,
+// see https://ieeexplore.ieee.org/document/8353899.
+struct HdrMasteringMetadata {
+  struct Chromaticity {
+    Chromaticity();
+
+    bool operator==(const Chromaticity& rhs) const {
+      return x == rhs.x && y == rhs.y;
+    }
+
+    bool Validate() const {
+      return x >= 0.0 && x <= 1.0 && y >= 0.0 && y <= 1.0;
+    }
+
+    // xy chromaticity coordinates must be calculated as specified in ISO
+    // 11664-3:2012 Section 7, and must be specified with four decimal places.
+    // The x coordinate should be in the range [0.0001, 0.7400] and the y
+    // coordinate should be in the range [0.0001, 0.8400]. Valid range [0.0000,
+    // 1.0000].
+    float x = 0.0f;
+    float y = 0.0f;
+  };
+
+  HdrMasteringMetadata();
+
+  bool operator==(const HdrMasteringMetadata& rhs) const {
+    return ((primary_r == rhs.primary_r) && (primary_g == rhs.primary_g) &&
+            (primary_b == rhs.primary_b) && (white_point == rhs.white_point) &&
+            (luminance_max == rhs.luminance_max) &&
+            (luminance_min == rhs.luminance_min));
+  }
+
+  bool Validate() const {
+    return luminance_max >= 0.0 && luminance_max <= 20000.0 &&
+           luminance_min >= 0.0 && luminance_min <= 5.0 &&
+           primary_r.Validate() && primary_g.Validate() &&
+           primary_b.Validate() && white_point.Validate();
+  }
+
+  // The nominal primaries of the mastering display.
+  Chromaticity primary_r;
+  Chromaticity primary_g;
+  Chromaticity primary_b;
+
+  // The nominal chromaticity of the white point of the mastering display.
+  Chromaticity white_point;
+
+  // The nominal maximum display luminance of the mastering display. Specified
+  // in the unit candela/m2. The value should be in the range [5, 10000] with
+  // zero decimal places. Valid range [0, 20000].
+  float luminance_max = 0.0f;
+
+  // The nominal minimum display luminance of the mastering display. Specified
+  // in the unit candela/m2. The value should be in the range [0.0001, 5.0000]
+  // with four decimal places. Valid range [0.0000, 5.0000].
+  float luminance_min = 0.0f;
+};
+
+// High dynamic range (HDR) metadata common for HDR10 and WebM/VP9-based HDR
+// formats. This struct replicates the HDRMetadata struct defined in
+// https://cs.chromium.org/chromium/src/media/base/hdr_metadata.h
+struct HdrMetadata {
+  HdrMetadata();
+
+  bool operator==(const HdrMetadata& rhs) const {
+    return (
+        (max_content_light_level == rhs.max_content_light_level) &&
+        (max_frame_average_light_level == rhs.max_frame_average_light_level) &&
+        (mastering_metadata == rhs.mastering_metadata));
+  }
+
+  bool Validate() const {
+    return max_content_light_level >= 0 && max_content_light_level <= 20000 &&
+           max_frame_average_light_level >= 0 &&
+           max_frame_average_light_level <= 20000 &&
+           mastering_metadata.Validate();
+  }
+
+  HdrMasteringMetadata mastering_metadata;
+  // Max content light level (CLL), i.e. maximum brightness level present in the
+  // stream, in nits. 1 nit = 1 candela/m2. Valid range [0, 20000].
+  int max_content_light_level = 0;
+  // Max frame-average light level (FALL), i.e. maximum average brightness of
+  // the brightest frame in the stream, in nits. Valid range [0, 20000].
+  int max_frame_average_light_level = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // API_VIDEO_HDR_METADATA_H_

VocieProcess/api/video/video_content_type.cc

@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/video/video_content_type.h"
+
+#include <cstdint>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace videocontenttypehelpers {
+
+namespace {
+static constexpr uint8_t kScreenshareBitsSize = 1;
+static constexpr uint8_t kScreenshareBitsMask =
+    (1u << kScreenshareBitsSize) - 1;
+}  // namespace
+
+bool IsScreenshare(const VideoContentType& content_type) {
+  // Ensure no bits apart from the screenshare bit is set.
+  // This CHECK is a temporary measure to detect code that introduces
+  // values according to old versions.
+  RTC_CHECK((static_cast<uint8_t>(content_type) & !kScreenshareBitsMask) == 0);
+  return (static_cast<uint8_t>(content_type) & kScreenshareBitsMask) > 0;
+}
+
+bool IsValidContentType(uint8_t value) {
+  // Only the screenshare bit is allowed.
+  // However, due to previous usage of the next 5 bits, we allow
+  // the lower 6 bits to be set.
+  return value < (1 << 6);
+}
+
+const char* ToString(const VideoContentType& content_type) {
+  return IsScreenshare(content_type) ? "screen" : "realtime";
+}
+}  // namespace videocontenttypehelpers
+}  // namespace webrtc

VocieProcess/api/video/video_content_type.h

@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_VIDEO_VIDEO_CONTENT_TYPE_H_
+#define API_VIDEO_VIDEO_CONTENT_TYPE_H_
+
+#include <stdint.h>
+
+namespace webrtc {
+
+// VideoContentType stored as a single byte, which is sent over the network
+// in the rtp-hdrext/video-content-type extension.
+// Only the lowest bit is used, per the enum.
+enum class VideoContentType : uint8_t {
+  UNSPECIFIED = 0,
+  SCREENSHARE = 1,
+};
+
+namespace videocontenttypehelpers {
+bool IsScreenshare(const VideoContentType& content_type);
+
+bool IsValidContentType(uint8_t value);
+
+const char* ToString(const VideoContentType& content_type);
+}  // namespace videocontenttypehelpers
+
+}  // namespace webrtc
+
+#endif  // API_VIDEO_VIDEO_CONTENT_TYPE_H_

VocieProcess/api/video/video_rotation.h

@@ -0,0 +1,26 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_VIDEO_VIDEO_ROTATION_H_
+#define API_VIDEO_VIDEO_ROTATION_H_
+
+namespace webrtc {
+
+// enum for clockwise rotation.
+enum VideoRotation {
+  kVideoRotation_0 = 0,
+  kVideoRotation_90 = 90,
+  kVideoRotation_180 = 180,
+  kVideoRotation_270 = 270
+};
+
+}  // namespace webrtc
+
+#endif  // API_VIDEO_VIDEO_ROTATION_H_

VocieProcess/api/video/video_timing.cc

@@ -0,0 +1,124 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/video/video_timing.h"
+
+#include <algorithm>
+#include <cstdint>
+#include <string>
+
+#include "api/array_view.h"
+#include "api/units/time_delta.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace webrtc {
+
+uint16_t VideoSendTiming::GetDeltaCappedMs(int64_t base_ms, int64_t time_ms) {
+  if (time_ms < base_ms) {
+    RTC_DLOG(LS_ERROR) << "Delta " << (time_ms - base_ms)
+                       << "ms expected to be positive";
+  }
+  return rtc::saturated_cast<uint16_t>(time_ms - base_ms);
+}
+
+uint16_t VideoSendTiming::GetDeltaCappedMs(TimeDelta delta) {
+  if (delta < TimeDelta::Zero()) {
+    RTC_DLOG(LS_ERROR) << "Delta " << delta.ms()
+                       << "ms expected to be positive";
+  }
+  return rtc::saturated_cast<uint16_t>(delta.ms());
+}
+
+TimingFrameInfo::TimingFrameInfo()
+    : rtp_timestamp(0),
+      capture_time_ms(-1),
+      encode_start_ms(-1),
+      encode_finish_ms(-1),
+      packetization_finish_ms(-1),
+      pacer_exit_ms(-1),
+      network_timestamp_ms(-1),
+      network2_timestamp_ms(-1),
+      receive_start_ms(-1),
+      receive_finish_ms(-1),
+      decode_start_ms(-1),
+      decode_finish_ms(-1),
+      render_time_ms(-1),
+      flags(VideoSendTiming::kNotTriggered) {}
+
+int64_t TimingFrameInfo::EndToEndDelay() const {
+  return capture_time_ms >= 0 ? decode_finish_ms - capture_time_ms : -1;
+}
+
+bool TimingFrameInfo::IsLongerThan(const TimingFrameInfo& other) const {
+  int64_t other_delay = other.EndToEndDelay();
+  return other_delay == -1 || EndToEndDelay() > other_delay;
+}
+
+bool TimingFrameInfo::operator<(const TimingFrameInfo& other) const {
+  return other.IsLongerThan(*this);
+}
+
+bool TimingFrameInfo::operator<=(const TimingFrameInfo& other) const {
+  return !IsLongerThan(other);
+}
+
+bool TimingFrameInfo::IsOutlier() const {
+  return !IsInvalid() && (flags & VideoSendTiming::kTriggeredBySize);
+}
+
+bool TimingFrameInfo::IsTimerTriggered() const {
+  return !IsInvalid() && (flags & VideoSendTiming::kTriggeredByTimer);
+}
+
+bool TimingFrameInfo::IsInvalid() const {
+  return flags == VideoSendTiming::kInvalid;
+}
+
+std::string TimingFrameInfo::ToString() const {
+  if (IsInvalid()) {
+    return "";
+  }
+
+  char buf[1024];
+  rtc::SimpleStringBuilder sb(buf);
+
+  sb << rtp_timestamp << ',' << capture_time_ms << ',' << encode_start_ms << ','
+     << encode_finish_ms << ',' << packetization_finish_ms << ','
+     << pacer_exit_ms << ',' << network_timestamp_ms << ','
+     << network2_timestamp_ms << ',' << receive_start_ms << ','
+     << receive_finish_ms << ',' << decode_start_ms << ',' << decode_finish_ms
+     << ',' << render_time_ms << ',' << IsOutlier() << ','
+     << IsTimerTriggered();
+
+  return sb.str();
+}
+
+VideoPlayoutDelay::VideoPlayoutDelay(TimeDelta min, TimeDelta max)
+    : min_(std::clamp(min, TimeDelta::Zero(), kMax)),
+      max_(std::clamp(max, min_, kMax)) {
+  if (!(TimeDelta::Zero() <= min && min <= max && max <= kMax)) {
+    RTC_LOG(LS_ERROR) << "Invalid video playout delay: [" << min << "," << max
+                      << "]. Clamped to [" << this->min() << "," << this->max()
+                      << "]";
+  }
+}
+
+bool VideoPlayoutDelay::Set(TimeDelta min, TimeDelta max) {
+  if (TimeDelta::Zero() <= min && min <= max && max <= kMax) {
+    min_ = min;
+    max_ = max;
+    return true;
+  }
+  return false;
+}
+
+}  // namespace webrtc

VocieProcess/api/video/video_timing.h

@@ -0,0 +1,150 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_VIDEO_VIDEO_TIMING_H_
+#define API_VIDEO_VIDEO_TIMING_H_
+
+#include <stdint.h>
+
+#include <limits>
+#include <string>
+
+#include "api/units/time_delta.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// Video timing timestamps in ms counted from capture_time_ms of a frame.
+// This structure represents data sent in video-timing RTP header extension.
+struct RTC_EXPORT VideoSendTiming {
+  enum TimingFrameFlags : uint8_t {
+    kNotTriggered = 0,  // Timing info valid, but not to be transmitted.
+                        // Used on send-side only.
+    kTriggeredByTimer = 1 << 0,  // Frame marked for tracing by periodic timer.
+    kTriggeredBySize = 1 << 1,   // Frame marked for tracing due to size.
+    kInvalid = std::numeric_limits<uint8_t>::max()  // Invalid, ignore!
+  };
+
+  // Returns |time_ms - base_ms| capped at max 16-bit value.
+  // Used to fill this data structure as per
+  // https://webrtc.org/experiments/rtp-hdrext/video-timing/ extension stores
+  // 16-bit deltas of timestamps from packet capture time.
+  static uint16_t GetDeltaCappedMs(int64_t base_ms, int64_t time_ms);
+  static uint16_t GetDeltaCappedMs(TimeDelta delta);
+
+  uint16_t encode_start_delta_ms;
+  uint16_t encode_finish_delta_ms;
+  uint16_t packetization_finish_delta_ms;
+  uint16_t pacer_exit_delta_ms;
+  uint16_t network_timestamp_delta_ms;
+  uint16_t network2_timestamp_delta_ms;
+  uint8_t flags = TimingFrameFlags::kInvalid;
+};
+
+// Used to report precise timings of a 'timing frames'. Contains all important
+// timestamps for a lifetime of that specific frame. Reported as a string via
+// GetStats(). Only frame which took the longest between two GetStats calls is
+// reported.
+struct RTC_EXPORT TimingFrameInfo {
+  TimingFrameInfo();
+
+  // Returns end-to-end delay of a frame, if sender and receiver timestamps are
+  // synchronized, -1 otherwise.
+  int64_t EndToEndDelay() const;
+
+  // Returns true if current frame took longer to process than `other` frame.
+  // If other frame's clocks are not synchronized, current frame is always
+  // preferred.
+  bool IsLongerThan(const TimingFrameInfo& other) const;
+
+  // Returns true if flags are set to indicate this frame was marked for tracing
+  // due to the size being outside some limit.
+  bool IsOutlier() const;
+
+  // Returns true if flags are set to indicate this frame was marked fro tracing
+  // due to cyclic timer.
+  bool IsTimerTriggered() const;
+
+  // Returns true if the timing data is marked as invalid, in which case it
+  // should be ignored.
+  bool IsInvalid() const;
+
+  std::string ToString() const;
+
+  bool operator<(const TimingFrameInfo& other) const;
+
+  bool operator<=(const TimingFrameInfo& other) const;
+
+  uint32_t rtp_timestamp;  // Identifier of a frame.
+  // All timestamps below are in local monotonous clock of a receiver.
+  // If sender clock is not yet estimated, sender timestamps
+  // (capture_time_ms ... pacer_exit_ms) are negative values, still
+  // relatively correct.
+  int64_t capture_time_ms;          // Captrue time of a frame.
+  int64_t encode_start_ms;          // Encode start time.
+  int64_t encode_finish_ms;         // Encode completion time.
+  int64_t packetization_finish_ms;  // Time when frame was passed to pacer.
+  int64_t pacer_exit_ms;  // Time when last packet was pushed out of pacer.
+  // Two in-network RTP processor timestamps: meaning is application specific.
+  int64_t network_timestamp_ms;
+  int64_t network2_timestamp_ms;
+  int64_t receive_start_ms;   // First received packet time.
+  int64_t receive_finish_ms;  // Last received packet time.
+  int64_t decode_start_ms;    // Decode start time.
+  int64_t decode_finish_ms;   // Decode completion time.
+  int64_t render_time_ms;     // Proposed render time to insure smooth playback.
+
+  uint8_t flags;  // Flags indicating validity and/or why tracing was triggered.
+};
+
+// Minimum and maximum playout delay values from capture to render.
+// These are best effort values.
+//
+// min = max = 0 indicates that the receiver should try and render
+// frame as soon as possible.
+//
+// min = x, max = y indicates that the receiver is free to adapt
+// in the range (x, y) based on network jitter.
+// This class ensures invariant 0 <= min <= max <= kMax.
+class RTC_EXPORT VideoPlayoutDelay {
+ public:
+  // Maximum supported value for the delay limit.
+  static constexpr TimeDelta kMax = TimeDelta::Millis(10) * 0xFFF;
+
+  // Creates delay limits that indicates receiver should try to render frame
+  // as soon as possible.
+  static VideoPlayoutDelay Minimal() {
+    return VideoPlayoutDelay(TimeDelta::Zero(), TimeDelta::Zero());
+  }
+
+  // Creates valid, but unspecified limits.
+  VideoPlayoutDelay() = default;
+  VideoPlayoutDelay(const VideoPlayoutDelay&) = default;
+  VideoPlayoutDelay& operator=(const VideoPlayoutDelay&) = default;
+  VideoPlayoutDelay(TimeDelta min, TimeDelta max);
+
+  bool Set(TimeDelta min, TimeDelta max);
+
+  TimeDelta min() const { return min_; }
+  TimeDelta max() const { return max_; }
+
+  friend bool operator==(const VideoPlayoutDelay& lhs,
+                         const VideoPlayoutDelay& rhs) {
+    return lhs.min_ == rhs.min_ && lhs.max_ == rhs.max_;
+  }
+
+ private:
+  TimeDelta min_ = TimeDelta::Zero();
+  TimeDelta max_ = kMax;
+};
+
+}  // namespace webrtc
+
+#endif  // API_VIDEO_VIDEO_TIMING_H_

VocieProcess/common_audio/audio_converter.cc

@@ -0,0 +1,219 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "common_audio/audio_converter.h"
+
+#include <cstring>
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "common_audio/channel_buffer.h"
+#include "common_audio/resampler/push_sinc_resampler.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+
+class CopyConverter : public AudioConverter {
+ public:
+  CopyConverter(size_t src_channels,
+                size_t src_frames,
+                size_t dst_channels,
+                size_t dst_frames)
+      : AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {}
+  ~CopyConverter() override {}
+
+  void Convert(const float* const* src,
+               size_t src_size,
+               float* const* dst,
+               size_t dst_capacity) override {
+    CheckSizes(src_size, dst_capacity);
+    if (src != dst) {
+      for (size_t i = 0; i < src_channels(); ++i)
+        std::memcpy(dst[i], src[i], dst_frames() * sizeof(*dst[i]));
+    }
+  }
+};
+
+class UpmixConverter : public AudioConverter {
+ public:
+  UpmixConverter(size_t src_channels,
+                 size_t src_frames,
+                 size_t dst_channels,
+                 size_t dst_frames)
+      : AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {}
+  ~UpmixConverter() override {}
+
+  void Convert(const float* const* src,
+               size_t src_size,
+               float* const* dst,
+               size_t dst_capacity) override {
+    CheckSizes(src_size, dst_capacity);
+    for (size_t i = 0; i < dst_frames(); ++i) {
+      const float value = src[0][i];
+      for (size_t j = 0; j < dst_channels(); ++j)
+        dst[j][i] = value;
+    }
+  }
+};
+
+class DownmixConverter : public AudioConverter {
+ public:
+  DownmixConverter(size_t src_channels,
+                   size_t src_frames,
+                   size_t dst_channels,
+                   size_t dst_frames)
+      : AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {}
+  ~DownmixConverter() override {}
+
+  void Convert(const float* const* src,
+               size_t src_size,
+               float* const* dst,
+               size_t dst_capacity) override {
+    CheckSizes(src_size, dst_capacity);
+    float* dst_mono = dst[0];
+    for (size_t i = 0; i < src_frames(); ++i) {
+      float sum = 0;
+      for (size_t j = 0; j < src_channels(); ++j)
+        sum += src[j][i];
+      dst_mono[i] = sum / src_channels();
+    }
+  }
+};
+
+class ResampleConverter : public AudioConverter {
+ public:
+  ResampleConverter(size_t src_channels,
+                    size_t src_frames,
+                    size_t dst_channels,
+                    size_t dst_frames)
+      : AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {
+    resamplers_.reserve(src_channels);
+    for (size_t i = 0; i < src_channels; ++i)
+      resamplers_.push_back(std::unique_ptr<PushSincResampler>(
+          new PushSincResampler(src_frames, dst_frames)));
+  }
+  ~ResampleConverter() override {}
+
+  void Convert(const float* const* src,
+               size_t src_size,
+               float* const* dst,
+               size_t dst_capacity) override {
+    CheckSizes(src_size, dst_capacity);
+    for (size_t i = 0; i < resamplers_.size(); ++i)
+      resamplers_[i]->Resample(src[i], src_frames(), dst[i], dst_frames());
+  }
+
+ private:
+  std::vector<std::unique_ptr<PushSincResampler>> resamplers_;
+};
+
+// Apply a vector of converters in serial, in the order given. At least two
+// converters must be provided.
+class CompositionConverter : public AudioConverter {
+ public:
+  explicit CompositionConverter(
+      std::vector<std::unique_ptr<AudioConverter>> converters)
+      : converters_(std::move(converters)) {
+    RTC_CHECK_GE(converters_.size(), 2);
+    // We need an intermediate buffer after every converter.
+    for (auto it = converters_.begin(); it != converters_.end() - 1; ++it)
+      buffers_.push_back(
+          std::unique_ptr<ChannelBuffer<float>>(new ChannelBuffer<float>(
+              (*it)->dst_frames(), (*it)->dst_channels())));
+  }
+  ~CompositionConverter() override {}
+
+  void Convert(const float* const* src,
+               size_t src_size,
+               float* const* dst,
+               size_t dst_capacity) override {
+    converters_.front()->Convert(src, src_size, buffers_.front()->channels(),
+                                 buffers_.front()->size());
+    for (size_t i = 2; i < converters_.size(); ++i) {
+      auto& src_buffer = buffers_[i - 2];
+      auto& dst_buffer = buffers_[i - 1];
+      converters_[i]->Convert(src_buffer->channels(), src_buffer->size(),
+                              dst_buffer->channels(), dst_buffer->size());
+    }
+    converters_.back()->Convert(buffers_.back()->channels(),
+                                buffers_.back()->size(), dst, dst_capacity);
+  }
+
+ private:
+  std::vector<std::unique_ptr<AudioConverter>> converters_;
+  std::vector<std::unique_ptr<ChannelBuffer<float>>> buffers_;
+};
+
+std::unique_ptr<AudioConverter> AudioConverter::Create(size_t src_channels,
+                                                       size_t src_frames,
+                                                       size_t dst_channels,
+                                                       size_t dst_frames) {
+  std::unique_ptr<AudioConverter> sp;
+  if (src_channels > dst_channels) {
+    if (src_frames != dst_frames) {
+      std::vector<std::unique_ptr<AudioConverter>> converters;
+      converters.push_back(std::unique_ptr<AudioConverter>(new DownmixConverter(
+          src_channels, src_frames, dst_channels, src_frames)));
+      converters.push_back(
+          std::unique_ptr<AudioConverter>(new ResampleConverter(
+              dst_channels, src_frames, dst_channels, dst_frames)));
+      sp.reset(new CompositionConverter(std::move(converters)));
+    } else {
+      sp.reset(new DownmixConverter(src_channels, src_frames, dst_channels,
+                                    dst_frames));
+    }
+  } else if (src_channels < dst_channels) {
+    if (src_frames != dst_frames) {
+      std::vector<std::unique_ptr<AudioConverter>> converters;
+      converters.push_back(
+          std::unique_ptr<AudioConverter>(new ResampleConverter(
+              src_channels, src_frames, src_channels, dst_frames)));
+      converters.push_back(std::unique_ptr<AudioConverter>(new UpmixConverter(
+          src_channels, dst_frames, dst_channels, dst_frames)));
+      sp.reset(new CompositionConverter(std::move(converters)));
+    } else {
+      sp.reset(new UpmixConverter(src_channels, src_frames, dst_channels,
+                                  dst_frames));
+    }
+  } else if (src_frames != dst_frames) {
+    sp.reset(new ResampleConverter(src_channels, src_frames, dst_channels,
+                                   dst_frames));
+  } else {
+    sp.reset(
+        new CopyConverter(src_channels, src_frames, dst_channels, dst_frames));
+  }
+
+  return sp;
+}
+
+// For CompositionConverter.
+AudioConverter::AudioConverter()
+    : src_channels_(0), src_frames_(0), dst_channels_(0), dst_frames_(0) {}
+
+AudioConverter::AudioConverter(size_t src_channels,
+                               size_t src_frames,
+                               size_t dst_channels,
+                               size_t dst_frames)
+    : src_channels_(src_channels),
+      src_frames_(src_frames),
+      dst_channels_(dst_channels),
+      dst_frames_(dst_frames) {
+  RTC_CHECK(dst_channels == src_channels || dst_channels == 1 ||
+            src_channels == 1);
+}
+
+void AudioConverter::CheckSizes(size_t src_size, size_t dst_capacity) const {
+  RTC_CHECK_EQ(src_size, src_channels() * src_frames());
+  RTC_CHECK_GE(dst_capacity, dst_channels() * dst_frames());
+}
+
+}  // namespace webrtc

VocieProcess/common_audio/audio_converter.h

@@ -0,0 +1,72 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef COMMON_AUDIO_AUDIO_CONVERTER_H_
+#define COMMON_AUDIO_AUDIO_CONVERTER_H_
+
+#include <stddef.h>
+
+#include <memory>
+
+namespace webrtc {
+
+// Format conversion (remixing and resampling) for audio. Only simple remixing
+// conversions are supported: downmix to mono (i.e. `dst_channels` == 1) or
+// upmix from mono (i.e. |src_channels == 1|).
+//
+// The source and destination chunks have the same duration in time; specifying
+// the number of frames is equivalent to specifying the sample rates.
+class AudioConverter {
+ public:
+  // Returns a new AudioConverter, which will use the supplied format for its
+  // lifetime. Caller is responsible for the memory.
+  static std::unique_ptr<AudioConverter> Create(size_t src_channels,
+                                                size_t src_frames,
+                                                size_t dst_channels,
+                                                size_t dst_frames);
+  virtual ~AudioConverter() {}
+
+  AudioConverter(const AudioConverter&) = delete;
+  AudioConverter& operator=(const AudioConverter&) = delete;
+
+  // Convert `src`, containing `src_size` samples, to `dst`, having a sample
+  // capacity of `dst_capacity`. Both point to a series of buffers containing
+  // the samples for each channel. The sizes must correspond to the format
+  // passed to Create().
+  virtual void Convert(const float* const* src,
+                       size_t src_size,
+                       float* const* dst,
+                       size_t dst_capacity) = 0;
+
+  size_t src_channels() const { return src_channels_; }
+  size_t src_frames() const { return src_frames_; }
+  size_t dst_channels() const { return dst_channels_; }
+  size_t dst_frames() const { return dst_frames_; }
+
+ protected:
+  AudioConverter();
+  AudioConverter(size_t src_channels,
+                 size_t src_frames,
+                 size_t dst_channels,
+                 size_t dst_frames);
+
+  // Helper to RTC_CHECK that inputs are correctly sized.
+  void CheckSizes(size_t src_size, size_t dst_capacity) const;
+
+ private:
+  const size_t src_channels_;
+  const size_t src_frames_;
+  const size_t dst_channels_;
+  const size_t dst_frames_;
+};
+
+}  // namespace webrtc
+
+#endif  // COMMON_AUDIO_AUDIO_CONVERTER_H_

VocieProcess/common_audio/resampler/include/push_resampler.h

@@ -0,0 +1,58 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_
+#define COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_
+
+#include <memory>
+#include <vector>
+
+#include "api/audio/audio_view.h"
+
+namespace webrtc {
+
+class PushSincResampler;
+
+// Wraps PushSincResampler to provide stereo support.
+// Note: This implementation assumes 10ms buffer sizes throughout.
+template <typename T>
+class PushResampler final {
+ public:
+  PushResampler();
+  PushResampler(size_t src_samples_per_channel,
+                size_t dst_samples_per_channel,
+                size_t num_channels);
+  ~PushResampler();
+
+  // Returns the total number of samples provided in destination (e.g. 32 kHz,
+  // 2 channel audio gives 640 samples).
+  int Resample(InterleavedView<const T> src, InterleavedView<T> dst);
+  // For when a deinterleaved/mono channel already exists and we can skip the
+  // deinterleaved operation.
+  int Resample(MonoView<const T> src, MonoView<T> dst);
+
+ private:
+  // Ensures that source and destination buffers for deinterleaving are
+  // correctly configured prior to resampling that requires deinterleaving.
+  void EnsureInitialized(size_t src_samples_per_channel,
+                         size_t dst_samples_per_channel,
+                         size_t num_channels);
+
+  // Buffers used for when a deinterleaving step is necessary.
+  std::unique_ptr<T[]> source_;
+  std::unique_ptr<T[]> destination_;
+  DeinterleavedView<T> source_view_;
+  DeinterleavedView<T> destination_view_;
+
+  std::vector<std::unique_ptr<PushSincResampler>> resamplers_;
+};
+}  // namespace webrtc
+
+#endif  // COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_

VocieProcess/common_audio/resampler/include/resampler.h

@@ -0,0 +1,99 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * A wrapper for resampling a numerous amount of sampling combinations.
+ */
+
+#ifndef COMMON_AUDIO_RESAMPLER_INCLUDE_RESAMPLER_H_
+#define COMMON_AUDIO_RESAMPLER_INCLUDE_RESAMPLER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+namespace webrtc {
+
+// All methods return 0 on success and -1 on failure.
+class Resampler {
+ public:
+  Resampler();
+  Resampler(int inFreq, int outFreq, size_t num_channels);
+  ~Resampler();
+
+  // Reset all states
+  int Reset(int inFreq, int outFreq, size_t num_channels);
+
+  // Reset all states if any parameter has changed
+  int ResetIfNeeded(int inFreq, int outFreq, size_t num_channels);
+
+  // Resample samplesIn to samplesOut.
+  int Push(const int16_t* samplesIn,
+           size_t lengthIn,
+           int16_t* samplesOut,
+           size_t maxLen,
+           size_t& outLen);  // NOLINT: to avoid changing APIs
+
+ private:
+  enum ResamplerMode {
+    kResamplerMode1To1,
+    kResamplerMode1To2,
+    kResamplerMode1To3,
+    kResamplerMode1To4,
+    kResamplerMode1To6,
+    kResamplerMode1To12,
+    kResamplerMode2To3,
+    kResamplerMode2To11,
+    kResamplerMode4To11,
+    kResamplerMode8To11,
+    kResamplerMode11To16,
+    kResamplerMode11To32,
+    kResamplerMode2To1,
+    kResamplerMode3To1,
+    kResamplerMode4To1,
+    kResamplerMode6To1,
+    kResamplerMode12To1,
+    kResamplerMode3To2,
+    kResamplerMode11To2,
+    kResamplerMode11To4,
+    kResamplerMode11To8
+  };
+
+  // Computes the resampler mode for a given sampling frequency pair.
+  // Returns -1 for unsupported frequency pairs.
+  static int ComputeResamplerMode(int in_freq_hz,
+                                  int out_freq_hz,
+                                  ResamplerMode* mode);
+
+  // Generic pointers since we don't know what states we'll need
+  void* state1_;
+  void* state2_;
+  void* state3_;
+
+  // Storage if needed
+  int16_t* in_buffer_;
+  int16_t* out_buffer_;
+  size_t in_buffer_size_;
+  size_t out_buffer_size_;
+  size_t in_buffer_size_max_;
+  size_t out_buffer_size_max_;
+
+  int my_in_frequency_khz_;
+  int my_out_frequency_khz_;
+  ResamplerMode my_mode_;
+  size_t num_channels_;
+
+  // Extra instance for stereo
+  Resampler* helper_left_;
+  Resampler* helper_right_;
+};
+
+}  // namespace webrtc
+
+#endif  // COMMON_AUDIO_RESAMPLER_INCLUDE_RESAMPLER_H_

VocieProcess/common_audio/vad/include/webrtc_vad.h

@@ -0,0 +1,87 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * This header file includes the VAD API calls. Specific function calls are
+ * given below.
+ */
+
+#ifndef COMMON_AUDIO_VAD_INCLUDE_WEBRTC_VAD_H_  // NOLINT
+#define COMMON_AUDIO_VAD_INCLUDE_WEBRTC_VAD_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+typedef struct WebRtcVadInst VadInst;
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+// Creates an instance to the VAD structure.
+VadInst* WebRtcVad_Create(void);
+
+// Frees the dynamic memory of a specified VAD instance.
+//
+// - handle [i] : Pointer to VAD instance that should be freed.
+void WebRtcVad_Free(VadInst* handle);
+
+// Initializes a VAD instance.
+//
+// - handle [i/o] : Instance that should be initialized.
+//
+// returns        : 0 - (OK),
+//                 -1 - (null pointer or Default mode could not be set).
+int WebRtcVad_Init(VadInst* handle);
+
+// Sets the VAD operating mode. A more aggressive (higher mode) VAD is more
+// restrictive in reporting speech. Put in other words the probability of being
+// speech when the VAD returns 1 is increased with increasing mode. As a
+// consequence also the missed detection rate goes up.
+//
+// - handle [i/o] : VAD instance.
+// - mode   [i]   : Aggressiveness mode (0, 1, 2, or 3).
+//
+// returns        : 0 - (OK),
+//                 -1 - (null pointer, mode could not be set or the VAD instance
+//                       has not been initialized).
+int WebRtcVad_set_mode(VadInst* handle, int mode);
+
+// Calculates a VAD decision for the `audio_frame`. For valid sampling rates
+// frame lengths, see the description of WebRtcVad_ValidRatesAndFrameLengths().
+//
+// - handle       [i/o] : VAD Instance. Needs to be initialized by
+//                        WebRtcVad_Init() before call.
+// - fs           [i]   : Sampling frequency (Hz): 8000, 16000, or 32000
+// - audio_frame  [i]   : Audio frame buffer.
+// - frame_length [i]   : Length of audio frame buffer in number of samples.
+//
+// returns              : 1 - (Active Voice),
+//                        0 - (Non-active Voice),
+//                       -1 - (Error)
+int WebRtcVad_Process(VadInst* handle,
+                      int fs,
+                      const int16_t* audio_frame,
+                      size_t frame_length);
+
+// Checks for valid combinations of `rate` and `frame_length`. We support 10,
+// 20 and 30 ms frames and the rates 8000, 16000 and 32000 Hz.
+//
+// - rate         [i] : Sampling frequency (Hz).
+// - frame_length [i] : Speech frame buffer length in number of samples.
+//
+// returns            : 0 - (valid combination), -1 - (invalid combination)
+int WebRtcVad_ValidRateAndFrameLength(int rate, size_t frame_length);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif  // COMMON_AUDIO_VAD_INCLUDE_WEBRTC_VAD_H_  // NOLINT

VocieProcess/modules/audio_coding/codecs/isac/bandwidth_info.h

@@ -0,0 +1,24 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_BANDWIDTH_INFO_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_BANDWIDTH_INFO_H_
+
+#include <stdint.h>
+
+typedef struct {
+  int in_use;
+  int32_t send_bw_avg;
+  int32_t send_max_delay_avg;
+  int16_t bottleneck_idx;
+  int16_t jitter_info;
+} IsacBandwidthInfo;
+
+#endif  // MODULES_AUDIO_CODING_CODECS_ISAC_BANDWIDTH_INFO_H_

VocieProcess/modules/audio_coding/codecs/isac/main/source/filter_functions.c

@@ -0,0 +1,195 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory.h>
+#include <string.h>
+#ifdef WEBRTC_ANDROID
+#include <stdlib.h>
+#endif
+
+#include "modules/audio_coding/codecs/isac/main/source/pitch_estimator.h"
+#include "modules/audio_coding/codecs/isac/main/source/isac_vad.h"
+
+static void WebRtcIsac_AllPoleFilter(double* InOut,
+                                     double* Coef,
+                                     size_t lengthInOut,
+                                     int orderCoef) {
+  /* the state of filter is assumed to be in InOut[-1] to InOut[-orderCoef] */
+  double scal;
+  double sum;
+  size_t n;
+  int k;
+
+  //if (fabs(Coef[0]-1.0)<0.001) {
+  if ( (Coef[0] > 0.9999) && (Coef[0] < 1.0001) )
+  {
+    for(n = 0; n < lengthInOut; n++)
+    {
+      sum = Coef[1] * InOut[-1];
+      for(k = 2; k <= orderCoef; k++){
+        sum += Coef[k] * InOut[-k];
+      }
+      *InOut++ -= sum;
+    }
+  }
+  else
+  {
+    scal = 1.0 / Coef[0];
+    for(n=0;n<lengthInOut;n++)
+    {
+      *InOut *= scal;
+      for(k=1;k<=orderCoef;k++){
+        *InOut -= scal*Coef[k]*InOut[-k];
+      }
+      InOut++;
+    }
+  }
+}
+
+static void WebRtcIsac_AllZeroFilter(double* In,
+                                     double* Coef,
+                                     size_t lengthInOut,
+                                     int orderCoef,
+                                     double* Out) {
+  /* the state of filter is assumed to be in In[-1] to In[-orderCoef] */
+
+  size_t n;
+  int k;
+  double tmp;
+
+  for(n = 0; n < lengthInOut; n++)
+  {
+    tmp = In[0] * Coef[0];
+
+    for(k = 1; k <= orderCoef; k++){
+      tmp += Coef[k] * In[-k];
+    }
+
+    *Out++ = tmp;
+    In++;
+  }
+}
+
+static void WebRtcIsac_ZeroPoleFilter(double* In,
+                                      double* ZeroCoef,
+                                      double* PoleCoef,
+                                      size_t lengthInOut,
+                                      int orderCoef,
+                                      double* Out) {
+  /* the state of the zero section is assumed to be in In[-1] to In[-orderCoef] */
+  /* the state of the pole section is assumed to be in Out[-1] to Out[-orderCoef] */
+
+  WebRtcIsac_AllZeroFilter(In,ZeroCoef,lengthInOut,orderCoef,Out);
+  WebRtcIsac_AllPoleFilter(Out,PoleCoef,lengthInOut,orderCoef);
+}
+
+
+void WebRtcIsac_AutoCorr(double* r, const double* x, size_t N, size_t order) {
+  size_t  lag, n;
+  double sum, prod;
+  const double *x_lag;
+
+  for (lag = 0; lag <= order; lag++)
+  {
+    sum = 0.0f;
+    x_lag = &x[lag];
+    prod = x[0] * x_lag[0];
+    for (n = 1; n < N - lag; n++) {
+      sum += prod;
+      prod = x[n] * x_lag[n];
+    }
+    sum += prod;
+    r[lag] = sum;
+  }
+
+}
+
+static void WebRtcIsac_BwExpand(double* out,
+                                double* in,
+                                double coef,
+                                size_t length) {
+  size_t i;
+  double  chirp;
+
+  chirp = coef;
+
+  out[0] = in[0];
+  for (i = 1; i < length; i++) {
+    out[i] = chirp * in[i];
+    chirp *= coef;
+  }
+}
+
+void WebRtcIsac_WeightingFilter(const double* in,
+                                double* weiout,
+                                double* whiout,
+                                WeightFiltstr* wfdata) {
+  double  tmpbuffer[PITCH_FRAME_LEN + PITCH_WLPCBUFLEN];
+  double  corr[PITCH_WLPCORDER+1], rc[PITCH_WLPCORDER+1];
+  double apol[PITCH_WLPCORDER+1], apolr[PITCH_WLPCORDER+1];
+  double  rho=0.9, *inp, *dp, *dp2;
+  double  whoutbuf[PITCH_WLPCBUFLEN + PITCH_WLPCORDER];
+  double  weoutbuf[PITCH_WLPCBUFLEN + PITCH_WLPCORDER];
+  double  *weo, *who, opol[PITCH_WLPCORDER+1], ext[PITCH_WLPCWINLEN];
+  int     k, n, endpos, start;
+
+  /* Set up buffer and states */
+  memcpy(tmpbuffer, wfdata->buffer, sizeof(double) * PITCH_WLPCBUFLEN);
+  memcpy(tmpbuffer+PITCH_WLPCBUFLEN, in, sizeof(double) * PITCH_FRAME_LEN);
+  memcpy(wfdata->buffer, tmpbuffer+PITCH_FRAME_LEN, sizeof(double) * PITCH_WLPCBUFLEN);
+
+  dp=weoutbuf;
+  dp2=whoutbuf;
+  for (k=0;k<PITCH_WLPCORDER;k++) {
+    *dp++ = wfdata->weostate[k];
+    *dp2++ = wfdata->whostate[k];
+    opol[k]=0.0;
+  }
+  opol[0]=1.0;
+  opol[PITCH_WLPCORDER]=0.0;
+  weo=dp;
+  who=dp2;
+
+  endpos=PITCH_WLPCBUFLEN + PITCH_SUBFRAME_LEN;
+  inp=tmpbuffer + PITCH_WLPCBUFLEN;
+
+  for (n=0; n<PITCH_SUBFRAMES; n++) {
+    /* Windowing */
+    start=endpos-PITCH_WLPCWINLEN;
+    for (k=0; k<PITCH_WLPCWINLEN; k++) {
+      ext[k]=wfdata->window[k]*tmpbuffer[start+k];
+    }
+
+    /* Get LPC polynomial */
+    WebRtcIsac_AutoCorr(corr, ext, PITCH_WLPCWINLEN, PITCH_WLPCORDER);
+    corr[0]=1.01*corr[0]+1.0; /* White noise correction */
+    WebRtcIsac_LevDurb(apol, rc, corr, PITCH_WLPCORDER);
+    WebRtcIsac_BwExpand(apolr, apol, rho, PITCH_WLPCORDER+1);
+
+    /* Filtering */
+    WebRtcIsac_ZeroPoleFilter(inp, apol, apolr, PITCH_SUBFRAME_LEN, PITCH_WLPCORDER, weo);
+    WebRtcIsac_ZeroPoleFilter(inp, apolr, opol, PITCH_SUBFRAME_LEN, PITCH_WLPCORDER, who);
+
+    inp+=PITCH_SUBFRAME_LEN;
+    endpos+=PITCH_SUBFRAME_LEN;
+    weo+=PITCH_SUBFRAME_LEN;
+    who+=PITCH_SUBFRAME_LEN;
+  }
+
+  /* Export filter states */
+  for (k=0;k<PITCH_WLPCORDER;k++) {
+    wfdata->weostate[k]=weoutbuf[PITCH_FRAME_LEN+k];
+    wfdata->whostate[k]=whoutbuf[PITCH_FRAME_LEN+k];
+  }
+
+  /* Export output data */
+  memcpy(weiout, weoutbuf+PITCH_WLPCORDER, sizeof(double) * PITCH_FRAME_LEN);
+  memcpy(whiout, whoutbuf+PITCH_WLPCORDER, sizeof(double) * PITCH_FRAME_LEN);
+}

VocieProcess/modules/audio_coding/codecs/isac/main/source/filter_functions.h

@@ -0,0 +1,25 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_FILTER_FUNCTIONS_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_FILTER_FUNCTIONS_H_
+
+#include <stddef.h>
+
+#include "modules/audio_coding/codecs/isac/main/source/structs.h"
+
+void WebRtcIsac_AutoCorr(double* r, const double* x, size_t N, size_t order);
+
+void WebRtcIsac_WeightingFilter(const double* in,
+                                double* weiout,
+                                double* whiout,
+                                WeightFiltstr* wfdata);
+
+#endif  // MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_FILTER_FUNCTIONS_H_

VocieProcess/modules/audio_coding/codecs/isac/main/source/isac_vad.c

@@ -0,0 +1,409 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/isac/main/source/isac_vad.h"
+
+#include <math.h>
+
+void WebRtcIsac_InitPitchFilter(PitchFiltstr* pitchfiltdata) {
+  int k;
+
+  for (k = 0; k < PITCH_BUFFSIZE; k++) {
+    pitchfiltdata->ubuf[k] = 0.0;
+  }
+  pitchfiltdata->ystate[0] = 0.0;
+  for (k = 1; k < (PITCH_DAMPORDER); k++) {
+    pitchfiltdata->ystate[k] = 0.0;
+  }
+  pitchfiltdata->oldlagp[0] = 50.0;
+  pitchfiltdata->oldgainp[0] = 0.0;
+}
+
+static void WebRtcIsac_InitWeightingFilter(WeightFiltstr* wfdata) {
+  int k;
+  double t, dtmp, dtmp2, denum, denum2;
+
+  for (k = 0; k < PITCH_WLPCBUFLEN; k++)
+    wfdata->buffer[k] = 0.0;
+
+  for (k = 0; k < PITCH_WLPCORDER; k++) {
+    wfdata->istate[k] = 0.0;
+    wfdata->weostate[k] = 0.0;
+    wfdata->whostate[k] = 0.0;
+  }
+
+  /* next part should be in Matlab, writing to a global table */
+  t = 0.5;
+  denum = 1.0 / ((double)PITCH_WLPCWINLEN);
+  denum2 = denum * denum;
+  for (k = 0; k < PITCH_WLPCWINLEN; k++) {
+    dtmp = PITCH_WLPCASYM * t * denum + (1 - PITCH_WLPCASYM) * t * t * denum2;
+    dtmp *= 3.14159265;
+    dtmp2 = sin(dtmp);
+    wfdata->window[k] = dtmp2 * dtmp2;
+    t++;
+  }
+}
+
+void WebRtcIsac_InitPitchAnalysis(PitchAnalysisStruct* State) {
+  int k;
+
+  for (k = 0; k < PITCH_CORR_LEN2 + PITCH_CORR_STEP2 + PITCH_MAX_LAG / 2 -
+                      PITCH_FRAME_LEN / 2 + 2;
+       k++)
+    State->dec_buffer[k] = 0.0;
+  for (k = 0; k < 2 * ALLPASSSECTIONS + 1; k++)
+    State->decimator_state[k] = 0.0;
+  for (k = 0; k < 2; k++)
+    State->hp_state[k] = 0.0;
+  for (k = 0; k < QLOOKAHEAD; k++)
+    State->whitened_buf[k] = 0.0;
+  for (k = 0; k < QLOOKAHEAD; k++)
+    State->inbuf[k] = 0.0;
+
+  WebRtcIsac_InitPitchFilter(&(State->PFstr_wght));
+
+  WebRtcIsac_InitPitchFilter(&(State->PFstr));
+
+  WebRtcIsac_InitWeightingFilter(&(State->Wghtstr));
+}
+
+void WebRtcIsac_InitPreFilterbank(PreFiltBankstr* prefiltdata) {
+  int k;
+
+  for (k = 0; k < QLOOKAHEAD; k++) {
+    prefiltdata->INLABUF1[k] = 0;
+    prefiltdata->INLABUF2[k] = 0;
+
+    prefiltdata->INLABUF1_float[k] = 0;
+    prefiltdata->INLABUF2_float[k] = 0;
+  }
+  for (k = 0; k < 2 * (QORDER - 1); k++) {
+    prefiltdata->INSTAT1[k] = 0;
+    prefiltdata->INSTAT2[k] = 0;
+    prefiltdata->INSTATLA1[k] = 0;
+    prefiltdata->INSTATLA2[k] = 0;
+
+    prefiltdata->INSTAT1_float[k] = 0;
+    prefiltdata->INSTAT2_float[k] = 0;
+    prefiltdata->INSTATLA1_float[k] = 0;
+    prefiltdata->INSTATLA2_float[k] = 0;
+  }
+
+  /* High pass filter states */
+  prefiltdata->HPstates[0] = 0.0;
+  prefiltdata->HPstates[1] = 0.0;
+
+  prefiltdata->HPstates_float[0] = 0.0f;
+  prefiltdata->HPstates_float[1] = 0.0f;
+
+  return;
+}
+
+double WebRtcIsac_LevDurb(double* a, double* k, double* r, size_t order) {
+  const double LEVINSON_EPS = 1.0e-10;
+
+  double sum, alpha;
+  size_t m, m_h, i;
+  alpha = 0;  // warning -DH
+  a[0] = 1.0;
+  if (r[0] < LEVINSON_EPS) { /* if r[0] <= 0, set LPC coeff. to zero */
+    for (i = 0; i < order; i++) {
+      k[i] = 0;
+      a[i + 1] = 0;
+    }
+  } else {
+    a[1] = k[0] = -r[1] / r[0];
+    alpha = r[0] + r[1] * k[0];
+    for (m = 1; m < order; m++) {
+      sum = r[m + 1];
+      for (i = 0; i < m; i++) {
+        sum += a[i + 1] * r[m - i];
+      }
+      k[m] = -sum / alpha;
+      alpha += k[m] * sum;
+      m_h = (m + 1) >> 1;
+      for (i = 0; i < m_h; i++) {
+        sum = a[i + 1] + k[m] * a[m - i];
+        a[m - i] += k[m] * a[i + 1];
+        a[i + 1] = sum;
+      }
+      a[m + 1] = k[m];
+    }
+  }
+  return alpha;
+}
+
+/* The upper channel all-pass filter factors */
+const float WebRtcIsac_kUpperApFactorsFloat[2] = {0.03470000000000f,
+                                                  0.38260000000000f};
+
+/* The lower channel all-pass filter factors */
+const float WebRtcIsac_kLowerApFactorsFloat[2] = {0.15440000000000f,
+                                                  0.74400000000000f};
+
+/* This function performs all-pass filtering--a series of first order all-pass
+ * sections are used to filter the input in a cascade manner.
+ * The input is overwritten!!
+ */
+void WebRtcIsac_AllPassFilter2Float(float* InOut,
+                                    const float* APSectionFactors,
+                                    int lengthInOut,
+                                    int NumberOfSections,
+                                    float* FilterState) {
+  int n, j;
+  float temp;
+  for (j = 0; j < NumberOfSections; j++) {
+    for (n = 0; n < lengthInOut; n++) {
+      temp = FilterState[j] + APSectionFactors[j] * InOut[n];
+      FilterState[j] = -APSectionFactors[j] * temp + InOut[n];
+      InOut[n] = temp;
+    }
+  }
+}
+
+/* The number of composite all-pass filter factors */
+#define NUMBEROFCOMPOSITEAPSECTIONS 4
+
+/* Function WebRtcIsac_SplitAndFilter
+ * This function creates low-pass and high-pass decimated versions of part of
+ the input signal, and part of the signal in the input 'lookahead buffer'.
+
+ INPUTS:
+ in: a length FRAMESAMPLES array of input samples
+ prefiltdata: input data structure containing the filterbank states
+ and lookahead samples from the previous encoding
+ iteration.
+ OUTPUTS:
+ LP: a FRAMESAMPLES_HALF array of low-pass filtered samples that
+ have been phase equalized.  The first QLOOKAHEAD samples are
+ based on the samples in the two prefiltdata->INLABUFx arrays
+ each of length QLOOKAHEAD.
+ The remaining FRAMESAMPLES_HALF-QLOOKAHEAD samples are based
+ on the first FRAMESAMPLES_HALF-QLOOKAHEAD samples of the input
+ array in[].
+ HP: a FRAMESAMPLES_HALF array of high-pass filtered samples that
+ have been phase equalized.  The first QLOOKAHEAD samples are
+ based on the samples in the two prefiltdata->INLABUFx arrays
+ each of length QLOOKAHEAD.
+ The remaining FRAMESAMPLES_HALF-QLOOKAHEAD samples are based
+ on the first FRAMESAMPLES_HALF-QLOOKAHEAD samples of the input
+ array in[].
+
+ LP_la: a FRAMESAMPLES_HALF array of low-pass filtered samples.
+ These samples are not phase equalized. They are computed
+ from the samples in the in[] array.
+ HP_la: a FRAMESAMPLES_HALF array of high-pass filtered samples
+ that are not phase equalized. They are computed from
+ the in[] vector.
+ prefiltdata: this input data structure's filterbank state and
+ lookahead sample buffers are updated for the next
+ encoding iteration.
+*/
+void WebRtcIsac_SplitAndFilterFloat(float* pin,
+                                    float* LP,
+                                    float* HP,
+                                    double* LP_la,
+                                    double* HP_la,
+                                    PreFiltBankstr* prefiltdata) {
+  int k, n;
+  float CompositeAPFilterState[NUMBEROFCOMPOSITEAPSECTIONS];
+  float ForTransform_CompositeAPFilterState[NUMBEROFCOMPOSITEAPSECTIONS];
+  float ForTransform_CompositeAPFilterState2[NUMBEROFCOMPOSITEAPSECTIONS];
+  float tempinoutvec[FRAMESAMPLES + MAX_AR_MODEL_ORDER];
+  float tempin_ch1[FRAMESAMPLES + MAX_AR_MODEL_ORDER];
+  float tempin_ch2[FRAMESAMPLES + MAX_AR_MODEL_ORDER];
+  float in[FRAMESAMPLES];
+  float ftmp;
+
+  /* HPstcoeff_in = {a1, a2, b1 - b0 * a1, b2 - b0 * a2}; */
+  static const float kHpStCoefInFloat[4] = {
+      -1.94895953203325f, 0.94984516000000f, -0.05101826139794f,
+      0.05015484000000f};
+
+  /* The composite all-pass filter factors */
+  static const float WebRtcIsac_kCompositeApFactorsFloat[4] = {
+      0.03470000000000f, 0.15440000000000f, 0.38260000000000f,
+      0.74400000000000f};
+
+  // The matrix for transforming the backward composite state to upper channel
+  // state.
+  static const float WebRtcIsac_kTransform1Float[8] = {
+      -0.00158678506084f, 0.00127157815343f, -0.00104805672709f,
+      0.00084837248079f,  0.00134467983258f, -0.00107756549387f,
+      0.00088814793277f,  -0.00071893072525f};
+
+  // The matrix for transforming the backward composite state to lower channel
+  // state.
+  static const float WebRtcIsac_kTransform2Float[8] = {
+      -0.00170686041697f, 0.00136780109829f, -0.00112736532350f,
+      0.00091257055385f,  0.00103094281812f, -0.00082615076557f,
+      0.00068092756088f,  -0.00055119165484f};
+
+  /* High pass filter */
+
+  for (k = 0; k < FRAMESAMPLES; k++) {
+    in[k] = pin[k] + kHpStCoefInFloat[2] * prefiltdata->HPstates_float[0] +
+            kHpStCoefInFloat[3] * prefiltdata->HPstates_float[1];
+    ftmp = pin[k] - kHpStCoefInFloat[0] * prefiltdata->HPstates_float[0] -
+           kHpStCoefInFloat[1] * prefiltdata->HPstates_float[1];
+    prefiltdata->HPstates_float[1] = prefiltdata->HPstates_float[0];
+    prefiltdata->HPstates_float[0] = ftmp;
+  }
+
+  /* First Channel */
+
+  /*initial state of composite filter is zero */
+  for (k = 0; k < NUMBEROFCOMPOSITEAPSECTIONS; k++) {
+    CompositeAPFilterState[k] = 0.0;
+  }
+  /* put every other sample of input into a temporary vector in reverse
+   * (backward) order*/
+  for (k = 0; k < FRAMESAMPLES_HALF; k++) {
+    tempinoutvec[k] = in[FRAMESAMPLES - 1 - 2 * k];
+  }
+
+  /* now all-pass filter the backwards vector.  Output values overwrite the
+   * input vector. */
+  WebRtcIsac_AllPassFilter2Float(
+      tempinoutvec, WebRtcIsac_kCompositeApFactorsFloat, FRAMESAMPLES_HALF,
+      NUMBEROFCOMPOSITEAPSECTIONS, CompositeAPFilterState);
+
+  /* save the backwards filtered output for later forward filtering,
+     but write it in forward order*/
+  for (k = 0; k < FRAMESAMPLES_HALF; k++) {
+    tempin_ch1[FRAMESAMPLES_HALF + QLOOKAHEAD - 1 - k] = tempinoutvec[k];
+  }
+
+  /* save the backwards filter state  becaue it will be transformed
+     later into a forward state */
+  for (k = 0; k < NUMBEROFCOMPOSITEAPSECTIONS; k++) {
+    ForTransform_CompositeAPFilterState[k] = CompositeAPFilterState[k];
+  }
+
+  /* now backwards filter the samples in the lookahead buffer. The samples were
+     placed there in the encoding of the previous frame.  The output samples
+     overwrite the input samples */
+  WebRtcIsac_AllPassFilter2Float(
+      prefiltdata->INLABUF1_float, WebRtcIsac_kCompositeApFactorsFloat,
+      QLOOKAHEAD, NUMBEROFCOMPOSITEAPSECTIONS, CompositeAPFilterState);
+
+  /* save the output, but write it in forward order */
+  /* write the lookahead samples for the next encoding iteration. Every other
+     sample at the end of the input frame is written in reverse order for the
+     lookahead length. Exported in the prefiltdata structure. */
+  for (k = 0; k < QLOOKAHEAD; k++) {
+    tempin_ch1[QLOOKAHEAD - 1 - k] = prefiltdata->INLABUF1_float[k];
+    prefiltdata->INLABUF1_float[k] = in[FRAMESAMPLES - 1 - 2 * k];
+  }
+
+  /* Second Channel.  This is exactly like the first channel, except that the
+     even samples are now filtered instead (lower channel). */
+  for (k = 0; k < NUMBEROFCOMPOSITEAPSECTIONS; k++) {
+    CompositeAPFilterState[k] = 0.0;
+  }
+
+  for (k = 0; k < FRAMESAMPLES_HALF; k++) {
+    tempinoutvec[k] = in[FRAMESAMPLES - 2 - 2 * k];
+  }
+
+  WebRtcIsac_AllPassFilter2Float(
+      tempinoutvec, WebRtcIsac_kCompositeApFactorsFloat, FRAMESAMPLES_HALF,
+      NUMBEROFCOMPOSITEAPSECTIONS, CompositeAPFilterState);
+
+  for (k = 0; k < FRAMESAMPLES_HALF; k++) {
+    tempin_ch2[FRAMESAMPLES_HALF + QLOOKAHEAD - 1 - k] = tempinoutvec[k];
+  }
+
+  for (k = 0; k < NUMBEROFCOMPOSITEAPSECTIONS; k++) {
+    ForTransform_CompositeAPFilterState2[k] = CompositeAPFilterState[k];
+  }
+
+  WebRtcIsac_AllPassFilter2Float(
+      prefiltdata->INLABUF2_float, WebRtcIsac_kCompositeApFactorsFloat,
+      QLOOKAHEAD, NUMBEROFCOMPOSITEAPSECTIONS, CompositeAPFilterState);
+
+  for (k = 0; k < QLOOKAHEAD; k++) {
+    tempin_ch2[QLOOKAHEAD - 1 - k] = prefiltdata->INLABUF2_float[k];
+    prefiltdata->INLABUF2_float[k] = in[FRAMESAMPLES - 2 - 2 * k];
+  }
+
+  /* Transform filter states from backward to forward */
+  /*At this point, each of the states of the backwards composite filters for the
+    two channels are transformed into forward filtering states for the
+    corresponding forward channel filters.  Each channel's forward filtering
+    state from the previous
+    encoding iteration is added to the transformed state to get a proper forward
+    state */
+
+  /* So the existing NUMBEROFCOMPOSITEAPSECTIONS x 1 (4x1) state vector is
+     multiplied by a NUMBEROFCHANNELAPSECTIONSxNUMBEROFCOMPOSITEAPSECTIONS (2x4)
+     transform matrix to get the new state that is added to the previous 2x1
+     input state */
+
+  for (k = 0; k < NUMBEROFCHANNELAPSECTIONS; k++) { /* k is row variable */
+    for (n = 0; n < NUMBEROFCOMPOSITEAPSECTIONS;
+         n++) { /* n is column variable */
+      prefiltdata->INSTAT1_float[k] +=
+          ForTransform_CompositeAPFilterState[n] *
+          WebRtcIsac_kTransform1Float[k * NUMBEROFCHANNELAPSECTIONS + n];
+      prefiltdata->INSTAT2_float[k] +=
+          ForTransform_CompositeAPFilterState2[n] *
+          WebRtcIsac_kTransform2Float[k * NUMBEROFCHANNELAPSECTIONS + n];
+    }
+  }
+
+  /*obtain polyphase components by forward all-pass filtering through each
+   * channel */
+  /* the backward filtered samples are now forward filtered with the
+   * corresponding channel filters */
+  /* The all pass filtering automatically updates the filter states which are
+     exported in the prefiltdata structure */
+  WebRtcIsac_AllPassFilter2Float(tempin_ch1, WebRtcIsac_kUpperApFactorsFloat,
+                                 FRAMESAMPLES_HALF, NUMBEROFCHANNELAPSECTIONS,
+                                 prefiltdata->INSTAT1_float);
+  WebRtcIsac_AllPassFilter2Float(tempin_ch2, WebRtcIsac_kLowerApFactorsFloat,
+                                 FRAMESAMPLES_HALF, NUMBEROFCHANNELAPSECTIONS,
+                                 prefiltdata->INSTAT2_float);
+
+  /* Now Construct low-pass and high-pass signals as combinations of polyphase
+   * components */
+  for (k = 0; k < FRAMESAMPLES_HALF; k++) {
+    LP[k] = 0.5f * (tempin_ch1[k] + tempin_ch2[k]); /* low pass signal*/
+    HP[k] = 0.5f * (tempin_ch1[k] - tempin_ch2[k]); /* high pass signal*/
+  }
+
+  /* Lookahead LP and HP signals */
+  /* now create low pass and high pass signals of the input vector.  However, no
+     backwards filtering is performed, and hence no phase equalization is
+     involved. Also, the input contains some samples that are lookahead samples.
+     The high pass and low pass signals that are created are used outside this
+     function for analysis (not encoding) purposes */
+
+  /* set up input */
+  for (k = 0; k < FRAMESAMPLES_HALF; k++) {
+    tempin_ch1[k] = in[2 * k + 1];
+    tempin_ch2[k] = in[2 * k];
+  }
+
+  /* the input filter states are passed in and updated by the all-pass filtering
+     routine and exported in the prefiltdata structure*/
+  WebRtcIsac_AllPassFilter2Float(tempin_ch1, WebRtcIsac_kUpperApFactorsFloat,
+                                 FRAMESAMPLES_HALF, NUMBEROFCHANNELAPSECTIONS,
+                                 prefiltdata->INSTATLA1_float);
+  WebRtcIsac_AllPassFilter2Float(tempin_ch2, WebRtcIsac_kLowerApFactorsFloat,
+                                 FRAMESAMPLES_HALF, NUMBEROFCHANNELAPSECTIONS,
+                                 prefiltdata->INSTATLA2_float);
+
+  for (k = 0; k < FRAMESAMPLES_HALF; k++) {
+    LP_la[k] = (float)(0.5f * (tempin_ch1[k] + tempin_ch2[k]));  /*low pass */
+    HP_la[k] = (double)(0.5f * (tempin_ch1[k] - tempin_ch2[k])); /* high pass */
+  }
+}

VocieProcess/modules/audio_coding/codecs/isac/main/source/isac_vad.h

@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ISAC_VAD_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ISAC_VAD_H_
+
+#include <stddef.h>
+
+#include "modules/audio_coding/codecs/isac/main/source/structs.h"
+
+void WebRtcIsac_InitPitchFilter(PitchFiltstr* pitchfiltdata);
+void WebRtcIsac_InitPitchAnalysis(PitchAnalysisStruct* state);
+void WebRtcIsac_InitPreFilterbank(PreFiltBankstr* prefiltdata);
+
+double WebRtcIsac_LevDurb(double* a, double* k, double* r, size_t order);
+
+/* The number of all-pass filter factors in an upper or lower channel*/
+#define NUMBEROFCHANNELAPSECTIONS 2
+
+/* The upper channel all-pass filter factors */
+extern const float WebRtcIsac_kUpperApFactorsFloat[2];
+
+/* The lower channel all-pass filter factors */
+extern const float WebRtcIsac_kLowerApFactorsFloat[2];
+
+void WebRtcIsac_AllPassFilter2Float(float* InOut,
+                                    const float* APSectionFactors,
+                                    int lengthInOut,
+                                    int NumberOfSections,
+                                    float* FilterState);
+void WebRtcIsac_SplitAndFilterFloat(float* in,
+                                    float* LP,
+                                    float* HP,
+                                    double* LP_la,
+                                    double* HP_la,
+                                    PreFiltBankstr* prefiltdata);
+
+#endif  // MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ISAC_VAD_H_

VocieProcess/modules/audio_coding/codecs/isac/main/source/os_specific_inline.h

@@ -0,0 +1,42 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_OS_SPECIFIC_INLINE_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_OS_SPECIFIC_INLINE_H_
+
+#include <math.h>
+
+#include "rtc_base/system/arch.h"
+
+#if defined(WEBRTC_POSIX)
+#define WebRtcIsac_lrint lrint
+#elif (defined(WEBRTC_ARCH_X86) && defined(WIN32))
+static __inline long int WebRtcIsac_lrint(double x_dbl) {
+  long int x_int;
+
+  __asm {
+    fld x_dbl
+    fistp x_int
+  }
+  ;
+
+  return x_int;
+}
+#else  // Do a slow but correct implementation of lrint
+
+static __inline long int WebRtcIsac_lrint(double x_dbl) {
+  long int x_int;
+  x_int = (long int)floor(x_dbl + 0.499999999999);
+  return x_int;
+}
+
+#endif
+
+#endif  // MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_OS_SPECIFIC_INLINE_H_

VocieProcess/modules/audio_coding/codecs/isac/main/source/pitch_estimator.c

@@ -0,0 +1,695 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/isac/main/source/pitch_estimator.h"
+
+#include <math.h>
+#include <memory.h>
+#include <string.h>
+#ifdef WEBRTC_ANDROID
+#include <stdlib.h>
+#endif
+
+#include "modules/audio_coding/codecs/isac/main/source/filter_functions.h"
+#include "modules/audio_coding/codecs/isac/main/source/pitch_filter.h"
+#include "rtc_base/system/ignore_warnings.h"
+
+static const double kInterpolWin[8] = {-0.00067556028640,  0.02184247643159, -0.12203175715679,  0.60086484101160,
+                                       0.60086484101160, -0.12203175715679,  0.02184247643159, -0.00067556028640};
+
+/* interpolation filter */
+__inline static void IntrepolFilter(double *data_ptr, double *intrp)
+{
+  *intrp = kInterpolWin[0] * data_ptr[-3];
+  *intrp += kInterpolWin[1] * data_ptr[-2];
+  *intrp += kInterpolWin[2] * data_ptr[-1];
+  *intrp += kInterpolWin[3] * data_ptr[0];
+  *intrp += kInterpolWin[4] * data_ptr[1];
+  *intrp += kInterpolWin[5] * data_ptr[2];
+  *intrp += kInterpolWin[6] * data_ptr[3];
+  *intrp += kInterpolWin[7] * data_ptr[4];
+}
+
+
+/* 2D parabolic interpolation */
+/* probably some 0.5 factors can be eliminated, and the square-roots can be removed from the Cholesky fact. */
+__inline static void Intrpol2D(double T[3][3], double *x, double *y, double *peak_val)
+{
+  double c, b[2], A[2][2];
+  double t1, t2, d;
+  double delta1, delta2;
+
+
+  // double T[3][3] = {{-1.25, -.25,-.25}, {-.25, .75, .75}, {-.25, .75, .75}};
+  // should result in: delta1 = 0.5;  delta2 = 0.0;  peak_val = 1.0
+
+  c = T[1][1];
+  b[0] = 0.5 * (T[1][2] + T[2][1] - T[0][1] - T[1][0]);
+  b[1] = 0.5 * (T[1][0] + T[2][1] - T[0][1] - T[1][2]);
+  A[0][1] = -0.5 * (T[0][1] + T[2][1] - T[1][0] - T[1][2]);
+  t1 = 0.5 * (T[0][0] + T[2][2]) - c;
+  t2 = 0.5 * (T[2][0] + T[0][2]) - c;
+  d = (T[0][1] + T[1][2] + T[1][0] + T[2][1]) - 4.0 * c - t1 - t2;
+  A[0][0] = -t1 - 0.5 * d;
+  A[1][1] = -t2 - 0.5 * d;
+
+  /* deal with singularities or ill-conditioned cases */
+  if ( (A[0][0] < 1e-7) || ((A[0][0] * A[1][1] - A[0][1] * A[0][1]) < 1e-7) ) {
+    *peak_val = T[1][1];
+    return;
+  }
+
+  /* Cholesky decomposition: replace A by upper-triangular factor */
+  A[0][0] = sqrt(A[0][0]);
+  A[0][1] = A[0][1] / A[0][0];
+  A[1][1] = sqrt(A[1][1] - A[0][1] * A[0][1]);
+
+  /* compute [x; y] = -0.5 * inv(A) * b */
+  t1 = b[0] / A[0][0];
+  t2 = (b[1] - t1 * A[0][1]) / A[1][1];
+  delta2 = t2 / A[1][1];
+  delta1 = 0.5 * (t1 - delta2 * A[0][1]) / A[0][0];
+  delta2 *= 0.5;
+
+  /* limit norm */
+  t1 = delta1 * delta1 + delta2 * delta2;
+  if (t1 > 1.0) {
+    delta1 /= t1;
+    delta2 /= t1;
+  }
+
+  *peak_val = 0.5 * (b[0] * delta1 + b[1] * delta2) + c;
+
+  *x += delta1;
+  *y += delta2;
+}
+
+
+static void PCorr(const double *in, double *outcorr)
+{
+  double sum, ysum, prod;
+  const double *x, *inptr;
+  int k, n;
+
+  //ysum = 1e-6;          /* use this with float (i.s.o. double)! */
+  ysum = 1e-13;
+  sum = 0.0;
+  x = in + PITCH_MAX_LAG/2 + 2;
+  for (n = 0; n < PITCH_CORR_LEN2; n++) {
+    ysum += in[n] * in[n];
+    sum += x[n] * in[n];
+  }
+
+  outcorr += PITCH_LAG_SPAN2 - 1;     /* index of last element in array */
+  *outcorr = sum / sqrt(ysum);
+
+  for (k = 1; k < PITCH_LAG_SPAN2; k++) {
+    ysum -= in[k-1] * in[k-1];
+    ysum += in[PITCH_CORR_LEN2 + k - 1] * in[PITCH_CORR_LEN2 + k - 1];
+    sum = 0.0;
+    inptr = &in[k];
+    prod = x[0] * inptr[0];
+    for (n = 1; n < PITCH_CORR_LEN2; n++) {
+      sum += prod;
+      prod = x[n] * inptr[n];
+    }
+    sum += prod;
+    outcorr--;
+    *outcorr = sum / sqrt(ysum);
+  }
+}
+
+static void WebRtcIsac_AllpassFilterForDec(double* InOut,
+                                           const double* APSectionFactors,
+                                           size_t lengthInOut,
+                                           double* FilterState) {
+  // This performs all-pass filtering--a series of first order all-pass
+  // sections are used to filter the input in a cascade manner.
+  size_t n, j;
+  double temp;
+  for (j = 0; j < ALLPASSSECTIONS; j++) {
+    for (n = 0; n < lengthInOut; n += 2) {
+      temp = InOut[n];  // store input
+      InOut[n] = FilterState[j] + APSectionFactors[j] * temp;
+      FilterState[j] = -APSectionFactors[j] * InOut[n] + temp;
+    }
+  }
+}
+
+static void WebRtcIsac_DecimateAllpass(
+    const double* in,
+    double* state_in,  // array of size: 2*ALLPASSSECTIONS+1
+    size_t N,          // number of input samples
+    double* out) {     // array of size N/2
+
+  static const double APupper[ALLPASSSECTIONS] = {0.0347, 0.3826};
+  static const double APlower[ALLPASSSECTIONS] = {0.1544, 0.744};
+
+  size_t n;
+  double data_vec[PITCH_FRAME_LEN];
+
+  /* copy input */
+  memcpy(data_vec + 1, in, sizeof(double) * (N - 1));
+
+  data_vec[0] = state_in[2 * ALLPASSSECTIONS];  // the z^(-1) state
+  state_in[2 * ALLPASSSECTIONS] = in[N - 1];
+
+  WebRtcIsac_AllpassFilterForDec(data_vec + 1, APupper, N, state_in);
+  WebRtcIsac_AllpassFilterForDec(data_vec, APlower, N,
+                                 state_in + ALLPASSSECTIONS);
+
+  for (n = 0; n < N / 2; n++)
+    out[n] = data_vec[2 * n] + data_vec[2 * n + 1];
+}
+
+RTC_PUSH_IGNORING_WFRAME_LARGER_THAN()
+
+static void WebRtcIsac_InitializePitch(const double* in,
+                                       const double old_lag,
+                                       const double old_gain,
+                                       PitchAnalysisStruct* State,
+                                       double* lags) {
+  double buf_dec[PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2+2];
+  double ratio, log_lag, gain_bias;
+  double bias;
+  double corrvec1[PITCH_LAG_SPAN2];
+  double corrvec2[PITCH_LAG_SPAN2];
+  int m, k;
+  // Allocating 10 extra entries at the begining of the CorrSurf
+  double corrSurfBuff[10 + (2*PITCH_BW+3)*(PITCH_LAG_SPAN2+4)];
+  double* CorrSurf[2*PITCH_BW+3];
+  double *CorrSurfPtr1, *CorrSurfPtr2;
+  double LagWin[3] = {0.2, 0.5, 0.98};
+  int ind1, ind2, peaks_ind, peak, max_ind;
+  int peaks[PITCH_MAX_NUM_PEAKS];
+  double adj, gain_tmp;
+  double corr, corr_max;
+  double intrp_a, intrp_b, intrp_c, intrp_d;
+  double peak_vals[PITCH_MAX_NUM_PEAKS];
+  double lags1[PITCH_MAX_NUM_PEAKS];
+  double lags2[PITCH_MAX_NUM_PEAKS];
+  double T[3][3];
+  int row;
+
+  for(k = 0; k < 2*PITCH_BW+3; k++)
+  {
+    CorrSurf[k] = &corrSurfBuff[10 + k * (PITCH_LAG_SPAN2+4)];
+  }
+  /* reset CorrSurf matrix */
+  memset(corrSurfBuff, 0, sizeof(double) * (10 + (2*PITCH_BW+3) * (PITCH_LAG_SPAN2+4)));
+
+  //warnings -DH
+  max_ind = 0;
+  peak = 0;
+
+  /* copy old values from state buffer */
+  memcpy(buf_dec, State->dec_buffer, sizeof(double) * (PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2-PITCH_FRAME_LEN/2+2));
+
+  /* decimation; put result after the old values */
+  WebRtcIsac_DecimateAllpass(in, State->decimator_state, PITCH_FRAME_LEN,
+                             &buf_dec[PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2-PITCH_FRAME_LEN/2+2]);
+
+  /* low-pass filtering */
+  for (k = PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2-PITCH_FRAME_LEN/2+2; k < PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2+2; k++)
+    buf_dec[k] += 0.75 * buf_dec[k-1] - 0.25 * buf_dec[k-2];
+
+  /* copy end part back into state buffer */
+  memcpy(State->dec_buffer, buf_dec+PITCH_FRAME_LEN/2, sizeof(double) * (PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2-PITCH_FRAME_LEN/2+2));
+
+  /* compute correlation for first and second half of the frame */
+  PCorr(buf_dec, corrvec1);
+  PCorr(buf_dec + PITCH_CORR_STEP2, corrvec2);
+
+  /* bias towards pitch lag of previous frame */
+  log_lag = log(0.5 * old_lag);
+  gain_bias = 4.0 * old_gain * old_gain;
+  if (gain_bias > 0.8) gain_bias = 0.8;
+  for (k = 0; k < PITCH_LAG_SPAN2; k++)
+  {
+    ratio = log((double) (k + (PITCH_MIN_LAG/2-2))) - log_lag;
+    bias = 1.0 + gain_bias * exp(-5.0 * ratio * ratio);
+    corrvec1[k] *= bias;
+  }
+
+  /* taper correlation functions */
+  for (k = 0; k < 3; k++) {
+    gain_tmp = LagWin[k];
+    corrvec1[k] *= gain_tmp;
+    corrvec2[k] *= gain_tmp;
+    corrvec1[PITCH_LAG_SPAN2-1-k] *= gain_tmp;
+    corrvec2[PITCH_LAG_SPAN2-1-k] *= gain_tmp;
+  }
+
+  corr_max = 0.0;
+  /* fill middle row of correlation surface */
+  ind1 = 0;
+  ind2 = 0;
+  CorrSurfPtr1 = &CorrSurf[PITCH_BW][2];
+  for (k = 0; k < PITCH_LAG_SPAN2; k++) {
+    corr = corrvec1[ind1++] + corrvec2[ind2++];
+    CorrSurfPtr1[k] = corr;
+    if (corr > corr_max) {
+      corr_max = corr;  /* update maximum */
+      max_ind = (int)(&CorrSurfPtr1[k] - &CorrSurf[0][0]);
+    }
+  }
+  /* fill first and last rows of correlation surface */
+  ind1 = 0;
+  ind2 = PITCH_BW;
+  CorrSurfPtr1 = &CorrSurf[0][2];
+  CorrSurfPtr2 = &CorrSurf[2*PITCH_BW][PITCH_BW+2];
+  for (k = 0; k < PITCH_LAG_SPAN2-PITCH_BW; k++) {
+    ratio = ((double) (ind1 + 12)) / ((double) (ind2 + 12));
+    adj = 0.2 * ratio * (2.0 - ratio);   /* adjustment factor; inverse parabola as a function of ratio */
+    corr = adj * (corrvec1[ind1] + corrvec2[ind2]);
+    CorrSurfPtr1[k] = corr;
+    if (corr > corr_max) {
+      corr_max = corr;  /* update maximum */
+      max_ind = (int)(&CorrSurfPtr1[k] - &CorrSurf[0][0]);
+    }
+    corr = adj * (corrvec1[ind2++] + corrvec2[ind1++]);
+    CorrSurfPtr2[k] = corr;
+    if (corr > corr_max) {
+      corr_max = corr;  /* update maximum */
+      max_ind = (int)(&CorrSurfPtr2[k] - &CorrSurf[0][0]);
+    }
+  }
+  /* fill second and next to last rows of correlation surface */
+  ind1 = 0;
+  ind2 = PITCH_BW-1;
+  CorrSurfPtr1 = &CorrSurf[1][2];
+  CorrSurfPtr2 = &CorrSurf[2*PITCH_BW-1][PITCH_BW+1];
+  for (k = 0; k < PITCH_LAG_SPAN2-PITCH_BW+1; k++) {
+    ratio = ((double) (ind1 + 12)) / ((double) (ind2 + 12));
+    adj = 0.9 * ratio * (2.0 - ratio);   /* adjustment factor; inverse parabola as a function of ratio */
+    corr = adj * (corrvec1[ind1] + corrvec2[ind2]);
+    CorrSurfPtr1[k] = corr;
+    if (corr > corr_max) {
+      corr_max = corr;  /* update maximum */
+      max_ind = (int)(&CorrSurfPtr1[k] - &CorrSurf[0][0]);
+    }
+    corr = adj * (corrvec1[ind2++] + corrvec2[ind1++]);
+    CorrSurfPtr2[k] = corr;
+    if (corr > corr_max) {
+      corr_max = corr;  /* update maximum */
+      max_ind = (int)(&CorrSurfPtr2[k] - &CorrSurf[0][0]);
+    }
+  }
+  /* fill remainder of correlation surface */
+  for (m = 2; m < PITCH_BW; m++) {
+    ind1 = 0;
+    ind2 = PITCH_BW - m;         /* always larger than ind1 */
+    CorrSurfPtr1 = &CorrSurf[m][2];
+    CorrSurfPtr2 = &CorrSurf[2*PITCH_BW-m][PITCH_BW+2-m];
+    for (k = 0; k < PITCH_LAG_SPAN2-PITCH_BW+m; k++) {
+      ratio = ((double) (ind1 + 12)) / ((double) (ind2 + 12));
+      adj = ratio * (2.0 - ratio);    /* adjustment factor; inverse parabola as a function of ratio */
+      corr = adj * (corrvec1[ind1] + corrvec2[ind2]);
+      CorrSurfPtr1[k] = corr;
+      if (corr > corr_max) {
+        corr_max = corr;  /* update maximum */
+        max_ind = (int)(&CorrSurfPtr1[k] - &CorrSurf[0][0]);
+      }
+      corr = adj * (corrvec1[ind2++] + corrvec2[ind1++]);
+      CorrSurfPtr2[k] = corr;
+      if (corr > corr_max) {
+        corr_max = corr;  /* update maximum */
+        max_ind = (int)(&CorrSurfPtr2[k] - &CorrSurf[0][0]);
+      }
+    }
+  }
+
+  /* threshold value to qualify as a peak */
+  corr_max *= 0.6;
+
+  peaks_ind = 0;
+  /* find peaks */
+  for (m = 1; m < PITCH_BW+1; m++) {
+    if (peaks_ind == PITCH_MAX_NUM_PEAKS) break;
+    CorrSurfPtr1 = &CorrSurf[m][2];
+    for (k = 2; k < PITCH_LAG_SPAN2-PITCH_BW-2+m; k++) {
+      corr = CorrSurfPtr1[k];
+      if (corr > corr_max) {
+        if ( (corr > CorrSurfPtr1[k - (PITCH_LAG_SPAN2+5)]) && (corr > CorrSurfPtr1[k - (PITCH_LAG_SPAN2+4)]) ) {
+          if ( (corr > CorrSurfPtr1[k + (PITCH_LAG_SPAN2+4)]) && (corr > CorrSurfPtr1[k + (PITCH_LAG_SPAN2+5)]) ) {
+            /* found a peak; store index into matrix */
+            peaks[peaks_ind++] = (int)(&CorrSurfPtr1[k] - &CorrSurf[0][0]);
+            if (peaks_ind == PITCH_MAX_NUM_PEAKS) break;
+          }
+        }
+      }
+    }
+  }
+  for (m = PITCH_BW+1; m < 2*PITCH_BW; m++) {
+    if (peaks_ind == PITCH_MAX_NUM_PEAKS) break;
+    CorrSurfPtr1 = &CorrSurf[m][2];
+    for (k = 2+m-PITCH_BW; k < PITCH_LAG_SPAN2-2; k++) {
+      corr = CorrSurfPtr1[k];
+      if (corr > corr_max) {
+        if ( (corr > CorrSurfPtr1[k - (PITCH_LAG_SPAN2+5)]) && (corr > CorrSurfPtr1[k - (PITCH_LAG_SPAN2+4)]) ) {
+          if ( (corr > CorrSurfPtr1[k + (PITCH_LAG_SPAN2+4)]) && (corr > CorrSurfPtr1[k + (PITCH_LAG_SPAN2+5)]) ) {
+            /* found a peak; store index into matrix */
+            peaks[peaks_ind++] = (int)(&CorrSurfPtr1[k] - &CorrSurf[0][0]);
+            if (peaks_ind == PITCH_MAX_NUM_PEAKS) break;
+          }
+        }
+      }
+    }
+  }
+
+  if (peaks_ind > 0) {
+    /* examine each peak */
+    CorrSurfPtr1 = &CorrSurf[0][0];
+    for (k = 0; k < peaks_ind; k++) {
+      peak = peaks[k];
+
+      /* compute four interpolated values around current peak */
+      IntrepolFilter(&CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+5)], &intrp_a);
+      IntrepolFilter(&CorrSurfPtr1[peak - 1            ], &intrp_b);
+      IntrepolFilter(&CorrSurfPtr1[peak                ], &intrp_c);
+      IntrepolFilter(&CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+4)], &intrp_d);
+
+      /* determine maximum of the interpolated values */
+      corr = CorrSurfPtr1[peak];
+      corr_max = intrp_a;
+      if (intrp_b > corr_max) corr_max = intrp_b;
+      if (intrp_c > corr_max) corr_max = intrp_c;
+      if (intrp_d > corr_max) corr_max = intrp_d;
+
+      /* determine where the peak sits and fill a 3x3 matrix around it */
+      row = peak / (PITCH_LAG_SPAN2+4);
+      lags1[k] = (double) ((peak - row * (PITCH_LAG_SPAN2+4)) + PITCH_MIN_LAG/2 - 4);
+      lags2[k] = (double) (lags1[k] + PITCH_BW - row);
+      if ( corr > corr_max ) {
+        T[0][0] = CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+5)];
+        T[2][0] = CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+4)];
+        T[1][1] = corr;
+        T[0][2] = CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+4)];
+        T[2][2] = CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+5)];
+        T[1][0] = intrp_a;
+        T[0][1] = intrp_b;
+        T[2][1] = intrp_c;
+        T[1][2] = intrp_d;
+      } else {
+        if (intrp_a == corr_max) {
+          lags1[k] -= 0.5;
+          lags2[k] += 0.5;
+          IntrepolFilter(&CorrSurfPtr1[peak - 2*(PITCH_LAG_SPAN2+5)], &T[0][0]);
+          IntrepolFilter(&CorrSurfPtr1[peak - (2*PITCH_LAG_SPAN2+9)], &T[2][0]);
+          T[1][1] = intrp_a;
+          T[0][2] = intrp_b;
+          T[2][2] = intrp_c;
+          T[1][0] = CorrSurfPtr1[peak - (2*PITCH_LAG_SPAN2+9)];
+          T[0][1] = CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+5)];
+          T[2][1] = CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+4)];
+          T[1][2] = corr;
+        } else if (intrp_b == corr_max) {
+          lags1[k] -= 0.5;
+          lags2[k] -= 0.5;
+          IntrepolFilter(&CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+6)], &T[0][0]);
+          T[2][0] = intrp_a;
+          T[1][1] = intrp_b;
+          IntrepolFilter(&CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+3)], &T[0][2]);
+          T[2][2] = intrp_d;
+          T[1][0] = CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+5)];
+          T[0][1] = CorrSurfPtr1[peak - 1];
+          T[2][1] = corr;
+          T[1][2] = CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+4)];
+        } else if (intrp_c == corr_max) {
+          lags1[k] += 0.5;
+          lags2[k] += 0.5;
+          T[0][0] = intrp_a;
+          IntrepolFilter(&CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+4)], &T[2][0]);
+          T[1][1] = intrp_c;
+          T[0][2] = intrp_d;
+          IntrepolFilter(&CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+5)], &T[2][2]);
+          T[1][0] = CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+4)];
+          T[0][1] = corr;
+          T[2][1] = CorrSurfPtr1[peak + 1];
+          T[1][2] = CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+5)];
+        } else {
+          lags1[k] += 0.5;
+          lags2[k] -= 0.5;
+          T[0][0] = intrp_b;
+          T[2][0] = intrp_c;
+          T[1][1] = intrp_d;
+          IntrepolFilter(&CorrSurfPtr1[peak + 2*(PITCH_LAG_SPAN2+4)], &T[0][2]);
+          IntrepolFilter(&CorrSurfPtr1[peak + (2*PITCH_LAG_SPAN2+9)], &T[2][2]);
+          T[1][0] = corr;
+          T[0][1] = CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+4)];
+          T[2][1] = CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+5)];
+          T[1][2] = CorrSurfPtr1[peak + (2*PITCH_LAG_SPAN2+9)];
+        }
+      }
+
+      /* 2D parabolic interpolation gives more accurate lags and peak value */
+      Intrpol2D(T, &lags1[k], &lags2[k], &peak_vals[k]);
+    }
+
+    /* determine the highest peak, after applying a bias towards short lags */
+    corr_max = 0.0;
+    for (k = 0; k < peaks_ind; k++) {
+      corr = peak_vals[k] * pow(PITCH_PEAK_DECAY, log(lags1[k] + lags2[k]));
+      if (corr > corr_max) {
+        corr_max = corr;
+        peak = k;
+      }
+    }
+
+    lags1[peak] *= 2.0;
+    lags2[peak] *= 2.0;
+
+    if (lags1[peak] < (double) PITCH_MIN_LAG) lags1[peak] = (double) PITCH_MIN_LAG;
+    if (lags2[peak] < (double) PITCH_MIN_LAG) lags2[peak] = (double) PITCH_MIN_LAG;
+    if (lags1[peak] > (double) PITCH_MAX_LAG) lags1[peak] = (double) PITCH_MAX_LAG;
+    if (lags2[peak] > (double) PITCH_MAX_LAG) lags2[peak] = (double) PITCH_MAX_LAG;
+
+    /* store lags of highest peak in output array */
+    lags[0] = lags1[peak];
+    lags[1] = lags1[peak];
+    lags[2] = lags2[peak];
+    lags[3] = lags2[peak];
+  }
+  else
+  {
+    row = max_ind / (PITCH_LAG_SPAN2+4);
+    lags1[0] = (double) ((max_ind - row * (PITCH_LAG_SPAN2+4)) + PITCH_MIN_LAG/2 - 4);
+    lags2[0] = (double) (lags1[0] + PITCH_BW - row);
+
+    if (lags1[0] < (double) PITCH_MIN_LAG) lags1[0] = (double) PITCH_MIN_LAG;
+    if (lags2[0] < (double) PITCH_MIN_LAG) lags2[0] = (double) PITCH_MIN_LAG;
+    if (lags1[0] > (double) PITCH_MAX_LAG) lags1[0] = (double) PITCH_MAX_LAG;
+    if (lags2[0] > (double) PITCH_MAX_LAG) lags2[0] = (double) PITCH_MAX_LAG;
+
+    /* store lags of highest peak in output array */
+    lags[0] = lags1[0];
+    lags[1] = lags1[0];
+    lags[2] = lags2[0];
+    lags[3] = lags2[0];
+  }
+}
+
+RTC_POP_IGNORING_WFRAME_LARGER_THAN()
+
+/* create weighting matrix by orthogonalizing a basis of polynomials of increasing order
+ * t = (0:4)';
+ * A = [t.^0, t.^1, t.^2, t.^3, t.^4];
+ * [Q, dummy] = qr(A);
+ * P.Weight = Q * diag([0, .1, .5, 1, 1]) * Q'; */
+static const double kWeight[5][5] = {
+  { 0.29714285714286,  -0.30857142857143,  -0.05714285714286,   0.05142857142857,  0.01714285714286},
+  {-0.30857142857143,   0.67428571428571,  -0.27142857142857,  -0.14571428571429,  0.05142857142857},
+  {-0.05714285714286,  -0.27142857142857,   0.65714285714286,  -0.27142857142857, -0.05714285714286},
+  { 0.05142857142857,  -0.14571428571429,  -0.27142857142857,   0.67428571428571, -0.30857142857143},
+  { 0.01714285714286,   0.05142857142857,  -0.05714285714286,  -0.30857142857143,  0.29714285714286}
+};
+
+/* second order high-pass filter */
+static void WebRtcIsac_Highpass(const double* in,
+                         double* out,
+                         double* state,
+                         size_t N) {
+  /* create high-pass filter ocefficients
+   * z = 0.998 * exp(j*2*pi*35/8000);
+   * p = 0.94 * exp(j*2*pi*140/8000);
+   * HP_b = [1, -2*real(z), abs(z)^2];
+   * HP_a = [1, -2*real(p), abs(p)^2]; */
+  static const double a_coef[2] = { 1.86864659625574, -0.88360000000000};
+  static const double b_coef[2] = {-1.99524591718270,  0.99600400000000};
+
+  size_t k;
+
+  for (k=0; k<N; k++) {
+    *out = *in + state[1];
+    state[1] = state[0] + b_coef[0] * *in + a_coef[0] * *out;
+    state[0] = b_coef[1] * *in++ + a_coef[1] * *out++;
+  }
+}
+
+RTC_PUSH_IGNORING_WFRAME_LARGER_THAN()
+
+void WebRtcIsac_PitchAnalysis(const double *in,               /* PITCH_FRAME_LEN samples */
+                              double *out,                    /* PITCH_FRAME_LEN+QLOOKAHEAD samples */
+                              PitchAnalysisStruct *State,
+                              double *lags,
+                              double *gains)
+{
+  double HPin[PITCH_FRAME_LEN];
+  double Weighted[PITCH_FRAME_LEN];
+  double Whitened[PITCH_FRAME_LEN + QLOOKAHEAD];
+  double inbuf[PITCH_FRAME_LEN + QLOOKAHEAD];
+  double out_G[PITCH_FRAME_LEN + QLOOKAHEAD];          // could be removed by using out instead
+  double out_dG[4][PITCH_FRAME_LEN + QLOOKAHEAD];
+  double old_lag, old_gain;
+  double nrg_wht, tmp;
+  double Wnrg, Wfluct, Wgain;
+  double H[4][4];
+  double grad[4];
+  double dG[4];
+  int k, m, n, iter;
+
+  /* high pass filtering using second order pole-zero filter */
+  WebRtcIsac_Highpass(in, HPin, State->hp_state, PITCH_FRAME_LEN);
+
+  /* copy from state into buffer */
+  memcpy(Whitened, State->whitened_buf, sizeof(double) * QLOOKAHEAD);
+
+  /* compute weighted and whitened signals */
+  WebRtcIsac_WeightingFilter(HPin, &Weighted[0], &Whitened[QLOOKAHEAD], &(State->Wghtstr));
+
+  /* copy from buffer into state */
+  memcpy(State->whitened_buf, Whitened+PITCH_FRAME_LEN, sizeof(double) * QLOOKAHEAD);
+
+  old_lag = State->PFstr_wght.oldlagp[0];
+  old_gain = State->PFstr_wght.oldgainp[0];
+
+  /* inital pitch estimate */
+  WebRtcIsac_InitializePitch(Weighted, old_lag, old_gain, State, lags);
+
+
+  /* Iterative optimization of lags - to be done */
+
+  /* compute energy of whitened signal */
+  nrg_wht = 0.0;
+  for (k = 0; k < PITCH_FRAME_LEN + QLOOKAHEAD; k++)
+    nrg_wht += Whitened[k] * Whitened[k];
+
+
+  /* Iterative optimization of gains */
+
+  /* set weights for energy, gain fluctiation, and spectral gain penalty functions */
+  Wnrg = 1.0 / nrg_wht;
+  Wgain = 0.005;
+  Wfluct = 3.0;
+
+  /* set initial gains */
+  for (k = 0; k < 4; k++)
+    gains[k] = PITCH_MAX_GAIN_06;
+
+  /* two iterations should be enough */
+  for (iter = 0; iter < 2; iter++) {
+    /* compute Jacobian of pre-filter output towards gains */
+    WebRtcIsac_PitchfilterPre_gains(Whitened, out_G, out_dG, &(State->PFstr_wght), lags, gains);
+
+    /* gradient and approximate Hessian (lower triangle) for minimizing the filter's output power */
+    for (k = 0; k < 4; k++) {
+      tmp = 0.0;
+      for (n = 0; n < PITCH_FRAME_LEN + QLOOKAHEAD; n++)
+        tmp += out_G[n] * out_dG[k][n];
+      grad[k] = tmp * Wnrg;
+    }
+    for (k = 0; k < 4; k++) {
+      for (m = 0; m <= k; m++) {
+        tmp = 0.0;
+        for (n = 0; n < PITCH_FRAME_LEN + QLOOKAHEAD; n++)
+          tmp += out_dG[m][n] * out_dG[k][n];
+        H[k][m] = tmp * Wnrg;
+      }
+    }
+
+    /* add gradient and Hessian (lower triangle) for dampening fast gain changes */
+    for (k = 0; k < 4; k++) {
+      tmp = kWeight[k+1][0] * old_gain;
+      for (m = 0; m < 4; m++)
+        tmp += kWeight[k+1][m+1] * gains[m];
+      grad[k] += tmp * Wfluct;
+    }
+    for (k = 0; k < 4; k++) {
+      for (m = 0; m <= k; m++) {
+        H[k][m] += kWeight[k+1][m+1] * Wfluct;
+      }
+    }
+
+    /* add gradient and Hessian for dampening gain */
+    for (k = 0; k < 3; k++) {
+      tmp = 1.0 / (1 - gains[k]);
+      grad[k] += tmp * tmp * Wgain;
+      H[k][k] += 2.0 * tmp * (tmp * tmp * Wgain);
+    }
+    tmp = 1.0 / (1 - gains[3]);
+    grad[3] += 1.33 * (tmp * tmp * Wgain);
+    H[3][3] += 2.66 * tmp * (tmp * tmp * Wgain);
+
+
+    /* compute Cholesky factorization of Hessian
+     * by overwritting the upper triangle; scale factors on diagonal
+     * (for non pc-platforms store the inverse of the diagonals seperately to minimize divisions) */
+    H[0][1] = H[1][0] / H[0][0];
+    H[0][2] = H[2][0] / H[0][0];
+    H[0][3] = H[3][0] / H[0][0];
+    H[1][1] -= H[0][0] * H[0][1] * H[0][1];
+    H[1][2] = (H[2][1] - H[0][1] * H[2][0]) / H[1][1];
+    H[1][3] = (H[3][1] - H[0][1] * H[3][0]) / H[1][1];
+    H[2][2] -= H[0][0] * H[0][2] * H[0][2] + H[1][1] * H[1][2] * H[1][2];
+    H[2][3] = (H[3][2] - H[0][2] * H[3][0] - H[1][2] * H[1][1] * H[1][3]) / H[2][2];
+    H[3][3] -= H[0][0] * H[0][3] * H[0][3] + H[1][1] * H[1][3] * H[1][3] + H[2][2] * H[2][3] * H[2][3];
+
+    /* Compute update as  delta_gains = -inv(H) * grad */
+    /* copy and negate */
+    for (k = 0; k < 4; k++)
+      dG[k] = -grad[k];
+    /* back substitution */
+    dG[1] -= dG[0] * H[0][1];
+    dG[2] -= dG[0] * H[0][2] + dG[1] * H[1][2];
+    dG[3] -= dG[0] * H[0][3] + dG[1] * H[1][3] + dG[2] * H[2][3];
+    /* scale */
+    for (k = 0; k < 4; k++)
+      dG[k] /= H[k][k];
+    /* back substitution */
+    dG[2] -= dG[3] * H[2][3];
+    dG[1] -= dG[3] * H[1][3] + dG[2] * H[1][2];
+    dG[0] -= dG[3] * H[0][3] + dG[2] * H[0][2] + dG[1] * H[0][1];
+
+    /* update gains and check range */
+    for (k = 0; k < 4; k++) {
+      gains[k] += dG[k];
+      if (gains[k] > PITCH_MAX_GAIN)
+        gains[k] = PITCH_MAX_GAIN;
+      else if (gains[k] < 0.0)
+        gains[k] = 0.0;
+    }
+  }
+
+  /* update state for next frame */
+  WebRtcIsac_PitchfilterPre(Whitened, out, &(State->PFstr_wght), lags, gains);
+
+  /* concatenate previous input's end and current input */
+  memcpy(inbuf, State->inbuf, sizeof(double) * QLOOKAHEAD);
+  memcpy(inbuf+QLOOKAHEAD, in, sizeof(double) * PITCH_FRAME_LEN);
+
+  /* lookahead pitch filtering for masking analysis */
+  WebRtcIsac_PitchfilterPre_la(inbuf, out, &(State->PFstr), lags, gains);
+
+  /* store last part of input */
+  for (k = 0; k < QLOOKAHEAD; k++)
+    State->inbuf[k] = inbuf[k + PITCH_FRAME_LEN];
+}
+
+RTC_POP_IGNORING_WFRAME_LARGER_THAN()

VocieProcess/modules/audio_coding/codecs/isac/main/source/pitch_estimator.h

@@ -0,0 +1,32 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * pitch_estimator.h
+ *
+ * Pitch functions
+ *
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_ESTIMATOR_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_ESTIMATOR_H_
+
+#include <stddef.h>
+
+#include "modules/audio_coding/codecs/isac/main/source/structs.h"
+
+void WebRtcIsac_PitchAnalysis(
+    const double* in, /* PITCH_FRAME_LEN samples */
+    double* out,      /* PITCH_FRAME_LEN+QLOOKAHEAD samples */
+    PitchAnalysisStruct* State,
+    double* lags,
+    double* gains);
+
+#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_ESTIMATOR_H_ */

VocieProcess/modules/audio_coding/codecs/isac/main/source/pitch_filter.c

@@ -0,0 +1,388 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <math.h>
+#include <memory.h>
+#include <stdlib.h>
+
+#include "modules/audio_coding/codecs/isac/main/source/pitch_estimator.h"
+#include "modules/audio_coding/codecs/isac/main/source/os_specific_inline.h"
+#include "rtc_base/compile_assert_c.h"
+
+/*
+ * We are implementing the following filters;
+ *
+ * Pre-filtering:
+ *   y(z) = x(z) + damper(z) * gain * (x(z) + y(z)) * z ^ (-lag);
+ *
+ * Post-filtering:
+ *   y(z) = x(z) - damper(z) * gain * (x(z) + y(z)) * z ^ (-lag);
+ *
+ * Note that `lag` is a floating number so we perform an interpolation to
+ * obtain the correct `lag`.
+ *
+ */
+
+static const double kDampFilter[PITCH_DAMPORDER] = {-0.07, 0.25, 0.64, 0.25,
+    -0.07};
+
+/* interpolation coefficients; generated by design_pitch_filter.m */
+static const double kIntrpCoef[PITCH_FRACS][PITCH_FRACORDER] = {
+    {-0.02239172458614,  0.06653315052934, -0.16515880017569,  0.60701333734125,
+     0.64671399919202, -0.20249000396417,  0.09926548334755, -0.04765933793109,
+     0.01754159521746},
+    {-0.01985640750434,  0.05816126837866, -0.13991265473714,  0.44560418147643,
+     0.79117042386876, -0.20266133815188,  0.09585268418555, -0.04533310458084,
+     0.01654127246314},
+    {-0.01463300534216,  0.04229888475060, -0.09897034715253,  0.28284326017787,
+     0.90385267956632, -0.16976950138649,  0.07704272393639, -0.03584218578311,
+     0.01295781500709},
+    {-0.00764851320885,  0.02184035544377, -0.04985561057281,  0.13083306574393,
+     0.97545011664662, -0.10177807997561,  0.04400901776474, -0.02010737175166,
+     0.00719783432422},
+    {-0.00000000000000,  0.00000000000000, -0.00000000000001,  0.00000000000001,
+     0.99999999999999,  0.00000000000001, -0.00000000000001,  0.00000000000000,
+     -0.00000000000000},
+    {0.00719783432422, -0.02010737175166,  0.04400901776474, -0.10177807997562,
+     0.97545011664663,  0.13083306574393, -0.04985561057280,  0.02184035544377,
+     -0.00764851320885},
+    {0.01295781500710, -0.03584218578312,  0.07704272393640, -0.16976950138650,
+     0.90385267956634,  0.28284326017785, -0.09897034715252,  0.04229888475059,
+     -0.01463300534216},
+    {0.01654127246315, -0.04533310458085,  0.09585268418557, -0.20266133815190,
+     0.79117042386878,  0.44560418147640, -0.13991265473712,  0.05816126837865,
+     -0.01985640750433}
+};
+
+/*
+ * Enumerating the operation of the filter.
+ * iSAC has 4 different pitch-filter which are very similar in their structure.
+ *
+ * kPitchFilterPre     : In this mode the filter is operating as pitch
+ *                       pre-filter. This is used at the encoder.
+ * kPitchFilterPost    : In this mode the filter is operating as pitch
+ *                       post-filter. This is the inverse of pre-filter and used
+ *                       in the decoder.
+ * kPitchFilterPreLa   : This is, in structure, similar to pre-filtering but
+ *                       utilizing 3 millisecond lookahead. It is used to
+ *                       obtain the signal for LPC analysis.
+ * kPitchFilterPreGain : This is, in structure, similar to pre-filtering but
+ *                       differential changes in gain is considered. This is
+ *                       used to find the optimal gain.
+ */
+typedef enum {
+  kPitchFilterPre, kPitchFilterPost, kPitchFilterPreLa, kPitchFilterPreGain
+} PitchFilterOperation;
+
+/*
+ * Structure with parameters used for pitch-filtering.
+ * buffer           : a buffer where the sum of previous inputs and outputs
+ *                    are stored.
+ * damper_state     : the state of the damping filter. The filter is defined by
+ *                    `kDampFilter`.
+ * interpol_coeff   : pointer to a set of coefficient which are used to utilize
+ *                    fractional pitch by interpolation.
+ * gain             : pitch-gain to be applied to the current segment of input.
+ * lag              : pitch-lag for the current segment of input.
+ * lag_offset       : the offset of lag w.r.t. current sample.
+ * sub_frame        : sub-frame index, there are 4 pitch sub-frames in an iSAC
+ *                    frame.
+ *                    This specifies the usage of the filter. See
+ *                    'PitchFilterOperation' for operational modes.
+ * num_samples      : number of samples to be processed in each segment.
+ * index            : index of the input and output sample.
+ * damper_state_dg  : state of damping filter for different trial gains.
+ * gain_mult        : differential changes to gain.
+ */
+typedef struct {
+  double buffer[PITCH_INTBUFFSIZE + QLOOKAHEAD];
+  double damper_state[PITCH_DAMPORDER];
+  const double *interpol_coeff;
+  double gain;
+  double lag;
+  int lag_offset;
+
+  int sub_frame;
+  PitchFilterOperation mode;
+  int num_samples;
+  int index;
+
+  double damper_state_dg[4][PITCH_DAMPORDER];
+  double gain_mult[4];
+} PitchFilterParam;
+
+/**********************************************************************
+ * FilterSegment()
+ * Filter one segment, a quarter of a frame.
+ *
+ * Inputs
+ *   in_data      : pointer to the input signal of 30 ms at 8 kHz sample-rate.
+ *   filter_param : pitch filter parameters.
+ *
+ * Outputs
+ *   out_data     : pointer to a buffer where the filtered signal is written to.
+ *   out_dg       : [only used in kPitchFilterPreGain] pointer to a buffer
+ *                  where the output of different gain values (differential
+ *                  change to gain) is written.
+ */
+static void FilterSegment(const double* in_data, PitchFilterParam* parameters,
+                          double* out_data,
+                          double out_dg[][PITCH_FRAME_LEN + QLOOKAHEAD]) {
+  int n;
+  int m;
+  int j;
+  double sum;
+  double sum2;
+  /* Index of `parameters->buffer` where the output is written to. */
+  int pos = parameters->index + PITCH_BUFFSIZE;
+  /* Index of `parameters->buffer` where samples are read for fractional-lag
+   * computation. */
+  int pos_lag = pos - parameters->lag_offset;
+
+  for (n = 0; n < parameters->num_samples; ++n) {
+    /* Shift low pass filter states. */
+    for (m = PITCH_DAMPORDER - 1; m > 0; --m) {
+      parameters->damper_state[m] = parameters->damper_state[m - 1];
+    }
+    /* Filter to get fractional pitch. */
+    sum = 0.0;
+    for (m = 0; m < PITCH_FRACORDER; ++m) {
+      sum += parameters->buffer[pos_lag + m] * parameters->interpol_coeff[m];
+    }
+    /* Multiply with gain. */
+    parameters->damper_state[0] = parameters->gain * sum;
+
+    if (parameters->mode == kPitchFilterPreGain) {
+      int lag_index = parameters->index - parameters->lag_offset;
+      int m_tmp = (lag_index < 0) ? -lag_index : 0;
+      /* Update the damper state for the new sample. */
+      for (m = PITCH_DAMPORDER - 1; m > 0; --m) {
+        for (j = 0; j < 4; ++j) {
+          parameters->damper_state_dg[j][m] =
+              parameters->damper_state_dg[j][m - 1];
+        }
+      }
+
+      for (j = 0; j < parameters->sub_frame + 1; ++j) {
+        /* Filter for fractional pitch. */
+        sum2 = 0.0;
+        for (m = PITCH_FRACORDER-1; m >= m_tmp; --m) {
+          /* `lag_index + m` is always larger than or equal to zero, see how
+           * m_tmp is computed. This is equivalent to assume samples outside
+           * `out_dg[j]` are zero. */
+          sum2 += out_dg[j][lag_index + m] * parameters->interpol_coeff[m];
+        }
+        /* Add the contribution of differential gain change. */
+        parameters->damper_state_dg[j][0] = parameters->gain_mult[j] * sum +
+            parameters->gain * sum2;
+      }
+
+      /* Filter with damping filter, and store the results. */
+      for (j = 0; j < parameters->sub_frame + 1; ++j) {
+        sum = 0.0;
+        for (m = 0; m < PITCH_DAMPORDER; ++m) {
+          sum -= parameters->damper_state_dg[j][m] * kDampFilter[m];
+        }
+        out_dg[j][parameters->index] = sum;
+      }
+    }
+    /* Filter with damping filter. */
+    sum = 0.0;
+    for (m = 0; m < PITCH_DAMPORDER; ++m) {
+      sum += parameters->damper_state[m] * kDampFilter[m];
+    }
+
+    /* Subtract from input and update buffer. */
+    out_data[parameters->index] = in_data[parameters->index] - sum;
+    parameters->buffer[pos] = in_data[parameters->index] +
+        out_data[parameters->index];
+
+    ++parameters->index;
+    ++pos;
+    ++pos_lag;
+  }
+  return;
+}
+
+/* Update filter parameters based on the pitch-gains and pitch-lags. */
+static void Update(PitchFilterParam* parameters) {
+  double fraction;
+  int fraction_index;
+  /* Compute integer lag-offset. */
+  parameters->lag_offset = WebRtcIsac_lrint(parameters->lag + PITCH_FILTDELAY +
+                                            0.5);
+  /* Find correct set of coefficients for computing fractional pitch. */
+  fraction = parameters->lag_offset - (parameters->lag + PITCH_FILTDELAY);
+  fraction_index = WebRtcIsac_lrint(PITCH_FRACS * fraction - 0.5);
+  parameters->interpol_coeff = kIntrpCoef[fraction_index];
+
+  if (parameters->mode == kPitchFilterPreGain) {
+    /* If in this mode make a differential change to pitch gain. */
+    parameters->gain_mult[parameters->sub_frame] += 0.2;
+    if (parameters->gain_mult[parameters->sub_frame] > 1.0) {
+      parameters->gain_mult[parameters->sub_frame] = 1.0;
+    }
+    if (parameters->sub_frame > 0) {
+      parameters->gain_mult[parameters->sub_frame - 1] -= 0.2;
+    }
+  }
+}
+
+/******************************************************************************
+ * FilterFrame()
+ * Filter a frame of 30 millisecond, given pitch-lags and pitch-gains.
+ *
+ * Inputs
+ *   in_data     : pointer to the input signal of 30 ms at 8 kHz sample-rate.
+ *   lags        : pointer to pitch-lags, 4 lags per frame.
+ *   gains       : pointer to pitch-gians, 4 gains per frame.
+ *   mode        : defining the functionality of the filter. It takes the
+ *                 following values.
+ *                 kPitchFilterPre:     Pitch pre-filter, used at encoder.
+ *                 kPitchFilterPost:    Pitch post-filter, used at decoder.
+ *                 kPitchFilterPreLa:   Pitch pre-filter with lookahead.
+ *                 kPitchFilterPreGain: Pitch pre-filter used to otain optimal
+ *                                      pitch-gains.
+ *
+ * Outputs
+ *   out_data    : pointer to a buffer where the filtered signal is written to.
+ *   out_dg      : [only used in kPitchFilterPreGain] pointer to a buffer
+ *                 where the output of different gain values (differential
+ *                 change to gain) is written.
+ */
+static void FilterFrame(const double* in_data, PitchFiltstr* filter_state,
+                        double* lags, double* gains, PitchFilterOperation mode,
+                        double* out_data,
+                        double out_dg[][PITCH_FRAME_LEN + QLOOKAHEAD]) {
+  PitchFilterParam filter_parameters;
+  double gain_delta, lag_delta;
+  double old_lag, old_gain;
+  int n;
+  int m;
+  const double kEnhancer = 1.3;
+
+  /* Set up buffer and states. */
+  filter_parameters.index = 0;
+  filter_parameters.lag_offset = 0;
+  filter_parameters.mode = mode;
+  /* Copy states to local variables. */
+  memcpy(filter_parameters.buffer, filter_state->ubuf,
+         sizeof(filter_state->ubuf));
+  RTC_COMPILE_ASSERT(sizeof(filter_parameters.buffer) >=
+                 sizeof(filter_state->ubuf));
+  memset(filter_parameters.buffer +
+             sizeof(filter_state->ubuf) / sizeof(filter_state->ubuf[0]),
+         0, sizeof(filter_parameters.buffer) - sizeof(filter_state->ubuf));
+  memcpy(filter_parameters.damper_state, filter_state->ystate,
+         sizeof(filter_state->ystate));
+
+  if (mode == kPitchFilterPreGain) {
+    /* Clear buffers. */
+    memset(filter_parameters.gain_mult, 0, sizeof(filter_parameters.gain_mult));
+    memset(filter_parameters.damper_state_dg, 0,
+           sizeof(filter_parameters.damper_state_dg));
+    for (n = 0; n < PITCH_SUBFRAMES; ++n) {
+      //memset(out_dg[n], 0, sizeof(double) * (PITCH_FRAME_LEN + QLOOKAHEAD));
+      memset(out_dg[n], 0, sizeof(out_dg[n]));
+    }
+  } else if (mode == kPitchFilterPost) {
+    /* Make output more periodic. Negative sign is to change the structure
+     * of the filter. */
+    for (n = 0; n < PITCH_SUBFRAMES; ++n) {
+      gains[n] *= -kEnhancer;
+    }
+  }
+
+  old_lag = *filter_state->oldlagp;
+  old_gain = *filter_state->oldgainp;
+
+  /* No interpolation if pitch lag step is big. */
+  if ((lags[0] > (PITCH_UPSTEP * old_lag)) ||
+      (lags[0] < (PITCH_DOWNSTEP * old_lag))) {
+    old_lag = lags[0];
+    old_gain = gains[0];
+
+    if (mode == kPitchFilterPreGain) {
+      filter_parameters.gain_mult[0] = 1.0;
+    }
+  }
+
+  filter_parameters.num_samples = PITCH_UPDATE;
+  for (m = 0; m < PITCH_SUBFRAMES; ++m) {
+    /* Set the sub-frame value. */
+    filter_parameters.sub_frame = m;
+    /* Calculate interpolation steps for pitch-lag and pitch-gain. */
+    lag_delta = (lags[m] - old_lag) / PITCH_GRAN_PER_SUBFRAME;
+    filter_parameters.lag = old_lag;
+    gain_delta = (gains[m] - old_gain) / PITCH_GRAN_PER_SUBFRAME;
+    filter_parameters.gain = old_gain;
+    /* Store for the next sub-frame. */
+    old_lag = lags[m];
+    old_gain = gains[m];
+
+    for (n = 0; n < PITCH_GRAN_PER_SUBFRAME; ++n) {
+      /* Step-wise interpolation of pitch gains and lags. As pitch-lag changes,
+       * some parameters of filter need to be update. */
+      filter_parameters.gain += gain_delta;
+      filter_parameters.lag += lag_delta;
+      /* Update parameters according to new lag value. */
+      Update(&filter_parameters);
+      /* Filter a segment of input. */
+      FilterSegment(in_data, &filter_parameters, out_data, out_dg);
+    }
+  }
+
+  if (mode != kPitchFilterPreGain) {
+    /* Export buffer and states. */
+    memcpy(filter_state->ubuf, &filter_parameters.buffer[PITCH_FRAME_LEN],
+           sizeof(filter_state->ubuf));
+    memcpy(filter_state->ystate, filter_parameters.damper_state,
+           sizeof(filter_state->ystate));
+
+    /* Store for the next frame. */
+    *filter_state->oldlagp = old_lag;
+    *filter_state->oldgainp = old_gain;
+  }
+
+  if ((mode == kPitchFilterPreGain) || (mode == kPitchFilterPreLa)) {
+    /* Filter the lookahead segment, this is treated as the last sub-frame. So
+     * set `pf_param` to last sub-frame. */
+    filter_parameters.sub_frame = PITCH_SUBFRAMES - 1;
+    filter_parameters.num_samples = QLOOKAHEAD;
+    FilterSegment(in_data, &filter_parameters, out_data, out_dg);
+  }
+}
+
+void WebRtcIsac_PitchfilterPre(double* in_data, double* out_data,
+                               PitchFiltstr* pf_state, double* lags,
+                               double* gains) {
+  FilterFrame(in_data, pf_state, lags, gains, kPitchFilterPre, out_data, NULL);
+}
+
+void WebRtcIsac_PitchfilterPre_la(double* in_data, double* out_data,
+                                  PitchFiltstr* pf_state, double* lags,
+                                  double* gains) {
+  FilterFrame(in_data, pf_state, lags, gains, kPitchFilterPreLa, out_data,
+              NULL);
+}
+
+void WebRtcIsac_PitchfilterPre_gains(
+    double* in_data, double* out_data,
+    double out_dg[][PITCH_FRAME_LEN + QLOOKAHEAD], PitchFiltstr *pf_state,
+    double* lags, double* gains) {
+  FilterFrame(in_data, pf_state, lags, gains, kPitchFilterPreGain, out_data,
+              out_dg);
+}
+
+void WebRtcIsac_PitchfilterPost(double* in_data, double* out_data,
+                                PitchFiltstr* pf_state, double* lags,
+                                double* gains) {
+  FilterFrame(in_data, pf_state, lags, gains, kPitchFilterPost, out_data, NULL);
+}

VocieProcess/modules/audio_coding/codecs/isac/main/source/pitch_filter.h

@@ -0,0 +1,42 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_FILTER_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_FILTER_H_
+
+#include "modules/audio_coding/codecs/isac/main/source/structs.h"
+
+void WebRtcIsac_PitchfilterPre(double* indat,
+                               double* outdat,
+                               PitchFiltstr* pfp,
+                               double* lags,
+                               double* gains);
+
+void WebRtcIsac_PitchfilterPost(double* indat,
+                                double* outdat,
+                                PitchFiltstr* pfp,
+                                double* lags,
+                                double* gains);
+
+void WebRtcIsac_PitchfilterPre_la(double* indat,
+                                  double* outdat,
+                                  PitchFiltstr* pfp,
+                                  double* lags,
+                                  double* gains);
+
+void WebRtcIsac_PitchfilterPre_gains(
+    double* indat,
+    double* outdat,
+    double out_dG[][PITCH_FRAME_LEN + QLOOKAHEAD],
+    PitchFiltstr* pfp,
+    double* lags,
+    double* gains);
+
+#endif  // MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_FILTER_H_

VocieProcess/modules/audio_coding/codecs/isac/main/source/settings.h

@@ -0,0 +1,196 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * settings.h
+ *
+ * Declaration of #defines used in the iSAC codec
+ *
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_SETTINGS_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_SETTINGS_H_
+
+/* sampling frequency (Hz) */
+#define FS 16000
+
+/* number of samples per frame (either 320 (20ms), 480 (30ms) or 960 (60ms)) */
+#define INITIAL_FRAMESAMPLES 960
+
+/* do not modify the following; this will have to be modified if we
+ * have a 20ms framesize option */
+/**********************************************************************/
+/* miliseconds */
+#define FRAMESIZE 30
+/* number of samples per frame processed in the encoder, 480 */
+#define FRAMESAMPLES 480 /* ((FRAMESIZE*FS)/1000) */
+#define FRAMESAMPLES_HALF 240
+#define FRAMESAMPLES_QUARTER 120
+/**********************************************************************/
+
+/* max number of samples per frame (= 60 ms frame) */
+#define MAX_FRAMESAMPLES 960
+#define MAX_SWBFRAMESAMPLES (MAX_FRAMESAMPLES * 2)
+/* number of samples per 10ms frame */
+#define FRAMESAMPLES_10ms ((10 * FS) / 1000)
+#define SWBFRAMESAMPLES_10ms (FRAMESAMPLES_10ms * 2)
+/* number of samples in 30 ms frame */
+#define FRAMESAMPLES_30ms 480
+/* number of subframes */
+#define SUBFRAMES 6
+/* length of a subframe */
+#define UPDATE 80
+/* length of half a subframe (low/high band) */
+#define HALF_SUBFRAMELEN (UPDATE / 2)
+/* samples of look ahead (in a half-band, so actually
+ * half the samples of look ahead @ FS) */
+#define QLOOKAHEAD 24 /* 3 ms */
+/* order of AR model in spectral entropy coder */
+#define AR_ORDER 6
+/* order of LP model in spectral entropy coder */
+#define LP_ORDER 0
+
+/* window length (masking analysis) */
+#define WINLEN 256
+/* order of low-band pole filter used to approximate masking curve */
+#define ORDERLO 12
+/* order of hi-band pole filter used to approximate masking curve */
+#define ORDERHI 6
+
+#define UB_LPC_ORDER 4
+#define UB_LPC_VEC_PER_FRAME 2
+#define UB16_LPC_VEC_PER_FRAME 4
+#define UB_ACTIVE_SUBFRAMES 2
+#define UB_MAX_LPC_ORDER 6
+#define UB_INTERPOL_SEGMENTS 1
+#define UB16_INTERPOL_SEGMENTS 3
+#define LB_TOTAL_DELAY_SAMPLES 48
+enum ISACBandwidth { isac8kHz = 8, isac12kHz = 12, isac16kHz = 16 };
+enum ISACBand {
+  kIsacLowerBand = 0,
+  kIsacUpperBand12 = 1,
+  kIsacUpperBand16 = 2
+};
+enum IsacSamplingRate { kIsacWideband = 16, kIsacSuperWideband = 32 };
+#define UB_LPC_GAIN_DIM SUBFRAMES
+#define FB_STATE_SIZE_WORD32 6
+
+/* order for post_filter_bank */
+#define POSTQORDER 3
+/* order for pre-filterbank */
+#define QORDER 3
+/* another order */
+#define QORDER_ALL (POSTQORDER + QORDER - 1)
+/* for decimator */
+#define ALLPASSSECTIONS 2
+
+/* array size for byte stream in number of bytes. */
+/* The old maximum size still needed for the decoding */
+#define STREAM_SIZE_MAX 600
+#define STREAM_SIZE_MAX_30 200 /* 200 bytes=53.4 kbps @ 30 ms.framelength */
+#define STREAM_SIZE_MAX_60 400 /* 400 bytes=53.4 kbps @ 60 ms.framelength */
+
+/* storage size for bit counts */
+#define BIT_COUNTER_SIZE 30
+/* maximum order of any AR model or filter */
+#define MAX_AR_MODEL_ORDER 12  // 50
+
+/* For pitch analysis */
+#define PITCH_FRAME_LEN (FRAMESAMPLES_HALF) /* 30 ms  */
+#define PITCH_MAX_LAG 140                   /* 57 Hz  */
+#define PITCH_MIN_LAG 20                    /* 400 Hz */
+#define PITCH_MAX_GAIN 0.45
+#define PITCH_MAX_GAIN_06 0.27 /* PITCH_MAX_GAIN*0.6 */
+#define PITCH_MAX_GAIN_Q12 1843
+#define PITCH_LAG_SPAN2 (PITCH_MAX_LAG / 2 - PITCH_MIN_LAG / 2 + 5)
+#define PITCH_CORR_LEN2 60 /* 15 ms  */
+#define PITCH_CORR_STEP2 (PITCH_FRAME_LEN / 4)
+#define PITCH_BW 11 /* half the band width of correlation surface */
+#define PITCH_SUBFRAMES 4
+#define PITCH_GRAN_PER_SUBFRAME 5
+#define PITCH_SUBFRAME_LEN (PITCH_FRAME_LEN / PITCH_SUBFRAMES)
+#define PITCH_UPDATE (PITCH_SUBFRAME_LEN / PITCH_GRAN_PER_SUBFRAME)
+/* maximum number of peaks to be examined in correlation surface */
+#define PITCH_MAX_NUM_PEAKS 10
+#define PITCH_PEAK_DECAY 0.85
+/* For weighting filter */
+#define PITCH_WLPCORDER 6
+#define PITCH_WLPCWINLEN PITCH_FRAME_LEN
+#define PITCH_WLPCASYM 0.3 /* asymmetry parameter */
+#define PITCH_WLPCBUFLEN PITCH_WLPCWINLEN
+/* For pitch filter */
+/* Extra 50 for fraction and LP filters */
+#define PITCH_BUFFSIZE (PITCH_MAX_LAG + 50)
+#define PITCH_INTBUFFSIZE (PITCH_FRAME_LEN + PITCH_BUFFSIZE)
+/* Max rel. step for interpolation */
+#define PITCH_UPSTEP 1.5
+/* Max rel. step for interpolation */
+#define PITCH_DOWNSTEP 0.67
+#define PITCH_FRACS 8
+#define PITCH_FRACORDER 9
+#define PITCH_DAMPORDER 5
+#define PITCH_FILTDELAY 1.5f
+/* stepsize for quantization of the pitch Gain */
+#define PITCH_GAIN_STEPSIZE 0.125
+
+/* Order of high pass filter */
+#define HPORDER 2
+
+/* some mathematical constants */
+/* log2(exp) */
+#define LOG2EXP 1.44269504088896
+#define PI 3.14159265358979
+
+/* Maximum number of iterations allowed to limit payload size */
+#define MAX_PAYLOAD_LIMIT_ITERATION 5
+
+/* Redundant Coding */
+#define RCU_BOTTLENECK_BPS 16000
+#define RCU_TRANSCODING_SCALE 0.40f
+#define RCU_TRANSCODING_SCALE_INVERSE 2.5f
+
+#define RCU_TRANSCODING_SCALE_UB 0.50f
+#define RCU_TRANSCODING_SCALE_UB_INVERSE 2.0f
+
+/* Define Error codes */
+/* 6000 General */
+#define ISAC_MEMORY_ALLOCATION_FAILED 6010
+#define ISAC_MODE_MISMATCH 6020
+#define ISAC_DISALLOWED_BOTTLENECK 6030
+#define ISAC_DISALLOWED_FRAME_LENGTH 6040
+#define ISAC_UNSUPPORTED_SAMPLING_FREQUENCY 6050
+
+/* 6200 Bandwidth estimator */
+#define ISAC_RANGE_ERROR_BW_ESTIMATOR 6240
+/* 6400 Encoder */
+#define ISAC_ENCODER_NOT_INITIATED 6410
+#define ISAC_DISALLOWED_CODING_MODE 6420
+#define ISAC_DISALLOWED_FRAME_MODE_ENCODER 6430
+#define ISAC_DISALLOWED_BITSTREAM_LENGTH 6440
+#define ISAC_PAYLOAD_LARGER_THAN_LIMIT 6450
+#define ISAC_DISALLOWED_ENCODER_BANDWIDTH 6460
+/* 6600 Decoder */
+#define ISAC_DECODER_NOT_INITIATED 6610
+#define ISAC_EMPTY_PACKET 6620
+#define ISAC_DISALLOWED_FRAME_MODE_DECODER 6630
+#define ISAC_RANGE_ERROR_DECODE_FRAME_LENGTH 6640
+#define ISAC_RANGE_ERROR_DECODE_BANDWIDTH 6650
+#define ISAC_RANGE_ERROR_DECODE_PITCH_GAIN 6660
+#define ISAC_RANGE_ERROR_DECODE_PITCH_LAG 6670
+#define ISAC_RANGE_ERROR_DECODE_LPC 6680
+#define ISAC_RANGE_ERROR_DECODE_SPECTRUM 6690
+#define ISAC_LENGTH_MISMATCH 6730
+#define ISAC_RANGE_ERROR_DECODE_BANDWITH 6740
+#define ISAC_DISALLOWED_BANDWIDTH_MODE_DECODER 6750
+#define ISAC_DISALLOWED_LPC_MODEL 6760
+/* 6800 Call setup formats */
+#define ISAC_INCOMPATIBLE_FORMATS 6810
+
+#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_SETTINGS_H_ */

VocieProcess/modules/audio_coding/codecs/isac/main/source/structs.h

@@ -0,0 +1,448 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * structs.h
+ *
+ * This header file contains all the structs used in the ISAC codec
+ *
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_STRUCTS_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_STRUCTS_H_
+
+#include "modules/audio_coding/codecs/isac/bandwidth_info.h"
+#include "modules/audio_coding/codecs/isac/main/source/settings.h"
+#include "modules/third_party/fft/fft.h"
+
+typedef struct Bitstreamstruct {
+  uint8_t stream[STREAM_SIZE_MAX];
+  uint32_t W_upper;
+  uint32_t streamval;
+  uint32_t stream_index;
+
+} Bitstr;
+
+typedef struct {
+  double DataBufferLo[WINLEN];
+  double DataBufferHi[WINLEN];
+
+  double CorrBufLo[ORDERLO + 1];
+  double CorrBufHi[ORDERHI + 1];
+
+  float PreStateLoF[ORDERLO + 1];
+  float PreStateLoG[ORDERLO + 1];
+  float PreStateHiF[ORDERHI + 1];
+  float PreStateHiG[ORDERHI + 1];
+  float PostStateLoF[ORDERLO + 1];
+  float PostStateLoG[ORDERLO + 1];
+  float PostStateHiF[ORDERHI + 1];
+  float PostStateHiG[ORDERHI + 1];
+
+  double OldEnergy;
+
+} MaskFiltstr;
+
+typedef struct {
+  // state vectors for each of the two analysis filters
+  double INSTAT1[2 * (QORDER - 1)];
+  double INSTAT2[2 * (QORDER - 1)];
+  double INSTATLA1[2 * (QORDER - 1)];
+  double INSTATLA2[2 * (QORDER - 1)];
+  double INLABUF1[QLOOKAHEAD];
+  double INLABUF2[QLOOKAHEAD];
+
+  float INSTAT1_float[2 * (QORDER - 1)];
+  float INSTAT2_float[2 * (QORDER - 1)];
+  float INSTATLA1_float[2 * (QORDER - 1)];
+  float INSTATLA2_float[2 * (QORDER - 1)];
+  float INLABUF1_float[QLOOKAHEAD];
+  float INLABUF2_float[QLOOKAHEAD];
+
+  /* High pass filter */
+  double HPstates[HPORDER];
+  float HPstates_float[HPORDER];
+
+} PreFiltBankstr;
+
+typedef struct {
+  // state vectors for each of the two analysis filters
+  double STATE_0_LOWER[2 * POSTQORDER];
+  double STATE_0_UPPER[2 * POSTQORDER];
+
+  /* High pass filter */
+  double HPstates1[HPORDER];
+  double HPstates2[HPORDER];
+
+  float STATE_0_LOWER_float[2 * POSTQORDER];
+  float STATE_0_UPPER_float[2 * POSTQORDER];
+
+  float HPstates1_float[HPORDER];
+  float HPstates2_float[HPORDER];
+
+} PostFiltBankstr;
+
+typedef struct {
+  // data buffer for pitch filter
+  double ubuf[PITCH_BUFFSIZE];
+
+  // low pass state vector
+  double ystate[PITCH_DAMPORDER];
+
+  // old lag and gain
+  double oldlagp[1];
+  double oldgainp[1];
+
+} PitchFiltstr;
+
+typedef struct {
+  // data buffer
+  double buffer[PITCH_WLPCBUFLEN];
+
+  // state vectors
+  double istate[PITCH_WLPCORDER];
+  double weostate[PITCH_WLPCORDER];
+  double whostate[PITCH_WLPCORDER];
+
+  // LPC window   -> should be a global array because constant
+  double window[PITCH_WLPCWINLEN];
+
+} WeightFiltstr;
+
+typedef struct {
+  // for inital estimator
+  double dec_buffer[PITCH_CORR_LEN2 + PITCH_CORR_STEP2 + PITCH_MAX_LAG / 2 -
+                    PITCH_FRAME_LEN / 2 + 2];
+  double decimator_state[2 * ALLPASSSECTIONS + 1];
+  double hp_state[2];
+
+  double whitened_buf[QLOOKAHEAD];
+
+  double inbuf[QLOOKAHEAD];
+
+  PitchFiltstr PFstr_wght;
+  PitchFiltstr PFstr;
+  WeightFiltstr Wghtstr;
+
+} PitchAnalysisStruct;
+
+/* Have instance of struct together with other iSAC structs */
+typedef struct {
+  /* Previous frame length (in ms)                                    */
+  int32_t prev_frame_length;
+
+  /* Previous RTP timestamp from received
+     packet (in samples relative beginning)                           */
+  int32_t prev_rec_rtp_number;
+
+  /* Send timestamp for previous packet (in ms using timeGetTime())   */
+  uint32_t prev_rec_send_ts;
+
+  /* Arrival time for previous packet (in ms using timeGetTime())     */
+  uint32_t prev_rec_arr_ts;
+
+  /* rate of previous packet, derived from RTP timestamps (in bits/s) */
+  float prev_rec_rtp_rate;
+
+  /* Time sinse the last update of the BN estimate (in ms)            */
+  uint32_t last_update_ts;
+
+  /* Time sinse the last reduction (in ms)                            */
+  uint32_t last_reduction_ts;
+
+  /* How many times the estimate was update in the beginning          */
+  int32_t count_tot_updates_rec;
+
+  /* The estimated bottle neck rate from there to here (in bits/s)    */
+  int32_t rec_bw;
+  float rec_bw_inv;
+  float rec_bw_avg;
+  float rec_bw_avg_Q;
+
+  /* The estimated mean absolute jitter value,
+     as seen on this side (in ms)                                     */
+  float rec_jitter;
+  float rec_jitter_short_term;
+  float rec_jitter_short_term_abs;
+  float rec_max_delay;
+  float rec_max_delay_avg_Q;
+
+  /* (assumed) bitrate for headers (bps)                              */
+  float rec_header_rate;
+
+  /* The estimated bottle neck rate from here to there (in bits/s)    */
+  float send_bw_avg;
+
+  /* The estimated mean absolute jitter value, as seen on
+     the other siee (in ms)                                           */
+  float send_max_delay_avg;
+
+  // number of packets received since last update
+  int num_pkts_rec;
+
+  int num_consec_rec_pkts_over_30k;
+
+  // flag for marking that a high speed network has been
+  // detected downstream
+  int hsn_detect_rec;
+
+  int num_consec_snt_pkts_over_30k;
+
+  // flag for marking that a high speed network has
+  // been detected upstream
+  int hsn_detect_snd;
+
+  uint32_t start_wait_period;
+
+  int in_wait_period;
+
+  int change_to_WB;
+
+  uint32_t senderTimestamp;
+  uint32_t receiverTimestamp;
+  // enum IsacSamplingRate incomingStreamSampFreq;
+  uint16_t numConsecLatePkts;
+  float consecLatency;
+  int16_t inWaitLatePkts;
+
+  IsacBandwidthInfo external_bw_info;
+} BwEstimatorstr;
+
+typedef struct {
+  /* boolean, flags if previous packet exceeded B.N. */
+  int PrevExceed;
+  /* ms */
+  int ExceedAgo;
+  /* packets left to send in current burst */
+  int BurstCounter;
+  /* packets */
+  int InitCounter;
+  /* ms remaining in buffer when next packet will be sent */
+  double StillBuffered;
+
+} RateModel;
+
+/* The following strutc is used to store data from encoding, to make it
+   fast and easy to construct a new bitstream with a different Bandwidth
+   estimate. All values (except framelength and minBytes) is double size to
+   handle 60 ms of data.
+*/
+typedef struct {
+  /* Used to keep track of if it is first or second part of 60 msec packet */
+  int startIdx;
+
+  /* Frame length in samples */
+  int16_t framelength;
+
+  /* Pitch Gain */
+  int pitchGain_index[2];
+
+  /* Pitch Lag */
+  double meanGain[2];
+  int pitchIndex[PITCH_SUBFRAMES * 2];
+
+  /* LPC */
+  int LPCindex_s[108 * 2]; /* KLT_ORDER_SHAPE = 108 */
+  int LPCindex_g[12 * 2];  /* KLT_ORDER_GAIN = 12 */
+  double LPCcoeffs_lo[(ORDERLO + 1) * SUBFRAMES * 2];
+  double LPCcoeffs_hi[(ORDERHI + 1) * SUBFRAMES * 2];
+
+  /* Encode Spec */
+  int16_t fre[FRAMESAMPLES];
+  int16_t fim[FRAMESAMPLES];
+  int16_t AvgPitchGain[2];
+
+  /* Used in adaptive mode only */
+  int minBytes;
+
+} IsacSaveEncoderData;
+
+typedef struct {
+  int indexLPCShape[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME];
+  double lpcGain[SUBFRAMES << 1];
+  int lpcGainIndex[SUBFRAMES << 1];
+
+  Bitstr bitStreamObj;
+
+  int16_t realFFT[FRAMESAMPLES_HALF];
+  int16_t imagFFT[FRAMESAMPLES_HALF];
+} ISACUBSaveEncDataStruct;
+
+typedef struct {
+  Bitstr bitstr_obj;
+  MaskFiltstr maskfiltstr_obj;
+  PreFiltBankstr prefiltbankstr_obj;
+  PitchFiltstr pitchfiltstr_obj;
+  PitchAnalysisStruct pitchanalysisstr_obj;
+  FFTstr fftstr_obj;
+  IsacSaveEncoderData SaveEnc_obj;
+
+  int buffer_index;
+  int16_t current_framesamples;
+
+  float data_buffer_float[FRAMESAMPLES_30ms];
+
+  int frame_nb;
+  double bottleneck;
+  int16_t new_framelength;
+  double s2nr;
+
+  /* Maximum allowed number of bits for a 30 msec packet */
+  int16_t payloadLimitBytes30;
+  /* Maximum allowed number of bits for a 30 msec packet */
+  int16_t payloadLimitBytes60;
+  /* Maximum allowed number of bits for both 30 and 60 msec packet */
+  int16_t maxPayloadBytes;
+  /* Maximum allowed rate in bytes per 30 msec packet */
+  int16_t maxRateInBytes;
+
+  /*---
+    If set to 1 iSAC will not adapt the frame-size, if used in
+    channel-adaptive mode. The initial value will be used for all rates.
+    ---*/
+  int16_t enforceFrameSize;
+
+  /*-----
+    This records the BWE index the encoder injected into the bit-stream.
+    It will be used in RCU. The same BWE index of main payload will be in
+    the redundant payload. We can not retrieve it from BWE because it is
+    a recursive procedure (WebRtcIsac_GetDownlinkBwJitIndexImpl) and has to be
+    called only once per each encode.
+    -----*/
+  int16_t lastBWIdx;
+} ISACLBEncStruct;
+
+typedef struct {
+  Bitstr bitstr_obj;
+  MaskFiltstr maskfiltstr_obj;
+  PreFiltBankstr prefiltbankstr_obj;
+  FFTstr fftstr_obj;
+  ISACUBSaveEncDataStruct SaveEnc_obj;
+
+  int buffer_index;
+  float data_buffer_float[MAX_FRAMESAMPLES + LB_TOTAL_DELAY_SAMPLES];
+  double bottleneck;
+  /* Maximum allowed number of bits for a 30 msec packet */
+  // int16_t        payloadLimitBytes30;
+  /* Maximum allowed number of bits for both 30 and 60 msec packet */
+  // int16_t        maxPayloadBytes;
+  int16_t maxPayloadSizeBytes;
+
+  double lastLPCVec[UB_LPC_ORDER];
+  int16_t numBytesUsed;
+  int16_t lastJitterInfo;
+} ISACUBEncStruct;
+
+typedef struct {
+  Bitstr bitstr_obj;
+  MaskFiltstr maskfiltstr_obj;
+  PostFiltBankstr postfiltbankstr_obj;
+  PitchFiltstr pitchfiltstr_obj;
+  FFTstr fftstr_obj;
+
+} ISACLBDecStruct;
+
+typedef struct {
+  Bitstr bitstr_obj;
+  MaskFiltstr maskfiltstr_obj;
+  PostFiltBankstr postfiltbankstr_obj;
+  FFTstr fftstr_obj;
+
+} ISACUBDecStruct;
+
+typedef struct {
+  ISACLBEncStruct ISACencLB_obj;
+  ISACLBDecStruct ISACdecLB_obj;
+} ISACLBStruct;
+
+typedef struct {
+  ISACUBEncStruct ISACencUB_obj;
+  ISACUBDecStruct ISACdecUB_obj;
+} ISACUBStruct;
+
+/*
+  This struct is used to take a snapshot of the entropy coder and LPC gains
+  right before encoding LPC gains. This allows us to go back to that state
+  if we like to limit the payload size.
+*/
+typedef struct {
+  /* 6 lower-band & 6 upper-band */
+  double loFiltGain[SUBFRAMES];
+  double hiFiltGain[SUBFRAMES];
+  /* Upper boundary of interval W */
+  uint32_t W_upper;
+  uint32_t streamval;
+  /* Index to the current position in bytestream */
+  uint32_t stream_index;
+  uint8_t stream[3];
+} transcode_obj;
+
+typedef struct {
+  // TODO(kwiberg): The size of these tables could be reduced by storing floats
+  // instead of doubles, and by making use of the identity cos(x) =
+  // sin(x+pi/2). They could also be made global constants that we fill in at
+  // compile time.
+  double costab1[FRAMESAMPLES_HALF];
+  double sintab1[FRAMESAMPLES_HALF];
+  double costab2[FRAMESAMPLES_QUARTER];
+  double sintab2[FRAMESAMPLES_QUARTER];
+} TransformTables;
+
+typedef struct {
+  // lower-band codec instance
+  ISACLBStruct instLB;
+  // upper-band codec instance
+  ISACUBStruct instUB;
+
+  // Bandwidth Estimator and model for the rate.
+  BwEstimatorstr bwestimator_obj;
+  RateModel rate_data_obj;
+  double MaxDelay;
+
+  /* 0 = adaptive; 1 = instantaneous */
+  int16_t codingMode;
+
+  // overall bottleneck of the codec
+  int32_t bottleneck;
+
+  // QMF Filter state
+  int32_t analysisFBState1[FB_STATE_SIZE_WORD32];
+  int32_t analysisFBState2[FB_STATE_SIZE_WORD32];
+  int32_t synthesisFBState1[FB_STATE_SIZE_WORD32];
+  int32_t synthesisFBState2[FB_STATE_SIZE_WORD32];
+
+  // Error Code
+  int16_t errorCode;
+
+  // bandwidth of the encoded audio 8, 12 or 16 kHz
+  enum ISACBandwidth bandwidthKHz;
+  // Sampling rate of audio, encoder and decode,  8 or 16 kHz
+  enum IsacSamplingRate encoderSamplingRateKHz;
+  enum IsacSamplingRate decoderSamplingRateKHz;
+  // Flag to keep track of initializations, lower & upper-band
+  // encoder and decoder.
+  int16_t initFlag;
+
+  // Flag to to indicate signal bandwidth switch
+  int16_t resetFlag_8kHz;
+
+  // Maximum allowed rate, measured in Bytes per 30 ms.
+  int16_t maxRateBytesPer30Ms;
+  // Maximum allowed payload-size, measured in Bytes.
+  int16_t maxPayloadSizeBytes;
+  /* The expected sampling rate of the input signal. Valid values are 16000
+   * and 32000. This is not the operation sampling rate of the codec. */
+  uint16_t in_sample_rate_hz;
+
+  // Trig tables for WebRtcIsac_Time2Spec and WebRtcIsac_Spec2time.
+  TransformTables transform_tables;
+} ISACMainStruct;
+
+#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_STRUCTS_H_ */

VocieProcess/modules/audio_processing/aec_dump/aec_dump_factory.h

@@ -0,0 +1,48 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC_DUMP_AEC_DUMP_FACTORY_H_
+#define MODULES_AUDIO_PROCESSING_AEC_DUMP_AEC_DUMP_FACTORY_H_
+
+#include <memory>
+
+#include "absl/base/nullability.h"
+#include "absl/strings/string_view.h"
+#include "api/task_queue/task_queue_base.h"
+#include "modules/audio_processing/include/aec_dump.h"
+#include "rtc_base/system/file_wrapper.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+class RTC_EXPORT AecDumpFactory {
+ public:
+  // The `worker_queue` must outlive the created AecDump instance.
+  // `max_log_size_bytes == -1` means the log size will be unlimited.
+  // The AecDump takes responsibility for `handle` and closes it in the
+  // destructor. A non-null return value indicates that the file has been
+  // sucessfully opened.
+  static absl::Nullable<std::unique_ptr<AecDump>> Create(
+      FileWrapper file,
+      int64_t max_log_size_bytes,
+      absl::Nonnull<TaskQueueBase*> worker_queue);
+  static absl::Nullable<std::unique_ptr<AecDump>> Create(
+      absl::string_view file_name,
+      int64_t max_log_size_bytes,
+      absl::Nonnull<TaskQueueBase*> worker_queue);
+  static absl::Nullable<std::unique_ptr<AecDump>> Create(
+      absl::Nonnull<FILE*> handle,
+      int64_t max_log_size_bytes,
+      absl::Nonnull<TaskQueueBase*> worker_queue);
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC_DUMP_AEC_DUMP_FACTORY_H_

VocieProcess/modules/audio_processing/capture_levels_adjuster/audio_samples_scaler.cc

@@ -0,0 +1,92 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/audio_processing/capture_levels_adjuster/audio_samples_scaler.h"
+
+#include <algorithm>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/audio_buffer.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_minmax.h"
+
+namespace webrtc {
+
+AudioSamplesScaler::AudioSamplesScaler(float initial_gain)
+    : previous_gain_(initial_gain), target_gain_(initial_gain) {}
+
+void AudioSamplesScaler::Process(AudioBuffer& audio_buffer) {
+  if (static_cast<int>(audio_buffer.num_frames()) != samples_per_channel_) {
+    // Update the members depending on audio-buffer length if needed.
+    RTC_DCHECK_GT(audio_buffer.num_frames(), 0);
+    samples_per_channel_ = static_cast<int>(audio_buffer.num_frames());
+    one_by_samples_per_channel_ = 1.f / samples_per_channel_;
+  }
+
+  if (target_gain_ == 1.f && previous_gain_ == target_gain_) {
+    // If only a gain of 1 is to be applied, do an early return without applying
+    // any gain.
+    return;
+  }
+
+  float gain = previous_gain_;
+  if (previous_gain_ == target_gain_) {
+    // Apply a non-changing gain.
+    for (size_t channel = 0; channel < audio_buffer.num_channels(); ++channel) {
+      rtc::ArrayView<float> channel_view(audio_buffer.channels()[channel],
+                                         samples_per_channel_);
+      for (float& sample : channel_view) {
+        sample *= gain;
+      }
+    }
+  } else {
+    const float increment =
+        (target_gain_ - previous_gain_) * one_by_samples_per_channel_;
+
+    if (increment > 0.f) {
+      // Apply an increasing gain.
+      for (size_t channel = 0; channel < audio_buffer.num_channels();
+           ++channel) {
+        gain = previous_gain_;
+        rtc::ArrayView<float> channel_view(audio_buffer.channels()[channel],
+                                           samples_per_channel_);
+        for (float& sample : channel_view) {
+          gain = std::min(gain + increment, target_gain_);
+          sample *= gain;
+        }
+      }
+    } else {
+      // Apply a decreasing gain.
+      for (size_t channel = 0; channel < audio_buffer.num_channels();
+           ++channel) {
+        gain = previous_gain_;
+        rtc::ArrayView<float> channel_view(audio_buffer.channels()[channel],
+                                           samples_per_channel_);
+        for (float& sample : channel_view) {
+          gain = std::max(gain + increment, target_gain_);
+          sample *= gain;
+        }
+      }
+    }
+  }
+  previous_gain_ = target_gain_;
+
+  // Saturate the samples to be in the S16 range.
+  for (size_t channel = 0; channel < audio_buffer.num_channels(); ++channel) {
+    rtc::ArrayView<float> channel_view(audio_buffer.channels()[channel],
+                                       samples_per_channel_);
+    for (float& sample : channel_view) {
+      constexpr float kMinFloatS16Value = -32768.f;
+      constexpr float kMaxFloatS16Value = 32767.f;
+      sample = rtc::SafeClamp(sample, kMinFloatS16Value, kMaxFloatS16Value);
+    }
+  }
+}
+
+}  // namespace webrtc

VocieProcess/modules/audio_processing/capture_levels_adjuster/audio_samples_scaler.h

@@ -0,0 +1,46 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_CAPTURE_LEVELS_ADJUSTER_AUDIO_SAMPLES_SCALER_H_
+#define MODULES_AUDIO_PROCESSING_CAPTURE_LEVELS_ADJUSTER_AUDIO_SAMPLES_SCALER_H_
+
+#include <stddef.h>
+
+#include "modules/audio_processing/audio_buffer.h"
+
+namespace webrtc {
+
+// Handles and applies a gain to the samples in an audio buffer.
+// The gain is applied for each sample and any changes in the gain take effect
+// gradually (in a linear manner) over one frame.
+class AudioSamplesScaler {
+ public:
+  // C-tor. The supplied `initial_gain` is used immediately at the first call to
+  // Process(), i.e., in contrast to the gain supplied by SetGain(...) there is
+  // no gradual change to the `initial_gain`.
+  explicit AudioSamplesScaler(float initial_gain);
+  AudioSamplesScaler(const AudioSamplesScaler&) = delete;
+  AudioSamplesScaler& operator=(const AudioSamplesScaler&) = delete;
+
+  // Applies the specified gain to the audio in `audio_buffer`.
+  void Process(AudioBuffer& audio_buffer);
+
+  // Sets the gain to apply to each sample.
+  void SetGain(float gain) { target_gain_ = gain; }
+
+ private:
+  float previous_gain_ = 1.f;
+  float target_gain_ = 1.f;
+  int samples_per_channel_ = -1;
+  float one_by_samples_per_channel_ = -1.f;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_CAPTURE_LEVELS_ADJUSTER_AUDIO_SAMPLES_SCALER_H_

VocieProcess/modules/audio_processing/capture_levels_adjuster/capture_levels_adjuster.cc

@@ -0,0 +1,96 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/audio_processing/capture_levels_adjuster/capture_levels_adjuster.h"
+
+#include "modules/audio_processing/audio_buffer.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_minmax.h"
+
+namespace webrtc {
+
+namespace {
+
+constexpr int kMinAnalogMicGainLevel = 0;
+constexpr int kMaxAnalogMicGainLevel = 255;
+
+float ComputeLevelBasedGain(int emulated_analog_mic_gain_level) {
+  static_assert(
+      kMinAnalogMicGainLevel == 0,
+      "The minimum gain level must be 0 for the maths below to work.");
+  static_assert(kMaxAnalogMicGainLevel > 0,
+                "The minimum gain level must be larger than 0 for the maths "
+                "below to work.");
+  constexpr float kGainToLevelMultiplier = 1.f / kMaxAnalogMicGainLevel;
+
+  RTC_DCHECK_GE(emulated_analog_mic_gain_level, kMinAnalogMicGainLevel);
+  RTC_DCHECK_LE(emulated_analog_mic_gain_level, kMaxAnalogMicGainLevel);
+  return kGainToLevelMultiplier * emulated_analog_mic_gain_level;
+}
+
+float ComputePreGain(float pre_gain,
+                     int emulated_analog_mic_gain_level,
+                     bool emulated_analog_mic_gain_enabled) {
+  return emulated_analog_mic_gain_enabled
+             ? pre_gain * ComputeLevelBasedGain(emulated_analog_mic_gain_level)
+             : pre_gain;
+}
+
+}  // namespace
+
+CaptureLevelsAdjuster::CaptureLevelsAdjuster(
+    bool emulated_analog_mic_gain_enabled,
+    int emulated_analog_mic_gain_level,
+    float pre_gain,
+    float post_gain)
+    : emulated_analog_mic_gain_enabled_(emulated_analog_mic_gain_enabled),
+      emulated_analog_mic_gain_level_(emulated_analog_mic_gain_level),
+      pre_gain_(pre_gain),
+      pre_adjustment_gain_(ComputePreGain(pre_gain_,
+                                          emulated_analog_mic_gain_level_,
+                                          emulated_analog_mic_gain_enabled_)),
+      pre_scaler_(pre_adjustment_gain_),
+      post_scaler_(post_gain) {}
+
+void CaptureLevelsAdjuster::ApplyPreLevelAdjustment(AudioBuffer& audio_buffer) {
+  pre_scaler_.Process(audio_buffer);
+}
+
+void CaptureLevelsAdjuster::ApplyPostLevelAdjustment(
+    AudioBuffer& audio_buffer) {
+  post_scaler_.Process(audio_buffer);
+}
+
+void CaptureLevelsAdjuster::SetPreGain(float pre_gain) {
+  pre_gain_ = pre_gain;
+  UpdatePreAdjustmentGain();
+}
+
+void CaptureLevelsAdjuster::SetPostGain(float post_gain) {
+  post_scaler_.SetGain(post_gain);
+}
+
+void CaptureLevelsAdjuster::SetAnalogMicGainLevel(int level) {
+  RTC_DCHECK_GE(level, kMinAnalogMicGainLevel);
+  RTC_DCHECK_LE(level, kMaxAnalogMicGainLevel);
+  int clamped_level =
+      rtc::SafeClamp(level, kMinAnalogMicGainLevel, kMaxAnalogMicGainLevel);
+
+  emulated_analog_mic_gain_level_ = clamped_level;
+  UpdatePreAdjustmentGain();
+}
+
+void CaptureLevelsAdjuster::UpdatePreAdjustmentGain() {
+  pre_adjustment_gain_ =
+      ComputePreGain(pre_gain_, emulated_analog_mic_gain_level_,
+                     emulated_analog_mic_gain_enabled_);
+  pre_scaler_.SetGain(pre_adjustment_gain_);
+}
+
+}  // namespace webrtc

VocieProcess/modules/audio_processing/capture_levels_adjuster/capture_levels_adjuster.h

@@ -0,0 +1,88 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_AUDIO_PROCESSING_CAPTURE_LEVELS_ADJUSTER_CAPTURE_LEVELS_ADJUSTER_H_
+#define MODULES_AUDIO_PROCESSING_CAPTURE_LEVELS_ADJUSTER_CAPTURE_LEVELS_ADJUSTER_H_
+
+#include <stddef.h>
+
+#include "modules/audio_processing/audio_buffer.h"
+#include "modules/audio_processing/capture_levels_adjuster/audio_samples_scaler.h"
+
+namespace webrtc {
+
+// Adjusts the level of the capture signal before and after all capture-side
+// processing is done using a combination of explicitly specified gains
+// and an emulated analog gain functionality where a specified analog level
+// results in an additional gain. The pre-adjustment is achieved by combining
+// the gain value `pre_gain` and the level `emulated_analog_mic_gain_level` to
+// form a combined gain of `pre_gain`*`emulated_analog_mic_gain_level`/255 which
+// is multiplied to each sample. The intention of the
+// `emulated_analog_mic_gain_level` is to be controlled by the analog AGC
+// functionality and to produce an emulated analog mic gain equal to
+// `emulated_analog_mic_gain_level`/255. The post level adjustment is achieved
+// by multiplying each sample with the value of `post_gain`. Any changes in the
+// gains take are done smoothly over one frame and the scaled samples are
+// clamped to fit into the allowed S16 sample range.
+class CaptureLevelsAdjuster {
+ public:
+  // C-tor. The values for the level and the gains must fulfill
+  // 0 <= emulated_analog_mic_gain_level <= 255.
+  // 0.f <= pre_gain.
+  // 0.f <= post_gain.
+  CaptureLevelsAdjuster(bool emulated_analog_mic_gain_enabled,
+                        int emulated_analog_mic_gain_level,
+                        float pre_gain,
+                        float post_gain);
+  CaptureLevelsAdjuster(const CaptureLevelsAdjuster&) = delete;
+  CaptureLevelsAdjuster& operator=(const CaptureLevelsAdjuster&) = delete;
+
+  // Adjusts the level of the signal. This should be called before any of the
+  // other processing is performed.
+  void ApplyPreLevelAdjustment(AudioBuffer& audio_buffer);
+
+  // Adjusts the level of the signal. This should be called after all of the
+  // other processing have been performed.
+  void ApplyPostLevelAdjustment(AudioBuffer& audio_buffer);
+
+  // Sets the gain to apply to each sample before any of the other processing is
+  // performed.
+  void SetPreGain(float pre_gain);
+
+  // Returns the total pre-adjustment gain applied, comprising both the pre_gain
+  // as well as the gain from the emulated analog mic, to each sample before any
+  // of the other processing is performed.
+  float GetPreAdjustmentGain() const { return pre_adjustment_gain_; }
+
+  // Sets the gain to apply to each sample after all of the other processing
+  // have been performed.
+  void SetPostGain(float post_gain);
+
+  // Sets the analog gain level to use for the emulated analog gain.
+  // `level` must be in the range [0...255].
+  void SetAnalogMicGainLevel(int level);
+
+  // Returns the current analog gain level used for the emulated analog gain.
+  int GetAnalogMicGainLevel() const { return emulated_analog_mic_gain_level_; }
+
+ private:
+  // Updates the value of `pre_adjustment_gain_` based on the supplied values
+  // for `pre_gain` and `emulated_analog_mic_gain_level_`.
+  void UpdatePreAdjustmentGain();
+
+  const bool emulated_analog_mic_gain_enabled_;
+  int emulated_analog_mic_gain_level_;
+  float pre_gain_;
+  float pre_adjustment_gain_;
+  AudioSamplesScaler pre_scaler_;
+  AudioSamplesScaler post_scaler_;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_CAPTURE_LEVELS_ADJUSTER_CAPTURE_LEVELS_ADJUSTER_H_

VocieProcess/modules/audio_processing/echo_control_mobile_impl.cc

@@ -0,0 +1,287 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/echo_control_mobile_impl.h"
+
+#include <string.h>
+
+#include <cstdint>
+
+#include "api/audio/audio_processing.h"
+#include "modules/audio_processing/aecm/echo_control_mobile.h"
+#include "modules/audio_processing/audio_buffer.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+namespace {
+int16_t MapSetting(EchoControlMobileImpl::RoutingMode mode) {
+  switch (mode) {
+    case EchoControlMobileImpl::kQuietEarpieceOrHeadset:
+      return 0;
+    case EchoControlMobileImpl::kEarpiece:
+      return 1;
+    case EchoControlMobileImpl::kLoudEarpiece:
+      return 2;
+    case EchoControlMobileImpl::kSpeakerphone:
+      return 3;
+    case EchoControlMobileImpl::kLoudSpeakerphone:
+      return 4;
+  }
+  RTC_DCHECK_NOTREACHED();
+  return -1;
+}
+
+AudioProcessing::Error MapError(int err) {
+  switch (err) {
+    case AECM_UNSUPPORTED_FUNCTION_ERROR:
+      return AudioProcessing::kUnsupportedFunctionError;
+    case AECM_NULL_POINTER_ERROR:
+      return AudioProcessing::kNullPointerError;
+    case AECM_BAD_PARAMETER_ERROR:
+      return AudioProcessing::kBadParameterError;
+    case AECM_BAD_PARAMETER_WARNING:
+      return AudioProcessing::kBadStreamParameterWarning;
+    default:
+      // AECM_UNSPECIFIED_ERROR
+      // AECM_UNINITIALIZED_ERROR
+      return AudioProcessing::kUnspecifiedError;
+  }
+}
+
+}  // namespace
+
+struct EchoControlMobileImpl::StreamProperties {
+  StreamProperties() = delete;
+  StreamProperties(int sample_rate_hz,
+                   size_t num_reverse_channels,
+                   size_t num_output_channels)
+      : sample_rate_hz(sample_rate_hz),
+        num_reverse_channels(num_reverse_channels),
+        num_output_channels(num_output_channels) {}
+
+  int sample_rate_hz;
+  size_t num_reverse_channels;
+  size_t num_output_channels;
+};
+
+class EchoControlMobileImpl::Canceller {
+ public:
+  Canceller() {
+    state_ = WebRtcAecm_Create();
+    RTC_CHECK(state_);
+  }
+
+  ~Canceller() {
+    RTC_DCHECK(state_);
+    WebRtcAecm_Free(state_);
+  }
+
+  Canceller(const Canceller&) = delete;
+  Canceller& operator=(const Canceller&) = delete;
+
+  void* state() {
+    RTC_DCHECK(state_);
+    return state_;
+  }
+
+  void Initialize(int sample_rate_hz) {
+    RTC_DCHECK(state_);
+    int error = WebRtcAecm_Init(state_, sample_rate_hz);
+    RTC_DCHECK_EQ(AudioProcessing::kNoError, error);
+  }
+
+ private:
+  void* state_;
+};
+
+EchoControlMobileImpl::EchoControlMobileImpl()
+    : routing_mode_(kSpeakerphone), comfort_noise_enabled_(false) {}
+
+EchoControlMobileImpl::~EchoControlMobileImpl() {}
+
+void EchoControlMobileImpl::ProcessRenderAudio(
+    rtc::ArrayView<const int16_t> packed_render_audio) {
+  RTC_DCHECK(stream_properties_);
+
+  size_t buffer_index = 0;
+  size_t num_frames_per_band =
+      packed_render_audio.size() / (stream_properties_->num_output_channels *
+                                    stream_properties_->num_reverse_channels);
+
+  for (auto& canceller : cancellers_) {
+    WebRtcAecm_BufferFarend(canceller->state(),
+                            &packed_render_audio[buffer_index],
+                            num_frames_per_band);
+
+    buffer_index += num_frames_per_band;
+  }
+}
+
+void EchoControlMobileImpl::PackRenderAudioBuffer(
+    const AudioBuffer* audio,
+    size_t num_output_channels,
+    size_t num_channels,
+    std::vector<int16_t>* packed_buffer) {
+  RTC_DCHECK_GE(AudioBuffer::kMaxSplitFrameLength,
+                audio->num_frames_per_band());
+  RTC_DCHECK_EQ(num_channels, audio->num_channels());
+
+  // The ordering convention must be followed to pass to the correct AECM.
+  packed_buffer->clear();
+  int render_channel = 0;
+  for (size_t i = 0; i < num_output_channels; i++) {
+    for (size_t j = 0; j < audio->num_channels(); j++) {
+      std::array<int16_t, AudioBuffer::kMaxSplitFrameLength> data_to_buffer;
+      FloatS16ToS16(audio->split_bands_const(render_channel)[kBand0To8kHz],
+                    audio->num_frames_per_band(), data_to_buffer.data());
+
+      // Buffer the samples in the render queue.
+      packed_buffer->insert(
+          packed_buffer->end(), data_to_buffer.data(),
+          data_to_buffer.data() + audio->num_frames_per_band());
+      render_channel = (render_channel + 1) % audio->num_channels();
+    }
+  }
+}
+
+size_t EchoControlMobileImpl::NumCancellersRequired(
+    size_t num_output_channels,
+    size_t num_reverse_channels) {
+  return num_output_channels * num_reverse_channels;
+}
+
+int EchoControlMobileImpl::ProcessCaptureAudio(AudioBuffer* audio,
+                                               int stream_delay_ms) {
+  RTC_DCHECK(stream_properties_);
+  RTC_DCHECK_GE(160, audio->num_frames_per_band());
+  RTC_DCHECK_EQ(audio->num_channels(), stream_properties_->num_output_channels);
+  RTC_DCHECK_GE(cancellers_.size(), stream_properties_->num_reverse_channels *
+                                        audio->num_channels());
+
+  int err = AudioProcessing::kNoError;
+
+  // The ordering convention must be followed to pass to the correct AECM.
+  size_t handle_index = 0;
+  for (size_t capture = 0; capture < audio->num_channels(); ++capture) {
+    // TODO(ajm): improve how this works, possibly inside AECM.
+    //            This is kind of hacked up.
+    RTC_DCHECK_LT(capture, low_pass_reference_.size());
+    const int16_t* noisy =
+        reference_copied_ ? low_pass_reference_[capture].data() : nullptr;
+
+    RTC_DCHECK_GE(AudioBuffer::kMaxSplitFrameLength,
+                  audio->num_frames_per_band());
+
+    std::array<int16_t, AudioBuffer::kMaxSplitFrameLength> split_bands_data;
+    int16_t* split_bands = split_bands_data.data();
+    const int16_t* clean = split_bands_data.data();
+    if (audio->split_bands(capture)[kBand0To8kHz]) {
+      FloatS16ToS16(audio->split_bands(capture)[kBand0To8kHz],
+                    audio->num_frames_per_band(), split_bands_data.data());
+    } else {
+      clean = nullptr;
+      split_bands = nullptr;
+    }
+
+    if (noisy == NULL) {
+      noisy = clean;
+      clean = NULL;
+    }
+    for (size_t render = 0; render < stream_properties_->num_reverse_channels;
+         ++render) {
+      err = WebRtcAecm_Process(cancellers_[handle_index]->state(), noisy, clean,
+                               split_bands, audio->num_frames_per_band(),
+                               stream_delay_ms);
+
+      if (split_bands) {
+        S16ToFloatS16(split_bands, audio->num_frames_per_band(),
+                      audio->split_bands(capture)[kBand0To8kHz]);
+      }
+
+      if (err != AudioProcessing::kNoError) {
+        return MapError(err);
+      }
+
+      ++handle_index;
+    }
+    for (size_t band = 1u; band < audio->num_bands(); ++band) {
+      memset(audio->split_bands_f(capture)[band], 0,
+             audio->num_frames_per_band() *
+                 sizeof(audio->split_bands_f(capture)[band][0]));
+    }
+  }
+  return AudioProcessing::kNoError;
+}
+
+int EchoControlMobileImpl::set_routing_mode(RoutingMode mode) {
+  if (MapSetting(mode) == -1) {
+    return AudioProcessing::kBadParameterError;
+  }
+  routing_mode_ = mode;
+  return Configure();
+}
+
+EchoControlMobileImpl::RoutingMode EchoControlMobileImpl::routing_mode() const {
+  return routing_mode_;
+}
+
+int EchoControlMobileImpl::enable_comfort_noise(bool enable) {
+  comfort_noise_enabled_ = enable;
+  return Configure();
+}
+
+bool EchoControlMobileImpl::is_comfort_noise_enabled() const {
+  return comfort_noise_enabled_;
+}
+
+void EchoControlMobileImpl::Initialize(int sample_rate_hz,
+                                       size_t num_reverse_channels,
+                                       size_t num_output_channels) {
+  low_pass_reference_.resize(num_output_channels);
+  for (auto& reference : low_pass_reference_) {
+    reference.fill(0);
+  }
+
+  stream_properties_.reset(new StreamProperties(
+      sample_rate_hz, num_reverse_channels, num_output_channels));
+
+  // AECM only supports 16 kHz or lower sample rates.
+  RTC_DCHECK_LE(stream_properties_->sample_rate_hz,
+                AudioProcessing::kSampleRate16kHz);
+
+  cancellers_.resize(
+      NumCancellersRequired(stream_properties_->num_output_channels,
+                            stream_properties_->num_reverse_channels));
+
+  for (auto& canceller : cancellers_) {
+    if (!canceller) {
+      canceller.reset(new Canceller());
+    }
+    canceller->Initialize(sample_rate_hz);
+  }
+  Configure();
+}
+
+int EchoControlMobileImpl::Configure() {
+  AecmConfig config;
+  config.cngMode = comfort_noise_enabled_;
+  config.echoMode = MapSetting(routing_mode_);
+  int error = AudioProcessing::kNoError;
+  for (auto& canceller : cancellers_) {
+    int handle_error = WebRtcAecm_set_config(canceller->state(), config);
+    if (handle_error != AudioProcessing::kNoError) {
+      error = handle_error;
+    }
+  }
+  return error;
+}
+
+}  // namespace webrtc

VocieProcess/modules/audio_processing/echo_control_mobile_impl.h

@@ -0,0 +1,86 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_ECHO_CONTROL_MOBILE_IMPL_H_
+#define MODULES_AUDIO_PROCESSING_ECHO_CONTROL_MOBILE_IMPL_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <memory>
+#include <vector>
+
+#include "api/array_view.h"
+
+namespace webrtc {
+
+class AudioBuffer;
+
+// The acoustic echo control for mobile (AECM) component is a low complexity
+// robust option intended for use on mobile devices.
+class EchoControlMobileImpl {
+ public:
+  EchoControlMobileImpl();
+
+  ~EchoControlMobileImpl();
+
+  // Recommended settings for particular audio routes. In general, the louder
+  // the echo is expected to be, the higher this value should be set. The
+  // preferred setting may vary from device to device.
+  enum RoutingMode {
+    kQuietEarpieceOrHeadset,
+    kEarpiece,
+    kLoudEarpiece,
+    kSpeakerphone,
+    kLoudSpeakerphone
+  };
+
+  // Sets echo control appropriate for the audio routing `mode` on the device.
+  // It can and should be updated during a call if the audio routing changes.
+  int set_routing_mode(RoutingMode mode);
+  RoutingMode routing_mode() const;
+
+  // Comfort noise replaces suppressed background noise to maintain a
+  // consistent signal level.
+  int enable_comfort_noise(bool enable);
+  bool is_comfort_noise_enabled() const;
+
+  void ProcessRenderAudio(rtc::ArrayView<const int16_t> packed_render_audio);
+  int ProcessCaptureAudio(AudioBuffer* audio, int stream_delay_ms);
+
+  void Initialize(int sample_rate_hz,
+                  size_t num_reverse_channels,
+                  size_t num_output_channels);
+
+  static void PackRenderAudioBuffer(const AudioBuffer* audio,
+                                    size_t num_output_channels,
+                                    size_t num_channels,
+                                    std::vector<int16_t>* packed_buffer);
+
+  static size_t NumCancellersRequired(size_t num_output_channels,
+                                      size_t num_reverse_channels);
+
+ private:
+  class Canceller;
+  struct StreamProperties;
+
+  int Configure();
+
+  RoutingMode routing_mode_;
+  bool comfort_noise_enabled_;
+
+  std::vector<std::unique_ptr<Canceller>> cancellers_;
+  std::unique_ptr<StreamProperties> stream_properties_;
+  std::vector<std::array<int16_t, 160>> low_pass_reference_;
+  bool reference_copied_ = false;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_ECHO_CONTROL_MOBILE_IMPL_H_

VocieProcess/modules/audio_processing/include/aec_dump.cc

@@ -0,0 +1,41 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/include/aec_dump.h"
+
+namespace webrtc {
+InternalAPMConfig::InternalAPMConfig() = default;
+InternalAPMConfig::InternalAPMConfig(const InternalAPMConfig&) = default;
+InternalAPMConfig::InternalAPMConfig(InternalAPMConfig&&) = default;
+InternalAPMConfig& InternalAPMConfig::operator=(const InternalAPMConfig&) =
+    default;
+
+bool InternalAPMConfig::operator==(const InternalAPMConfig& other) const {
+  return aec_enabled == other.aec_enabled &&
+         aec_delay_agnostic_enabled == other.aec_delay_agnostic_enabled &&
+         aec_drift_compensation_enabled ==
+             other.aec_drift_compensation_enabled &&
+         aec_extended_filter_enabled == other.aec_extended_filter_enabled &&
+         aec_suppression_level == other.aec_suppression_level &&
+         aecm_enabled == other.aecm_enabled &&
+         aecm_comfort_noise_enabled == other.aecm_comfort_noise_enabled &&
+         aecm_routing_mode == other.aecm_routing_mode &&
+         agc_enabled == other.agc_enabled && agc_mode == other.agc_mode &&
+         agc_limiter_enabled == other.agc_limiter_enabled &&
+         hpf_enabled == other.hpf_enabled && ns_enabled == other.ns_enabled &&
+         ns_level == other.ns_level &&
+         transient_suppression_enabled == other.transient_suppression_enabled &&
+         noise_robust_agc_enabled == other.noise_robust_agc_enabled &&
+         pre_amplifier_enabled == other.pre_amplifier_enabled &&
+         pre_amplifier_fixed_gain_factor ==
+             other.pre_amplifier_fixed_gain_factor &&
+         experiments_description == other.experiments_description;
+}
+}  // namespace webrtc

VocieProcess/modules/audio_processing/include/aec_dump.h

@@ -0,0 +1,116 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_INCLUDE_AEC_DUMP_H_
+#define MODULES_AUDIO_PROCESSING_INCLUDE_AEC_DUMP_H_
+
+#include <stdint.h>
+
+#include <string>
+
+#include "absl/base/attributes.h"
+#include "absl/types/optional.h"
+#include "api/audio/audio_processing.h"
+#include "modules/audio_processing/include/audio_frame_view.h"
+
+namespace webrtc {
+
+// Struct for passing current config from APM without having to
+// include protobuf headers.
+struct InternalAPMConfig {
+  InternalAPMConfig();
+  InternalAPMConfig(const InternalAPMConfig&);
+  InternalAPMConfig(InternalAPMConfig&&);
+
+  InternalAPMConfig& operator=(const InternalAPMConfig&);
+  InternalAPMConfig& operator=(InternalAPMConfig&&) = delete;
+
+  bool operator==(const InternalAPMConfig& other) const;
+
+  bool aec_enabled = false;
+  bool aec_delay_agnostic_enabled = false;
+  bool aec_drift_compensation_enabled = false;
+  bool aec_extended_filter_enabled = false;
+  int aec_suppression_level = 0;
+  bool aecm_enabled = false;
+  bool aecm_comfort_noise_enabled = false;
+  int aecm_routing_mode = 0;
+  bool agc_enabled = false;
+  int agc_mode = 0;
+  bool agc_limiter_enabled = false;
+  bool hpf_enabled = false;
+  bool ns_enabled = false;
+  int ns_level = 0;
+  bool transient_suppression_enabled = false;
+  bool noise_robust_agc_enabled = false;
+  bool pre_amplifier_enabled = false;
+  float pre_amplifier_fixed_gain_factor = 1.f;
+  std::string experiments_description = "";
+};
+
+// An interface for recording configuration and input/output streams
+// of the Audio Processing Module. The recordings are called
+// 'aec-dumps' and are stored in a protobuf format defined in
+// debug.proto.
+// The Write* methods are always safe to call concurrently or
+// otherwise for all implementing subclasses. The intended mode of
+// operation is to create a protobuf object from the input, and send
+// it away to be written to file asynchronously.
+class AecDump {
+ public:
+  struct AudioProcessingState {
+    int delay;
+    int drift;
+    absl::optional<int> applied_input_volume;
+    bool keypress;
+  };
+
+  virtual ~AecDump() = default;
+
+  // Logs Event::Type INIT message.
+  virtual void WriteInitMessage(const ProcessingConfig& api_format,
+                                int64_t time_now_ms) = 0;
+  ABSL_DEPRECATED("")
+  void WriteInitMessage(const ProcessingConfig& api_format) {
+    WriteInitMessage(api_format, 0);
+  }
+
+  // Logs Event::Type STREAM message. To log an input/output pair,
+  // call the AddCapture* and AddAudioProcessingState methods followed
+  // by a WriteCaptureStreamMessage call.
+  virtual void AddCaptureStreamInput(
+      const AudioFrameView<const float>& src) = 0;
+  virtual void AddCaptureStreamOutput(
+      const AudioFrameView<const float>& src) = 0;
+  virtual void AddCaptureStreamInput(const int16_t* const data,
+                                     int num_channels,
+                                     int samples_per_channel) = 0;
+  virtual void AddCaptureStreamOutput(const int16_t* const data,
+                                      int num_channels,
+                                      int samples_per_channel) = 0;
+  virtual void AddAudioProcessingState(const AudioProcessingState& state) = 0;
+  virtual void WriteCaptureStreamMessage() = 0;
+
+  // Logs Event::Type REVERSE_STREAM message.
+  virtual void WriteRenderStreamMessage(const int16_t* const data,
+                                        int num_channels,
+                                        int samples_per_channel) = 0;
+  virtual void WriteRenderStreamMessage(
+      const AudioFrameView<const float>& src) = 0;
+
+  virtual void WriteRuntimeSetting(
+      const AudioProcessing::RuntimeSetting& runtime_setting) = 0;
+
+  // Logs Event::Type CONFIG message.
+  virtual void WriteConfig(const InternalAPMConfig& config) = 0;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_INCLUDE_AEC_DUMP_H_

VocieProcess/modules/audio_processing/include/audio_frame_proxies.cc

@@ -0,0 +1,66 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/include/audio_frame_proxies.h"
+
+#include "api/audio/audio_frame.h"
+#include "api/audio/audio_processing.h"
+
+namespace webrtc {
+
+int ProcessAudioFrame(AudioProcessing* ap, AudioFrame* frame) {
+  if (!frame || !ap) {
+    return AudioProcessing::Error::kNullPointerError;
+  }
+
+  StreamConfig input_config(frame->sample_rate_hz_, frame->num_channels_);
+  StreamConfig output_config(frame->sample_rate_hz_, frame->num_channels_);
+  RTC_DCHECK_EQ(frame->samples_per_channel(), input_config.num_frames());
+
+  int result = ap->ProcessStream(frame->data(), input_config, output_config,
+                                 frame->mutable_data());
+
+  AudioProcessingStats stats = ap->GetStatistics();
+
+  if (stats.voice_detected) {
+    frame->vad_activity_ = *stats.voice_detected
+                               ? AudioFrame::VADActivity::kVadActive
+                               : AudioFrame::VADActivity::kVadPassive;
+  }
+
+  return result;
+}
+
+int ProcessReverseAudioFrame(AudioProcessing* ap, AudioFrame* frame) {
+  if (!frame || !ap) {
+    return AudioProcessing::Error::kNullPointerError;
+  }
+
+  // Must be a native rate.
+  if (frame->sample_rate_hz_ != AudioProcessing::NativeRate::kSampleRate8kHz &&
+      frame->sample_rate_hz_ != AudioProcessing::NativeRate::kSampleRate16kHz &&
+      frame->sample_rate_hz_ != AudioProcessing::NativeRate::kSampleRate32kHz &&
+      frame->sample_rate_hz_ != AudioProcessing::NativeRate::kSampleRate48kHz) {
+    return AudioProcessing::Error::kBadSampleRateError;
+  }
+
+  if (frame->num_channels_ <= 0) {
+    return AudioProcessing::Error::kBadNumberChannelsError;
+  }
+
+  StreamConfig input_config(frame->sample_rate_hz_, frame->num_channels_);
+  StreamConfig output_config(frame->sample_rate_hz_, frame->num_channels_);
+
+  int result = ap->ProcessReverseStream(frame->data(), input_config,
+                                        output_config, frame->mutable_data());
+  return result;
+}
+
+}  // namespace webrtc

VocieProcess/modules/audio_processing/include/audio_frame_proxies.h

@@ -0,0 +1,41 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_FRAME_PROXIES_H_
+#define MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_FRAME_PROXIES_H_
+
+namespace webrtc {
+
+class AudioFrame;
+class AudioProcessing;
+
+// Processes a 10 ms `frame` of the primary audio stream using the provided
+// AudioProcessing object. On the client-side, this is the near-end (or
+// captured) audio. The `sample_rate_hz_`, `num_channels_`, and
+// `samples_per_channel_` members of `frame` must be valid. If changed from the
+// previous call to this function, it will trigger an initialization of the
+// provided AudioProcessing object.
+// The function returns any error codes passed from the AudioProcessing
+// ProcessStream method.
+int ProcessAudioFrame(AudioProcessing* ap, AudioFrame* frame);
+
+// Processes a 10 ms `frame` of the reverse direction audio stream using the
+// provided AudioProcessing object. The frame may be modified. On the
+// client-side, this is the far-end (or to be rendered) audio. The
+// `sample_rate_hz_`, `num_channels_`, and `samples_per_channel_` members of
+// `frame` must be valid. If changed from the previous call to this function, it
+// will trigger an initialization of the provided AudioProcessing object.
+// The function returns any error codes passed from the AudioProcessing
+// ProcessReverseStream method.
+int ProcessReverseAudioFrame(AudioProcessing* ap, AudioFrame* frame);
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_FRAME_PROXIES_H_

VocieProcess/modules/audio_processing/include/audio_frame_view.h

@@ -0,0 +1,66 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_FRAME_VIEW_H_
+#define MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_FRAME_VIEW_H_
+
+#include "api/audio/audio_view.h"
+
+namespace webrtc {
+
+// Class to pass audio data in T** format, where T is a numeric type.
+template <class T>
+class AudioFrameView {
+ public:
+  // `num_channels` and `channel_size` describe the T**
+  // `audio_samples`. `audio_samples` is assumed to point to a
+  // two-dimensional |num_channels * channel_size| array of floats.
+  //
+  // Note: The implementation now only requires the first channel pointer.
+  // The previous implementation retained a pointer to externally owned array
+  // of channel pointers, but since the channel size and count are provided
+  // and the array is assumed to be a single two-dimensional array, the other
+  // channel pointers can be calculated based on that (which is what the class
+  // now uses `DeinterleavedView<>` internally for).
+  AudioFrameView(T* const* audio_samples, int num_channels, int channel_size)
+      : view_(num_channels && channel_size ? audio_samples[0] : nullptr,
+              channel_size,
+              num_channels) {
+    RTC_DCHECK_GE(view_.num_channels(), 0);
+    RTC_DCHECK_GE(view_.samples_per_channel(), 0);
+  }
+
+  // Implicit cast to allow converting AudioFrameView<float> to
+  // AudioFrameView<const float>.
+  template <class U>
+  AudioFrameView(AudioFrameView<U> other) : view_(other.view()) {}
+
+  // Allow constructing AudioFrameView from a DeinterleavedView.
+  template <class U>
+  explicit AudioFrameView(DeinterleavedView<U> view) : view_(view) {}
+
+  AudioFrameView() = delete;
+
+  int num_channels() const { return view_.num_channels(); }
+  int samples_per_channel() const { return view_.samples_per_channel(); }
+  MonoView<T> channel(int idx) { return view_[idx]; }
+  MonoView<const T> channel(int idx) const { return view_[idx]; }
+  MonoView<T> operator[](int idx) { return view_[idx]; }
+  MonoView<const T> operator[](int idx) const { return view_[idx]; }
+
+  DeinterleavedView<T> view() { return view_; }
+  DeinterleavedView<const T> view() const { return view_; }
+
+ private:
+  DeinterleavedView<T> view_;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_FRAME_VIEW_H_

VocieProcess/modules/audio_processing/render_queue_item_verifier.h

@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_RENDER_QUEUE_ITEM_VERIFIER_H_
+#define MODULES_AUDIO_PROCESSING_RENDER_QUEUE_ITEM_VERIFIER_H_
+
+#include <vector>
+
+namespace webrtc {
+
+// Functor to use when supplying a verifier function for the queue item
+// verifcation.
+template <typename T>
+class RenderQueueItemVerifier {
+ public:
+  explicit RenderQueueItemVerifier(size_t minimum_capacity)
+      : minimum_capacity_(minimum_capacity) {}
+
+  bool operator()(const std::vector<T>& v) const {
+    return v.capacity() >= minimum_capacity_;
+  }
+
+ private:
+  size_t minimum_capacity_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_RENDER_QUEUE_ITEM_VERIFIER_H__

VocieProcess/modules/audio_processing/rms_level.cc

@@ -0,0 +1,138 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/rms_level.h"
+
+#include <algorithm>
+#include <cmath>
+#include <numeric>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace {
+static constexpr float kMaxSquaredLevel = 32768 * 32768;
+// kMinLevel is the level corresponding to kMinLevelDb, that is 10^(-127/10).
+static constexpr float kMinLevel = 1.995262314968883e-13f;
+
+// Calculates the normalized RMS value from a mean square value. The input
+// should be the sum of squared samples divided by the number of samples. The
+// value will be normalized to full range before computing the RMS, wich is
+// returned as a negated dBfs. That is, 0 is full amplitude while 127 is very
+// faint.
+int ComputeRms(float mean_square) {
+  if (mean_square <= kMinLevel * kMaxSquaredLevel) {
+    // Very faint; simply return the minimum value.
+    return RmsLevel::kMinLevelDb;
+  }
+  // Normalize by the max level.
+  const float mean_square_norm = mean_square / kMaxSquaredLevel;
+  RTC_DCHECK_GT(mean_square_norm, kMinLevel);
+  // 20log_10(x^0.5) = 10log_10(x)
+  const float rms = 10.f * std::log10(mean_square_norm);
+  RTC_DCHECK_LE(rms, 0.f);
+  RTC_DCHECK_GT(rms, -RmsLevel::kMinLevelDb);
+  // Return the negated value.
+  return static_cast<int>(-rms + 0.5f);
+}
+}  // namespace
+
+RmsLevel::RmsLevel() {
+  Reset();
+}
+
+RmsLevel::~RmsLevel() = default;
+
+void RmsLevel::Reset() {
+  sum_square_ = 0.f;
+  sample_count_ = 0;
+  max_sum_square_ = 0.f;
+  block_size_ = absl::nullopt;
+}
+
+void RmsLevel::Analyze(rtc::ArrayView<const int16_t> data) {
+  if (data.empty()) {
+    return;
+  }
+
+  CheckBlockSize(data.size());
+
+  const float sum_square =
+      std::accumulate(data.begin(), data.end(), 0.f,
+                      [](float a, int16_t b) { return a + b * b; });
+  RTC_DCHECK_GE(sum_square, 0.f);
+  sum_square_ += sum_square;
+  sample_count_ += data.size();
+
+  max_sum_square_ = std::max(max_sum_square_, sum_square);
+}
+
+void RmsLevel::Analyze(rtc::ArrayView<const float> data) {
+  if (data.empty()) {
+    return;
+  }
+
+  CheckBlockSize(data.size());
+
+  float sum_square = 0.f;
+
+  for (float data_k : data) {
+    int16_t tmp =
+        static_cast<int16_t>(std::min(std::max(data_k, -32768.f), 32767.f));
+    sum_square += tmp * tmp;
+  }
+  RTC_DCHECK_GE(sum_square, 0.f);
+  sum_square_ += sum_square;
+  sample_count_ += data.size();
+
+  max_sum_square_ = std::max(max_sum_square_, sum_square);
+}
+
+void RmsLevel::AnalyzeMuted(size_t length) {
+  CheckBlockSize(length);
+  sample_count_ += length;
+}
+
+int RmsLevel::Average() {
+  const bool have_samples = (sample_count_ != 0);
+  int rms = have_samples ? ComputeRms(sum_square_ / sample_count_)
+                         : RmsLevel::kMinLevelDb;
+
+  // To ensure that kMinLevelDb represents digital silence (muted audio
+  // sources) we'll check here if the sum_square is actually 0. If it's not
+  // we'll bump up the return value to `kInaudibleButNotMuted`.
+  // https://datatracker.ietf.org/doc/html/rfc6464
+  if (have_samples && rms == RmsLevel::kMinLevelDb && sum_square_ != 0.0f) {
+    rms = kInaudibleButNotMuted;
+  }
+
+  Reset();
+  return rms;
+}
+
+RmsLevel::Levels RmsLevel::AverageAndPeak() {
+  // Note that block_size_ should by design always be non-empty when
+  // sample_count_ != 0. Also, the * operator of absl::optional enforces this
+  // with a DCHECK.
+  Levels levels = (sample_count_ == 0)
+                      ? Levels{RmsLevel::kMinLevelDb, RmsLevel::kMinLevelDb}
+                      : Levels{ComputeRms(sum_square_ / sample_count_),
+                               ComputeRms(max_sum_square_ / *block_size_)};
+  Reset();
+  return levels;
+}
+
+void RmsLevel::CheckBlockSize(size_t block_size) {
+  if (block_size_ != block_size) {
+    Reset();
+    block_size_ = block_size;
+  }
+}
+}  // namespace webrtc

VocieProcess/modules/audio_processing/rms_level.h

@@ -0,0 +1,77 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
+#define MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+
+namespace webrtc {
+
+// Computes the root mean square (RMS) level in dBFs (decibels from digital
+// full-scale) of audio data. The computation follows RFC 6465:
+// https://tools.ietf.org/html/rfc6465
+// with the intent that it can provide the RTP audio level indication.
+//
+// The expected approach is to provide constant-sized chunks of audio to
+// Analyze(). When enough chunks have been accumulated to form a packet, call
+// Average() to get the audio level indicator for the RTP header.
+class RmsLevel {
+ public:
+  struct Levels {
+    int average;
+    int peak;
+  };
+
+  enum : int { kMinLevelDb = 127, kInaudibleButNotMuted = 126 };
+
+  RmsLevel();
+  ~RmsLevel();
+
+  // Can be called to reset internal states, but is not required during normal
+  // operation.
+  void Reset();
+
+  // Pass each chunk of audio to Analyze() to accumulate the level.
+  void Analyze(rtc::ArrayView<const int16_t> data);
+  void Analyze(rtc::ArrayView<const float> data);
+
+  // If all samples with the given `length` have a magnitude of zero, this is
+  // a shortcut to avoid some computation.
+  void AnalyzeMuted(size_t length);
+
+  // Computes the RMS level over all data passed to Analyze() since the last
+  // call to Average(). The returned value is positive but should be interpreted
+  // as negative as per the RFC. It is constrained to [0, 127]. Resets the
+  // internal state to start a new measurement period.
+  int Average();
+
+  // Like Average() above, but also returns the RMS peak value. Resets the
+  // internal state to start a new measurement period.
+  Levels AverageAndPeak();
+
+ private:
+  // Compares `block_size` with `block_size_`. If they are different, calls
+  // Reset() and stores the new size.
+  void CheckBlockSize(size_t block_size);
+
+  float sum_square_;
+  size_t sample_count_;
+  float max_sum_square_;
+  absl::optional<size_t> block_size_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_

VocieProcess/modules/audio_processing/vad/common.h

@@ -0,0 +1,29 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_VAD_COMMON_H_
+#define MODULES_AUDIO_PROCESSING_VAD_COMMON_H_
+
+#include <stddef.h>
+
+static const int kSampleRateHz = 16000;
+static const size_t kLength10Ms = kSampleRateHz / 100;
+static const size_t kMaxNumFrames = 4;
+
+struct AudioFeatures {
+  double log_pitch_gain[kMaxNumFrames];
+  double pitch_lag_hz[kMaxNumFrames];
+  double spectral_peak[kMaxNumFrames];
+  double rms[kMaxNumFrames];
+  size_t num_frames;
+  bool silence;
+};
+
+#endif  // MODULES_AUDIO_PROCESSING_VAD_COMMON_H_

VocieProcess/modules/audio_processing/vad/gmm.cc

@@ -0,0 +1,61 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/vad/gmm.h"
+
+#include <math.h>
+
+namespace webrtc {
+
+static const int kMaxDimension = 10;
+
+static void RemoveMean(const double* in,
+                       const double* mean_vec,
+                       int dimension,
+                       double* out) {
+  for (int n = 0; n < dimension; ++n)
+    out[n] = in[n] - mean_vec[n];
+}
+
+static double ComputeExponent(const double* in,
+                              const double* covar_inv,
+                              int dimension) {
+  double q = 0;
+  for (int i = 0; i < dimension; ++i) {
+    double v = 0;
+    for (int j = 0; j < dimension; j++)
+      v += (*covar_inv++) * in[j];
+    q += v * in[i];
+  }
+  q *= -0.5;
+  return q;
+}
+
+double EvaluateGmm(const double* x, const GmmParameters& gmm_parameters) {
+  if (gmm_parameters.dimension > kMaxDimension) {
+    return -1;  // This is invalid pdf so the caller can check this.
+  }
+  double f = 0;
+  double v[kMaxDimension];
+  const double* mean_vec = gmm_parameters.mean;
+  const double* covar_inv = gmm_parameters.covar_inverse;
+
+  for (int n = 0; n < gmm_parameters.num_mixtures; n++) {
+    RemoveMean(x, mean_vec, gmm_parameters.dimension, v);
+    double q = ComputeExponent(v, covar_inv, gmm_parameters.dimension) +
+               gmm_parameters.weight[n];
+    f += exp(q);
+    mean_vec += gmm_parameters.dimension;
+    covar_inv += gmm_parameters.dimension * gmm_parameters.dimension;
+  }
+  return f;
+}
+
+}  // namespace webrtc

VocieProcess/modules/audio_processing/vad/gmm.h

@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_VAD_GMM_H_
+#define MODULES_AUDIO_PROCESSING_VAD_GMM_H_
+
+namespace webrtc {
+
+// A structure that specifies a GMM.
+// A GMM is formulated as
+//  f(x) = w[0] * mixture[0] + w[1] * mixture[1] + ... +
+//         w[num_mixtures - 1] * mixture[num_mixtures - 1];
+// Where a 'mixture' is a Gaussian density.
+
+struct GmmParameters {
+  // weight[n] = log(w[n]) - `dimension`/2 * log(2*pi) - 1/2 * log(det(cov[n]));
+  // where cov[n] is the covariance matrix of mixture n;
+  const double* weight;
+  // pointer to the first element of a `num_mixtures`x`dimension` matrix
+  // where kth row is the mean of the kth mixture.
+  const double* mean;
+  // pointer to the first element of a `num_mixtures`x`dimension`x`dimension`
+  // 3D-matrix, where the kth 2D-matrix is the inverse of the covariance
+  // matrix of the kth mixture.
+  const double* covar_inverse;
+  // Dimensionality of the mixtures.
+  int dimension;
+  // number of the mixtures.
+  int num_mixtures;
+};
+
+// Evaluate the given GMM, according to `gmm_parameters`, at the given point
+// `x`. If the dimensionality of the given GMM is larger that the maximum
+// acceptable dimension by the following function -1 is returned.
+double EvaluateGmm(const double* x, const GmmParameters& gmm_parameters);
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_PROCESSING_VAD_GMM_H_

VocieProcess/modules/audio_processing/vad/noise_gmm_tables.h

@@ -0,0 +1,82 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// GMM tables for inactive segments. Generated by MakeGmmTables.m.
+
+#ifndef MODULES_AUDIO_PROCESSING_VAD_NOISE_GMM_TABLES_H_
+#define MODULES_AUDIO_PROCESSING_VAD_NOISE_GMM_TABLES_H_
+
+namespace webrtc {
+
+static const int kNoiseGmmNumMixtures = 12;
+static const int kNoiseGmmDim = 3;
+
+static const double
+    kNoiseGmmCovarInverse[kNoiseGmmNumMixtures][kNoiseGmmDim][kNoiseGmmDim] = {
+        {{7.36219567592941e+00, 4.83060785179861e-03, 1.23335151497610e-02},
+         {4.83060785179861e-03, 1.65289507047817e-04, -2.41490588169997e-04},
+         {1.23335151497610e-02, -2.41490588169997e-04, 6.59472060689382e-03}},
+        {{8.70265239309140e+00, -5.30636201431086e-04, 5.44014966585347e-03},
+         {-5.30636201431086e-04, 3.11095453521008e-04, -1.86287206836035e-04},
+         {5.44014966585347e-03, -1.86287206836035e-04, 6.29493388790744e-04}},
+        {{4.53467851955055e+00, -3.92977536695197e-03, -2.46521420693317e-03},
+         {-3.92977536695197e-03, 4.94650752632750e-05, -1.08587438501826e-05},
+         {-2.46521420693317e-03, -1.08587438501826e-05, 9.28793975422261e-05}},
+        {{9.26817997114275e-01, -4.03976069276753e-04, -3.56441427392165e-03},
+         {-4.03976069276753e-04, 2.51976251631430e-06, 1.46914206734572e-07},
+         {-3.56441427392165e-03, 1.46914206734572e-07, 8.19914567685373e-05}},
+        {{7.61715986787441e+00, -1.54889041216888e-04, 2.41756280071656e-02},
+         {-1.54889041216888e-04, 3.50282550461672e-07, -6.27251196972490e-06},
+         {2.41756280071656e-02, -6.27251196972490e-06, 1.45061847649872e-02}},
+        {{8.31193642663158e+00, -3.84070508164323e-04, -3.09750630821876e-02},
+         {-3.84070508164323e-04, 3.80433432277336e-07, -1.14321142836636e-06},
+         {-3.09750630821876e-02, -1.14321142836636e-06, 8.35091486289997e-04}},
+        {{9.67283151270894e-01, 5.82465812445039e-05, -3.18350798617053e-03},
+         {5.82465812445039e-05, 2.23762672000318e-07, -7.74196587408623e-07},
+         {-3.18350798617053e-03, -7.74196587408623e-07, 3.85120938338325e-04}},
+        {{8.28066236985388e+00, 5.87634508319763e-05, 6.99303090891743e-03},
+         {5.87634508319763e-05, 2.93746018618058e-07, 3.40843332882272e-07},
+         {6.99303090891743e-03, 3.40843332882272e-07, 1.99379171190344e-04}},
+        {{6.07488998675646e+00, -1.11494526618473e-02, 5.10013111123381e-03},
+         {-1.11494526618473e-02, 6.99238879921751e-04, 5.36718550370870e-05},
+         {5.10013111123381e-03, 5.36718550370870e-05, 5.26909853276753e-04}},
+        {{6.90492021419175e+00, 4.20639355257863e-04, -2.38612752336481e-03},
+         {4.20639355257863e-04, 3.31246767338153e-06, -2.42052288150859e-08},
+         {-2.38612752336481e-03, -2.42052288150859e-08, 4.46608368363412e-04}},
+        {{1.31069150869715e+01, -1.73718583865670e-04, -1.97591814508578e-02},
+         {-1.73718583865670e-04, 2.80451716300124e-07, 9.96570755379865e-07},
+         {-1.97591814508578e-02, 9.96570755379865e-07, 2.41361900868847e-03}},
+        {{4.69566344239814e+00, -2.61077567563690e-04, 5.26359000761433e-03},
+         {-2.61077567563690e-04, 1.82420859823767e-06, -7.83645887541601e-07},
+         {5.26359000761433e-03, -7.83645887541601e-07, 1.33586288288802e-02}}};
+
+static const double kNoiseGmmMean[kNoiseGmmNumMixtures][kNoiseGmmDim] = {
+    {-2.01386094766163e+00, 1.69702162045397e+02, 7.41715804872181e+01},
+    {-1.94684591777290e+00, 1.42398396732668e+02, 1.64186321157831e+02},
+    {-2.29319297562437e+00, 3.86415425589868e+02, 2.13452215267125e+02},
+    {-3.25487177070268e+00, 1.08668712553616e+03, 2.33119949467419e+02},
+    {-2.13159632447467e+00, 4.83821702557717e+03, 6.86786166673740e+01},
+    {-2.26171410780526e+00, 4.79420193982422e+03, 1.53222513286450e+02},
+    {-3.32166740703185e+00, 4.35161135834358e+03, 1.33206448431316e+02},
+    {-2.19290322814343e+00, 3.98325506609408e+03, 2.13249167359934e+02},
+    {-2.02898459255404e+00, 7.37039893155007e+03, 1.12518527491926e+02},
+    {-2.26150236399500e+00, 1.54896745196145e+03, 1.49717357868579e+02},
+    {-2.00417668301790e+00, 3.82434760310304e+03, 1.07438913004312e+02},
+    {-2.30193040814533e+00, 1.43953696546439e+03, 7.04085275122649e+01}};
+
+static const double kNoiseGmmWeights[kNoiseGmmNumMixtures] = {
+    -1.09422832086193e+01, -1.10847897513425e+01, -1.36767587732187e+01,
+    -1.79789356118641e+01, -1.42830169160894e+01, -1.56500228061379e+01,
+    -1.83124990950113e+01, -1.69979436177477e+01, -1.12329424387828e+01,
+    -1.41311785780639e+01, -1.47171861448585e+01, -1.35963362781839e+01};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_VAD_NOISE_GMM_TABLES_H_

VocieProcess/modules/audio_processing/vad/pitch_based_vad.cc

@@ -0,0 +1,120 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/vad/pitch_based_vad.h"
+
+#include <string.h>
+
+#include "modules/audio_processing/vad/common.h"
+#include "modules/audio_processing/vad/noise_gmm_tables.h"
+#include "modules/audio_processing/vad/vad_circular_buffer.h"
+#include "modules/audio_processing/vad/voice_gmm_tables.h"
+
+namespace webrtc {
+
+static_assert(kNoiseGmmDim == kVoiceGmmDim,
+              "noise and voice gmm dimension not equal");
+
+// These values should match MATLAB counterparts for unit-tests to pass.
+static const int kPosteriorHistorySize = 500;  // 5 sec of 10 ms frames.
+static const double kInitialPriorProbability = 0.3;
+static const int kTransientWidthThreshold = 7;
+static const double kLowProbabilityThreshold = 0.2;
+
+static double LimitProbability(double p) {
+  const double kLimHigh = 0.99;
+  const double kLimLow = 0.01;
+
+  if (p > kLimHigh)
+    p = kLimHigh;
+  else if (p < kLimLow)
+    p = kLimLow;
+  return p;
+}
+
+PitchBasedVad::PitchBasedVad()
+    : p_prior_(kInitialPriorProbability),
+      circular_buffer_(VadCircularBuffer::Create(kPosteriorHistorySize)) {
+  // Setup noise GMM.
+  noise_gmm_.dimension = kNoiseGmmDim;
+  noise_gmm_.num_mixtures = kNoiseGmmNumMixtures;
+  noise_gmm_.weight = kNoiseGmmWeights;
+  noise_gmm_.mean = &kNoiseGmmMean[0][0];
+  noise_gmm_.covar_inverse = &kNoiseGmmCovarInverse[0][0][0];
+
+  // Setup voice GMM.
+  voice_gmm_.dimension = kVoiceGmmDim;
+  voice_gmm_.num_mixtures = kVoiceGmmNumMixtures;
+  voice_gmm_.weight = kVoiceGmmWeights;
+  voice_gmm_.mean = &kVoiceGmmMean[0][0];
+  voice_gmm_.covar_inverse = &kVoiceGmmCovarInverse[0][0][0];
+}
+
+PitchBasedVad::~PitchBasedVad() {}
+
+int PitchBasedVad::VoicingProbability(const AudioFeatures& features,
+                                      double* p_combined) {
+  double p;
+  double gmm_features[3];
+  double pdf_features_given_voice;
+  double pdf_features_given_noise;
+  // These limits are the same in matlab implementation 'VoicingProbGMM().'
+  const double kLimLowLogPitchGain = -2.0;
+  const double kLimHighLogPitchGain = -0.9;
+  const double kLimLowSpectralPeak = 200;
+  const double kLimHighSpectralPeak = 2000;
+  const double kEps = 1e-12;
+  for (size_t n = 0; n < features.num_frames; n++) {
+    gmm_features[0] = features.log_pitch_gain[n];
+    gmm_features[1] = features.spectral_peak[n];
+    gmm_features[2] = features.pitch_lag_hz[n];
+
+    pdf_features_given_voice = EvaluateGmm(gmm_features, voice_gmm_);
+    pdf_features_given_noise = EvaluateGmm(gmm_features, noise_gmm_);
+
+    if (features.spectral_peak[n] < kLimLowSpectralPeak ||
+        features.spectral_peak[n] > kLimHighSpectralPeak ||
+        features.log_pitch_gain[n] < kLimLowLogPitchGain) {
+      pdf_features_given_voice = kEps * pdf_features_given_noise;
+    } else if (features.log_pitch_gain[n] > kLimHighLogPitchGain) {
+      pdf_features_given_noise = kEps * pdf_features_given_voice;
+    }
+
+    p = p_prior_ * pdf_features_given_voice /
+        (pdf_features_given_voice * p_prior_ +
+         pdf_features_given_noise * (1 - p_prior_));
+
+    p = LimitProbability(p);
+
+    // Combine pitch-based probability with standalone probability, before
+    // updating prior probabilities.
+    double prod_active = p * p_combined[n];
+    double prod_inactive = (1 - p) * (1 - p_combined[n]);
+    p_combined[n] = prod_active / (prod_active + prod_inactive);
+
+    if (UpdatePrior(p_combined[n]) < 0)
+      return -1;
+    // Limit prior probability. With a zero prior probability the posterior
+    // probability is always zero.
+    p_prior_ = LimitProbability(p_prior_);
+  }
+  return 0;
+}
+
+int PitchBasedVad::UpdatePrior(double p) {
+  circular_buffer_->Insert(p);
+  if (circular_buffer_->RemoveTransient(kTransientWidthThreshold,
+                                        kLowProbabilityThreshold) < 0)
+    return -1;
+  p_prior_ = circular_buffer_->Mean();
+  return 0;
+}
+
+}  // namespace webrtc

VocieProcess/modules/audio_processing/vad/pitch_based_vad.h

@@ -0,0 +1,57 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_VAD_PITCH_BASED_VAD_H_
+#define MODULES_AUDIO_PROCESSING_VAD_PITCH_BASED_VAD_H_
+
+#include <memory>
+
+#include "modules/audio_processing/vad/common.h"
+#include "modules/audio_processing/vad/gmm.h"
+
+namespace webrtc {
+
+class VadCircularBuffer;
+
+// Computes the probability of the input audio frame to be active given
+// the corresponding pitch-gain and lag of the frame.
+class PitchBasedVad {
+ public:
+  PitchBasedVad();
+  ~PitchBasedVad();
+
+  // Compute pitch-based voicing probability, given the features.
+  //   features: a structure containing features required for computing voicing
+  //             probabilities.
+  //
+  //   p_combined: an array which contains the combined activity probabilities
+  //               computed prior to the call of this function. The method,
+  //               then, computes the voicing probabilities and combine them
+  //               with the given values. The result are returned in `p`.
+  int VoicingProbability(const AudioFeatures& features, double* p_combined);
+
+ private:
+  int UpdatePrior(double p);
+
+  // TODO(turajs): maybe defining this at a higher level (maybe enum) so that
+  // all the code recognize it as "no-error."
+  static const int kNoError = 0;
+
+  GmmParameters noise_gmm_;
+  GmmParameters voice_gmm_;
+
+  double p_prior_;
+
+  std::unique_ptr<VadCircularBuffer> circular_buffer_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_VAD_PITCH_BASED_VAD_H_

VocieProcess/modules/audio_processing/vad/pitch_internal.cc

@@ -0,0 +1,55 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/vad/pitch_internal.h"
+
+#include <cmath>
+
+namespace webrtc {
+
+// A 4-to-3 linear interpolation.
+// The interpolation constants are derived as following:
+// Input pitch parameters are updated every 7.5 ms. Within a 30-ms interval
+// we are interested in pitch parameters of 0-5 ms, 10-15ms and 20-25ms. This is
+// like interpolating 4-to-6 and keep the odd samples.
+// The reason behind this is that LPC coefficients are computed for the first
+// half of each 10ms interval.
+static void PitchInterpolation(double old_val, const double* in, double* out) {
+  out[0] = 1. / 6. * old_val + 5. / 6. * in[0];
+  out[1] = 5. / 6. * in[1] + 1. / 6. * in[2];
+  out[2] = 0.5 * in[2] + 0.5 * in[3];
+}
+
+void GetSubframesPitchParameters(int sampling_rate_hz,
+                                 double* gains,
+                                 double* lags,
+                                 int num_in_frames,
+                                 int num_out_frames,
+                                 double* log_old_gain,
+                                 double* old_lag,
+                                 double* log_pitch_gain,
+                                 double* pitch_lag_hz) {
+  // Gain interpolation is in log-domain, also returned in log-domain.
+  for (int n = 0; n < num_in_frames; n++)
+    gains[n] = log(gains[n] + 1e-12);
+
+  // Interpolate lags and gains.
+  PitchInterpolation(*log_old_gain, gains, log_pitch_gain);
+  *log_old_gain = gains[num_in_frames - 1];
+  PitchInterpolation(*old_lag, lags, pitch_lag_hz);
+  *old_lag = lags[num_in_frames - 1];
+
+  // Convert pitch-lags to Hertz.
+  for (int n = 0; n < num_out_frames; n++) {
+    pitch_lag_hz[n] = (sampling_rate_hz) / (pitch_lag_hz[n]);
+  }
+}
+
+}  // namespace webrtc

VocieProcess/modules/audio_processing/vad/pitch_internal.h

@@ -0,0 +1,30 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_VAD_PITCH_INTERNAL_H_
+#define MODULES_AUDIO_PROCESSING_VAD_PITCH_INTERNAL_H_
+
+namespace webrtc {
+
+// TODO(turajs): Write a description of this function. Also be consistent with
+// usage of `sampling_rate_hz` vs `kSamplingFreqHz`.
+void GetSubframesPitchParameters(int sampling_rate_hz,
+                                 double* gains,
+                                 double* lags,
+                                 int num_in_frames,
+                                 int num_out_frames,
+                                 double* log_old_gain,
+                                 double* old_lag,
+                                 double* log_pitch_gain,
+                                 double* pitch_lag_hz);
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_VAD_PITCH_INTERNAL_H_

VocieProcess/modules/audio_processing/vad/pole_zero_filter.cc

@@ -0,0 +1,107 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/vad/pole_zero_filter.h"
+
+#include <string.h>
+
+#include <algorithm>
+
+namespace webrtc {
+
+PoleZeroFilter* PoleZeroFilter::Create(const float* numerator_coefficients,
+                                       size_t order_numerator,
+                                       const float* denominator_coefficients,
+                                       size_t order_denominator) {
+  if (order_numerator > kMaxFilterOrder ||
+      order_denominator > kMaxFilterOrder || denominator_coefficients[0] == 0 ||
+      numerator_coefficients == NULL || denominator_coefficients == NULL)
+    return NULL;
+  return new PoleZeroFilter(numerator_coefficients, order_numerator,
+                            denominator_coefficients, order_denominator);
+}
+
+PoleZeroFilter::PoleZeroFilter(const float* numerator_coefficients,
+                               size_t order_numerator,
+                               const float* denominator_coefficients,
+                               size_t order_denominator)
+    : past_input_(),
+      past_output_(),
+      numerator_coefficients_(),
+      denominator_coefficients_(),
+      order_numerator_(order_numerator),
+      order_denominator_(order_denominator),
+      highest_order_(std::max(order_denominator, order_numerator)) {
+  memcpy(numerator_coefficients_, numerator_coefficients,
+         sizeof(numerator_coefficients_[0]) * (order_numerator_ + 1));
+  memcpy(denominator_coefficients_, denominator_coefficients,
+         sizeof(denominator_coefficients_[0]) * (order_denominator_ + 1));
+
+  if (denominator_coefficients_[0] != 1) {
+    for (size_t n = 0; n <= order_numerator_; n++)
+      numerator_coefficients_[n] /= denominator_coefficients_[0];
+    for (size_t n = 0; n <= order_denominator_; n++)
+      denominator_coefficients_[n] /= denominator_coefficients_[0];
+  }
+}
+
+template <typename T>
+static float FilterArPast(const T* past,
+                          size_t order,
+                          const float* coefficients) {
+  float sum = 0.0f;
+  size_t past_index = order - 1;
+  for (size_t k = 1; k <= order; k++, past_index--)
+    sum += coefficients[k] * past[past_index];
+  return sum;
+}
+
+int PoleZeroFilter::Filter(const int16_t* in,
+                           size_t num_input_samples,
+                           float* output) {
+  if (in == NULL || output == NULL)
+    return -1;
+  // This is the typical case, just a memcpy.
+  const size_t k = std::min(num_input_samples, highest_order_);
+  size_t n;
+  for (n = 0; n < k; n++) {
+    output[n] = in[n] * numerator_coefficients_[0];
+    output[n] += FilterArPast(&past_input_[n], order_numerator_,
+                              numerator_coefficients_);
+    output[n] -= FilterArPast(&past_output_[n], order_denominator_,
+                              denominator_coefficients_);
+
+    past_input_[n + order_numerator_] = in[n];
+    past_output_[n + order_denominator_] = output[n];
+  }
+  if (highest_order_ < num_input_samples) {
+    for (size_t m = 0; n < num_input_samples; n++, m++) {
+      output[n] = in[n] * numerator_coefficients_[0];
+      output[n] +=
+          FilterArPast(&in[m], order_numerator_, numerator_coefficients_);
+      output[n] -= FilterArPast(&output[m], order_denominator_,
+                                denominator_coefficients_);
+    }
+    // Record into the past signal.
+    memcpy(past_input_, &in[num_input_samples - order_numerator_],
+           sizeof(in[0]) * order_numerator_);
+    memcpy(past_output_, &output[num_input_samples - order_denominator_],
+           sizeof(output[0]) * order_denominator_);
+  } else {
+    // Odd case that the length of the input is shorter that filter order.
+    memmove(past_input_, &past_input_[num_input_samples],
+            order_numerator_ * sizeof(past_input_[0]));
+    memmove(past_output_, &past_output_[num_input_samples],
+            order_denominator_ * sizeof(past_output_[0]));
+  }
+  return 0;
+}
+
+}  // namespace webrtc

VocieProcess/modules/audio_processing/vad/pole_zero_filter.h

@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_VAD_POLE_ZERO_FILTER_H_
+#define MODULES_AUDIO_PROCESSING_VAD_POLE_ZERO_FILTER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+namespace webrtc {
+
+class PoleZeroFilter {
+ public:
+  ~PoleZeroFilter() {}
+
+  static PoleZeroFilter* Create(const float* numerator_coefficients,
+                                size_t order_numerator,
+                                const float* denominator_coefficients,
+                                size_t order_denominator);
+
+  int Filter(const int16_t* in, size_t num_input_samples, float* output);
+
+ private:
+  PoleZeroFilter(const float* numerator_coefficients,
+                 size_t order_numerator,
+                 const float* denominator_coefficients,
+                 size_t order_denominator);
+
+  static const int kMaxFilterOrder = 24;
+
+  int16_t past_input_[kMaxFilterOrder * 2];
+  float past_output_[kMaxFilterOrder * 2];
+
+  float numerator_coefficients_[kMaxFilterOrder + 1];
+  float denominator_coefficients_[kMaxFilterOrder + 1];
+
+  size_t order_numerator_;
+  size_t order_denominator_;
+  size_t highest_order_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_VAD_POLE_ZERO_FILTER_H_

VocieProcess/modules/audio_processing/vad/standalone_vad.cc

@@ -0,0 +1,91 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/vad/standalone_vad.h"
+
+#include <string.h>
+
+#include "common_audio/vad/include/webrtc_vad.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+static const int kDefaultStandaloneVadMode = 3;
+
+StandaloneVad::StandaloneVad(VadInst* vad)
+    : vad_(vad), buffer_(), index_(0), mode_(kDefaultStandaloneVadMode) {}
+
+StandaloneVad::~StandaloneVad() {
+  WebRtcVad_Free(vad_);
+}
+
+StandaloneVad* StandaloneVad::Create() {
+  VadInst* vad = WebRtcVad_Create();
+  if (!vad)
+    return nullptr;
+
+  int err = WebRtcVad_Init(vad);
+  err |= WebRtcVad_set_mode(vad, kDefaultStandaloneVadMode);
+  if (err != 0) {
+    WebRtcVad_Free(vad);
+    return nullptr;
+  }
+  return new StandaloneVad(vad);
+}
+
+int StandaloneVad::AddAudio(const int16_t* data, size_t length) {
+  if (length != kLength10Ms)
+    return -1;
+
+  if (index_ + length > kLength10Ms * kMaxNum10msFrames)
+    // Reset the buffer if it's full.
+    // TODO(ajm): Instead, consider just processing every 10 ms frame. Then we
+    // can forgo the buffering.
+    index_ = 0;
+
+  memcpy(&buffer_[index_], data, sizeof(int16_t) * length);
+  index_ += length;
+  return 0;
+}
+
+int StandaloneVad::GetActivity(double* p, size_t length_p) {
+  if (index_ == 0)
+    return -1;
+
+  const size_t num_frames = index_ / kLength10Ms;
+  if (num_frames > length_p)
+    return -1;
+  RTC_DCHECK_EQ(0, WebRtcVad_ValidRateAndFrameLength(kSampleRateHz, index_));
+
+  int activity = WebRtcVad_Process(vad_, kSampleRateHz, buffer_, index_);
+  if (activity < 0)
+    return -1;
+  else if (activity == 0)
+    p[0] = 0.01;  // Arbitrary but small and non-zero.
+  else
+    p[0] = 0.5;  // 0.5 is neutral values when combinned by other probabilities.
+  for (size_t n = 1; n < num_frames; n++)
+    p[n] = p[0];
+  // Reset the buffer to start from the beginning.
+  index_ = 0;
+  return activity;
+}
+
+int StandaloneVad::set_mode(int mode) {
+  if (mode < 0 || mode > 3)
+    return -1;
+  if (WebRtcVad_set_mode(vad_, mode) != 0)
+    return -1;
+
+  mode_ = mode;
+  return 0;
+}
+
+}  // namespace webrtc

VocieProcess/modules/audio_processing/vad/standalone_vad.h

@@ -0,0 +1,69 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC_STANDALONE_VAD_H_
+#define MODULES_AUDIO_PROCESSING_AGC_STANDALONE_VAD_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "common_audio/vad/include/webrtc_vad.h"
+#include "modules/audio_processing/vad/common.h"
+
+namespace webrtc {
+
+class StandaloneVad {
+ public:
+  static StandaloneVad* Create();
+  ~StandaloneVad();
+
+  // Outputs
+  //   p: a buffer where probabilities are written to.
+  //   length_p: number of elements of `p`.
+  //
+  // return value:
+  //    -1: if no audio is stored or VAD returns error.
+  //     0: in success.
+  // In case of error the content of `activity` is unchanged.
+  //
+  // Note that due to a high false-positive (VAD decision is active while the
+  // processed audio is just background noise) rate, stand-alone VAD is used as
+  // a one-sided indicator. The activity probability is 0.5 if the frame is
+  // classified as active, and the probability is 0.01 if the audio is
+  // classified as passive. In this way, when probabilities are combined, the
+  // effect of the stand-alone VAD is neutral if the input is classified as
+  // active.
+  int GetActivity(double* p, size_t length_p);
+
+  // Expecting 10 ms of 16 kHz audio to be pushed in.
+  int AddAudio(const int16_t* data, size_t length);
+
+  // Set aggressiveness of VAD, 0 is the least aggressive and 3 is the most
+  // aggressive mode. Returns -1 if the input is less than 0 or larger than 3,
+  // otherwise 0 is returned.
+  int set_mode(int mode);
+  // Get the agressiveness of the current VAD.
+  int mode() const { return mode_; }
+
+ private:
+  explicit StandaloneVad(VadInst* vad);
+
+  static const size_t kMaxNum10msFrames = 3;
+
+  // TODO(turajs): Is there a way to use scoped-pointer here?
+  VadInst* vad_;
+  int16_t buffer_[kMaxNum10msFrames * kLength10Ms];
+  size_t index_;
+  int mode_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC_STANDALONE_VAD_H_

VocieProcess/modules/audio_processing/vad/vad_audio_proc.cc

@@ -0,0 +1,275 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/vad/vad_audio_proc.h"
+
+#include <math.h>
+#include <stdio.h>
+#include <string.h>
+
+#include "common_audio/third_party/ooura/fft_size_256/fft4g.h"
+#include "modules/audio_processing/vad/pitch_internal.h"
+#include "modules/audio_processing/vad/pole_zero_filter.h"
+#include "modules/audio_processing/vad/vad_audio_proc_internal.h"
+#include "rtc_base/checks.h"
+extern "C" {
+#include "modules/audio_coding/codecs/isac/main/source/filter_functions.h"
+#include "modules/audio_coding/codecs/isac/main/source/isac_vad.h"
+#include "modules/audio_coding/codecs/isac/main/source/pitch_estimator.h"
+#include "modules/audio_coding/codecs/isac/main/source/structs.h"
+}
+
+namespace webrtc {
+
+// The following structures are declared anonymous in iSAC's structs.h. To
+// forward declare them, we use this derived class trick.
+struct VadAudioProc::PitchAnalysisStruct : public ::PitchAnalysisStruct {};
+struct VadAudioProc::PreFiltBankstr : public ::PreFiltBankstr {};
+
+static constexpr float kFrequencyResolution =
+    kSampleRateHz / static_cast<float>(VadAudioProc::kDftSize);
+static constexpr int kSilenceRms = 5;
+
+// TODO(turajs): Make a Create or Init for VadAudioProc.
+VadAudioProc::VadAudioProc()
+    : audio_buffer_(),
+      num_buffer_samples_(kNumPastSignalSamples),
+      log_old_gain_(-2),
+      old_lag_(50),  // Arbitrary but valid as pitch-lag (in samples).
+      pitch_analysis_handle_(new PitchAnalysisStruct),
+      pre_filter_handle_(new PreFiltBankstr),
+      high_pass_filter_(PoleZeroFilter::Create(kCoeffNumerator,
+                                               kFilterOrder,
+                                               kCoeffDenominator,
+                                               kFilterOrder)) {
+  static_assert(kNumPastSignalSamples + kNumSubframeSamples ==
+                    sizeof(kLpcAnalWin) / sizeof(kLpcAnalWin[0]),
+                "lpc analysis window incorrect size");
+  static_assert(kLpcOrder + 1 == sizeof(kCorrWeight) / sizeof(kCorrWeight[0]),
+                "correlation weight incorrect size");
+
+  // TODO(turajs): Are we doing too much in the constructor?
+  float data[kDftSize];
+  // Make FFT to initialize.
+  ip_[0] = 0;
+  WebRtc_rdft(kDftSize, 1, data, ip_, w_fft_);
+  // TODO(turajs): Need to initialize high-pass filter.
+
+  // Initialize iSAC components.
+  WebRtcIsac_InitPreFilterbank(pre_filter_handle_.get());
+  WebRtcIsac_InitPitchAnalysis(pitch_analysis_handle_.get());
+}
+
+VadAudioProc::~VadAudioProc() {}
+
+void VadAudioProc::ResetBuffer() {
+  memcpy(audio_buffer_, &audio_buffer_[kNumSamplesToProcess],
+         sizeof(audio_buffer_[0]) * kNumPastSignalSamples);
+  num_buffer_samples_ = kNumPastSignalSamples;
+}
+
+int VadAudioProc::ExtractFeatures(const int16_t* frame,
+                                  size_t length,
+                                  AudioFeatures* features) {
+  features->num_frames = 0;
+  if (length != kNumSubframeSamples) {
+    return -1;
+  }
+
+  // High-pass filter to remove the DC component and very low frequency content.
+  // We have experienced that this high-pass filtering improves voice/non-voiced
+  // classification.
+  if (high_pass_filter_->Filter(frame, kNumSubframeSamples,
+                                &audio_buffer_[num_buffer_samples_]) != 0) {
+    return -1;
+  }
+
+  num_buffer_samples_ += kNumSubframeSamples;
+  if (num_buffer_samples_ < kBufferLength) {
+    return 0;
+  }
+  RTC_DCHECK_EQ(num_buffer_samples_, kBufferLength);
+  features->num_frames = kNum10msSubframes;
+  features->silence = false;
+
+  Rms(features->rms, kMaxNumFrames);
+  for (size_t i = 0; i < kNum10msSubframes; ++i) {
+    if (features->rms[i] < kSilenceRms) {
+      // PitchAnalysis can cause NaNs in the pitch gain if it's fed silence.
+      // Bail out here instead.
+      features->silence = true;
+      ResetBuffer();
+      return 0;
+    }
+  }
+
+  PitchAnalysis(features->log_pitch_gain, features->pitch_lag_hz,
+                kMaxNumFrames);
+  FindFirstSpectralPeaks(features->spectral_peak, kMaxNumFrames);
+  ResetBuffer();
+  return 0;
+}
+
+// Computes |kLpcOrder + 1| correlation coefficients.
+void VadAudioProc::SubframeCorrelation(double* corr,
+                                       size_t length_corr,
+                                       size_t subframe_index) {
+  RTC_DCHECK_GE(length_corr, kLpcOrder + 1);
+  double windowed_audio[kNumSubframeSamples + kNumPastSignalSamples];
+  size_t buffer_index = subframe_index * kNumSubframeSamples;
+
+  for (size_t n = 0; n < kNumSubframeSamples + kNumPastSignalSamples; n++)
+    windowed_audio[n] = audio_buffer_[buffer_index++] * kLpcAnalWin[n];
+
+  WebRtcIsac_AutoCorr(corr, windowed_audio,
+                      kNumSubframeSamples + kNumPastSignalSamples, kLpcOrder);
+}
+
+// Compute `kNum10msSubframes` sets of LPC coefficients, one per 10 ms input.
+// The analysis window is 15 ms long and it is centered on the first half of
+// each 10ms sub-frame. This is equivalent to computing LPC coefficients for the
+// first half of each 10 ms subframe.
+void VadAudioProc::GetLpcPolynomials(double* lpc, size_t length_lpc) {
+  RTC_DCHECK_GE(length_lpc, kNum10msSubframes * (kLpcOrder + 1));
+  double corr[kLpcOrder + 1];
+  double reflec_coeff[kLpcOrder];
+  for (size_t i = 0, offset_lpc = 0; i < kNum10msSubframes;
+       i++, offset_lpc += kLpcOrder + 1) {
+    SubframeCorrelation(corr, kLpcOrder + 1, i);
+    corr[0] *= 1.0001;
+    // This makes Lev-Durb a bit more stable.
+    for (size_t k = 0; k < kLpcOrder + 1; k++) {
+      corr[k] *= kCorrWeight[k];
+    }
+    WebRtcIsac_LevDurb(&lpc[offset_lpc], reflec_coeff, corr, kLpcOrder);
+  }
+}
+
+// Fit a second order curve to these 3 points and find the location of the
+// extremum. The points are inverted before curve fitting.
+static float QuadraticInterpolation(float prev_val,
+                                    float curr_val,
+                                    float next_val) {
+  // Doing the interpolation in |1 / A(z)|^2.
+  float fractional_index = 0;
+  next_val = 1.0f / next_val;
+  prev_val = 1.0f / prev_val;
+  curr_val = 1.0f / curr_val;
+
+  fractional_index =
+      -(next_val - prev_val) * 0.5f / (next_val + prev_val - 2.f * curr_val);
+  RTC_DCHECK_LT(fabs(fractional_index), 1);
+  return fractional_index;
+}
+
+// 1 / A(z), where A(z) is defined by `lpc` is a model of the spectral envelope
+// of the input signal. The local maximum of the spectral envelope corresponds
+// with the local minimum of A(z). It saves complexity, as we save one
+// inversion. Furthermore, we find the first local maximum of magnitude squared,
+// to save on one square root.
+void VadAudioProc::FindFirstSpectralPeaks(double* f_peak,
+                                          size_t length_f_peak) {
+  RTC_DCHECK_GE(length_f_peak, kNum10msSubframes);
+  double lpc[kNum10msSubframes * (kLpcOrder + 1)];
+  // For all sub-frames.
+  GetLpcPolynomials(lpc, kNum10msSubframes * (kLpcOrder + 1));
+
+  const size_t kNumDftCoefficients = kDftSize / 2 + 1;
+  float data[kDftSize];
+
+  for (size_t i = 0; i < kNum10msSubframes; i++) {
+    // Convert to float with zero pad.
+    memset(data, 0, sizeof(data));
+    for (size_t n = 0; n < kLpcOrder + 1; n++) {
+      data[n] = static_cast<float>(lpc[i * (kLpcOrder + 1) + n]);
+    }
+    // Transform to frequency domain.
+    WebRtc_rdft(kDftSize, 1, data, ip_, w_fft_);
+
+    size_t index_peak = 0;
+    float prev_magn_sqr = data[0] * data[0];
+    float curr_magn_sqr = data[2] * data[2] + data[3] * data[3];
+    float next_magn_sqr;
+    bool found_peak = false;
+    for (size_t n = 2; n < kNumDftCoefficients - 1; n++) {
+      next_magn_sqr =
+          data[2 * n] * data[2 * n] + data[2 * n + 1] * data[2 * n + 1];
+      if (curr_magn_sqr < prev_magn_sqr && curr_magn_sqr < next_magn_sqr) {
+        found_peak = true;
+        index_peak = n - 1;
+        break;
+      }
+      prev_magn_sqr = curr_magn_sqr;
+      curr_magn_sqr = next_magn_sqr;
+    }
+    float fractional_index = 0;
+    if (!found_peak) {
+      // Checking if |kNumDftCoefficients - 1| is the local minimum.
+      next_magn_sqr = data[1] * data[1];
+      if (curr_magn_sqr < prev_magn_sqr && curr_magn_sqr < next_magn_sqr) {
+        index_peak = kNumDftCoefficients - 1;
+      }
+    } else {
+      // A peak is found, do a simple quadratic interpolation to get a more
+      // accurate estimate of the peak location.
+      fractional_index =
+          QuadraticInterpolation(prev_magn_sqr, curr_magn_sqr, next_magn_sqr);
+    }
+    f_peak[i] = (index_peak + fractional_index) * kFrequencyResolution;
+  }
+}
+
+// Using iSAC functions to estimate pitch gains & lags.
+void VadAudioProc::PitchAnalysis(double* log_pitch_gains,
+                                 double* pitch_lags_hz,
+                                 size_t length) {
+  // TODO(turajs): This can be "imported" from iSAC & and the next two
+  // constants.
+  RTC_DCHECK_GE(length, kNum10msSubframes);
+  const int kNumPitchSubframes = 4;
+  double gains[kNumPitchSubframes];
+  double lags[kNumPitchSubframes];
+
+  const int kNumSubbandFrameSamples = 240;
+  const int kNumLookaheadSamples = 24;
+
+  float lower[kNumSubbandFrameSamples];
+  float upper[kNumSubbandFrameSamples];
+  double lower_lookahead[kNumSubbandFrameSamples];
+  double upper_lookahead[kNumSubbandFrameSamples];
+  double lower_lookahead_pre_filter[kNumSubbandFrameSamples +
+                                    kNumLookaheadSamples];
+
+  // Split signal to lower and upper bands
+  WebRtcIsac_SplitAndFilterFloat(&audio_buffer_[kNumPastSignalSamples], lower,
+                                 upper, lower_lookahead, upper_lookahead,
+                                 pre_filter_handle_.get());
+  WebRtcIsac_PitchAnalysis(lower_lookahead, lower_lookahead_pre_filter,
+                           pitch_analysis_handle_.get(), lags, gains);
+
+  // Lags are computed on lower-band signal with sampling rate half of the
+  // input signal.
+  GetSubframesPitchParameters(
+      kSampleRateHz / 2, gains, lags, kNumPitchSubframes, kNum10msSubframes,
+      &log_old_gain_, &old_lag_, log_pitch_gains, pitch_lags_hz);
+}
+
+void VadAudioProc::Rms(double* rms, size_t length_rms) {
+  RTC_DCHECK_GE(length_rms, kNum10msSubframes);
+  size_t offset = kNumPastSignalSamples;
+  for (size_t i = 0; i < kNum10msSubframes; i++) {
+    rms[i] = 0;
+    for (size_t n = 0; n < kNumSubframeSamples; n++, offset++)
+      rms[i] += audio_buffer_[offset] * audio_buffer_[offset];
+    rms[i] = sqrt(rms[i] / kNumSubframeSamples);
+  }
+}
+
+}  // namespace webrtc

VocieProcess/modules/audio_processing/vad/vad_audio_proc.h

@@ -0,0 +1,88 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_VAD_VAD_AUDIO_PROC_H_
+#define MODULES_AUDIO_PROCESSING_VAD_VAD_AUDIO_PROC_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <memory>
+
+#include "modules/audio_processing/vad/common.h"  // AudioFeatures, kSampleR...
+
+namespace webrtc {
+
+class PoleZeroFilter;
+
+class VadAudioProc {
+ public:
+  // Forward declare iSAC structs.
+  struct PitchAnalysisStruct;
+  struct PreFiltBankstr;
+
+  VadAudioProc();
+  ~VadAudioProc();
+
+  int ExtractFeatures(const int16_t* audio_frame,
+                      size_t length,
+                      AudioFeatures* audio_features);
+
+  static constexpr size_t kDftSize = 512;
+
+ private:
+  void PitchAnalysis(double* pitch_gains, double* pitch_lags_hz, size_t length);
+  void SubframeCorrelation(double* corr,
+                           size_t length_corr,
+                           size_t subframe_index);
+  void GetLpcPolynomials(double* lpc, size_t length_lpc);
+  void FindFirstSpectralPeaks(double* f_peak, size_t length_f_peak);
+  void Rms(double* rms, size_t length_rms);
+  void ResetBuffer();
+
+  // To compute spectral peak we perform LPC analysis to get spectral envelope.
+  // For every 30 ms we compute 3 spectral peak there for 3 LPC analysis.
+  // LPC is computed over 15 ms of windowed audio. For every 10 ms sub-frame
+  // we need 5 ms of past signal to create the input of LPC analysis.
+  static constexpr size_t kNumPastSignalSamples = size_t{kSampleRateHz / 200};
+
+  // TODO(turajs): maybe defining this at a higher level (maybe enum) so that
+  // all the code recognize it as "no-error."
+  static constexpr int kNoError = 0;
+
+  static constexpr size_t kNum10msSubframes = 3;
+  static constexpr size_t kNumSubframeSamples = size_t{kSampleRateHz / 100};
+  // Samples in 30 ms @ given sampling rate.
+  static constexpr size_t kNumSamplesToProcess =
+      kNum10msSubframes * kNumSubframeSamples;
+  static constexpr size_t kBufferLength =
+      kNumPastSignalSamples + kNumSamplesToProcess;
+  static constexpr size_t kIpLength = kDftSize >> 1;
+  static constexpr size_t kWLength = kDftSize >> 1;
+  static constexpr size_t kLpcOrder = 16;
+
+  size_t ip_[kIpLength];
+  float w_fft_[kWLength];
+
+  // A buffer of 5 ms (past audio) + 30 ms (one iSAC frame ).
+  float audio_buffer_[kBufferLength];
+  size_t num_buffer_samples_;
+
+  double log_old_gain_;
+  double old_lag_;
+
+  std::unique_ptr<PitchAnalysisStruct> pitch_analysis_handle_;
+  std::unique_ptr<PreFiltBankstr> pre_filter_handle_;
+  std::unique_ptr<PoleZeroFilter> high_pass_filter_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_VAD_VAD_AUDIO_PROC_H_

VocieProcess/modules/audio_processing/vad/vad_audio_proc_internal.h

@@ -0,0 +1,81 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_VAD_VAD_AUDIO_PROC_INTERNAL_H_
+#define MODULES_AUDIO_PROCESSING_VAD_VAD_AUDIO_PROC_INTERNAL_H_
+
+#include <stddef.h>
+
+namespace webrtc {
+
+// These values should match MATLAB counterparts for unit-tests to pass.
+static const double kCorrWeight[] = {
+    1.000000, 0.985000, 0.970225, 0.955672, 0.941337, 0.927217,
+    0.913308, 0.899609, 0.886115, 0.872823, 0.859730, 0.846834,
+    0.834132, 0.821620, 0.809296, 0.797156, 0.785199};
+
+static const double kLpcAnalWin[] = {
+    0.00000000, 0.01314436, 0.02628645, 0.03942400, 0.05255473, 0.06567639,
+    0.07878670, 0.09188339, 0.10496421, 0.11802689, 0.13106918, 0.14408883,
+    0.15708358, 0.17005118, 0.18298941, 0.19589602, 0.20876878, 0.22160547,
+    0.23440387, 0.24716177, 0.25987696, 0.27254725, 0.28517045, 0.29774438,
+    0.31026687, 0.32273574, 0.33514885, 0.34750406, 0.35979922, 0.37203222,
+    0.38420093, 0.39630327, 0.40833713, 0.42030043, 0.43219112, 0.44400713,
+    0.45574642, 0.46740697, 0.47898676, 0.49048379, 0.50189608, 0.51322164,
+    0.52445853, 0.53560481, 0.54665854, 0.55761782, 0.56848075, 0.57924546,
+    0.58991008, 0.60047278, 0.61093173, 0.62128512, 0.63153117, 0.64166810,
+    0.65169416, 0.66160761, 0.67140676, 0.68108990, 0.69065536, 0.70010148,
+    0.70942664, 0.71862923, 0.72770765, 0.73666033, 0.74548573, 0.75418233,
+    0.76274862, 0.77118312, 0.77948437, 0.78765094, 0.79568142, 0.80357442,
+    0.81132858, 0.81894256, 0.82641504, 0.83374472, 0.84093036, 0.84797069,
+    0.85486451, 0.86161063, 0.86820787, 0.87465511, 0.88095122, 0.88709512,
+    0.89308574, 0.89892206, 0.90460306, 0.91012776, 0.91549520, 0.92070447,
+    0.92575465, 0.93064488, 0.93537432, 0.93994213, 0.94434755, 0.94858979,
+    0.95266814, 0.95658189, 0.96033035, 0.96391289, 0.96732888, 0.97057773,
+    0.97365889, 0.97657181, 0.97931600, 0.98189099, 0.98429632, 0.98653158,
+    0.98859639, 0.99049038, 0.99221324, 0.99376466, 0.99514438, 0.99635215,
+    0.99738778, 0.99825107, 0.99894188, 0.99946010, 0.99980562, 0.99997840,
+    0.99997840, 0.99980562, 0.99946010, 0.99894188, 0.99825107, 0.99738778,
+    0.99635215, 0.99514438, 0.99376466, 0.99221324, 0.99049038, 0.98859639,
+    0.98653158, 0.98429632, 0.98189099, 0.97931600, 0.97657181, 0.97365889,
+    0.97057773, 0.96732888, 0.96391289, 0.96033035, 0.95658189, 0.95266814,
+    0.94858979, 0.94434755, 0.93994213, 0.93537432, 0.93064488, 0.92575465,
+    0.92070447, 0.91549520, 0.91012776, 0.90460306, 0.89892206, 0.89308574,
+    0.88709512, 0.88095122, 0.87465511, 0.86820787, 0.86161063, 0.85486451,
+    0.84797069, 0.84093036, 0.83374472, 0.82641504, 0.81894256, 0.81132858,
+    0.80357442, 0.79568142, 0.78765094, 0.77948437, 0.77118312, 0.76274862,
+    0.75418233, 0.74548573, 0.73666033, 0.72770765, 0.71862923, 0.70942664,
+    0.70010148, 0.69065536, 0.68108990, 0.67140676, 0.66160761, 0.65169416,
+    0.64166810, 0.63153117, 0.62128512, 0.61093173, 0.60047278, 0.58991008,
+    0.57924546, 0.56848075, 0.55761782, 0.54665854, 0.53560481, 0.52445853,
+    0.51322164, 0.50189608, 0.49048379, 0.47898676, 0.46740697, 0.45574642,
+    0.44400713, 0.43219112, 0.42030043, 0.40833713, 0.39630327, 0.38420093,
+    0.37203222, 0.35979922, 0.34750406, 0.33514885, 0.32273574, 0.31026687,
+    0.29774438, 0.28517045, 0.27254725, 0.25987696, 0.24716177, 0.23440387,
+    0.22160547, 0.20876878, 0.19589602, 0.18298941, 0.17005118, 0.15708358,
+    0.14408883, 0.13106918, 0.11802689, 0.10496421, 0.09188339, 0.07878670,
+    0.06567639, 0.05255473, 0.03942400, 0.02628645, 0.01314436, 0.00000000};
+
+static const size_t kFilterOrder = 2;
+static const float kCoeffNumerator[kFilterOrder + 1] = {0.974827f, -1.949650f,
+                                                        0.974827f};
+static const float kCoeffDenominator[kFilterOrder + 1] = {1.0f, -1.971999f,
+                                                          0.972457f};
+
+static_assert(kFilterOrder + 1 ==
+                  sizeof(kCoeffNumerator) / sizeof(kCoeffNumerator[0]),
+              "numerator coefficients incorrect size");
+static_assert(kFilterOrder + 1 ==
+                  sizeof(kCoeffDenominator) / sizeof(kCoeffDenominator[0]),
+              "denominator coefficients incorrect size");
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_VAD_VAD_AUDIO_PROCESSING_H_

VocieProcess/modules/audio_processing/vad/vad_circular_buffer.cc

@@ -0,0 +1,135 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/vad/vad_circular_buffer.h"
+
+#include <stdlib.h>
+
+namespace webrtc {
+
+VadCircularBuffer::VadCircularBuffer(int buffer_size)
+    : buffer_(new double[buffer_size]),
+      is_full_(false),
+      index_(0),
+      buffer_size_(buffer_size),
+      sum_(0) {}
+
+VadCircularBuffer::~VadCircularBuffer() {}
+
+void VadCircularBuffer::Reset() {
+  is_full_ = false;
+  index_ = 0;
+  sum_ = 0;
+}
+
+VadCircularBuffer* VadCircularBuffer::Create(int buffer_size) {
+  if (buffer_size <= 0)
+    return NULL;
+  return new VadCircularBuffer(buffer_size);
+}
+
+double VadCircularBuffer::Oldest() const {
+  if (!is_full_)
+    return buffer_[0];
+  else
+    return buffer_[index_];
+}
+
+double VadCircularBuffer::Mean() {
+  double m;
+  if (is_full_) {
+    m = sum_ / buffer_size_;
+  } else {
+    if (index_ > 0)
+      m = sum_ / index_;
+    else
+      m = 0;
+  }
+  return m;
+}
+
+void VadCircularBuffer::Insert(double value) {
+  if (is_full_) {
+    sum_ -= buffer_[index_];
+  }
+  sum_ += value;
+  buffer_[index_] = value;
+  index_++;
+  if (index_ >= buffer_size_) {
+    is_full_ = true;
+    index_ = 0;
+  }
+}
+int VadCircularBuffer::BufferLevel() {
+  if (is_full_)
+    return buffer_size_;
+  return index_;
+}
+
+int VadCircularBuffer::Get(int index, double* value) const {
+  int err = ConvertToLinearIndex(&index);
+  if (err < 0)
+    return -1;
+  *value = buffer_[index];
+  return 0;
+}
+
+int VadCircularBuffer::Set(int index, double value) {
+  int err = ConvertToLinearIndex(&index);
+  if (err < 0)
+    return -1;
+
+  sum_ -= buffer_[index];
+  buffer_[index] = value;
+  sum_ += value;
+  return 0;
+}
+
+int VadCircularBuffer::ConvertToLinearIndex(int* index) const {
+  if (*index < 0 || *index >= buffer_size_)
+    return -1;
+
+  if (!is_full_ && *index >= index_)
+    return -1;
+
+  *index = index_ - 1 - *index;
+  if (*index < 0)
+    *index += buffer_size_;
+  return 0;
+}
+
+int VadCircularBuffer::RemoveTransient(int width_threshold,
+                                       double val_threshold) {
+  if (!is_full_ && index_ < width_threshold + 2)
+    return 0;
+
+  int index_1 = 0;
+  int index_2 = width_threshold + 1;
+  double v = 0;
+  if (Get(index_1, &v) < 0)
+    return -1;
+  if (v < val_threshold) {
+    Set(index_1, 0);
+    int index;
+    for (index = index_2; index > index_1; index--) {
+      if (Get(index, &v) < 0)
+        return -1;
+      if (v < val_threshold)
+        break;
+    }
+    for (; index > index_1; index--) {
+      if (Set(index, 0.0) < 0)
+        return -1;
+    }
+  }
+  return 0;
+}
+
+}  // namespace webrtc

VocieProcess/modules/audio_processing/vad/vad_circular_buffer.h

@@ -0,0 +1,69 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_VAD_VAD_CIRCULAR_BUFFER_H_
+#define MODULES_AUDIO_PROCESSING_VAD_VAD_CIRCULAR_BUFFER_H_
+
+#include <memory>
+
+namespace webrtc {
+
+// A circular buffer tailored to the need of this project. It stores last
+// K samples of the input, and keeps track of the mean of the last samples.
+//
+// It is used in class "PitchBasedActivity" to keep track of posterior
+// probabilities in the past few seconds. The posterior probabilities are used
+// to recursively update prior probabilities.
+class VadCircularBuffer {
+ public:
+  static VadCircularBuffer* Create(int buffer_size);
+  ~VadCircularBuffer();
+
+  // If buffer is wrapped around.
+  bool is_full() const { return is_full_; }
+  // Get the oldest entry in the buffer.
+  double Oldest() const;
+  // Insert new value into the buffer.
+  void Insert(double value);
+  // Reset buffer, forget the past, start fresh.
+  void Reset();
+
+  // The mean value of the elements in the buffer. The return value is zero if
+  // buffer is empty, i.e. no value is inserted.
+  double Mean();
+  // Remove transients. If the values exceed `val_threshold` for a period
+  // shorter then or equal to `width_threshold`, then that period is considered
+  // transient and set to zero.
+  int RemoveTransient(int width_threshold, double val_threshold);
+
+ private:
+  explicit VadCircularBuffer(int buffer_size);
+  // Get previous values. |index = 0| corresponds to the most recent
+  // insertion. |index = 1| is the one before the most recent insertion, and
+  // so on.
+  int Get(int index, double* value) const;
+  // Set a given position to `value`. `index` is interpreted as above.
+  int Set(int index, double value);
+  // Return the number of valid elements in the buffer.
+  int BufferLevel();
+
+  // Convert an index with the interpretation as get() method to the
+  // corresponding linear index.
+  int ConvertToLinearIndex(int* index) const;
+
+  std::unique_ptr<double[]> buffer_;
+  bool is_full_;
+  int index_;
+  int buffer_size_;
+  double sum_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_PROCESSING_VAD_VAD_CIRCULAR_BUFFER_H_

VocieProcess/modules/audio_processing/vad/voice_activity_detector.cc

@@ -0,0 +1,85 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/vad/voice_activity_detector.h"
+
+#include <algorithm>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace {
+
+const size_t kNumChannels = 1;
+
+const double kDefaultVoiceValue = 1.0;
+const double kNeutralProbability = 0.5;
+const double kLowProbability = 0.01;
+
+}  // namespace
+
+VoiceActivityDetector::VoiceActivityDetector()
+    : last_voice_probability_(kDefaultVoiceValue),
+      standalone_vad_(StandaloneVad::Create()) {}
+
+VoiceActivityDetector::~VoiceActivityDetector() = default;
+
+// Because ISAC has a different chunk length, it updates
+// `chunkwise_voice_probabilities_` and `chunkwise_rms_` when there is new data.
+// Otherwise it clears them.
+void VoiceActivityDetector::ProcessChunk(const int16_t* audio,
+                                         size_t length,
+                                         int sample_rate_hz) {
+  RTC_DCHECK_EQ(length, sample_rate_hz / 100);
+  // TODO(bugs.webrtc.org/7494): Remove resampling and force 16 kHz audio.
+  // Resample to the required rate.
+  const int16_t* resampled_ptr = audio;
+  if (sample_rate_hz != kSampleRateHz) {
+    RTC_CHECK_EQ(
+        resampler_.ResetIfNeeded(sample_rate_hz, kSampleRateHz, kNumChannels),
+        0);
+    resampler_.Push(audio, length, resampled_, kLength10Ms, length);
+    resampled_ptr = resampled_;
+  }
+  RTC_DCHECK_EQ(length, kLength10Ms);
+
+  // Each chunk needs to be passed into `standalone_vad_`, because internally it
+  // buffers the audio and processes it all at once when GetActivity() is
+  // called.
+  RTC_CHECK_EQ(standalone_vad_->AddAudio(resampled_ptr, length), 0);
+
+  audio_processing_.ExtractFeatures(resampled_ptr, length, &features_);
+
+  chunkwise_voice_probabilities_.resize(features_.num_frames);
+  chunkwise_rms_.resize(features_.num_frames);
+  std::copy(features_.rms, features_.rms + chunkwise_rms_.size(),
+            chunkwise_rms_.begin());
+  if (features_.num_frames > 0) {
+    if (features_.silence) {
+      // The other features are invalid, so set the voice probabilities to an
+      // arbitrary low value.
+      std::fill(chunkwise_voice_probabilities_.begin(),
+                chunkwise_voice_probabilities_.end(), kLowProbability);
+    } else {
+      std::fill(chunkwise_voice_probabilities_.begin(),
+                chunkwise_voice_probabilities_.end(), kNeutralProbability);
+      RTC_CHECK_GE(
+          standalone_vad_->GetActivity(&chunkwise_voice_probabilities_[0],
+                                       chunkwise_voice_probabilities_.size()),
+          0);
+      RTC_CHECK_GE(pitch_based_vad_.VoicingProbability(
+                       features_, &chunkwise_voice_probabilities_[0]),
+                   0);
+    }
+    last_voice_probability_ = chunkwise_voice_probabilities_.back();
+  }
+}
+
+}  // namespace webrtc

VocieProcess/modules/audio_processing/vad/voice_activity_detector.h

@@ -0,0 +1,74 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_VAD_VOICE_ACTIVITY_DETECTOR_H_
+#define MODULES_AUDIO_PROCESSING_VAD_VOICE_ACTIVITY_DETECTOR_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <memory>
+#include <vector>
+
+#include "common_audio/resampler/include/resampler.h"
+#include "modules/audio_processing/vad/common.h"
+#include "modules/audio_processing/vad/pitch_based_vad.h"
+#include "modules/audio_processing/vad/standalone_vad.h"
+#include "modules/audio_processing/vad/vad_audio_proc.h"
+
+namespace webrtc {
+
+// A Voice Activity Detector (VAD) that combines the voice probability from the
+// StandaloneVad and PitchBasedVad to get a more robust estimation.
+class VoiceActivityDetector {
+ public:
+  VoiceActivityDetector();
+  ~VoiceActivityDetector();
+
+  // Processes each audio chunk and estimates the voice probability.
+  // TODO(bugs.webrtc.org/7494): Switch to rtc::ArrayView and remove
+  // `sample_rate_hz`.
+  void ProcessChunk(const int16_t* audio, size_t length, int sample_rate_hz);
+
+  // Returns a vector of voice probabilities for each chunk. It can be empty for
+  // some chunks, but it catches up afterwards returning multiple values at
+  // once.
+  const std::vector<double>& chunkwise_voice_probabilities() const {
+    return chunkwise_voice_probabilities_;
+  }
+
+  // Returns a vector of RMS values for each chunk. It has the same length as
+  // chunkwise_voice_probabilities().
+  const std::vector<double>& chunkwise_rms() const { return chunkwise_rms_; }
+
+  // Returns the last voice probability, regardless of the internal
+  // implementation, although it has a few chunks of delay.
+  float last_voice_probability() const { return last_voice_probability_; }
+
+ private:
+  // TODO(aluebs): Change these to float.
+  std::vector<double> chunkwise_voice_probabilities_;
+  std::vector<double> chunkwise_rms_;
+
+  float last_voice_probability_;
+
+  Resampler resampler_;
+  VadAudioProc audio_processing_;
+
+  std::unique_ptr<StandaloneVad> standalone_vad_;
+  PitchBasedVad pitch_based_vad_;
+
+  int16_t resampled_[kLength10Ms];
+  AudioFeatures features_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_VAD_VOICE_ACTIVITY_DETECTOR_H_

VocieProcess/modules/audio_processing/vad/voice_gmm_tables.h

@@ -0,0 +1,77 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// GMM tables for active segments. Generated by MakeGmmTables.m.
+
+#ifndef MODULES_AUDIO_PROCESSING_VAD_VOICE_GMM_TABLES_H_
+#define MODULES_AUDIO_PROCESSING_VAD_VOICE_GMM_TABLES_H_
+
+static const int kVoiceGmmNumMixtures = 12;
+static const int kVoiceGmmDim = 3;
+
+static const double
+    kVoiceGmmCovarInverse[kVoiceGmmNumMixtures][kVoiceGmmDim][kVoiceGmmDim] = {
+        {{1.83673825579513e+00, -8.09791637570095e-04, 4.60106414365986e-03},
+         {-8.09791637570095e-04, 8.89351738394608e-04, -9.80188953277734e-04},
+         {4.60106414365986e-03, -9.80188953277734e-04, 1.38706060206582e-03}},
+        {{6.76228912850703e+01, -1.98893120119660e-02, -3.53548357253551e-03},
+         {-1.98893120119660e-02, 3.96216858500530e-05, -4.08492938394097e-05},
+         {-3.53548357253551e-03, -4.08492938394097e-05, 9.31864352856416e-04}},
+        {{9.98612435944558e+00, -5.27880954316893e-03, -6.30342541619017e-03},
+         {-5.27880954316893e-03, 4.54359480225226e-05, 6.30804591626044e-05},
+         {-6.30342541619017e-03, 6.30804591626044e-05, 5.36466441382942e-04}},
+        {{3.39917474216349e+01, -1.56213579433191e-03, -4.01459014990225e-02},
+         {-1.56213579433191e-03, 6.40415424897724e-05, 6.20076342427833e-05},
+         {-4.01459014990225e-02, 6.20076342427833e-05, 3.51199070103063e-03}},
+        {{1.34545062271428e+01, -7.94513610147144e-03, -5.34401019341728e-02},
+         {-7.94513610147144e-03, 1.16511820098649e-04, 4.66063702069293e-05},
+         {-5.34401019341728e-02, 4.66063702069293e-05, 2.72354323774163e-03}},
+        {{1.08557844314806e+02, -1.54885805673668e-02, -1.88029692674851e-02},
+         {-1.54885805673668e-02, 1.16404042786406e-04, 6.45579292702802e-06},
+         {-1.88029692674851e-02, 6.45579292702802e-06, 4.32330478391416e-04}},
+        {{8.22940066541450e+01, -1.15903110231303e-02, -4.92166764865343e-02},
+         {-1.15903110231303e-02, 7.42510742165261e-05, 3.73007314191290e-06},
+         {-4.92166764865343e-02, 3.73007314191290e-06, 3.64005221593244e-03}},
+        {{2.31133605685660e+00, -7.83261568950254e-04, 7.45744012346313e-04},
+         {-7.83261568950254e-04, 1.29460648214142e-05, -2.22774455093730e-06},
+         {7.45744012346313e-04, -2.22774455093730e-06, 1.05117294093010e-04}},
+        {{3.78767849189611e+02, 1.57759761011568e-03, -2.08551217988774e-02},
+         {1.57759761011568e-03, 4.76066236886865e-05, -2.33977412299324e-05},
+         {-2.08551217988774e-02, -2.33977412299324e-05, 5.24261005371196e-04}},
+        {{6.98580096506135e-01, -5.13850255217378e-04, -4.01124551717056e-04},
+         {-5.13850255217378e-04, 1.40501021984840e-06, -2.09496928716569e-06},
+         {-4.01124551717056e-04, -2.09496928716569e-06, 2.82879357740037e-04}},
+        {{2.62770945162399e+00, -2.31825753241430e-03, -5.30447217466318e-03},
+         {-2.31825753241430e-03, 4.59108572227649e-05, 7.67631886355405e-05},
+         {-5.30447217466318e-03, 7.67631886355405e-05, 2.28521601674098e-03}},
+        {{1.89940391362152e+02, -4.23280856852379e-03, -2.70608873541399e-02},
+         {-4.23280856852379e-03, 6.77547582742563e-05, 2.69154203800467e-05},
+         {-2.70608873541399e-02, 2.69154203800467e-05, 3.88574543373470e-03}}};
+
+static const double kVoiceGmmMean[kVoiceGmmNumMixtures][kVoiceGmmDim] = {
+    {-2.15020241646536e+00, 4.97079062999877e+02, 4.77078119504505e+02},
+    {-8.92097680029190e-01, 5.92064964199921e+02, 1.81045145941059e+02},
+    {-1.29435784144398e+00, 4.98450293410611e+02, 1.71991263804064e+02},
+    {-1.03925228397884e+00, 4.99511274321571e+02, 1.05838336539105e+02},
+    {-1.29229047206129e+00, 4.15026762566707e+02, 1.12861119017125e+02},
+    {-7.88748114599810e-01, 4.48739336688113e+02, 1.89784216956337e+02},
+    {-8.77777402332642e-01, 4.86620285054533e+02, 1.13477708016491e+02},
+    {-2.06465957063057e+00, 6.33385049870607e+02, 2.32758546796149e+02},
+    {-6.98893789231685e-01, 5.93622051503385e+02, 1.92536982473203e+02},
+    {-2.55901217508894e+00, 1.55914919756205e+03, 1.39769980835570e+02},
+    {-1.92070024165837e+00, 4.87983940444185e+02, 1.02745468128289e+02},
+    {-7.29187507662854e-01, 5.22717685022855e+02, 1.16377942283991e+02}};
+
+static const double kVoiceGmmWeights[kVoiceGmmNumMixtures] = {
+    -1.39789694361035e+01, -1.19527720202104e+01, -1.32396317929055e+01,
+    -1.09436815209238e+01, -1.13440027478149e+01, -1.12200721834504e+01,
+    -1.02537324043693e+01, -1.60789861938302e+01, -1.03394494048344e+01,
+    -1.83207938586818e+01, -1.31186044948288e+01, -9.52479998673554e+00};
+#endif  // MODULES_AUDIO_PROCESSING_VAD_VOICE_GMM_TABLES_H_

VocieProcess/modules/third_party/fft/fft.c

@@ -0,0 +1,942 @@
+/*
+ * Copyright(c)1995,97 Mark Olesen <olesen@me.QueensU.CA>
+ *    Queen's Univ at Kingston (Canada)
+ *
+ * Permission to use, copy, modify, and distribute this software for
+ * any purpose without fee is hereby granted, provided that this
+ * entire notice is included in all copies of any software which is
+ * or includes a copy or modification of this software and in all
+ * copies of the supporting documentation for such software.
+ *
+ * THIS SOFTWARE IS BEING PROVIDED "AS IS", WITHOUT ANY EXPRESS OR
+ * IMPLIED WARRANTY.  IN PARTICULAR, NEITHER THE AUTHOR NOR QUEEN'S
+ * UNIVERSITY AT KINGSTON MAKES ANY REPRESENTATION OR WARRANTY OF ANY
+ * KIND CONCERNING THE MERCHANTABILITY OF THIS SOFTWARE OR ITS
+ * FITNESS FOR ANY PARTICULAR PURPOSE.
+ *
+ * All of which is to say that you can do what you like with this
+ * source code provided you don't try to sell it as your own and you
+ * include an unaltered copy of this message (including the
+ * copyright).
+ *
+ * It is also implicitly understood that bug fixes and improvements
+ * should make their way back to the general Internet community so
+ * that everyone benefits.
+ *
+ * Changes:
+ *   Trivial type modifications by the WebRTC authors.
+ */
+
+
+/*
+ * File:
+ * WebRtcIsac_Fftn.c
+ *
+ * Public:
+ * WebRtcIsac_Fftn / fftnf ();
+ *
+ * Private:
+ * WebRtcIsac_Fftradix / fftradixf ();
+ *
+ * Descript:
+ * multivariate complex Fourier transform, computed in place
+ * using mixed-radix Fast Fourier Transform algorithm.
+ *
+ * Fortran code by:
+ * RC Singleton, Stanford Research Institute, Sept. 1968
+ *
+ * translated by f2c (version 19950721).
+ *
+ * int WebRtcIsac_Fftn (int ndim, const int dims[], REAL Re[], REAL Im[],
+ *     int iSign, double scaling);
+ *
+ * NDIM = the total number dimensions
+ * DIMS = a vector of array sizes
+ * if NDIM is zero then DIMS must be zero-terminated
+ *
+ * RE and IM hold the real and imaginary components of the data, and return
+ * the resulting real and imaginary Fourier coefficients.  Multidimensional
+ * data *must* be allocated contiguously.  There is no limit on the number
+ * of dimensions.
+ *
+ * ISIGN = the sign of the complex exponential (ie, forward or inverse FFT)
+ * the magnitude of ISIGN (normally 1) is used to determine the
+ * correct indexing increment (see below).
+ *
+ * SCALING = normalizing constant by which the final result is *divided*
+ * if SCALING == -1, normalize by total dimension of the transform
+ * if SCALING <  -1, normalize by the square-root of the total dimension
+ *
+ * example:
+ * tri-variate transform with Re[n1][n2][n3], Im[n1][n2][n3]
+ *
+ * int dims[3] = {n1,n2,n3}
+ * WebRtcIsac_Fftn (3, dims, Re, Im, 1, scaling);
+ *
+ *-----------------------------------------------------------------------*
+ * int WebRtcIsac_Fftradix (REAL Re[], REAL Im[], size_t nTotal, size_t nPass,
+ *   size_t nSpan, int iSign, size_t max_factors,
+ *   size_t max_perm);
+ *
+ * RE, IM - see above documentation
+ *
+ * Although there is no limit on the number of dimensions, WebRtcIsac_Fftradix() must
+ * be called once for each dimension, but the calls may be in any order.
+ *
+ * NTOTAL = the total number of complex data values
+ * NPASS  = the dimension of the current variable
+ * NSPAN/NPASS = the spacing of consecutive data values while indexing the
+ * current variable
+ * ISIGN - see above documentation
+ *
+ * example:
+ * tri-variate transform with Re[n1][n2][n3], Im[n1][n2][n3]
+ *
+ * WebRtcIsac_Fftradix (Re, Im, n1*n2*n3, n1,       n1, 1, maxf, maxp);
+ * WebRtcIsac_Fftradix (Re, Im, n1*n2*n3, n2,    n1*n2, 1, maxf, maxp);
+ * WebRtcIsac_Fftradix (Re, Im, n1*n2*n3, n3, n1*n2*n3, 1, maxf, maxp);
+ *
+ * single-variate transform,
+ *    NTOTAL = N = NSPAN = (number of complex data values),
+ *
+ * WebRtcIsac_Fftradix (Re, Im, n, n, n, 1, maxf, maxp);
+ *
+ * The data can also be stored in a single array with alternating real and
+ * imaginary parts, the magnitude of ISIGN is changed to 2 to give correct
+ * indexing increment, and data [0] and data [1] used to pass the initial
+ * addresses for the sequences of real and imaginary values,
+ *
+ * example:
+ * REAL data [2*NTOTAL];
+ * WebRtcIsac_Fftradix ( &data[0], &data[1], NTOTAL, nPass, nSpan, 2, maxf, maxp);
+ *
+ * for temporary allocation:
+ *
+ * MAX_FACTORS >= the maximum prime factor of NPASS
+ * MAX_PERM >= the number of prime factors of NPASS.  In addition,
+ * if the square-free portion K of NPASS has two or more prime
+ * factors, then MAX_PERM >= (K-1)
+ *
+ * storage in FACTOR for a maximum of 15 prime factors of NPASS. if NPASS
+ * has more than one square-free factor, the product of the square-free
+ * factors must be <= 210 array storage for maximum prime factor of 23 the
+ * following two constants should agree with the array dimensions.
+ *
+ *----------------------------------------------------------------------*/
+
+#include <stdlib.h>
+#include <math.h>
+
+#include "modules/third_party/fft/fft.h"
+
+/* double precision routine */
+static int
+WebRtcIsac_Fftradix (double Re[], double Im[],
+                    size_t nTotal, size_t nPass, size_t nSpan, int isign,
+                    int max_factors, unsigned int max_perm,
+                    FFTstr *fftstate);
+
+
+
+#ifndef M_PI
+# define M_PI 3.14159265358979323846264338327950288
+#endif
+
+#ifndef SIN60
+# define SIN60 0.86602540378443865 /* sin(60 deg) */
+# define COS72 0.30901699437494742 /* cos(72 deg) */
+# define SIN72 0.95105651629515357 /* sin(72 deg) */
+#endif
+
+# define REAL  double
+# define FFTN  WebRtcIsac_Fftn
+# define FFTNS  "fftn"
+# define FFTRADIX WebRtcIsac_Fftradix
+# define FFTRADIXS "fftradix"
+
+
+int  WebRtcIsac_Fftns(unsigned int ndim, const int dims[],
+                     double Re[],
+                     double Im[],
+                     int iSign,
+                     double scaling,
+                     FFTstr *fftstate)
+{
+
+  size_t nSpan, nPass, nTotal;
+  unsigned int i;
+  int ret, max_factors, max_perm;
+
+  /*
+   * tally the number of elements in the data array
+   * and determine the number of dimensions
+   */
+  nTotal = 1;
+  if (ndim && dims [0])
+  {
+    for (i = 0; i < ndim; i++)
+    {
+      if (dims [i] <= 0)
+      {
+        return -1;
+      }
+      nTotal *= dims [i];
+    }
+  }
+  else
+  {
+    ndim = 0;
+    for (i = 0; dims [i]; i++)
+    {
+      if (dims [i] <= 0)
+      {
+        return -1;
+      }
+      nTotal *= dims [i];
+      ndim++;
+    }
+  }
+
+  /* determine maximum number of factors and permuations */
+#if 1
+  /*
+   * follow John Beale's example, just use the largest dimension and don't
+   * worry about excess allocation.  May be someone else will do it?
+   */
+  max_factors = max_perm = 1;
+  for (i = 0; i < ndim; i++)
+  {
+    nSpan = dims [i];
+    if ((int)nSpan > max_factors)
+    {
+      max_factors = (int)nSpan;
+    }
+    if ((int)nSpan > max_perm)
+    {
+      max_perm = (int)nSpan;
+    }
+  }
+#else
+  /* use the constants used in the original Fortran code */
+  max_factors = 23;
+  max_perm = 209;
+#endif
+  /* loop over the dimensions: */
+  nPass = 1;
+  for (i = 0; i < ndim; i++)
+  {
+    nSpan = dims [i];
+    nPass *= nSpan;
+    ret = FFTRADIX (Re, Im, nTotal, nSpan, nPass, iSign,
+                    max_factors, max_perm, fftstate);
+    /* exit, clean-up already done */
+    if (ret)
+      return ret;
+  }
+
+  /* Divide through by the normalizing constant: */
+  if (scaling && scaling != 1.0)
+  {
+    if (iSign < 0) iSign = -iSign;
+    if (scaling < 0.0)
+    {
+      scaling = (double)nTotal;
+      if (scaling < -1.0)
+        scaling = sqrt (scaling);
+    }
+    scaling = 1.0 / scaling; /* multiply is often faster */
+    for (i = 0; i < nTotal; i += iSign)
+    {
+      Re [i] *= scaling;
+      Im [i] *= scaling;
+    }
+  }
+  return 0;
+}
+
+/*
+ * singleton's mixed radix routine
+ *
+ * could move allocation out to WebRtcIsac_Fftn(), but leave it here so that it's
+ * possible to make this a standalone function
+ */
+
+static int   FFTRADIX (REAL Re[],
+                       REAL Im[],
+                       size_t nTotal,
+                       size_t nPass,
+                       size_t nSpan,
+                       int iSign,
+                       int max_factors,
+                       unsigned int max_perm,
+                       FFTstr *fftstate)
+{
+  int ii, mfactor, kspan, ispan, inc;
+  int j, jc, jf, jj, k, k1, k2, k3, k4, kk, kt, nn, ns, nt;
+
+
+  REAL radf;
+  REAL c1, c2, c3, cd, aa, aj, ak, ajm, ajp, akm, akp;
+  REAL s1, s2, s3, sd, bb, bj, bk, bjm, bjp, bkm, bkp;
+
+  REAL *Rtmp = NULL; /* temp space for real part*/
+  REAL *Itmp = NULL; /* temp space for imaginary part */
+  REAL *Cos = NULL; /* Cosine values */
+  REAL *Sin = NULL; /* Sine values */
+
+  REAL s60 = SIN60;  /* sin(60 deg) */
+  REAL c72 = COS72;  /* cos(72 deg) */
+  REAL s72 = SIN72;  /* sin(72 deg) */
+  REAL pi2 = M_PI;  /* use PI first, 2 PI later */
+
+
+  fftstate->SpaceAlloced = 0;
+  fftstate->MaxPermAlloced = 0;
+
+
+  // initialize to avoid warnings
+  k3 = c2 = c3 = s2 = s3 = 0.0;
+
+  if (nPass < 2)
+    return 0;
+
+  /*  allocate storage */
+  if (fftstate->SpaceAlloced < max_factors * sizeof (REAL))
+  {
+#ifdef SUN_BROKEN_REALLOC
+    if (!fftstate->SpaceAlloced) /* first time */
+    {
+      fftstate->SpaceAlloced = max_factors * sizeof (REAL);
+    }
+    else
+    {
+#endif
+      fftstate->SpaceAlloced = max_factors * sizeof (REAL);
+#ifdef SUN_BROKEN_REALLOC
+    }
+#endif
+  }
+  else
+  {
+    /* allow full use of alloc'd space */
+    max_factors = fftstate->SpaceAlloced / sizeof (REAL);
+  }
+  if (fftstate->MaxPermAlloced < max_perm)
+  {
+#ifdef SUN_BROKEN_REALLOC
+    if (!fftstate->MaxPermAlloced) /* first time */
+    else
+#endif
+      fftstate->MaxPermAlloced = max_perm;
+  }
+  else
+  {
+    /* allow full use of alloc'd space */
+    max_perm = fftstate->MaxPermAlloced;
+  }
+
+  /* assign pointers */
+  Rtmp = (REAL *) fftstate->Tmp0;
+  Itmp = (REAL *) fftstate->Tmp1;
+  Cos  = (REAL *) fftstate->Tmp2;
+  Sin  = (REAL *) fftstate->Tmp3;
+
+  /*
+   * Function Body
+   */
+  inc = iSign;
+  if (iSign < 0) {
+    s72 = -s72;
+    s60 = -s60;
+    pi2 = -pi2;
+    inc = -inc;  /* absolute value */
+  }
+
+  /* adjust for strange increments */
+  nt = inc * (int)nTotal;
+  ns = inc * (int)nSpan;
+  kspan = ns;
+
+  nn = nt - inc;
+  jc = ns / (int)nPass;
+  radf = pi2 * (double) jc;
+  pi2 *= 2.0;   /* use 2 PI from here on */
+
+  ii = 0;
+  jf = 0;
+  /*  determine the factors of n */
+  mfactor = 0;
+  k = (int)nPass;
+  while (k % 16 == 0) {
+    mfactor++;
+    fftstate->factor [mfactor - 1] = 4;
+    k /= 16;
+  }
+  j = 3;
+  jj = 9;
+  do {
+    while (k % jj == 0) {
+      mfactor++;
+      fftstate->factor [mfactor - 1] = j;
+      k /= jj;
+    }
+    j += 2;
+    jj = j * j;
+  } while (jj <= k);
+  if (k <= 4) {
+    kt = mfactor;
+    fftstate->factor [mfactor] = k;
+    if (k != 1)
+      mfactor++;
+  } else {
+    if (k - (k / 4 << 2) == 0) {
+      mfactor++;
+      fftstate->factor [mfactor - 1] = 2;
+      k /= 4;
+    }
+    kt = mfactor;
+    j = 2;
+    do {
+      if (k % j == 0) {
+        mfactor++;
+        fftstate->factor [mfactor - 1] = j;
+        k /= j;
+      }
+      j = ((j + 1) / 2 << 1) + 1;
+    } while (j <= k);
+  }
+  if (kt) {
+    j = kt;
+    do {
+      mfactor++;
+      fftstate->factor [mfactor - 1] = fftstate->factor [j - 1];
+      j--;
+    } while (j);
+  }
+
+  /* test that mfactors is in range */
+  if (mfactor > FFT_NFACTOR)
+  {
+    return -1;
+  }
+
+  /* compute fourier transform */
+  for (;;) {
+    sd = radf / (double) kspan;
+    cd = sin(sd);
+    cd = 2.0 * cd * cd;
+    sd = sin(sd + sd);
+    kk = 0;
+    ii++;
+
+    switch (fftstate->factor [ii - 1]) {
+      case 2:
+        /* transform for factor of 2 (including rotation factor) */
+        kspan /= 2;
+        k1 = kspan + 2;
+        do {
+          do {
+            k2 = kk + kspan;
+            ak = Re [k2];
+            bk = Im [k2];
+            Re [k2] = Re [kk] - ak;
+            Im [k2] = Im [kk] - bk;
+            Re [kk] += ak;
+            Im [kk] += bk;
+            kk = k2 + kspan;
+          } while (kk < nn);
+          kk -= nn;
+        } while (kk < jc);
+        if (kk >= kspan)
+          goto Permute_Results_Label;  /* exit infinite loop */
+        do {
+          c1 = 1.0 - cd;
+          s1 = sd;
+          do {
+            do {
+              do {
+                k2 = kk + kspan;
+                ak = Re [kk] - Re [k2];
+                bk = Im [kk] - Im [k2];
+                Re [kk] += Re [k2];
+                Im [kk] += Im [k2];
+                Re [k2] = c1 * ak - s1 * bk;
+                Im [k2] = s1 * ak + c1 * bk;
+                kk = k2 + kspan;
+              } while (kk < (nt-1));
+              k2 = kk - nt;
+              c1 = -c1;
+              kk = k1 - k2;
+            } while (kk > k2);
+            ak = c1 - (cd * c1 + sd * s1);
+            s1 = sd * c1 - cd * s1 + s1;
+            c1 = 2.0 - (ak * ak + s1 * s1);
+            s1 *= c1;
+            c1 *= ak;
+            kk += jc;
+          } while (kk < k2);
+          k1 += inc + inc;
+          kk = (k1 - kspan + 1) / 2 + jc - 1;
+        } while (kk < (jc + jc));
+        break;
+
+      case 4:   /* transform for factor of 4 */
+        ispan = kspan;
+        kspan /= 4;
+
+        do {
+          c1 = 1.0;
+          s1 = 0.0;
+          do {
+            do {
+              k1 = kk + kspan;
+              k2 = k1 + kspan;
+              k3 = k2 + kspan;
+              akp = Re [kk] + Re [k2];
+              akm = Re [kk] - Re [k2];
+              ajp = Re [k1] + Re [k3];
+              ajm = Re [k1] - Re [k3];
+              bkp = Im [kk] + Im [k2];
+              bkm = Im [kk] - Im [k2];
+              bjp = Im [k1] + Im [k3];
+              bjm = Im [k1] - Im [k3];
+              Re [kk] = akp + ajp;
+              Im [kk] = bkp + bjp;
+              ajp = akp - ajp;
+              bjp = bkp - bjp;
+              if (iSign < 0) {
+                akp = akm + bjm;
+                bkp = bkm - ajm;
+                akm -= bjm;
+                bkm += ajm;
+              } else {
+                akp = akm - bjm;
+                bkp = bkm + ajm;
+                akm += bjm;
+                bkm -= ajm;
+              }
+              /* avoid useless multiplies */
+              if (s1 == 0.0) {
+                Re [k1] = akp;
+                Re [k2] = ajp;
+                Re [k3] = akm;
+                Im [k1] = bkp;
+                Im [k2] = bjp;
+                Im [k3] = bkm;
+              } else {
+                Re [k1] = akp * c1 - bkp * s1;
+                Re [k2] = ajp * c2 - bjp * s2;
+                Re [k3] = akm * c3 - bkm * s3;
+                Im [k1] = akp * s1 + bkp * c1;
+                Im [k2] = ajp * s2 + bjp * c2;
+                Im [k3] = akm * s3 + bkm * c3;
+              }
+              kk = k3 + kspan;
+            } while (kk < nt);
+
+            c2 = c1 - (cd * c1 + sd * s1);
+            s1 = sd * c1 - cd * s1 + s1;
+            c1 = 2.0 - (c2 * c2 + s1 * s1);
+            s1 *= c1;
+            c1 *= c2;
+            /* values of c2, c3, s2, s3 that will get used next time */
+            c2 = c1 * c1 - s1 * s1;
+            s2 = 2.0 * c1 * s1;
+            c3 = c2 * c1 - s2 * s1;
+            s3 = c2 * s1 + s2 * c1;
+            kk = kk - nt + jc;
+          } while (kk < kspan);
+          kk = kk - kspan + inc;
+        } while (kk < jc);
+        if (kspan == jc)
+          goto Permute_Results_Label;  /* exit infinite loop */
+        break;
+
+      default:
+        /*  transform for odd factors */
+#ifdef FFT_RADIX4
+        return -1;
+        break;
+#else /* FFT_RADIX4 */
+        k = fftstate->factor [ii - 1];
+        ispan = kspan;
+        kspan /= k;
+
+        switch (k) {
+          case 3: /* transform for factor of 3 (optional code) */
+            do {
+              do {
+                k1 = kk + kspan;
+                k2 = k1 + kspan;
+                ak = Re [kk];
+                bk = Im [kk];
+                aj = Re [k1] + Re [k2];
+                bj = Im [k1] + Im [k2];
+                Re [kk] = ak + aj;
+                Im [kk] = bk + bj;
+                ak -= 0.5 * aj;
+                bk -= 0.5 * bj;
+                aj = (Re [k1] - Re [k2]) * s60;
+                bj = (Im [k1] - Im [k2]) * s60;
+                Re [k1] = ak - bj;
+                Re [k2] = ak + bj;
+                Im [k1] = bk + aj;
+                Im [k2] = bk - aj;
+                kk = k2 + kspan;
+              } while (kk < (nn - 1));
+              kk -= nn;
+            } while (kk < kspan);
+            break;
+
+          case 5: /*  transform for factor of 5 (optional code) */
+            c2 = c72 * c72 - s72 * s72;
+            s2 = 2.0 * c72 * s72;
+            do {
+              do {
+                k1 = kk + kspan;
+                k2 = k1 + kspan;
+                k3 = k2 + kspan;
+                k4 = k3 + kspan;
+                akp = Re [k1] + Re [k4];
+                akm = Re [k1] - Re [k4];
+                bkp = Im [k1] + Im [k4];
+                bkm = Im [k1] - Im [k4];
+                ajp = Re [k2] + Re [k3];
+                ajm = Re [k2] - Re [k3];
+                bjp = Im [k2] + Im [k3];
+                bjm = Im [k2] - Im [k3];
+                aa = Re [kk];
+                bb = Im [kk];
+                Re [kk] = aa + akp + ajp;
+                Im [kk] = bb + bkp + bjp;
+                ak = akp * c72 + ajp * c2 + aa;
+                bk = bkp * c72 + bjp * c2 + bb;
+                aj = akm * s72 + ajm * s2;
+                bj = bkm * s72 + bjm * s2;
+                Re [k1] = ak - bj;
+                Re [k4] = ak + bj;
+                Im [k1] = bk + aj;
+                Im [k4] = bk - aj;
+                ak = akp * c2 + ajp * c72 + aa;
+                bk = bkp * c2 + bjp * c72 + bb;
+                aj = akm * s2 - ajm * s72;
+                bj = bkm * s2 - bjm * s72;
+                Re [k2] = ak - bj;
+                Re [k3] = ak + bj;
+                Im [k2] = bk + aj;
+                Im [k3] = bk - aj;
+                kk = k4 + kspan;
+              } while (kk < (nn-1));
+              kk -= nn;
+            } while (kk < kspan);
+            break;
+
+          default:
+            if (k != jf) {
+              jf = k;
+              s1 = pi2 / (double) k;
+              c1 = cos(s1);
+              s1 = sin(s1);
+              if (jf > max_factors){
+                return -1;
+              }
+              Cos [jf - 1] = 1.0;
+              Sin [jf - 1] = 0.0;
+              j = 1;
+              do {
+                Cos [j - 1] = Cos [k - 1] * c1 + Sin [k - 1] * s1;
+                Sin [j - 1] = Cos [k - 1] * s1 - Sin [k - 1] * c1;
+                k--;
+                Cos [k - 1] = Cos [j - 1];
+                Sin [k - 1] = -Sin [j - 1];
+                j++;
+              } while (j < k);
+            }
+            do {
+              do {
+                k1 = kk;
+                k2 = kk + ispan;
+                ak = aa = Re [kk];
+                bk = bb = Im [kk];
+                j = 1;
+                k1 += kspan;
+                do {
+                  k2 -= kspan;
+                  j++;
+                  Rtmp [j - 1] = Re [k1] + Re [k2];
+                  ak += Rtmp [j - 1];
+                  Itmp [j - 1] = Im [k1] + Im [k2];
+                  bk += Itmp [j - 1];
+                  j++;
+                  Rtmp [j - 1] = Re [k1] - Re [k2];
+                  Itmp [j - 1] = Im [k1] - Im [k2];
+                  k1 += kspan;
+                } while (k1 < k2);
+                Re [kk] = ak;
+                Im [kk] = bk;
+                k1 = kk;
+                k2 = kk + ispan;
+                j = 1;
+                do {
+                  k1 += kspan;
+                  k2 -= kspan;
+                  jj = j;
+                  ak = aa;
+                  bk = bb;
+                  aj = 0.0;
+                  bj = 0.0;
+                  k = 1;
+                  do {
+                    k++;
+                    ak += Rtmp [k - 1] * Cos [jj - 1];
+                    bk += Itmp [k - 1] * Cos [jj - 1];
+                    k++;
+                    aj += Rtmp [k - 1] * Sin [jj - 1];
+                    bj += Itmp [k - 1] * Sin [jj - 1];
+                    jj += j;
+                    if (jj > jf) {
+                      jj -= jf;
+                    }
+                  } while (k < jf);
+                  k = jf - j;
+                  Re [k1] = ak - bj;
+                  Im [k1] = bk + aj;
+                  Re [k2] = ak + bj;
+                  Im [k2] = bk - aj;
+                  j++;
+                } while (j < k);
+                kk += ispan;
+              } while (kk < nn);
+              kk -= nn;
+            } while (kk < kspan);
+            break;
+        }
+
+        /*  multiply by rotation factor (except for factors of 2 and 4) */
+        if (ii == mfactor)
+          goto Permute_Results_Label;  /* exit infinite loop */
+        kk = jc;
+        do {
+          c2 = 1.0 - cd;
+          s1 = sd;
+          do {
+            c1 = c2;
+            s2 = s1;
+            kk += kspan;
+            do {
+              do {
+                ak = Re [kk];
+                Re [kk] = c2 * ak - s2 * Im [kk];
+                Im [kk] = s2 * ak + c2 * Im [kk];
+                kk += ispan;
+              } while (kk < nt);
+              ak = s1 * s2;
+              s2 = s1 * c2 + c1 * s2;
+              c2 = c1 * c2 - ak;
+              kk = kk - nt + kspan;
+            } while (kk < ispan);
+            c2 = c1 - (cd * c1 + sd * s1);
+            s1 += sd * c1 - cd * s1;
+            c1 = 2.0 - (c2 * c2 + s1 * s1);
+            s1 *= c1;
+            c2 *= c1;
+            kk = kk - ispan + jc;
+          } while (kk < kspan);
+          kk = kk - kspan + jc + inc;
+        } while (kk < (jc + jc));
+        break;
+#endif /* FFT_RADIX4 */
+    }
+  }
+
+  /*  permute the results to normal order---done in two stages */
+  /*  permutation for square factors of n */
+Permute_Results_Label:
+  fftstate->Perm [0] = ns;
+  if (kt) {
+    k = kt + kt + 1;
+    if (mfactor < k)
+      k--;
+    j = 1;
+    fftstate->Perm [k] = jc;
+    do {
+      fftstate->Perm [j] = fftstate->Perm [j - 1] / fftstate->factor [j - 1];
+      fftstate->Perm [k - 1] = fftstate->Perm [k] * fftstate->factor [j - 1];
+      j++;
+      k--;
+    } while (j < k);
+    k3 = fftstate->Perm [k];
+    kspan = fftstate->Perm [1];
+    kk = jc;
+    k2 = kspan;
+    j = 1;
+    if (nPass != nTotal) {
+      /*  permutation for multivariate transform */
+   Permute_Multi_Label:
+      do {
+        do {
+          k = kk + jc;
+          do {
+            /* swap Re [kk] <> Re [k2], Im [kk] <> Im [k2] */
+            ak = Re [kk]; Re [kk] = Re [k2]; Re [k2] = ak;
+            bk = Im [kk]; Im [kk] = Im [k2]; Im [k2] = bk;
+            kk += inc;
+            k2 += inc;
+          } while (kk < (k-1));
+          kk += ns - jc;
+          k2 += ns - jc;
+        } while (kk < (nt-1));
+        k2 = k2 - nt + kspan;
+        kk = kk - nt + jc;
+      } while (k2 < (ns-1));
+      do {
+        do {
+          k2 -= fftstate->Perm [j - 1];
+          j++;
+          k2 = fftstate->Perm [j] + k2;
+        } while (k2 > fftstate->Perm [j - 1]);
+        j = 1;
+        do {
+          if (kk < (k2-1))
+            goto Permute_Multi_Label;
+          kk += jc;
+          k2 += kspan;
+        } while (k2 < (ns-1));
+      } while (kk < (ns-1));
+    } else {
+      /*  permutation for single-variate transform (optional code) */
+   Permute_Single_Label:
+      do {
+        /* swap Re [kk] <> Re [k2], Im [kk] <> Im [k2] */
+        ak = Re [kk]; Re [kk] = Re [k2]; Re [k2] = ak;
+        bk = Im [kk]; Im [kk] = Im [k2]; Im [k2] = bk;
+        kk += inc;
+        k2 += kspan;
+      } while (k2 < (ns-1));
+      do {
+        do {
+          k2 -= fftstate->Perm [j - 1];
+          j++;
+          k2 = fftstate->Perm [j] + k2;
+        } while (k2 >= fftstate->Perm [j - 1]);
+        j = 1;
+        do {
+          if (kk < k2)
+            goto Permute_Single_Label;
+          kk += inc;
+          k2 += kspan;
+        } while (k2 < (ns-1));
+      } while (kk < (ns-1));
+    }
+    jc = k3;
+  }
+
+  if ((kt << 1) + 1 >= mfactor)
+    return 0;
+  ispan = fftstate->Perm [kt];
+  /* permutation for square-free factors of n */
+  j = mfactor - kt;
+  fftstate->factor [j] = 1;
+  do {
+    fftstate->factor [j - 1] *= fftstate->factor [j];
+    j--;
+  } while (j != kt);
+  kt++;
+  nn = fftstate->factor [kt - 1] - 1;
+  if (nn > (int) max_perm) {
+    return -1;
+  }
+  j = jj = 0;
+  for (;;) {
+    k = kt + 1;
+    k2 = fftstate->factor [kt - 1];
+    kk = fftstate->factor [k - 1];
+    j++;
+    if (j > nn)
+      break;    /* exit infinite loop */
+    jj += kk;
+    while (jj >= k2) {
+      jj -= k2;
+      k2 = kk;
+      k++;
+      kk = fftstate->factor [k - 1];
+      jj += kk;
+    }
+    fftstate->Perm [j - 1] = jj;
+  }
+  /*  determine the permutation cycles of length greater than 1 */
+  j = 0;
+  for (;;) {
+    do {
+      j++;
+      kk = fftstate->Perm [j - 1];
+    } while (kk < 0);
+    if (kk != j) {
+      do {
+        k = kk;
+        kk = fftstate->Perm [k - 1];
+        fftstate->Perm [k - 1] = -kk;
+      } while (kk != j);
+      k3 = kk;
+    } else {
+      fftstate->Perm [j - 1] = -j;
+      if (j == nn)
+        break;  /* exit infinite loop */
+    }
+  }
+  max_factors *= inc;
+  /*  reorder a and b, following the permutation cycles */
+  for (;;) {
+    j = k3 + 1;
+    nt -= ispan;
+    ii = nt - inc + 1;
+    if (nt < 0)
+      break;   /* exit infinite loop */
+    do {
+      do {
+        j--;
+      } while (fftstate->Perm [j - 1] < 0);
+      jj = jc;
+      do {
+        kspan = jj;
+        if (jj > max_factors) {
+          kspan = max_factors;
+        }
+        jj -= kspan;
+        k = fftstate->Perm [j - 1];
+        kk = jc * k + ii + jj;
+        k1 = kk + kspan - 1;
+        k2 = 0;
+        do {
+          k2++;
+          Rtmp [k2 - 1] = Re [k1];
+          Itmp [k2 - 1] = Im [k1];
+          k1 -= inc;
+        } while (k1 != (kk-1));
+        do {
+          k1 = kk + kspan - 1;
+          k2 = k1 - jc * (k + fftstate->Perm [k - 1]);
+          k = -fftstate->Perm [k - 1];
+          do {
+            Re [k1] = Re [k2];
+            Im [k1] = Im [k2];
+            k1 -= inc;
+            k2 -= inc;
+          } while (k1 != (kk-1));
+          kk = k2 + 1;
+        } while (k != j);
+        k1 = kk + kspan - 1;
+        k2 = 0;
+        do {
+          k2++;
+          Re [k1] = Rtmp [k2 - 1];
+          Im [k1] = Itmp [k2 - 1];
+          k1 -= inc;
+        } while (k1 != (kk-1));
+      } while (jj);
+    } while (j != 1);
+  }
+  return 0;   /* exit point here */
+}
+/* ---------------------- end-of-file (c source) ---------------------- */
+

VocieProcess/modules/third_party/fft/fft.h

@@ -0,0 +1,58 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the ../../../LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*--------------------------------*-C-*---------------------------------*
+ * File:
+ *       fftn.h
+ * ---------------------------------------------------------------------*
+ * Re[]:        real value array
+ * Im[]:        imaginary value array
+ * nTotal:      total number of complex values
+ * nPass:       number of elements involved in this pass of transform
+ * nSpan:       nspan/nPass = number of bytes to increment pointer
+ *              in Re[] and Im[]
+ * isign: exponent: +1 = forward  -1 = reverse
+ * scaling: normalizing constant by which the final result is *divided*
+ * scaling == -1, normalize by total dimension of the transform
+ * scaling <  -1, normalize by the square-root of the total dimension
+ *
+ * ----------------------------------------------------------------------
+ * See the comments in the code for correct usage!
+ */
+
+#ifndef MODULES_THIRD_PARTY_FFT_FFT_H_
+#define MODULES_THIRD_PARTY_FFT_FFT_H_
+
+#define FFT_MAXFFTSIZE 2048
+#define FFT_NFACTOR 11
+
+typedef struct {
+  unsigned int SpaceAlloced;
+  unsigned int MaxPermAlloced;
+  double Tmp0[FFT_MAXFFTSIZE];
+  double Tmp1[FFT_MAXFFTSIZE];
+  double Tmp2[FFT_MAXFFTSIZE];
+  double Tmp3[FFT_MAXFFTSIZE];
+  int Perm[FFT_MAXFFTSIZE];
+  int factor[FFT_NFACTOR];
+
+} FFTstr;
+
+/* double precision routine */
+
+int WebRtcIsac_Fftns(unsigned int ndim,
+                     const int dims[],
+                     double Re[],
+                     double Im[],
+                     int isign,
+                     double scaling,
+                     FFTstr* fftstate);
+
+#endif /* MODULES_THIRD_PARTY_FFT_FFT_H_ */

VocieProcess/rtc_base/event.cc

@@ -0,0 +1,210 @@
+/*
+ *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "rtc_base/event.h"
+
+#if defined(WEBRTC_WIN)
+#include <windows.h>
+#elif defined(WEBRTC_POSIX)
+#include <errno.h>
+#include <pthread.h>
+#include <sys/time.h>
+#include <time.h>
+#else
+#error "Must define either WEBRTC_WIN or WEBRTC_POSIX."
+#endif
+
+#include "absl/types/optional.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/synchronization/yield_policy.h"
+#include "rtc_base/system/warn_current_thread_is_deadlocked.h"
+#include "rtc_base/time_utils.h"
+
+namespace rtc {
+
+using ::webrtc::TimeDelta;
+
+Event::Event() : Event(false, false) {}
+
+#if defined(WEBRTC_WIN)
+
+Event::Event(bool manual_reset, bool initially_signaled) {
+  event_handle_ = ::CreateEvent(nullptr,  // Security attributes.
+                                manual_reset, initially_signaled,
+                                nullptr);  // Name.
+  RTC_CHECK(event_handle_);
+}
+
+Event::~Event() {
+  CloseHandle(event_handle_);
+}
+
+void Event::Set() {
+  SetEvent(event_handle_);
+}
+
+void Event::Reset() {
+  ResetEvent(event_handle_);
+}
+
+bool Event::Wait(TimeDelta give_up_after, TimeDelta /*warn_after*/) {
+  ScopedYieldPolicy::YieldExecution();
+  const DWORD ms =
+      give_up_after.IsPlusInfinity()
+          ? INFINITE
+          : give_up_after.RoundUpTo(webrtc::TimeDelta::Millis(1)).ms();
+  return (WaitForSingleObject(event_handle_, ms) == WAIT_OBJECT_0);
+}
+
+#elif defined(WEBRTC_POSIX)
+
+// On MacOS, clock_gettime is available from version 10.12, and on
+// iOS, from version 10.0. So we can't use it yet.
+#if defined(WEBRTC_MAC) || defined(WEBRTC_IOS)
+#define USE_CLOCK_GETTIME 0
+#define USE_PTHREAD_COND_TIMEDWAIT_MONOTONIC_NP 0
+// On Android, pthread_condattr_setclock is available from version 21. By
+// default, we target a new enough version for 64-bit platforms but not for
+// 32-bit platforms. For older versions, use
+// pthread_cond_timedwait_monotonic_np.
+#elif defined(WEBRTC_ANDROID) && (__ANDROID_API__ < 21)
+#define USE_CLOCK_GETTIME 1
+#define USE_PTHREAD_COND_TIMEDWAIT_MONOTONIC_NP 1
+#else
+#define USE_CLOCK_GETTIME 1
+#define USE_PTHREAD_COND_TIMEDWAIT_MONOTONIC_NP 0
+#endif
+
+Event::Event(bool manual_reset, bool initially_signaled)
+    : is_manual_reset_(manual_reset), event_status_(initially_signaled) {
+  RTC_CHECK(pthread_mutex_init(&event_mutex_, nullptr) == 0);
+  pthread_condattr_t cond_attr;
+  RTC_CHECK(pthread_condattr_init(&cond_attr) == 0);
+#if USE_CLOCK_GETTIME && !USE_PTHREAD_COND_TIMEDWAIT_MONOTONIC_NP
+  RTC_CHECK(pthread_condattr_setclock(&cond_attr, CLOCK_MONOTONIC) == 0);
+#endif
+  RTC_CHECK(pthread_cond_init(&event_cond_, &cond_attr) == 0);
+  pthread_condattr_destroy(&cond_attr);
+}
+
+Event::~Event() {
+  pthread_mutex_destroy(&event_mutex_);
+  pthread_cond_destroy(&event_cond_);
+}
+
+void Event::Set() {
+  pthread_mutex_lock(&event_mutex_);
+  event_status_ = true;
+  pthread_cond_broadcast(&event_cond_);
+  pthread_mutex_unlock(&event_mutex_);
+}
+
+void Event::Reset() {
+  pthread_mutex_lock(&event_mutex_);
+  event_status_ = false;
+  pthread_mutex_unlock(&event_mutex_);
+}
+
+namespace {
+
+timespec GetTimespec(TimeDelta duration_from_now) {
+  timespec ts;
+
+  // Get the current time.
+#if USE_CLOCK_GETTIME
+  clock_gettime(CLOCK_MONOTONIC, &ts);
+#else
+  timeval tv;
+  gettimeofday(&tv, nullptr);
+  ts.tv_sec = tv.tv_sec;
+  ts.tv_nsec = tv.tv_usec * kNumNanosecsPerMicrosec;
+#endif
+
+  // Add the specified number of milliseconds to it.
+  int64_t microsecs_from_now = duration_from_now.us();
+  ts.tv_sec += microsecs_from_now / kNumMicrosecsPerSec;
+  ts.tv_nsec +=
+      (microsecs_from_now % kNumMicrosecsPerSec) * kNumNanosecsPerMicrosec;
+
+  // Normalize.
+  if (ts.tv_nsec >= kNumNanosecsPerSec) {
+    ts.tv_sec++;
+    ts.tv_nsec -= kNumNanosecsPerSec;
+  }
+
+  return ts;
+}
+
+}  // namespace
+
+bool Event::Wait(TimeDelta give_up_after, TimeDelta warn_after) {
+  // Instant when we'll log a warning message (because we've been waiting so
+  // long it might be a bug), but not yet give up waiting. nullopt if we
+  // shouldn't log a warning.
+  const absl::optional<timespec> warn_ts =
+      warn_after >= give_up_after
+          ? absl::nullopt
+          : absl::make_optional(GetTimespec(warn_after));
+
+  // Instant when we'll stop waiting and return an error. nullopt if we should
+  // never give up.
+  const absl::optional<timespec> give_up_ts =
+      give_up_after.IsPlusInfinity()
+          ? absl::nullopt
+          : absl::make_optional(GetTimespec(give_up_after));
+
+  ScopedYieldPolicy::YieldExecution();
+  pthread_mutex_lock(&event_mutex_);
+
+  // Wait for `event_cond_` to trigger and `event_status_` to be set, with the
+  // given timeout (or without a timeout if none is given).
+  const auto wait = [&](const absl::optional<timespec> timeout_ts) {
+    int error = 0;
+    while (!event_status_ && error == 0) {
+      if (timeout_ts == absl::nullopt) {
+        error = pthread_cond_wait(&event_cond_, &event_mutex_);
+      } else {
+#if USE_PTHREAD_COND_TIMEDWAIT_MONOTONIC_NP
+        error = pthread_cond_timedwait_monotonic_np(&event_cond_, &event_mutex_,
+                                                    &*timeout_ts);
+#else
+        error =
+            pthread_cond_timedwait(&event_cond_, &event_mutex_, &*timeout_ts);
+#endif
+      }
+    }
+    return error;
+  };
+
+  int error;
+  if (warn_ts == absl::nullopt) {
+    error = wait(give_up_ts);
+  } else {
+    error = wait(warn_ts);
+    if (error == ETIMEDOUT) {
+      webrtc::WarnThatTheCurrentThreadIsProbablyDeadlocked();
+      error = wait(give_up_ts);
+    }
+  }
+
+  // NOTE(liulk): Exactly one thread will auto-reset this event. All
+  // the other threads will think it's unsignaled.  This seems to be
+  // consistent with auto-reset events in WEBRTC_WIN
+  if (error == 0 && !is_manual_reset_)
+    event_status_ = false;
+
+  pthread_mutex_unlock(&event_mutex_);
+
+  return (error == 0);
+}
+
+#endif
+
+}  // namespace rtc

VocieProcess/rtc_base/event.h

@@ -0,0 +1,137 @@
+/*
+ *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef RTC_BASE_EVENT_H_
+#define RTC_BASE_EVENT_H_
+
+#include "api/units/time_delta.h"
+
+#if defined(WEBRTC_WIN)
+#include <windows.h>
+#elif defined(WEBRTC_POSIX)
+#include <pthread.h>
+#else
+#error "Must define either WEBRTC_WIN or WEBRTC_POSIX."
+#endif
+
+#include "rtc_base/synchronization/yield_policy.h"
+
+namespace rtc {
+
+// RTC_DISALLOW_WAIT() utility
+//
+// Sets a stack-scoped flag that disallows use of `rtc::Event::Wait` by means
+// of raising a DCHECK when a call to `rtc::Event::Wait()` is made..
+// This is useful to guard synchronization-free scopes against regressions.
+//
+// Example of what this would catch (`ScopeToProtect` calls `Foo`):
+//
+//  void Foo(TaskQueue* tq) {
+//    Event event;
+//    tq->PostTask([&event]() {
+//      event.Set();
+//    });
+//    event.Wait(Event::kForever);  // <- Will trigger a DCHECK.
+//  }
+//
+//  void ScopeToProtect() {
+//    TaskQueue* tq = GetSomeTaskQueue();
+//    RTC_DISALLOW_WAIT();  // Policy takes effect.
+//    Foo(tq);
+//  }
+//
+#if RTC_DCHECK_IS_ON
+#define RTC_DISALLOW_WAIT() ScopedDisallowWait disallow_wait_##__LINE__
+#else
+#define RTC_DISALLOW_WAIT()
+#endif
+
+class Event {
+ public:
+  // TODO(bugs.webrtc.org/14366): Consider removing this redundant alias.
+  static constexpr webrtc::TimeDelta kForever =
+      webrtc::TimeDelta::PlusInfinity();
+
+  Event();
+  Event(bool manual_reset, bool initially_signaled);
+  Event(const Event&) = delete;
+  Event& operator=(const Event&) = delete;
+  ~Event();
+
+  void Set();
+  void Reset();
+
+  // Waits for the event to become signaled, but logs a warning if it takes more
+  // than `warn_after`, and gives up completely if it takes more than
+  // `give_up_after`. (If `warn_after >= give_up_after`, no warning will be
+  // logged.) Either or both may be `kForever`, which means wait indefinitely.
+  //
+  // Care is taken so that the underlying OS wait call isn't requested to sleep
+  // shorter than `give_up_after`.
+  //
+  // Returns true if the event was signaled, false if there was a timeout or
+  // some other error.
+  bool Wait(webrtc::TimeDelta give_up_after, webrtc::TimeDelta warn_after);
+
+  // Waits with the given timeout and a reasonable default warning timeout.
+  bool Wait(webrtc::TimeDelta give_up_after) {
+    return Wait(give_up_after, give_up_after.IsPlusInfinity()
+                                   ? webrtc::TimeDelta::Seconds(3)
+                                   : kForever);
+  }
+
+ private:
+#if defined(WEBRTC_WIN)
+  HANDLE event_handle_;
+#elif defined(WEBRTC_POSIX)
+  pthread_mutex_t event_mutex_;
+  pthread_cond_t event_cond_;
+  const bool is_manual_reset_;
+  bool event_status_;
+#endif
+};
+
+// These classes are provided for compatibility with Chromium.
+// The rtc::Event implementation is overriden inside of Chromium for the
+// purposes of detecting when threads are blocked that shouldn't be as well as
+// to use the more accurate event implementation that's there than is provided
+// by default on some platforms (e.g. Windows).
+// When building with standalone WebRTC, this class is a noop.
+// For further information, please see the
+// ScopedAllowBaseSyncPrimitives(ForTesting) classes in Chromium.
+class ScopedAllowBaseSyncPrimitives {
+ public:
+  ScopedAllowBaseSyncPrimitives() {}
+  ~ScopedAllowBaseSyncPrimitives() {}
+};
+
+class ScopedAllowBaseSyncPrimitivesForTesting {
+ public:
+  ScopedAllowBaseSyncPrimitivesForTesting() {}
+  ~ScopedAllowBaseSyncPrimitivesForTesting() {}
+};
+
+#if RTC_DCHECK_IS_ON
+class ScopedDisallowWait {
+ public:
+  ScopedDisallowWait() = default;
+
+ private:
+  class DisallowYieldHandler : public YieldInterface {
+   public:
+    void YieldExecution() override { RTC_DCHECK_NOTREACHED(); }
+  } handler_;
+  rtc::ScopedYieldPolicy policy{&handler_};
+};
+#endif
+
+}  // namespace rtc
+
+#endif  // RTC_BASE_EVENT_H_

VocieProcess/rtc_base/event_tracer.cc

@@ -0,0 +1,449 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "rtc_base/event_tracer.h"
+
+#include <stdio.h>
+
+#include "rtc_base/trace_event.h"
+
+#if defined(RTC_USE_PERFETTO)
+#include "rtc_base/trace_categories.h"
+#include "third_party/perfetto/include/perfetto/tracing/tracing.h"
+#else
+#include <inttypes.h>
+#include <stdint.h>
+#include <string.h>
+
+#include <atomic>
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/sequence_checker.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/event.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/platform_thread.h"
+#include "rtc_base/platform_thread_types.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+#include "rtc_base/time_utils.h"
+#endif
+
+namespace webrtc {
+
+namespace {
+
+#if !defined(RTC_USE_PERFETTO)
+GetCategoryEnabledPtr g_get_category_enabled_ptr = nullptr;
+AddTraceEventPtr g_add_trace_event_ptr = nullptr;
+#endif
+
+}  // namespace
+
+#if defined(RTC_USE_PERFETTO)
+void RegisterPerfettoTrackEvents() {
+  if (perfetto::Tracing::IsInitialized()) {
+    webrtc::TrackEvent::Register();
+  }
+}
+#else
+
+void SetupEventTracer(GetCategoryEnabledPtr get_category_enabled_ptr,
+                      AddTraceEventPtr add_trace_event_ptr) {
+  g_get_category_enabled_ptr = get_category_enabled_ptr;
+  g_add_trace_event_ptr = add_trace_event_ptr;
+}
+
+const unsigned char* EventTracer::GetCategoryEnabled(const char* name) {
+  if (g_get_category_enabled_ptr)
+    return g_get_category_enabled_ptr(name);
+
+  // A string with null terminator means category is disabled.
+  return reinterpret_cast<const unsigned char*>("\0");
+}
+
+// Arguments to this function (phase, etc.) are as defined in
+// webrtc/rtc_base/trace_event.h.
+void EventTracer::AddTraceEvent(char phase,
+                                const unsigned char* category_enabled,
+                                const char* name,
+                                unsigned long long id,
+                                int num_args,
+                                const char** arg_names,
+                                const unsigned char* arg_types,
+                                const unsigned long long* arg_values,
+                                unsigned char flags) {
+  if (g_add_trace_event_ptr) {
+    g_add_trace_event_ptr(phase, category_enabled, name, id, num_args,
+                          arg_names, arg_types, arg_values, flags);
+  }
+}
+#endif
+
+}  // namespace webrtc
+
+#if defined(RTC_USE_PERFETTO)
+// TODO(bugs.webrtc.org/15917): Implement for perfetto.
+namespace rtc::tracing {
+void SetupInternalTracer(bool enable_all_categories) {}
+bool StartInternalCapture(absl::string_view filename) {
+  return false;
+}
+void StartInternalCaptureToFile(FILE* file) {}
+void StopInternalCapture() {}
+void ShutdownInternalTracer() {}
+
+}  // namespace rtc::tracing
+#else
+
+// This is a guesstimate that should be enough in most cases.
+static const size_t kEventLoggerArgsStrBufferInitialSize = 256;
+static const size_t kTraceArgBufferLength = 32;
+
+namespace rtc {
+namespace tracing {
+namespace {
+
+// Atomic-int fast path for avoiding logging when disabled.
+static std::atomic<int> g_event_logging_active(0);
+
+// TODO(pbos): Log metadata for all threads, etc.
+class EventLogger final {
+ public:
+  ~EventLogger() { RTC_DCHECK(thread_checker_.IsCurrent()); }
+
+  void AddTraceEvent(const char* name,
+                     const unsigned char* category_enabled,
+                     char phase,
+                     int num_args,
+                     const char** arg_names,
+                     const unsigned char* arg_types,
+                     const unsigned long long* arg_values,
+                     uint64_t timestamp,
+                     int pid,
+                     rtc::PlatformThreadId thread_id) {
+    std::vector<TraceArg> args(num_args);
+    for (int i = 0; i < num_args; ++i) {
+      TraceArg& arg = args[i];
+      arg.name = arg_names[i];
+      arg.type = arg_types[i];
+      arg.value.as_uint = arg_values[i];
+
+      // Value is a pointer to a temporary string, so we have to make a copy.
+      if (arg.type == TRACE_VALUE_TYPE_COPY_STRING) {
+        // Space for the string and for the terminating null character.
+        size_t str_length = strlen(arg.value.as_string) + 1;
+        char* str_copy = new char[str_length];
+        memcpy(str_copy, arg.value.as_string, str_length);
+        arg.value.as_string = str_copy;
+      }
+    }
+    webrtc::MutexLock lock(&mutex_);
+    trace_events_.push_back(
+        {name, category_enabled, phase, args, timestamp, 1, thread_id});
+  }
+
+  // The TraceEvent format is documented here:
+  // https://docs.google.com/document/d/1CvAClvFfyA5R-PhYUmn5OOQtYMH4h6I0nSsKchNAySU/preview
+  void Log() {
+    RTC_DCHECK(output_file_);
+    static constexpr webrtc::TimeDelta kLoggingInterval =
+        webrtc::TimeDelta::Millis(100);
+    fprintf(output_file_, "{ \"traceEvents\": [\n");
+    bool has_logged_event = false;
+    while (true) {
+      bool shutting_down = shutdown_event_.Wait(kLoggingInterval);
+      std::vector<TraceEvent> events;
+      {
+        webrtc::MutexLock lock(&mutex_);
+        trace_events_.swap(events);
+      }
+      std::string args_str;
+      args_str.reserve(kEventLoggerArgsStrBufferInitialSize);
+      for (TraceEvent& e : events) {
+        args_str.clear();
+        if (!e.args.empty()) {
+          args_str += ", \"args\": {";
+          bool is_first_argument = true;
+          for (TraceArg& arg : e.args) {
+            if (!is_first_argument)
+              args_str += ",";
+            is_first_argument = false;
+            args_str += " \"";
+            args_str += arg.name;
+            args_str += "\": ";
+            args_str += TraceArgValueAsString(arg);
+
+            // Delete our copy of the string.
+            if (arg.type == TRACE_VALUE_TYPE_COPY_STRING) {
+              delete[] arg.value.as_string;
+              arg.value.as_string = nullptr;
+            }
+          }
+          args_str += " }";
+        }
+        fprintf(output_file_,
+                "%s{ \"name\": \"%s\""
+                ", \"cat\": \"%s\""
+                ", \"ph\": \"%c\""
+                ", \"ts\": %" PRIu64
+                ", \"pid\": %d"
+#if defined(WEBRTC_WIN)
+                ", \"tid\": %lu"
+#else
+                ", \"tid\": %d"
+#endif  // defined(WEBRTC_WIN)
+                "%s"
+                "}\n",
+                has_logged_event ? "," : " ", e.name, e.category_enabled,
+                e.phase, e.timestamp, e.pid, e.tid, args_str.c_str());
+        has_logged_event = true;
+      }
+      if (shutting_down)
+        break;
+    }
+    fprintf(output_file_, "]}\n");
+    if (output_file_owned_)
+      fclose(output_file_);
+    output_file_ = nullptr;
+  }
+
+  void Start(FILE* file, bool owned) {
+    RTC_DCHECK(thread_checker_.IsCurrent());
+    RTC_DCHECK(file);
+    RTC_DCHECK(!output_file_);
+    output_file_ = file;
+    output_file_owned_ = owned;
+    {
+      webrtc::MutexLock lock(&mutex_);
+      // Since the atomic fast-path for adding events to the queue can be
+      // bypassed while the logging thread is shutting down there may be some
+      // stale events in the queue, hence the vector needs to be cleared to not
+      // log events from a previous logging session (which may be days old).
+      trace_events_.clear();
+    }
+    // Enable event logging (fast-path). This should be disabled since starting
+    // shouldn't be done twice.
+    int zero = 0;
+    RTC_CHECK(g_event_logging_active.compare_exchange_strong(zero, 1));
+
+    // Finally start, everything should be set up now.
+    logging_thread_ =
+        PlatformThread::SpawnJoinable([this] { Log(); }, "EventTracingThread");
+    TRACE_EVENT_INSTANT0("webrtc", "EventLogger::Start",
+                         TRACE_EVENT_SCOPE_GLOBAL);
+  }
+
+  void Stop() {
+    RTC_DCHECK(thread_checker_.IsCurrent());
+    TRACE_EVENT_INSTANT0("webrtc", "EventLogger::Stop",
+                         TRACE_EVENT_SCOPE_GLOBAL);
+    // Try to stop. Abort if we're not currently logging.
+    int one = 1;
+    if (g_event_logging_active.compare_exchange_strong(one, 0))
+      return;
+
+    // Wake up logging thread to finish writing.
+    shutdown_event_.Set();
+    // Join the logging thread.
+    logging_thread_.Finalize();
+  }
+
+ private:
+  struct TraceArg {
+    const char* name;
+    unsigned char type;
+    // Copied from webrtc/rtc_base/trace_event.h TraceValueUnion.
+    union TraceArgValue {
+      bool as_bool;
+      unsigned long long as_uint;
+      long long as_int;
+      double as_double;
+      const void* as_pointer;
+      const char* as_string;
+    } value;
+
+    // Assert that the size of the union is equal to the size of the as_uint
+    // field since we are assigning to arbitrary types using it.
+    static_assert(sizeof(TraceArgValue) == sizeof(unsigned long long),
+                  "Size of TraceArg value union is not equal to the size of "
+                  "the uint field of that union.");
+  };
+
+  struct TraceEvent {
+    const char* name;
+    const unsigned char* category_enabled;
+    char phase;
+    std::vector<TraceArg> args;
+    uint64_t timestamp;
+    int pid;
+    rtc::PlatformThreadId tid;
+  };
+
+  static std::string TraceArgValueAsString(TraceArg arg) {
+    std::string output;
+
+    if (arg.type == TRACE_VALUE_TYPE_STRING ||
+        arg.type == TRACE_VALUE_TYPE_COPY_STRING) {
+      // Space for every character to be an espaced character + two for
+      // quatation marks.
+      output.reserve(strlen(arg.value.as_string) * 2 + 2);
+      output += '\"';
+      const char* c = arg.value.as_string;
+      do {
+        if (*c == '"' || *c == '\\') {
+          output += '\\';
+          output += *c;
+        } else {
+          output += *c;
+        }
+      } while (*++c);
+      output += '\"';
+    } else {
+      output.resize(kTraceArgBufferLength);
+      size_t print_length = 0;
+      switch (arg.type) {
+        case TRACE_VALUE_TYPE_BOOL:
+          if (arg.value.as_bool) {
+            strcpy(&output[0], "true");
+            print_length = 4;
+          } else {
+            strcpy(&output[0], "false");
+            print_length = 5;
+          }
+          break;
+        case TRACE_VALUE_TYPE_UINT:
+          print_length = snprintf(&output[0], kTraceArgBufferLength, "%llu",
+                                  arg.value.as_uint);
+          break;
+        case TRACE_VALUE_TYPE_INT:
+          print_length = snprintf(&output[0], kTraceArgBufferLength, "%lld",
+                                  arg.value.as_int);
+          break;
+        case TRACE_VALUE_TYPE_DOUBLE:
+          print_length = snprintf(&output[0], kTraceArgBufferLength, "%f",
+                                  arg.value.as_double);
+          break;
+        case TRACE_VALUE_TYPE_POINTER:
+          print_length = snprintf(&output[0], kTraceArgBufferLength, "\"%p\"",
+                                  arg.value.as_pointer);
+          break;
+      }
+      size_t output_length = print_length < kTraceArgBufferLength
+                                 ? print_length
+                                 : kTraceArgBufferLength - 1;
+      // This will hopefully be very close to nop. On most implementations, it
+      // just writes null byte and sets the length field of the string.
+      output.resize(output_length);
+    }
+
+    return output;
+  }
+
+  webrtc::Mutex mutex_;
+  std::vector<TraceEvent> trace_events_ RTC_GUARDED_BY(mutex_);
+  rtc::PlatformThread logging_thread_;
+  rtc::Event shutdown_event_;
+  webrtc::SequenceChecker thread_checker_;
+  FILE* output_file_ = nullptr;
+  bool output_file_owned_ = false;
+};
+
+static std::atomic<EventLogger*> g_event_logger(nullptr);
+static const char* const kDisabledTracePrefix = TRACE_DISABLED_BY_DEFAULT("");
+const unsigned char* InternalGetCategoryEnabled(const char* name) {
+  const char* prefix_ptr = &kDisabledTracePrefix[0];
+  const char* name_ptr = name;
+  // Check whether name contains the default-disabled prefix.
+  while (*prefix_ptr == *name_ptr && *prefix_ptr != '\0') {
+    ++prefix_ptr;
+    ++name_ptr;
+  }
+  return reinterpret_cast<const unsigned char*>(*prefix_ptr == '\0' ? ""
+                                                                    : name);
+}
+
+const unsigned char* InternalEnableAllCategories(const char* name) {
+  return reinterpret_cast<const unsigned char*>(name);
+}
+
+void InternalAddTraceEvent(char phase,
+                           const unsigned char* category_enabled,
+                           const char* name,
+                           unsigned long long id,
+                           int num_args,
+                           const char** arg_names,
+                           const unsigned char* arg_types,
+                           const unsigned long long* arg_values,
+                           unsigned char flags) {
+  // Fast path for when event tracing is inactive.
+  if (g_event_logging_active.load() == 0)
+    return;
+
+  g_event_logger.load()->AddTraceEvent(
+      name, category_enabled, phase, num_args, arg_names, arg_types, arg_values,
+      rtc::TimeMicros(), 1, rtc::CurrentThreadId());
+}
+
+}  // namespace
+
+void SetupInternalTracer(bool enable_all_categories) {
+  EventLogger* null_logger = nullptr;
+  RTC_CHECK(
+      g_event_logger.compare_exchange_strong(null_logger, new EventLogger()));
+  webrtc::SetupEventTracer(enable_all_categories ? InternalEnableAllCategories
+                                                 : InternalGetCategoryEnabled,
+                           InternalAddTraceEvent);
+}
+
+void StartInternalCaptureToFile(FILE* file) {
+  EventLogger* event_logger = g_event_logger.load();
+  if (event_logger) {
+    event_logger->Start(file, false);
+  }
+}
+
+bool StartInternalCapture(absl::string_view filename) {
+  EventLogger* event_logger = g_event_logger.load();
+  if (!event_logger)
+    return false;
+
+  FILE* file = fopen(std::string(filename).c_str(), "w");
+  if (!file) {
+    RTC_LOG(LS_ERROR) << "Failed to open trace file '" << filename
+                      << "' for writing.";
+    return false;
+  }
+  event_logger->Start(file, true);
+  return true;
+}
+
+void StopInternalCapture() {
+  EventLogger* event_logger = g_event_logger.load();
+  if (event_logger) {
+    event_logger->Stop();
+  }
+}
+
+void ShutdownInternalTracer() {
+  StopInternalCapture();
+  EventLogger* old_logger = g_event_logger.load(std::memory_order_acquire);
+  RTC_DCHECK(old_logger);
+  RTC_CHECK(g_event_logger.compare_exchange_strong(old_logger, nullptr));
+  delete old_logger;
+  webrtc::SetupEventTracer(nullptr, nullptr);
+}
+
+}  // namespace tracing
+}  // namespace rtc
+
+#endif  // defined(RTC_USE_PERFETTO)

VocieProcess/rtc_base/event_tracer.h

@@ -0,0 +1,88 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef RTC_BASE_EVENT_TRACER_H_
+#define RTC_BASE_EVENT_TRACER_H_
+
+// This file defines the interface for event tracing in WebRTC.
+//
+// Event log handlers are set through SetupEventTracer(). User of this API will
+// provide two function pointers to handle event tracing calls.
+//
+// * GetCategoryEnabledPtr
+//   Event tracing system calls this function to determine if a particular
+//   event category is enabled.
+//
+// * AddTraceEventPtr
+//   Adds a tracing event. It is the user's responsibility to log the data
+//   provided.
+//
+// Parameters for the above two functions are described in trace_event.h.
+
+#include <stdio.h>
+
+#include "absl/strings/string_view.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+#if defined(RTC_USE_PERFETTO)
+void RegisterPerfettoTrackEvents();
+#else
+typedef const unsigned char* (*GetCategoryEnabledPtr)(const char* name);
+typedef void (*AddTraceEventPtr)(char phase,
+                                 const unsigned char* category_enabled,
+                                 const char* name,
+                                 unsigned long long id,
+                                 int num_args,
+                                 const char** arg_names,
+                                 const unsigned char* arg_types,
+                                 const unsigned long long* arg_values,
+                                 unsigned char flags);
+
+// User of WebRTC can call this method to setup event tracing.
+//
+// This method must be called before any WebRTC methods. Functions
+// provided should be thread-safe.
+void SetupEventTracer(GetCategoryEnabledPtr get_category_enabled_ptr,
+                      AddTraceEventPtr add_trace_event_ptr);
+
+// This class defines interface for the event tracing system to call
+// internally. Do not call these methods directly.
+class EventTracer {
+ public:
+  static const unsigned char* GetCategoryEnabled(const char* name);
+
+  static void AddTraceEvent(char phase,
+                            const unsigned char* category_enabled,
+                            const char* name,
+                            unsigned long long id,
+                            int num_args,
+                            const char** arg_names,
+                            const unsigned char* arg_types,
+                            const unsigned long long* arg_values,
+                            unsigned char flags);
+};
+#endif
+
+}  // namespace webrtc
+
+namespace rtc::tracing {
+// Set up internal event tracer.
+// TODO(webrtc:15917): Implement for perfetto.
+RTC_EXPORT void SetupInternalTracer(bool enable_all_categories = true);
+RTC_EXPORT bool StartInternalCapture(absl::string_view filename);
+RTC_EXPORT void StartInternalCaptureToFile(FILE* file);
+RTC_EXPORT void StopInternalCapture();
+// Make sure we run this, this will tear down the internal tracing.
+RTC_EXPORT void ShutdownInternalTracer();
+}  // namespace rtc::tracing
+
+#endif  // RTC_BASE_EVENT_TRACER_H_

VocieProcess/rtc_base/platform_thread.cc

@@ -0,0 +1,213 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "rtc_base/platform_thread.h"
+
+#include <algorithm>
+#include <memory>
+
+#if !defined(WEBRTC_WIN)
+#include <sched.h>
+#endif
+
+#include "rtc_base/checks.h"
+
+namespace rtc {
+namespace {
+
+#if defined(WEBRTC_WIN)
+int Win32PriorityFromThreadPriority(ThreadPriority priority) {
+  switch (priority) {
+    case ThreadPriority::kLow:
+      return THREAD_PRIORITY_BELOW_NORMAL;
+    case ThreadPriority::kNormal:
+      return THREAD_PRIORITY_NORMAL;
+    case ThreadPriority::kHigh:
+      return THREAD_PRIORITY_ABOVE_NORMAL;
+    case ThreadPriority::kRealtime:
+      return THREAD_PRIORITY_TIME_CRITICAL;
+  }
+}
+#endif
+
+bool SetPriority(ThreadPriority priority) {
+#if defined(WEBRTC_WIN)
+  return SetThreadPriority(GetCurrentThread(),
+                           Win32PriorityFromThreadPriority(priority)) != FALSE;
+#elif defined(__native_client__) || defined(WEBRTC_FUCHSIA) || \
+    (defined(__EMSCRIPTEN__) && !defined(__EMSCRIPTEN_PTHREADS__))
+  // Setting thread priorities is not supported in NaCl, Fuchsia or Emscripten
+  // without pthreads.
+  return true;
+#elif defined(WEBRTC_CHROMIUM_BUILD) && defined(WEBRTC_LINUX)
+  // TODO(tommi): Switch to the same mechanism as Chromium uses for changing
+  // thread priorities.
+  return true;
+#else
+  const int policy = SCHED_FIFO;
+  const int min_prio = sched_get_priority_min(policy);
+  const int max_prio = sched_get_priority_max(policy);
+  if (min_prio == -1 || max_prio == -1) {
+    return false;
+  }
+
+  if (max_prio - min_prio <= 2)
+    return false;
+
+  // Convert webrtc priority to system priorities:
+  sched_param param;
+  const int top_prio = max_prio - 1;
+  const int low_prio = min_prio + 1;
+  switch (priority) {
+    case ThreadPriority::kLow:
+      param.sched_priority = low_prio;
+      break;
+    case ThreadPriority::kNormal:
+      // The -1 ensures that the kHighPriority is always greater or equal to
+      // kNormalPriority.
+      param.sched_priority = (low_prio + top_prio - 1) / 2;
+      break;
+    case ThreadPriority::kHigh:
+      param.sched_priority = std::max(top_prio - 2, low_prio);
+      break;
+    case ThreadPriority::kRealtime:
+      param.sched_priority = top_prio;
+      break;
+  }
+  return pthread_setschedparam(pthread_self(), policy, &param) == 0;
+#endif  // defined(WEBRTC_WIN)
+}
+
+#if defined(WEBRTC_WIN)
+DWORD WINAPI RunPlatformThread(void* param) {
+  // The GetLastError() function only returns valid results when it is called
+  // after a Win32 API function that returns a "failed" result. A crash dump
+  // contains the result from GetLastError() and to make sure it does not
+  // falsely report a Windows error we call SetLastError here.
+  ::SetLastError(ERROR_SUCCESS);
+  auto function = static_cast<std::function<void()>*>(param);
+  (*function)();
+  delete function;
+  return 0;
+}
+#else
+void* RunPlatformThread(void* param) {
+  auto function = static_cast<std::function<void()>*>(param);
+  (*function)();
+  delete function;
+  return 0;
+}
+#endif  // defined(WEBRTC_WIN)
+
+}  // namespace
+
+PlatformThread::PlatformThread(Handle handle, bool joinable)
+    : handle_(handle), joinable_(joinable) {}
+
+PlatformThread::PlatformThread(PlatformThread&& rhs)
+    : handle_(rhs.handle_), joinable_(rhs.joinable_) {
+  rhs.handle_ = absl::nullopt;
+}
+
+PlatformThread& PlatformThread::operator=(PlatformThread&& rhs) {
+  Finalize();
+  handle_ = rhs.handle_;
+  joinable_ = rhs.joinable_;
+  rhs.handle_ = absl::nullopt;
+  return *this;
+}
+
+PlatformThread::~PlatformThread() {
+  Finalize();
+}
+
+PlatformThread PlatformThread::SpawnJoinable(
+    std::function<void()> thread_function,
+    absl::string_view name,
+    ThreadAttributes attributes) {
+  return SpawnThread(std::move(thread_function), name, attributes,
+                     /*joinable=*/true);
+}
+
+PlatformThread PlatformThread::SpawnDetached(
+    std::function<void()> thread_function,
+    absl::string_view name,
+    ThreadAttributes attributes) {
+  return SpawnThread(std::move(thread_function), name, attributes,
+                     /*joinable=*/false);
+}
+
+absl::optional<PlatformThread::Handle> PlatformThread::GetHandle() const {
+  return handle_;
+}
+
+#if defined(WEBRTC_WIN)
+bool PlatformThread::QueueAPC(PAPCFUNC function, ULONG_PTR data) {
+  RTC_DCHECK(handle_.has_value());
+  return handle_.has_value() ? QueueUserAPC(function, *handle_, data) != FALSE
+                             : false;
+}
+#endif
+
+void PlatformThread::Finalize() {
+  if (!handle_.has_value())
+    return;
+#if defined(WEBRTC_WIN)
+  if (joinable_)
+    WaitForSingleObject(*handle_, INFINITE);
+  CloseHandle(*handle_);
+#else
+  if (joinable_)
+    RTC_CHECK_EQ(0, pthread_join(*handle_, nullptr));
+#endif
+  handle_ = absl::nullopt;
+}
+
+PlatformThread PlatformThread::SpawnThread(
+    std::function<void()> thread_function,
+    absl::string_view name,
+    ThreadAttributes attributes,
+    bool joinable) {
+  RTC_DCHECK(thread_function);
+  RTC_DCHECK(!name.empty());
+  // TODO(tommi): Consider lowering the limit to 15 (limit on Linux).
+  RTC_DCHECK(name.length() < 64);
+  auto start_thread_function_ptr =
+      new std::function<void()>([thread_function = std::move(thread_function),
+                                 name = std::string(name), attributes] {
+        rtc::SetCurrentThreadName(name.c_str());
+        SetPriority(attributes.priority);
+        thread_function();
+      });
+#if defined(WEBRTC_WIN)
+  // See bug 2902 for background on STACK_SIZE_PARAM_IS_A_RESERVATION.
+  // Set the reserved stack stack size to 1M, which is the default on Windows
+  // and Linux.
+  DWORD thread_id = 0;
+  PlatformThread::Handle handle = ::CreateThread(
+      nullptr, 1024 * 1024, &RunPlatformThread, start_thread_function_ptr,
+      STACK_SIZE_PARAM_IS_A_RESERVATION, &thread_id);
+  RTC_CHECK(handle) << "CreateThread failed";
+#else
+  pthread_attr_t attr;
+  pthread_attr_init(&attr);
+  // Set the stack stack size to 1M.
+  pthread_attr_setstacksize(&attr, 1024 * 1024);
+  pthread_attr_setdetachstate(
+      &attr, joinable ? PTHREAD_CREATE_JOINABLE : PTHREAD_CREATE_DETACHED);
+  PlatformThread::Handle handle;
+  RTC_CHECK_EQ(0, pthread_create(&handle, &attr, &RunPlatformThread,
+                                 start_thread_function_ptr));
+  pthread_attr_destroy(&attr);
+#endif  // defined(WEBRTC_WIN)
+  return PlatformThread(handle, joinable);
+}
+
+}  // namespace rtc

VocieProcess/rtc_base/platform_thread.h

@@ -0,0 +1,120 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef RTC_BASE_PLATFORM_THREAD_H_
+#define RTC_BASE_PLATFORM_THREAD_H_
+
+#include <functional>
+#include <string>
+#if !defined(WEBRTC_WIN)
+#include <pthread.h>
+#endif
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "rtc_base/platform_thread_types.h"
+
+namespace rtc {
+
+enum class ThreadPriority {
+  kLow = 1,
+  kNormal,
+  kHigh,
+  kRealtime,
+};
+
+struct ThreadAttributes {
+  ThreadPriority priority = ThreadPriority::kNormal;
+  ThreadAttributes& SetPriority(ThreadPriority priority_param) {
+    priority = priority_param;
+    return *this;
+  }
+};
+
+// Represents a simple worker thread.
+class PlatformThread final {
+ public:
+  // Handle is the base platform thread handle.
+#if defined(WEBRTC_WIN)
+  using Handle = HANDLE;
+#else
+  using Handle = pthread_t;
+#endif  // defined(WEBRTC_WIN)
+  // This ctor creates the PlatformThread with an unset handle (returning true
+  // in empty()) and is provided for convenience.
+  // TODO(bugs.webrtc.org/12727) Look into if default and move support can be
+  // removed.
+  PlatformThread() = default;
+
+  // Moves `rhs` into this, storing an empty state in `rhs`.
+  // TODO(bugs.webrtc.org/12727) Look into if default and move support can be
+  // removed.
+  PlatformThread(PlatformThread&& rhs);
+
+  // Copies won't work since we'd have problems with joinable threads.
+  PlatformThread(const PlatformThread&) = delete;
+  PlatformThread& operator=(const PlatformThread&) = delete;
+
+  // Moves `rhs` into this, storing an empty state in `rhs`.
+  // TODO(bugs.webrtc.org/12727) Look into if default and move support can be
+  // removed.
+  PlatformThread& operator=(PlatformThread&& rhs);
+
+  // For a PlatformThread that's been spawned joinable, the destructor suspends
+  // the calling thread until the created thread exits unless the thread has
+  // already exited.
+  virtual ~PlatformThread();
+
+  // Finalizes any allocated resources.
+  // For a PlatformThread that's been spawned joinable, Finalize() suspends
+  // the calling thread until the created thread exits unless the thread has
+  // already exited.
+  // empty() returns true after completion.
+  void Finalize();
+
+  // Returns true if default constructed, moved from, or Finalize()ed.
+  bool empty() const { return !handle_.has_value(); }
+
+  // Creates a started joinable thread which will be joined when the returned
+  // PlatformThread destructs or Finalize() is called.
+  static PlatformThread SpawnJoinable(
+      std::function<void()> thread_function,
+      absl::string_view name,
+      ThreadAttributes attributes = ThreadAttributes());
+
+  // Creates a started detached thread. The caller has to use external
+  // synchronization as nothing is provided by the PlatformThread construct.
+  static PlatformThread SpawnDetached(
+      std::function<void()> thread_function,
+      absl::string_view name,
+      ThreadAttributes attributes = ThreadAttributes());
+
+  // Returns the base platform thread handle of this thread.
+  absl::optional<Handle> GetHandle() const;
+
+#if defined(WEBRTC_WIN)
+  // Queue a Windows APC function that runs when the thread is alertable.
+  bool QueueAPC(PAPCFUNC apc_function, ULONG_PTR data);
+#endif
+
+ private:
+  PlatformThread(Handle handle, bool joinable);
+  static PlatformThread SpawnThread(std::function<void()> thread_function,
+                                    absl::string_view name,
+                                    ThreadAttributes attributes,
+                                    bool joinable);
+
+  absl::optional<Handle> handle_;
+  bool joinable_ = false;
+};
+
+}  // namespace rtc
+
+#endif  // RTC_BASE_PLATFORM_THREAD_H_

VocieProcess/rtc_base/ref_count.h

@@ -0,0 +1,29 @@
+/*
+ *  Copyright 2011 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef RTC_BASE_REF_COUNT_H_
+#define RTC_BASE_REF_COUNT_H_
+
+// Transition file for backwards compatibility with source code
+// that includes the non-API file.
+
+#include "api/ref_count.h"
+
+namespace rtc {
+
+// TODO(bugs.webrtc.org/15622): Deprecate and remove these aliases.
+using RefCountInterface [[deprecated("Use webrtc::RefCountInterface")]] =
+    webrtc::RefCountInterface;
+using RefCountReleaseStatus
+    [[deprecated("Use webrtc::RefCountReleaseStatus")]] =
+        webrtc::RefCountReleaseStatus;
+
+}  // namespace rtc
+
+#endif  // RTC_BASE_REF_COUNT_H_

VocieProcess/rtc_base/ref_counted_object.h

@@ -0,0 +1,142 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef RTC_BASE_REF_COUNTED_OBJECT_H_
+#define RTC_BASE_REF_COUNTED_OBJECT_H_
+
+#include "api/scoped_refptr.h"
+#include "rtc_base/ref_count.h"
+#include "rtc_base/ref_counter.h"
+
+namespace webrtc {
+
+template <class T>
+class RefCountedObject : public T {
+ public:
+  RefCountedObject() {}
+
+  RefCountedObject(const RefCountedObject&) = delete;
+  RefCountedObject& operator=(const RefCountedObject&) = delete;
+
+  template <class P0>
+  explicit RefCountedObject(P0&& p0) : T(std::forward<P0>(p0)) {}
+
+  template <class P0, class P1, class... Args>
+  RefCountedObject(P0&& p0, P1&& p1, Args&&... args)
+      : T(std::forward<P0>(p0),
+          std::forward<P1>(p1),
+          std::forward<Args>(args)...) {}
+
+  void AddRef() const override { ref_count_.IncRef(); }
+
+  RefCountReleaseStatus Release() const override {
+    const auto status = ref_count_.DecRef();
+    if (status == RefCountReleaseStatus::kDroppedLastRef) {
+      delete this;
+    }
+    return status;
+  }
+
+  // Return whether the reference count is one. If the reference count is used
+  // in the conventional way, a reference count of 1 implies that the current
+  // thread owns the reference and no other thread shares it. This call
+  // performs the test for a reference count of one, and performs the memory
+  // barrier needed for the owning thread to act on the object, knowing that it
+  // has exclusive access to the object.
+  virtual bool HasOneRef() const { return ref_count_.HasOneRef(); }
+
+ protected:
+  ~RefCountedObject() override {}
+
+  mutable webrtc::webrtc_impl::RefCounter ref_count_{0};
+};
+
+template <class T>
+class FinalRefCountedObject final : public T {
+ public:
+  using T::T;
+  // Above using declaration propagates a default move constructor
+  // FinalRefCountedObject(FinalRefCountedObject&& other), but we also need
+  // move construction from T.
+  explicit FinalRefCountedObject(T&& other) : T(std::move(other)) {}
+  FinalRefCountedObject(const FinalRefCountedObject&) = delete;
+  FinalRefCountedObject& operator=(const FinalRefCountedObject&) = delete;
+
+  void AddRef() const { ref_count_.IncRef(); }
+  RefCountReleaseStatus Release() const {
+    const auto status = ref_count_.DecRef();
+    if (status == RefCountReleaseStatus::kDroppedLastRef) {
+      delete this;
+    }
+    return status;
+  }
+  bool HasOneRef() const { return ref_count_.HasOneRef(); }
+
+ private:
+  ~FinalRefCountedObject() = default;
+
+  mutable webrtc::webrtc_impl::RefCounter ref_count_{0};
+};
+
+}  // namespace webrtc
+
+// Backwards compatibe aliases.
+// TODO: https://issues.webrtc.org/42225969 - deprecate and remove.
+namespace rtc {
+// Because there are users of this template that use "friend
+// rtc::RefCountedObject<>" to give access to a private destructor, some
+// indirection is needed; "friend" declarations cannot span an "using"
+// declaration. Since a templated class on top of a templated class can't access
+// the subclass' protected members, we duplicate the entire class instead.
+template <class T>
+class RefCountedObject : public T {
+ public:
+  RefCountedObject() {}
+
+  RefCountedObject(const RefCountedObject&) = delete;
+  RefCountedObject& operator=(const RefCountedObject&) = delete;
+
+  template <class P0>
+  explicit RefCountedObject(P0&& p0) : T(std::forward<P0>(p0)) {}
+
+  template <class P0, class P1, class... Args>
+  RefCountedObject(P0&& p0, P1&& p1, Args&&... args)
+      : T(std::forward<P0>(p0),
+          std::forward<P1>(p1),
+          std::forward<Args>(args)...) {}
+
+  void AddRef() const override { ref_count_.IncRef(); }
+
+  webrtc::RefCountReleaseStatus Release() const override {
+    const auto status = ref_count_.DecRef();
+    if (status == webrtc::RefCountReleaseStatus::kDroppedLastRef) {
+      delete this;
+    }
+    return status;
+  }
+
+  // Return whether the reference count is one. If the reference count is used
+  // in the conventional way, a reference count of 1 implies that the current
+  // thread owns the reference and no other thread shares it. This call
+  // performs the test for a reference count of one, and performs the memory
+  // barrier needed for the owning thread to act on the object, knowing that it
+  // has exclusive access to the object.
+  virtual bool HasOneRef() const { return ref_count_.HasOneRef(); }
+
+ protected:
+  ~RefCountedObject() override {}
+
+  mutable webrtc::webrtc_impl::RefCounter ref_count_{0};
+};
+
+template <typename T>
+using FinalRefCountedObject = webrtc::FinalRefCountedObject<T>;
+}  // namespace rtc
+
+#endif  // RTC_BASE_REF_COUNTED_OBJECT_H_

VocieProcess/rtc_base/ref_counter.h

@@ -0,0 +1,75 @@
+/*
+ *  Copyright 2017 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef RTC_BASE_REF_COUNTER_H_
+#define RTC_BASE_REF_COUNTER_H_
+
+#include <atomic>
+
+#include "rtc_base/ref_count.h"
+
+namespace webrtc {
+namespace webrtc_impl {
+
+class RefCounter {
+ public:
+  explicit RefCounter(int ref_count) : ref_count_(ref_count) {}
+  RefCounter() = delete;
+
+  void IncRef() {
+    // Relaxed memory order: The current thread is allowed to act on the
+    // resource protected by the reference counter both before and after the
+    // atomic op, so this function doesn't prevent memory access reordering.
+    ref_count_.fetch_add(1, std::memory_order_relaxed);
+  }
+
+  // Returns kDroppedLastRef if this call dropped the last reference; the caller
+  // should therefore free the resource protected by the reference counter.
+  // Otherwise, returns kOtherRefsRemained (note that in case of multithreading,
+  // some other caller may have dropped the last reference by the time this call
+  // returns; all we know is that we didn't do it).
+  RefCountReleaseStatus DecRef() {
+    // Use release-acquire barrier to ensure all actions on the protected
+    // resource are finished before the resource can be freed.
+    // When ref_count_after_subtract > 0, this function require
+    // std::memory_order_release part of the barrier.
+    // When ref_count_after_subtract == 0, this function require
+    // std::memory_order_acquire part of the barrier.
+    // In addition std::memory_order_release is used for synchronization with
+    // the HasOneRef function to make sure all actions on the protected resource
+    // are finished before the resource is assumed to have exclusive access.
+    int ref_count_after_subtract =
+        ref_count_.fetch_sub(1, std::memory_order_acq_rel) - 1;
+    return ref_count_after_subtract == 0
+               ? RefCountReleaseStatus::kDroppedLastRef
+               : RefCountReleaseStatus::kOtherRefsRemained;
+  }
+
+  // Return whether the reference count is one. If the reference count is used
+  // in the conventional way, a reference count of 1 implies that the current
+  // thread owns the reference and no other thread shares it. This call performs
+  // the test for a reference count of one, and performs the memory barrier
+  // needed for the owning thread to act on the resource protected by the
+  // reference counter, knowing that it has exclusive access.
+  bool HasOneRef() const {
+    // To ensure resource protected by the reference counter has exclusive
+    // access, all changes to the resource before it was released by other
+    // threads must be visible by current thread. That is provided by release
+    // (in DecRef) and acquire (in this function) ordering.
+    return ref_count_.load(std::memory_order_acquire) == 1;
+  }
+
+ private:
+  std::atomic<int> ref_count_;
+};
+
+}  // namespace webrtc_impl
+}  // namespace webrtc
+
+#endif  // RTC_BASE_REF_COUNTER_H_

VocieProcess/rtc_base/synchronization/sequence_checker_internal.cc

@@ -0,0 +1,86 @@
+/*
+ *  Copyright 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "rtc_base/synchronization/sequence_checker_internal.h"
+
+#include <string>
+
+#include "rtc_base/checks.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace webrtc {
+namespace webrtc_sequence_checker_internal {
+
+SequenceCheckerImpl::SequenceCheckerImpl(bool attach_to_current_thread)
+    : attached_(attach_to_current_thread),
+      valid_thread_(rtc::CurrentThreadRef()),
+      valid_queue_(TaskQueueBase::Current()) {}
+
+SequenceCheckerImpl::SequenceCheckerImpl(TaskQueueBase* attached_queue)
+    : attached_(attached_queue != nullptr),
+      valid_thread_(rtc::PlatformThreadRef()),
+      valid_queue_(attached_queue) {}
+
+bool SequenceCheckerImpl::IsCurrent() const {
+  const TaskQueueBase* const current_queue = TaskQueueBase::Current();
+  const rtc::PlatformThreadRef current_thread = rtc::CurrentThreadRef();
+  MutexLock scoped_lock(&lock_);
+  if (!attached_) {  // Previously detached.
+    attached_ = true;
+    valid_thread_ = current_thread;
+    valid_queue_ = current_queue;
+    return true;
+  }
+  if (valid_queue_) {
+    return valid_queue_ == current_queue;
+  }
+  return rtc::IsThreadRefEqual(valid_thread_, current_thread);
+}
+
+void SequenceCheckerImpl::Detach() {
+  MutexLock scoped_lock(&lock_);
+  attached_ = false;
+  // We don't need to touch the other members here, they will be
+  // reset on the next call to IsCurrent().
+}
+
+#if RTC_DCHECK_IS_ON
+std::string SequenceCheckerImpl::ExpectationToString() const {
+  const TaskQueueBase* const current_queue = TaskQueueBase::Current();
+  const rtc::PlatformThreadRef current_thread = rtc::CurrentThreadRef();
+  MutexLock scoped_lock(&lock_);
+  if (!attached_)
+    return "Checker currently not attached.";
+
+  // The format of the string is meant to compliment the one we have inside of
+  // FatalLog() (checks.cc).  Example:
+  //
+  // # Expected: TQ: 0x0 SysQ: 0x7fff69541330 Thread: 0x11dcf6dc0
+  // # Actual:   TQ: 0x7fa8f0604190 SysQ: 0x7fa8f0604a30 Thread: 0x700006f1a000
+  // TaskQueue doesn't match
+
+  rtc::StringBuilder message;
+  message.AppendFormat(
+      "# Expected: TQ: %p Thread: %p\n"
+      "# Actual:   TQ: %p Thread: %p\n",
+      valid_queue_, reinterpret_cast<const void*>(valid_thread_), current_queue,
+      reinterpret_cast<const void*>(current_thread));
+
+  if ((valid_queue_ || current_queue) && valid_queue_ != current_queue) {
+    message << "TaskQueue doesn't match\n";
+  } else if (!rtc::IsThreadRefEqual(valid_thread_, current_thread)) {
+    message << "Threads don't match\n";
+  }
+
+  return message.Release();
+}
+#endif  // RTC_DCHECK_IS_ON
+
+}  // namespace webrtc_sequence_checker_internal
+}  // namespace webrtc

VocieProcess/rtc_base/synchronization/sequence_checker_internal.h

@@ -0,0 +1,90 @@
+/*
+ *  Copyright 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef RTC_BASE_SYNCHRONIZATION_SEQUENCE_CHECKER_INTERNAL_H_
+#define RTC_BASE_SYNCHRONIZATION_SEQUENCE_CHECKER_INTERNAL_H_
+
+#include <string>
+#include <type_traits>
+
+#include "api/task_queue/task_queue_base.h"
+#include "rtc_base/platform_thread_types.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/system/rtc_export.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+namespace webrtc_sequence_checker_internal {
+
+// Real implementation of SequenceChecker, for use in debug mode, or
+// for temporary use in release mode (e.g. to RTC_CHECK on a threading issue
+// seen only in the wild).
+//
+// Note: You should almost always use the SequenceChecker class to get the
+// right version for your build configuration.
+class RTC_EXPORT SequenceCheckerImpl {
+ public:
+  explicit SequenceCheckerImpl(bool attach_to_current_thread);
+  explicit SequenceCheckerImpl(TaskQueueBase* attached_queue);
+  ~SequenceCheckerImpl() = default;
+
+  bool IsCurrent() const;
+  // Changes the task queue or thread that is checked for in IsCurrent. This can
+  // be useful when an object may be created on one task queue / thread and then
+  // used exclusively on another thread.
+  void Detach();
+
+  // Returns a string that is formatted to match with the error string printed
+  // by RTC_CHECK() when a condition is not met.
+  // This is used in conjunction with the RTC_DCHECK_RUN_ON() macro.
+  std::string ExpectationToString() const;
+
+ private:
+  mutable Mutex lock_;
+  // These are mutable so that IsCurrent can set them.
+  mutable bool attached_ RTC_GUARDED_BY(lock_);
+  mutable rtc::PlatformThreadRef valid_thread_ RTC_GUARDED_BY(lock_);
+  mutable const TaskQueueBase* valid_queue_ RTC_GUARDED_BY(lock_);
+};
+
+// Do nothing implementation, for use in release mode.
+//
+// Note: You should almost always use the SequenceChecker class to get the
+// right version for your build configuration.
+class SequenceCheckerDoNothing {
+ public:
+  explicit SequenceCheckerDoNothing(bool attach_to_current_thread) {}
+  explicit SequenceCheckerDoNothing(TaskQueueBase* attached_queue) {}
+  bool IsCurrent() const { return true; }
+  void Detach() {}
+};
+
+template <typename ThreadLikeObject>
+std::enable_if_t<std::is_base_of_v<SequenceCheckerImpl, ThreadLikeObject>,
+                 std::string>
+ExpectationToString(const ThreadLikeObject* checker) {
+#if RTC_DCHECK_IS_ON
+  return checker->ExpectationToString();
+#else
+  return std::string();
+#endif
+}
+
+// Catch-all implementation for types other than explicitly supported above.
+template <typename ThreadLikeObject>
+std::enable_if_t<!std::is_base_of_v<SequenceCheckerImpl, ThreadLikeObject>,
+                 std::string>
+ExpectationToString(const ThreadLikeObject*) {
+  return std::string();
+}
+
+}  // namespace webrtc_sequence_checker_internal
+}  // namespace webrtc
+
+#endif  // RTC_BASE_SYNCHRONIZATION_SEQUENCE_CHECKER_INTERNAL_H_

VocieProcess/rtc_base/synchronization/yield_policy.cc

@@ -0,0 +1,82 @@
+/*
+ *  Copyright 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "rtc_base/synchronization/yield_policy.h"
+
+#include "absl/base/attributes.h"
+#include "absl/base/config.h"
+#include "rtc_base/checks.h"
+#if !defined(ABSL_HAVE_THREAD_LOCAL) && defined(WEBRTC_POSIX)
+#include <pthread.h>
+#endif
+
+namespace rtc {
+namespace {
+
+#if defined(ABSL_HAVE_THREAD_LOCAL)
+
+ABSL_CONST_INIT thread_local YieldInterface* current_yield_policy = nullptr;
+
+YieldInterface* GetCurrentYieldPolicy() {
+  return current_yield_policy;
+}
+
+void SetCurrentYieldPolicy(YieldInterface* ptr) {
+  current_yield_policy = ptr;
+}
+
+#elif defined(WEBRTC_POSIX)
+
+// Emscripten does not support the C++11 thread_local keyword but does support
+// the pthread thread-local storage API.
+// https://github.com/emscripten-core/emscripten/issues/3502
+
+ABSL_CONST_INIT pthread_key_t g_current_yield_policy_tls = 0;
+
+void InitializeTls() {
+  RTC_CHECK_EQ(pthread_key_create(&g_current_yield_policy_tls, nullptr), 0);
+}
+
+pthread_key_t GetCurrentYieldPolicyTls() {
+  static pthread_once_t init_once = PTHREAD_ONCE_INIT;
+  RTC_CHECK_EQ(pthread_once(&init_once, &InitializeTls), 0);
+  return g_current_yield_policy_tls;
+}
+
+YieldInterface* GetCurrentYieldPolicy() {
+  return static_cast<YieldInterface*>(
+      pthread_getspecific(GetCurrentYieldPolicyTls()));
+}
+
+void SetCurrentYieldPolicy(YieldInterface* ptr) {
+  pthread_setspecific(GetCurrentYieldPolicyTls(), ptr);
+}
+
+#else
+#error Unsupported platform
+#endif
+
+}  // namespace
+
+ScopedYieldPolicy::ScopedYieldPolicy(YieldInterface* policy)
+    : previous_(GetCurrentYieldPolicy()) {
+  SetCurrentYieldPolicy(policy);
+}
+
+ScopedYieldPolicy::~ScopedYieldPolicy() {
+  SetCurrentYieldPolicy(previous_);
+}
+
+void ScopedYieldPolicy::YieldExecution() {
+  YieldInterface* current = GetCurrentYieldPolicy();
+  if (current)
+    current->YieldExecution();
+}
+
+}  // namespace rtc

VocieProcess/rtc_base/synchronization/yield_policy.h

@@ -0,0 +1,38 @@
+/*
+ *  Copyright 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef RTC_BASE_SYNCHRONIZATION_YIELD_POLICY_H_
+#define RTC_BASE_SYNCHRONIZATION_YIELD_POLICY_H_
+
+namespace rtc {
+class YieldInterface {
+ public:
+  virtual ~YieldInterface() = default;
+  virtual void YieldExecution() = 0;
+};
+
+// Sets the current thread-local yield policy while it's in scope and reverts
+// to the previous policy when it leaves the scope.
+class ScopedYieldPolicy final {
+ public:
+  explicit ScopedYieldPolicy(YieldInterface* policy);
+  ScopedYieldPolicy(const ScopedYieldPolicy&) = delete;
+  ScopedYieldPolicy& operator=(const ScopedYieldPolicy&) = delete;
+  ~ScopedYieldPolicy();
+  // Will yield as specified by the currently active thread-local yield policy
+  // (which by default is a no-op).
+  static void YieldExecution();
+
+ private:
+  YieldInterface* const previous_;
+};
+
+}  // namespace rtc
+
+#endif  // RTC_BASE_SYNCHRONIZATION_YIELD_POLICY_H_

VocieProcess/rtc_base/system/ignore_warnings.h

@@ -0,0 +1,29 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef RTC_BASE_SYSTEM_IGNORE_WARNINGS_H_
+#define RTC_BASE_SYSTEM_IGNORE_WARNINGS_H_
+
+#ifdef __clang__
+#define RTC_PUSH_IGNORING_WFRAME_LARGER_THAN() \
+  _Pragma("clang diagnostic push")             \
+      _Pragma("clang diagnostic ignored \"-Wframe-larger-than=\"")
+#define RTC_POP_IGNORING_WFRAME_LARGER_THAN() _Pragma("clang diagnostic pop")
+#elif __GNUC__
+#define RTC_PUSH_IGNORING_WFRAME_LARGER_THAN() \
+  _Pragma("GCC diagnostic push")               \
+      _Pragma("GCC diagnostic ignored \"-Wframe-larger-than=\"")
+#define RTC_POP_IGNORING_WFRAME_LARGER_THAN() _Pragma("GCC diagnostic pop")
+#else
+#define RTC_PUSH_IGNORING_WFRAME_LARGER_THAN()
+#define RTC_POP_IGNORING_WFRAME_LARGER_THAN()
+#endif
+
+#endif  // RTC_BASE_SYSTEM_IGNORE_WARNINGS_H_

VocieProcess/rtc_base/system/warn_current_thread_is_deadlocked.cc

@@ -0,0 +1,19 @@
+/*
+ *  Copyright 2019 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "rtc_base/system/warn_current_thread_is_deadlocked.h"
+
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+
+
+}  // namespace webrtc

VocieProcess/rtc_base/system/warn_current_thread_is_deadlocked.h

@@ -0,0 +1,24 @@
+/*
+ *  Copyright 2019 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef RTC_BASE_SYSTEM_WARN_CURRENT_THREAD_IS_DEADLOCKED_H_
+#define RTC_BASE_SYSTEM_WARN_CURRENT_THREAD_IS_DEADLOCKED_H_
+
+namespace webrtc {
+
+#if defined(WEBRTC_ANDROID) && !defined(WEBRTC_CHROMIUM_BUILD)
+void WarnThatTheCurrentThreadIsProbablyDeadlocked();
+#else
+inline void WarnThatTheCurrentThreadIsProbablyDeadlocked() {}
+#endif
+
+}  // namespace webrtc
+
+#endif  // RTC_BASE_SYSTEM_WARN_CURRENT_THREAD_IS_DEADLOCKED_H_

VocieProcess/rtc_base/trace_event.h

@@ -0,0 +1,807 @@
+/*
+ *  Copyright (c) 2024 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef RTC_BASE_TRACE_EVENT_H_
+#define RTC_BASE_TRACE_EVENT_H_
+
+#if defined(RTC_DISABLE_TRACE_EVENTS)
+#define RTC_TRACE_EVENTS_ENABLED 0
+#else
+#define RTC_TRACE_EVENTS_ENABLED 1
+#endif
+
+// IWYU pragma: begin_exports
+#if defined(RTC_USE_PERFETTO)
+#include "rtc_base/trace_categories.h"
+#endif
+// IWYU pragma: end_exports
+
+#if !defined(RTC_USE_PERFETTO)
+#include <string>
+
+#include "rtc_base/event_tracer.h"
+
+#define RTC_NOOP() \
+  do {             \
+  } while (0)
+
+// TODO(b/42226290): Add implementation for these events with Perfetto.
+#define TRACE_EVENT_BEGIN(category, name, ...) RTC_NOOP();
+#define TRACE_EVENT_END(category, ...) RTC_NOOP();
+#define TRACE_EVENT(category, name, ...) RTC_NOOP();
+#define TRACE_EVENT_INSTANT(category, name, ...) RTC_NOOP();
+#define TRACE_EVENT_CATEGORY_ENABLED(category) RTC_NOOP();
+#define TRACE_COUNTER(category, track, ...) RTC_NOOP();
+
+// Type values for identifying types in the TraceValue union.
+#define TRACE_VALUE_TYPE_BOOL (static_cast<unsigned char>(1))
+#define TRACE_VALUE_TYPE_UINT (static_cast<unsigned char>(2))
+#define TRACE_VALUE_TYPE_INT (static_cast<unsigned char>(3))
+#define TRACE_VALUE_TYPE_DOUBLE (static_cast<unsigned char>(4))
+#define TRACE_VALUE_TYPE_POINTER (static_cast<unsigned char>(5))
+#define TRACE_VALUE_TYPE_STRING (static_cast<unsigned char>(6))
+#define TRACE_VALUE_TYPE_COPY_STRING (static_cast<unsigned char>(7))
+
+#if defined(TRACE_EVENT0)
+#error "Another copy of trace_event.h has already been included."
+#endif
+
+#if RTC_TRACE_EVENTS_ENABLED
+
+// Extracted from Chromium's src/base/debug/trace_event.h.
+
+// This header is designed to give you trace_event macros without specifying
+// how the events actually get collected and stored. If you need to expose trace
+// event to some other universe, you can copy-and-paste this file,
+// implement the TRACE_EVENT_API macros, and do any other necessary fixup for
+// the target platform. The end result is that multiple libraries can funnel
+// events through to a shared trace event collector.
+
+// Trace events are for tracking application performance and resource usage.
+// Macros are provided to track:
+//    Begin and end of function calls
+//    Counters
+//
+// Events are issued against categories. Whereas RTC_LOG's
+// categories are statically defined, TRACE categories are created
+// implicitly with a string. For example:
+//   TRACE_EVENT_INSTANT0("MY_SUBSYSTEM", "SomeImportantEvent")
+//
+// Events can be INSTANT, or can be pairs of BEGIN and END in the same scope:
+//   TRACE_EVENT_BEGIN0("MY_SUBSYSTEM", "SomethingCostly")
+//   doSomethingCostly()
+//   TRACE_EVENT_END0("MY_SUBSYSTEM", "SomethingCostly")
+// Note: our tools can't always determine the correct BEGIN/END pairs unless
+// these are used in the same scope. Use ASYNC_BEGIN/ASYNC_END macros if you
+// need them to be in separate scopes.
+//
+// A common use case is to trace entire function scopes. This
+// issues a trace BEGIN and END automatically:
+//   void doSomethingCostly() {
+//     TRACE_EVENT0("MY_SUBSYSTEM", "doSomethingCostly");
+//     ...
+//   }
+//
+// Additional parameters can be associated with an event:
+//   void doSomethingCostly2(int howMuch) {
+//     TRACE_EVENT1("MY_SUBSYSTEM", "doSomethingCostly",
+//         "howMuch", howMuch);
+//     ...
+//   }
+//
+// The trace system will automatically add to this information the
+// current process id, thread id, and a timestamp in microseconds.
+//
+// To trace an asynchronous procedure such as an IPC send/receive, use
+// ASYNC_BEGIN and ASYNC_END:
+//   [single threaded sender code]
+//     static int send_count = 0;
+//     ++send_count;
+//     TRACE_EVENT_ASYNC_BEGIN0("ipc", "message", send_count);
+//     Send(new MyMessage(send_count));
+//   [receive code]
+//     void OnMyMessage(send_count) {
+//       TRACE_EVENT_ASYNC_END0("ipc", "message", send_count);
+//     }
+// The third parameter is a unique ID to match ASYNC_BEGIN/ASYNC_END pairs.
+// ASYNC_BEGIN and ASYNC_END can occur on any thread of any traced process.
+// Pointers can be used for the ID parameter, and they will be mangled
+// internally so that the same pointer on two different processes will not
+// match. For example:
+//   class MyTracedClass {
+//    public:
+//     MyTracedClass() {
+//       TRACE_EVENT_ASYNC_BEGIN0("category", "MyTracedClass", this);
+//     }
+//     ~MyTracedClass() {
+//       TRACE_EVENT_ASYNC_END0("category", "MyTracedClass", this);
+//     }
+//   }
+//
+// Trace event also supports counters, which is a way to track a quantity
+// as it varies over time. Counters are created with the following macro:
+//   TRACE_COUNTER1("MY_SUBSYSTEM", "myCounter", g_myCounterValue);
+//
+// Counters are process-specific. The macro itself can be issued from any
+// thread, however.
+//
+// Sometimes, you want to track two counters at once. You can do this with two
+// counter macros:
+//   TRACE_COUNTER1("MY_SUBSYSTEM", "myCounter0", g_myCounterValue[0]);
+//   TRACE_COUNTER1("MY_SUBSYSTEM", "myCounter1", g_myCounterValue[1]);
+// Or you can do it with a combined macro:
+//   TRACE_COUNTER2("MY_SUBSYSTEM", "myCounter",
+//       "bytesPinned", g_myCounterValue[0],
+//       "bytesAllocated", g_myCounterValue[1]);
+// This indicates to the tracing UI that these counters should be displayed
+// in a single graph, as a summed area chart.
+//
+// Since counters are in a global namespace, you may want to disembiguate with a
+// unique ID, by using the TRACE_COUNTER_ID* variations.
+//
+// By default, trace collection is compiled in, but turned off at runtime.
+// Collecting trace data is the responsibility of the embedding
+// application. In Chrome's case, navigating to about:tracing will turn on
+// tracing and display data collected across all active processes.
+//
+// When are string argument values copied:
+// const char* arg_values are only referenced by default:
+//     TRACE_EVENT1("category", "name",
+//                  "arg1", "literal string is only referenced");
+// Use TRACE_STR_COPY to force copying of a const char*:
+//     TRACE_EVENT1("category", "name",
+//                  "arg1", TRACE_STR_COPY("string will be copied"));
+// std::string arg_values are always copied:
+//     TRACE_EVENT1("category", "name",
+//                  "arg1", std::string("string will be copied"));
+//
+//
+// Thread Safety:
+// Thread safety is provided by methods defined in event_tracer.h. See the file
+// for details.
+
+// By default, const char* argument values are assumed to have long-lived scope
+// and will not be copied. Use this macro to force a const char* to be copied.
+#define TRACE_STR_COPY(str) \
+  webrtc::trace_event_internal::TraceStringWithCopy(str)
+
+// This will mark the trace event as disabled by default. The user will need
+// to explicitly enable the event.
+#define TRACE_DISABLED_BY_DEFAULT(name) "disabled-by-default-" name
+
+// By default, uint64 ID argument values are not mangled with the Process ID in
+// TRACE_EVENT_ASYNC macros. Use this macro to force Process ID mangling.
+#define TRACE_ID_MANGLE(id) \
+  webrtc::trace_event_internal::TraceID::ForceMangle(id)
+
+// Records a pair of begin and end events called "name" for the current
+// scope, with 0, 1 or 2 associated arguments. If the category is not
+// enabled, then this does nothing.
+// - category and name strings must have application lifetime (statics or
+//   literals). They may not include " chars.
+#define TRACE_EVENT0(category, name) \
+  INTERNAL_TRACE_EVENT_ADD_SCOPED(category, name)
+#define TRACE_EVENT1(category, name, arg1_name, arg1_val) \
+  INTERNAL_TRACE_EVENT_ADD_SCOPED(category, name, arg1_name, arg1_val)
+#define TRACE_EVENT2(category, name, arg1_name, arg1_val, arg2_name, arg2_val) \
+  INTERNAL_TRACE_EVENT_ADD_SCOPED(category, name, arg1_name, arg1_val,         \
+                                  arg2_name, arg2_val)
+
+// Enum reflecting the scope of an INSTANT event. Must fit within
+// TRACE_EVENT_FLAG_SCOPE_MASK.
+static constexpr uint8_t TRACE_EVENT_SCOPE_GLOBAL = 0u << 2;
+static constexpr uint8_t TRACE_EVENT_SCOPE_PROCESS = 1u << 2;
+static constexpr uint8_t TRACE_EVENT_SCOPE_THREAD = 2u << 2;
+
+// Records a single event called "name" immediately, with 0, 1 or 2
+// associated arguments. If the category is not enabled, then this
+// does nothing.
+// - category and name strings must have application lifetime (statics or
+//   literals). They may not include " chars.
+#define TRACE_EVENT_INSTANT0(category, name, scope)                   \
+  INTERNAL_TRACE_EVENT_ADD(TRACE_EVENT_PHASE_INSTANT, category, name, \
+                           TRACE_EVENT_FLAG_NONE)
+#define TRACE_EVENT_INSTANT1(category, name, scope, arg1_name, arg1_val) \
+  INTERNAL_TRACE_EVENT_ADD(TRACE_EVENT_PHASE_INSTANT, category, name,    \
+                           TRACE_EVENT_FLAG_NONE, arg1_name, arg1_val)
+#define TRACE_EVENT_INSTANT2(category, name, scope, arg1_name, arg1_val, \
+                             arg2_name, arg2_val)                        \
+  INTERNAL_TRACE_EVENT_ADD(TRACE_EVENT_PHASE_INSTANT, category, name,    \
+                           TRACE_EVENT_FLAG_NONE, arg1_name, arg1_val,   \
+                           arg2_name, arg2_val)
+
+// Records a single BEGIN event called "name" immediately, with 0, 1 or 2
+// associated arguments. If the category is not enabled, then this
+// does nothing.
+// - category and name strings must have application lifetime (statics or
+//   literals). They may not include " chars.
+#define TRACE_EVENT_BEGIN0(category, name)                          \
+  INTERNAL_TRACE_EVENT_ADD(TRACE_EVENT_PHASE_BEGIN, category, name, \
+                           TRACE_EVENT_FLAG_NONE)
+#define TRACE_EVENT_BEGIN1(category, name, arg1_name, arg1_val)     \
+  INTERNAL_TRACE_EVENT_ADD(TRACE_EVENT_PHASE_BEGIN, category, name, \
+                           TRACE_EVENT_FLAG_NONE, arg1_name, arg1_val)
+#define TRACE_EVENT_BEGIN2(category, name, arg1_name, arg1_val, arg2_name, \
+                           arg2_val)                                       \
+  INTERNAL_TRACE_EVENT_ADD(TRACE_EVENT_PHASE_BEGIN, category, name,        \
+                           TRACE_EVENT_FLAG_NONE, arg1_name, arg1_val,     \
+                           arg2_name, arg2_val)
+
+// Records a single END event for "name" immediately. If the category
+// is not enabled, then this does nothing.
+// - category and name strings must have application lifetime (statics or
+//   literals). They may not include " chars.
+#define TRACE_EVENT_END0(category, name)                          \
+  INTERNAL_TRACE_EVENT_ADD(TRACE_EVENT_PHASE_END, category, name, \
+                           TRACE_EVENT_FLAG_NONE)
+#define TRACE_EVENT_END1(category, name, arg1_name, arg1_val)     \
+  INTERNAL_TRACE_EVENT_ADD(TRACE_EVENT_PHASE_END, category, name, \
+                           TRACE_EVENT_FLAG_NONE, arg1_name, arg1_val)
+#define TRACE_EVENT_END2(category, name, arg1_name, arg1_val, arg2_name, \
+                         arg2_val)                                       \
+  INTERNAL_TRACE_EVENT_ADD(TRACE_EVENT_PHASE_END, category, name,        \
+                           TRACE_EVENT_FLAG_NONE, arg1_name, arg1_val,   \
+                           arg2_name, arg2_val)
+
+// Records the value of a counter called "name" immediately. Value
+// must be representable as a 32 bit integer.
+// - category and name strings must have application lifetime (statics or
+//   literals). They may not include " chars.
+#define TRACE_COUNTER1(category, name, value)                         \
+  INTERNAL_TRACE_EVENT_ADD(TRACE_EVENT_PHASE_COUNTER, category, name, \
+                           TRACE_EVENT_FLAG_NONE, "value",            \
+                           static_cast<int>(value))
+
+// Records the values of a multi-parted counter called "name" immediately.
+// The UI will treat value1 and value2 as parts of a whole, displaying their
+// values as a stacked-bar chart.
+// - category and name strings must have application lifetime (statics or
+//   literals). They may not include " chars.
+#define TRACE_COUNTER2(category, name, value1_name, value1_val, value2_name, \
+                       value2_val)                                           \
+  INTERNAL_TRACE_EVENT_ADD(TRACE_EVENT_PHASE_COUNTER, category, name,        \
+                           TRACE_EVENT_FLAG_NONE, value1_name,               \
+                           static_cast<int>(value1_val), value2_name,        \
+                           static_cast<int>(value2_val))
+
+// Records the value of a counter called "name" immediately. Value
+// must be representable as a 32 bit integer.
+// - category and name strings must have application lifetime (statics or
+//   literals). They may not include " chars.
+// - `id` is used to disambiguate counters with the same name. It must either
+//   be a pointer or an integer value up to 64 bits. If it's a pointer, the bits
+//   will be xored with a hash of the process ID so that the same pointer on
+//   two different processes will not collide.
+#define TRACE_COUNTER_ID1(category, name, id, value)                          \
+  INTERNAL_TRACE_EVENT_ADD_WITH_ID(TRACE_EVENT_PHASE_COUNTER, category, name, \
+                                   id, TRACE_EVENT_FLAG_NONE, "value",        \
+                                   static_cast<int>(value))
+
+// Records the values of a multi-parted counter called "name" immediately.
+// The UI will treat value1 and value2 as parts of a whole, displaying their
+// values as a stacked-bar chart.
+// - category and name strings must have application lifetime (statics or
+//   literals). They may not include " chars.
+// - `id` is used to disambiguate counters with the same name. It must either
+//   be a pointer or an integer value up to 64 bits. If it's a pointer, the bits
+//   will be xored with a hash of the process ID so that the same pointer on
+//   two different processes will not collide.
+#define TRACE_COUNTER_ID2(category, name, id, value1_name, value1_val,        \
+                          value2_name, value2_val)                            \
+  INTERNAL_TRACE_EVENT_ADD_WITH_ID(TRACE_EVENT_PHASE_COUNTER, category, name, \
+                                   id, TRACE_EVENT_FLAG_NONE, value1_name,    \
+                                   static_cast<int>(value1_val), value2_name, \
+                                   static_cast<int>(value2_val))
+
+// Records a single ASYNC_BEGIN event called "name" immediately, with 0, 1 or 2
+// associated arguments. If the category is not enabled, then this
+// does nothing.
+// - category and name strings must have application lifetime (statics or
+//   literals). They may not include " chars.
+// - `id` is used to match the ASYNC_BEGIN event with the ASYNC_END event. ASYNC
+//   events are considered to match if their category, name and id values all
+//   match. `id` must either be a pointer or an integer value up to 64 bits. If
+//   it's a pointer, the bits will be xored with a hash of the process ID so
+//   that the same pointer on two different processes will not collide.
+// An asynchronous operation can consist of multiple phases. The first phase is
+// defined by the ASYNC_BEGIN calls. Additional phases can be defined using the
+// ASYNC_STEP macros. When the operation completes, call ASYNC_END.
+// An ASYNC trace typically occur on a single thread (if not, they will only be
+// drawn on the thread defined in the ASYNC_BEGIN event), but all events in that
+// operation must use the same `name` and `id`. Each event can have its own
+// args.
+#define TRACE_EVENT_ASYNC_BEGIN0(category, name, id)                        \
+  INTERNAL_TRACE_EVENT_ADD_WITH_ID(TRACE_EVENT_PHASE_ASYNC_BEGIN, category, \
+                                   name, id, TRACE_EVENT_FLAG_NONE)
+#define TRACE_EVENT_ASYNC_BEGIN1(category, name, id, arg1_name, arg1_val)      \
+  INTERNAL_TRACE_EVENT_ADD_WITH_ID(TRACE_EVENT_PHASE_ASYNC_BEGIN, category,    \
+                                   name, id, TRACE_EVENT_FLAG_NONE, arg1_name, \
+                                   arg1_val)
+#define TRACE_EVENT_ASYNC_BEGIN2(category, name, id, arg1_name, arg1_val,      \
+                                 arg2_name, arg2_val)                          \
+  INTERNAL_TRACE_EVENT_ADD_WITH_ID(TRACE_EVENT_PHASE_ASYNC_BEGIN, category,    \
+                                   name, id, TRACE_EVENT_FLAG_NONE, arg1_name, \
+                                   arg1_val, arg2_name, arg2_val)
+
+// Records a single ASYNC_STEP event for `step` immediately. If the category
+// is not enabled, then this does nothing. The `name` and `id` must match the
+// ASYNC_BEGIN event above. The `step` param identifies this step within the
+// async event. This should be called at the beginning of the next phase of an
+// asynchronous operation.
+#define TRACE_EVENT_ASYNC_STEP_INTO0(category, name, id, step)              \
+  INTERNAL_TRACE_EVENT_ADD_WITH_ID(TRACE_EVENT_PHASE_ASYNC_STEP, category,  \
+                                   name, id, TRACE_EVENT_FLAG_NONE, "step", \
+                                   step)
+#define TRACE_EVENT_ASYNC_STEP_INTO1(category, name, id, step, arg1_name,   \
+                                     arg1_val)                              \
+  INTERNAL_TRACE_EVENT_ADD_WITH_ID(TRACE_EVENT_PHASE_ASYNC_STEP, category,  \
+                                   name, id, TRACE_EVENT_FLAG_NONE, "step", \
+                                   step, arg1_name, arg1_val)
+
+// Records a single ASYNC_END event for "name" immediately. If the category
+// is not enabled, then this does nothing.
+#define TRACE_EVENT_ASYNC_END0(category, name, id)                        \
+  INTERNAL_TRACE_EVENT_ADD_WITH_ID(TRACE_EVENT_PHASE_ASYNC_END, category, \
+                                   name, id, TRACE_EVENT_FLAG_NONE)
+#define TRACE_EVENT_ASYNC_END1(category, name, id, arg1_name, arg1_val)        \
+  INTERNAL_TRACE_EVENT_ADD_WITH_ID(TRACE_EVENT_PHASE_ASYNC_END, category,      \
+                                   name, id, TRACE_EVENT_FLAG_NONE, arg1_name, \
+                                   arg1_val)
+#define TRACE_EVENT_ASYNC_END2(category, name, id, arg1_name, arg1_val,        \
+                               arg2_name, arg2_val)                            \
+  INTERNAL_TRACE_EVENT_ADD_WITH_ID(TRACE_EVENT_PHASE_ASYNC_END, category,      \
+                                   name, id, TRACE_EVENT_FLAG_NONE, arg1_name, \
+                                   arg1_val, arg2_name, arg2_val)
+
+////////////////////////////////////////////////////////////////////////////////
+// Implementation specific tracing API definitions.
+
+// Get a pointer to the enabled state of the given trace category. Only
+// long-lived literal strings should be given as the category name. The returned
+// pointer can be held permanently in a local static for example. If the
+// unsigned char is non-zero, tracing is enabled. If tracing is enabled,
+// TRACE_EVENT_API_ADD_TRACE_EVENT can be called. It's OK if tracing is disabled
+// between the load of the tracing state and the call to
+// TRACE_EVENT_API_ADD_TRACE_EVENT, because this flag only provides an early out
+// for best performance when tracing is disabled.
+// const unsigned char*
+//     TRACE_EVENT_API_GET_CATEGORY_ENABLED(const char* category_name)
+#define TRACE_EVENT_API_GET_CATEGORY_ENABLED \
+  webrtc::EventTracer::GetCategoryEnabled
+
+// Add a trace event to the platform tracing system.
+// void TRACE_EVENT_API_ADD_TRACE_EVENT(
+//                    char phase,
+//                    const unsigned char* category_enabled,
+//                    const char* name,
+//                    unsigned long long id,
+//                    int num_args,
+//                    const char** arg_names,
+//                    const unsigned char* arg_types,
+//                    const unsigned long long* arg_values,
+//                    unsigned char flags)
+#define TRACE_EVENT_API_ADD_TRACE_EVENT webrtc::EventTracer::AddTraceEvent
+
+////////////////////////////////////////////////////////////////////////////////
+
+// Implementation detail: trace event macros create temporary variables
+// to keep instrumentation overhead low. These macros give each temporary
+// variable a unique name based on the line number to prevent name collissions.
+#define INTERNAL_TRACE_EVENT_UID3(a, b) trace_event_unique_##a##b
+#define INTERNAL_TRACE_EVENT_UID2(a, b) INTERNAL_TRACE_EVENT_UID3(a, b)
+#define INTERNAL_TRACE_EVENT_UID(name_prefix) \
+  INTERNAL_TRACE_EVENT_UID2(name_prefix, __LINE__)
+
+#if WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS
+#define INTERNAL_TRACE_EVENT_INFO_TYPE const unsigned char*
+#else
+#define INTERNAL_TRACE_EVENT_INFO_TYPE static const unsigned char*
+#endif  // WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS
+
+// Implementation detail: internal macro to create static category.
+#define INTERNAL_TRACE_EVENT_GET_CATEGORY_INFO(category)               \
+  INTERNAL_TRACE_EVENT_INFO_TYPE INTERNAL_TRACE_EVENT_UID(catstatic) = \
+      TRACE_EVENT_API_GET_CATEGORY_ENABLED(category);
+
+// Implementation detail: internal macro to create static category and add
+// event if the category is enabled.
+#define INTERNAL_TRACE_EVENT_ADD(phase, category, name, flags, ...)        \
+  do {                                                                     \
+    INTERNAL_TRACE_EVENT_GET_CATEGORY_INFO(category);                      \
+    if (*INTERNAL_TRACE_EVENT_UID(catstatic)) {                            \
+      webrtc::trace_event_internal::AddTraceEvent(                         \
+          phase, INTERNAL_TRACE_EVENT_UID(catstatic), name,                \
+          webrtc::trace_event_internal::kNoEventId, flags, ##__VA_ARGS__); \
+    }                                                                      \
+  } while (0)
+
+// Implementation detail: internal macro to create static category and add begin
+// event if the category is enabled. Also adds the end event when the scope
+// ends.
+#define INTERNAL_TRACE_EVENT_ADD_SCOPED(category, name, ...)                   \
+  INTERNAL_TRACE_EVENT_GET_CATEGORY_INFO(category);                            \
+  webrtc::trace_event_internal::TraceEndOnScopeClose INTERNAL_TRACE_EVENT_UID( \
+      profileScope);                                                           \
+  if (*INTERNAL_TRACE_EVENT_UID(catstatic)) {                                  \
+    webrtc::trace_event_internal::AddTraceEvent(                               \
+        TRACE_EVENT_PHASE_BEGIN, INTERNAL_TRACE_EVENT_UID(catstatic), name,    \
+        webrtc::trace_event_internal::kNoEventId, TRACE_EVENT_FLAG_NONE,       \
+        ##__VA_ARGS__);                                                        \
+    INTERNAL_TRACE_EVENT_UID(profileScope)                                     \
+        .Initialize(INTERNAL_TRACE_EVENT_UID(catstatic), name);                \
+  }
+
+// Implementation detail: internal macro to create static category and add
+// event if the category is enabled.
+#define INTERNAL_TRACE_EVENT_ADD_WITH_ID(phase, category, name, id, flags, \
+                                         ...)                              \
+  do {                                                                     \
+    INTERNAL_TRACE_EVENT_GET_CATEGORY_INFO(category);                      \
+    if (*INTERNAL_TRACE_EVENT_UID(catstatic)) {                            \
+      unsigned char trace_event_flags = flags | TRACE_EVENT_FLAG_HAS_ID;   \
+      webrtc::trace_event_internal::TraceID trace_event_trace_id(          \
+          id, &trace_event_flags);                                         \
+      webrtc::trace_event_internal::AddTraceEvent(                         \
+          phase, INTERNAL_TRACE_EVENT_UID(catstatic), name,                \
+          trace_event_trace_id.data(), trace_event_flags, ##__VA_ARGS__);  \
+    }                                                                      \
+  } while (0)
+
+// Notes regarding the following definitions:
+// New values can be added and propagated to third party libraries, but existing
+// definitions must never be changed, because third party libraries may use old
+// definitions.
+
+// Phase indicates the nature of an event entry. E.g. part of a begin/end pair.
+#define TRACE_EVENT_PHASE_BEGIN ('B')
+#define TRACE_EVENT_PHASE_END ('E')
+#define TRACE_EVENT_PHASE_INSTANT ('I')
+#define TRACE_EVENT_PHASE_ASYNC_BEGIN ('S')
+#define TRACE_EVENT_PHASE_ASYNC_STEP ('T')
+#define TRACE_EVENT_PHASE_ASYNC_END ('F')
+#define TRACE_EVENT_PHASE_METADATA ('M')
+#define TRACE_EVENT_PHASE_COUNTER ('C')
+
+// Flags for changing the behavior of TRACE_EVENT_API_ADD_TRACE_EVENT.
+#define TRACE_EVENT_FLAG_NONE (static_cast<unsigned char>(0))
+#define TRACE_EVENT_FLAG_HAS_ID (static_cast<unsigned char>(1 << 1))
+#define TRACE_EVENT_FLAG_MANGLE_ID (static_cast<unsigned char>(1 << 2))
+
+namespace webrtc {
+namespace trace_event_internal {
+
+// Specify these values when the corresponding argument of AddTraceEvent is not
+// used.
+const int kZeroNumArgs = 0;
+const unsigned long long kNoEventId = 0;
+
+// TraceID encapsulates an ID that can either be an integer or pointer. Pointers
+// are mangled with the Process ID so that they are unlikely to collide when the
+// same pointer is used on different processes.
+class TraceID {
+ public:
+  class ForceMangle {
+   public:
+    explicit ForceMangle(unsigned long long id) : data_(id) {}
+    explicit ForceMangle(unsigned long id) : data_(id) {}
+    explicit ForceMangle(unsigned int id) : data_(id) {}
+    explicit ForceMangle(unsigned short id) : data_(id) {}
+    explicit ForceMangle(unsigned char id) : data_(id) {}
+    explicit ForceMangle(long long id)
+        : data_(static_cast<unsigned long long>(id)) {}
+    explicit ForceMangle(long id)
+        : data_(static_cast<unsigned long long>(id)) {}
+    explicit ForceMangle(int id) : data_(static_cast<unsigned long long>(id)) {}
+    explicit ForceMangle(short id)
+        : data_(static_cast<unsigned long long>(id)) {}
+    explicit ForceMangle(signed char id)
+        : data_(static_cast<unsigned long long>(id)) {}
+
+    unsigned long long data() const { return data_; }
+
+   private:
+    unsigned long long data_;
+  };
+
+  explicit TraceID(const void* id, unsigned char* flags)
+      : data_(
+            static_cast<unsigned long long>(reinterpret_cast<uintptr_t>(id))) {
+    *flags |= TRACE_EVENT_FLAG_MANGLE_ID;
+  }
+  explicit TraceID(ForceMangle id, unsigned char* flags) : data_(id.data()) {
+    *flags |= TRACE_EVENT_FLAG_MANGLE_ID;
+  }
+  explicit TraceID(unsigned long long id, unsigned char* flags) : data_(id) {
+    (void)flags;
+  }
+  explicit TraceID(unsigned long id, unsigned char* flags) : data_(id) {
+    (void)flags;
+  }
+  explicit TraceID(unsigned int id, unsigned char* flags) : data_(id) {
+    (void)flags;
+  }
+  explicit TraceID(unsigned short id, unsigned char* flags) : data_(id) {
+    (void)flags;
+  }
+  explicit TraceID(unsigned char id, unsigned char* flags) : data_(id) {
+    (void)flags;
+  }
+  explicit TraceID(long long id, unsigned char* flags)
+      : data_(static_cast<unsigned long long>(id)) {
+    (void)flags;
+  }
+  explicit TraceID(long id, unsigned char* flags)
+      : data_(static_cast<unsigned long long>(id)) {
+    (void)flags;
+  }
+  explicit TraceID(int id, unsigned char* flags)
+      : data_(static_cast<unsigned long long>(id)) {
+    (void)flags;
+  }
+  explicit TraceID(short id, unsigned char* flags)
+      : data_(static_cast<unsigned long long>(id)) {
+    (void)flags;
+  }
+  explicit TraceID(signed char id, unsigned char* flags)
+      : data_(static_cast<unsigned long long>(id)) {
+    (void)flags;
+  }
+
+  unsigned long long data() const { return data_; }
+
+ private:
+  unsigned long long data_;
+};
+
+// Simple union to store various types as unsigned long long.
+union TraceValueUnion {
+  bool as_bool;
+  unsigned long long as_uint;
+  long long as_int;
+  double as_double;
+  const void* as_pointer;
+  const char* as_string;
+};
+
+// Simple container for const char* that should be copied instead of retained.
+class TraceStringWithCopy {
+ public:
+  explicit TraceStringWithCopy(const char* str) : str_(str) {}
+  operator const char*() const { return str_; }
+
+ private:
+  const char* str_;
+};
+
+// Define SetTraceValue for each allowed type. It stores the type and
+// value in the return arguments. This allows this API to avoid declaring any
+// structures so that it is portable to third_party libraries.
+#define INTERNAL_DECLARE_SET_TRACE_VALUE(actual_type, union_member,      \
+                                         value_type_id)                  \
+  static inline void SetTraceValue(actual_type arg, unsigned char* type, \
+                                   unsigned long long* value) {          \
+    TraceValueUnion type_value;                                          \
+    type_value.union_member = arg;                                       \
+    *type = value_type_id;                                               \
+    *value = type_value.as_uint;                                         \
+  }
+// Simpler form for int types that can be safely casted.
+#define INTERNAL_DECLARE_SET_TRACE_VALUE_INT(actual_type, value_type_id) \
+  static inline void SetTraceValue(actual_type arg, unsigned char* type, \
+                                   unsigned long long* value) {          \
+    *type = value_type_id;                                               \
+    *value = static_cast<unsigned long long>(arg);                       \
+  }
+
+INTERNAL_DECLARE_SET_TRACE_VALUE_INT(unsigned long long, TRACE_VALUE_TYPE_UINT)
+INTERNAL_DECLARE_SET_TRACE_VALUE_INT(unsigned long, TRACE_VALUE_TYPE_UINT)
+INTERNAL_DECLARE_SET_TRACE_VALUE_INT(unsigned int, TRACE_VALUE_TYPE_UINT)
+INTERNAL_DECLARE_SET_TRACE_VALUE_INT(unsigned short, TRACE_VALUE_TYPE_UINT)
+INTERNAL_DECLARE_SET_TRACE_VALUE_INT(unsigned char, TRACE_VALUE_TYPE_UINT)
+INTERNAL_DECLARE_SET_TRACE_VALUE_INT(long long, TRACE_VALUE_TYPE_INT)
+INTERNAL_DECLARE_SET_TRACE_VALUE_INT(long, TRACE_VALUE_TYPE_INT)
+INTERNAL_DECLARE_SET_TRACE_VALUE_INT(int, TRACE_VALUE_TYPE_INT)
+INTERNAL_DECLARE_SET_TRACE_VALUE_INT(short, TRACE_VALUE_TYPE_INT)
+INTERNAL_DECLARE_SET_TRACE_VALUE_INT(signed char, TRACE_VALUE_TYPE_INT)
+INTERNAL_DECLARE_SET_TRACE_VALUE(bool, as_bool, TRACE_VALUE_TYPE_BOOL)
+INTERNAL_DECLARE_SET_TRACE_VALUE(double, as_double, TRACE_VALUE_TYPE_DOUBLE)
+INTERNAL_DECLARE_SET_TRACE_VALUE(const void*,
+                                 as_pointer,
+                                 TRACE_VALUE_TYPE_POINTER)
+INTERNAL_DECLARE_SET_TRACE_VALUE(const char*,
+                                 as_string,
+                                 TRACE_VALUE_TYPE_STRING)
+INTERNAL_DECLARE_SET_TRACE_VALUE(const TraceStringWithCopy&,
+                                 as_string,
+                                 TRACE_VALUE_TYPE_COPY_STRING)
+
+#undef INTERNAL_DECLARE_SET_TRACE_VALUE
+#undef INTERNAL_DECLARE_SET_TRACE_VALUE_INT
+
+// std::string version of SetTraceValue so that trace arguments can be strings.
+static inline void SetTraceValue(const std::string& arg,
+                                 unsigned char* type,
+                                 unsigned long long* value) {
+  TraceValueUnion type_value;
+  type_value.as_string = arg.c_str();
+  *type = TRACE_VALUE_TYPE_COPY_STRING;
+  *value = type_value.as_uint;
+}
+
+// These AddTraceEvent template functions are defined here instead of in the
+// macro, because the arg_values could be temporary objects, such as
+// std::string. In order to store pointers to the internal c_str and pass
+// through to the tracing API, the arg_values must live throughout
+// these procedures.
+
+static inline void AddTraceEvent(char phase,
+                                 const unsigned char* category_enabled,
+                                 const char* name,
+                                 unsigned long long id,
+                                 unsigned char flags) {
+  TRACE_EVENT_API_ADD_TRACE_EVENT(phase, category_enabled, name, id,
+                                  kZeroNumArgs, nullptr, nullptr, nullptr,
+                                  flags);
+}
+
+template <class ARG1_TYPE>
+static inline void AddTraceEvent(char phase,
+                                 const unsigned char* category_enabled,
+                                 const char* name,
+                                 unsigned long long id,
+                                 unsigned char flags,
+                                 const char* arg1_name,
+                                 const ARG1_TYPE& arg1_val) {
+  const int num_args = 1;
+  unsigned char arg_types[1];
+  unsigned long long arg_values[1];
+  SetTraceValue(arg1_val, &arg_types[0], &arg_values[0]);
+  TRACE_EVENT_API_ADD_TRACE_EVENT(phase, category_enabled, name, id, num_args,
+                                  &arg1_name, arg_types, arg_values, flags);
+}
+
+template <class ARG1_TYPE, class ARG2_TYPE>
+static inline void AddTraceEvent(char phase,
+                                 const unsigned char* category_enabled,
+                                 const char* name,
+                                 unsigned long long id,
+                                 unsigned char flags,
+                                 const char* arg1_name,
+                                 const ARG1_TYPE& arg1_val,
+                                 const char* arg2_name,
+                                 const ARG2_TYPE& arg2_val) {
+  const int num_args = 2;
+  const char* arg_names[2] = {arg1_name, arg2_name};
+  unsigned char arg_types[2];
+  unsigned long long arg_values[2];
+  SetTraceValue(arg1_val, &arg_types[0], &arg_values[0]);
+  SetTraceValue(arg2_val, &arg_types[1], &arg_values[1]);
+  TRACE_EVENT_API_ADD_TRACE_EVENT(phase, category_enabled, name, id, num_args,
+                                  arg_names, arg_types, arg_values, flags);
+}
+
+// Used by TRACE_EVENTx macro. Do not use directly.
+class TraceEndOnScopeClose {
+ public:
+  // Note: members of data_ intentionally left uninitialized. See Initialize.
+  TraceEndOnScopeClose() : p_data_(nullptr) {}
+  ~TraceEndOnScopeClose() {
+    if (p_data_)
+      AddEventIfEnabled();
+  }
+
+  void Initialize(const unsigned char* category_enabled, const char* name) {
+    data_.category_enabled = category_enabled;
+    data_.name = name;
+    p_data_ = &data_;
+  }
+
+ private:
+  // Add the end event if the category is still enabled.
+  void AddEventIfEnabled() {
+    // Only called when p_data_ is non-null.
+    if (*p_data_->category_enabled) {
+      TRACE_EVENT_API_ADD_TRACE_EVENT(TRACE_EVENT_PHASE_END,
+                                      p_data_->category_enabled, p_data_->name,
+                                      kNoEventId, kZeroNumArgs, nullptr,
+                                      nullptr, nullptr, TRACE_EVENT_FLAG_NONE);
+    }
+  }
+
+  // This Data struct workaround is to avoid initializing all the members
+  // in Data during construction of this object, since this object is always
+  // constructed, even when tracing is disabled. If the members of Data were
+  // members of this class instead, compiler warnings occur about potential
+  // uninitialized accesses.
+  struct Data {
+    const unsigned char* category_enabled;
+    const char* name;
+  };
+  Data* p_data_;
+  Data data_;
+};
+
+}  // namespace trace_event_internal
+}  // namespace webrtc
+
+#else
+
+////////////////////////////////////////////////////////////////////////////////
+// This section defines no-op alternatives to the tracing macros when
+// RTC_DISABLE_TRACE_EVENTS is defined.
+
+#define TRACE_DISABLED_BY_DEFAULT(name) "disabled-by-default-" name
+
+#define TRACE_ID_MANGLE(id) 0
+
+#define TRACE_EVENT0(category, name) RTC_NOOP()
+#define TRACE_EVENT1(category, name, arg1_name, arg1_val) RTC_NOOP()
+#define TRACE_EVENT2(category, name, arg1_name, arg1_val, arg2_name, arg2_val) \
+  RTC_NOOP()
+
+#define TRACE_EVENT_INSTANT0(category, name, scope) RTC_NOOP()
+#define TRACE_EVENT_INSTANT1(category, name, scope, arg1_name, arg1_val) \
+  RTC_NOOP()
+
+#define TRACE_EVENT_INSTANT2(category, name, scope, arg1_name, arg1_val, \
+                             arg2_name, arg2_val)                        \
+  RTC_NOOP()
+
+#define TRACE_EVENT_BEGIN0(category, name) RTC_NOOP()
+#define TRACE_EVENT_BEGIN1(category, name, arg1_name, arg1_val) RTC_NOOP()
+#define TRACE_EVENT_BEGIN2(category, name, arg1_name, arg1_val, arg2_name, \
+                           arg2_val)                                       \
+  RTC_NOOP()
+
+#define TRACE_EVENT_END0(category, name) RTC_NOOP()
+#define TRACE_EVENT_END1(category, name, arg1_name, arg1_val) RTC_NOOP()
+#define TRACE_EVENT_END2(category, name, arg1_name, arg1_val, arg2_name, \
+                         arg2_val)                                       \
+  RTC_NOOP()
+
+#define TRACE_COUNTER1(category, name, value) RTC_NOOP()
+
+#define TRACE_COUNTER2(category, name, value1_name, value1_val, value2_name, \
+                       value2_val)                                           \
+  RTC_NOOP()
+
+#define TRACE_COUNTER_ID1(category, name, id, value) RTC_NOOP()
+
+#define TRACE_COUNTER_ID2(category, name, id, value1_name, value1_val, \
+                          value2_name, value2_val)                     \
+  RTC_NOOP()
+
+#define TRACE_EVENT_ASYNC_BEGIN0(category, name, id) RTC_NOOP()
+#define TRACE_EVENT_ASYNC_BEGIN1(category, name, id, arg1_name, arg1_val) \
+  RTC_NOOP()
+#define TRACE_EVENT_ASYNC_BEGIN2(category, name, id, arg1_name, arg1_val, \
+                                 arg2_name, arg2_val)                     \
+  RTC_NOOP()
+
+#define TRACE_EVENT_ASYNC_STEP_INTO0(category, name, id, step) RTC_NOOP()
+#define TRACE_EVENT_ASYNC_STEP_INTO1(category, name, id, step, arg1_name, \
+                                     arg1_val)                            \
+  RTC_NOOP()
+
+#define TRACE_EVENT_ASYNC_END0(category, name, id) RTC_NOOP()
+#define TRACE_EVENT_ASYNC_END1(category, name, id, arg1_name, arg1_val) \
+  RTC_NOOP()
+#define TRACE_EVENT_ASYNC_END2(category, name, id, arg1_name, arg1_val, \
+                               arg2_name, arg2_val)                     \
+  RTC_NOOP()
+
+#define TRACE_EVENT_API_GET_CATEGORY_ENABLED ""
+
+#define TRACE_EVENT_API_ADD_TRACE_EVENT RTC_NOOP()
+
+#endif  // RTC_TRACE_EVENTS_ENABLED
+#endif  // RTC_USE_PERFETTO
+
+#endif  // RTC_BASE_TRACE_EVENT_H_

VocieProcess/system_wrappers/include/denormal_disabler.h

@@ -0,0 +1,56 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef SYSTEM_WRAPPERS_INCLUDE_DENORMAL_DISABLER_H_
+#define SYSTEM_WRAPPERS_INCLUDE_DENORMAL_DISABLER_H_
+
+#include "rtc_base/system/arch.h"
+
+namespace webrtc {
+
+// Activates the hardware (HW) way to flush denormals (see [1]) to zero as they
+// can very seriously impact performance. At destruction time restores the
+// denormals handling state read by the ctor; hence, supports nested calls.
+// Equals a no-op if the architecture is not x86 or ARM or if the compiler is
+// not CLANG.
+// [1] https://en.wikipedia.org/wiki/Denormal_number
+//
+// Example usage:
+//
+// void Foo() {
+//   DenormalDisabler d;
+//   ...
+// }
+class DenormalDisabler {
+ public:
+  // Ctor. If architecture and compiler are supported, stores the HW settings
+  // for denormals, disables denormals and sets `disabling_activated_` to true.
+  // Otherwise, only sets `disabling_activated_` to false.
+  DenormalDisabler();
+  // Ctor. Same as above, but also requires `enabled` to be true to disable
+  // denormals.
+  explicit DenormalDisabler(bool enabled);
+  DenormalDisabler(const DenormalDisabler&) = delete;
+  DenormalDisabler& operator=(const DenormalDisabler&) = delete;
+  // Dtor. If `disabling_activated_` is true, restores the denormals HW settings
+  // read by the ctor before denormals were disabled. Otherwise it's a no-op.
+  ~DenormalDisabler();
+
+  // Returns true if architecture and compiler are supported.
+  static bool IsSupported();
+
+ private:
+  const int status_word_;
+  const bool disabling_activated_;
+};
+
+}  // namespace webrtc
+
+#endif  // SYSTEM_WRAPPERS_INCLUDE_DENORMAL_DISABLER_H_