add ns code.
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2bed1dacf2
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@ -6,6 +6,7 @@
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#include "api/audio/echo_canceller3_factory.h"
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#include "main.h"
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#include "modules/audio_processing/aec3/echo_canceller3.h"
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#include "modules/audio_processing/ns/noise_suppressor.h"
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#include <boost/asio/post.hpp>
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#include <sstream>
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@ -19,6 +20,10 @@ public:
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linearOutputBuffer = std::make_unique<webrtc::AudioBuffer>(sampleRate, channels, sampleRate, channels, sampleRate, channels);
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// RKAP_3A_Init(&m_vqe, AEC_TX_TYPE);
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webrtc::NsConfig config;
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config.target_level = webrtc::NsConfig::SuppressionLevel::k12dB;
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noiseSuppressor = std::make_unique<webrtc::NoiseSuppressor>(config, sampleRate, channels);
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noiseSuppressorBuffer = std::make_unique<webrtc::AudioBuffer>(sampleRate, channels, sampleRate, channels, sampleRate, channels);
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}
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std::unique_ptr<webrtc::EchoControl> echoCanceller;
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@ -26,6 +31,9 @@ public:
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std::unique_ptr<webrtc::AudioBuffer> farendBuffer;
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std::unique_ptr<webrtc::AudioBuffer> linearOutputBuffer;
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std::unique_ptr<webrtc::NoiseSuppressor> noiseSuppressor;
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std::unique_ptr<webrtc::AudioBuffer> noiseSuppressorBuffer;
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// RKAP_AEC_State m_vqe;
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};
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@ -54,6 +62,10 @@ void ProcessFileTask::run() {
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m_outFilename = oss.str();
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m_ofs = std::make_shared<std::ofstream>(m_outFilename, std::ofstream::binary);
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oss.str("");
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oss << DumpPath << "/ns_" << dspToString(m_dsp) << "_16k.pcm";
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m_nsOfs = std::make_shared<std::ofstream>(oss.str(), std::ofstream::binary);
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// m_speakerIfs = std::make_shared<std::ifstream>("/sdcard/speaker_8k.pcm", std::ifstream::binary);
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// m_micIfs = std::make_shared<std::ifstream>("/sdcard/micin_8k.pcm", std::ifstream::binary);
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// m_ofs = std::make_shared<std::ofstream>("/sdcard/out_8k.pcm", std::ofstream::binary);
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@ -75,23 +87,32 @@ void ProcessFileTask::process() {
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if (m_speakerIfs && m_micIfs && *m_speakerIfs && *m_micIfs) {
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char farendBuffer[sizeof(int16_t) * 16000 / 1000 * 10] = {0};
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char nearendBuffer[sizeof(int16_t) * 16000 / 1000 * 10] = {0};
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char noiseSuppressorOutBuffer[sizeof(int16_t) * 16000 / 1000 * 10] = {0};
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char outBuffer[sizeof(int16_t) * 16000 / 1000 * 10] = {0};
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webrtc::StreamConfig config(16000, 1); // 单声道
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// char farendBuffer[sizeof(int16_t) * 8000 / 1000 * 10] = {0};
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// char nearendBuffer[sizeof(int16_t) * 8000 / 1000 * 10] = {0};
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// char outBuffer[sizeof(int16_t) * 8000 / 1000 * 10] = {0};
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m_speakerIfs->read(farendBuffer, sizeof(farendBuffer));
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m_micIfs->read(nearendBuffer, sizeof(nearendBuffer));
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m_d->nearendBuffer->CopyFrom(reinterpret_cast<const int16_t *>(nearendBuffer), config);
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if (m_dsp == Speex) {
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m_speex->echoPlayback(reinterpret_cast<const int16_t *>(farendBuffer));
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m_speex->echoCapture(reinterpret_cast<const int16_t *>(nearendBuffer), reinterpret_cast<int16_t *>(outBuffer));
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} else if (m_dsp == AecMobile) {
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m_d->noiseSuppressor->Analyze(*m_d->nearendBuffer);
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m_d->noiseSuppressor->Process(m_d->nearendBuffer.get());
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m_d->nearendBuffer->CopyTo(config, reinterpret_cast<int16_t *>(noiseSuppressorOutBuffer));
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m_nsOfs->write(noiseSuppressorOutBuffer, sizeof(noiseSuppressorOutBuffer));
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// LOG(info) << " " << m_d->noiseSuppressorBuffer->num_frames() << " " << m_d->nearendBuffer->num_frames();
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m_webRtcAecm->echoPlayback(reinterpret_cast<const int16_t *>(farendBuffer), sizeof(farendBuffer) / 2);
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m_webRtcAecm->echoCancellation(reinterpret_cast<int16_t *>(nearendBuffer), nullptr, reinterpret_cast<int16_t *>(outBuffer),
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m_webRtcAecm->echoCancellation(reinterpret_cast<int16_t *>(nearendBuffer),
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reinterpret_cast<int16_t *>(noiseSuppressorOutBuffer), reinterpret_cast<int16_t *>(outBuffer),
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sizeof(farendBuffer) / 2);
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} else if (m_dsp == Aec3) {
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webrtc::StreamConfig config(16000, 1); // 单声道
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m_d->nearendBuffer->CopyFrom(reinterpret_cast<const int16_t *>(nearendBuffer), config);
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m_d->farendBuffer->CopyFrom(reinterpret_cast<const int16_t *>(farendBuffer), config);
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@ -96,6 +96,7 @@ private:
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std::shared_ptr<std::ifstream> m_speakerIfs;
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std::shared_ptr<std::ifstream> m_micIfs;
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std::shared_ptr<std::ofstream> m_ofs;
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std::shared_ptr<std::ofstream> m_nsOfs; // 降噪后的文件
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std::string m_outFilename;
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std::chrono::system_clock::time_point m_begin;
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};
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