mirror of
https://github.com/ZLMediaKit/ZLMediaKit.git
synced 2024-11-22 19:00:01 +08:00
增加FFmpegEncoder和test_audio_transcode, 主要修改有:
- 使用FFmpegAudioFifo来简化代码 - 修复转码时间戳不规则递增,导致的音频破音和不连续问题 - 使用double来保存_tsp,避免出现2.5ms的frame_size(ffmpeg内置的opus)
This commit is contained in:
parent
b33c227cf4
commit
5c92b4e48f
@ -13,10 +13,17 @@
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#include <dlfcn.h>
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#endif
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#include "Util/File.h"
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#include "Util/util.h"
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#include "Util/uv_errno.h"
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#include <float.h>
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#include "Transcode.h"
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#include "Extension/AAC.h"
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#include "Common/config.h"
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#include "Extension/Opus.h"
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#include "Extension/G711.h"
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#include "Extension/H264.h"
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#include "Extension/H265.h"
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#define MAX_DELAY_SECOND 3
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using namespace std;
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@ -66,6 +73,9 @@ static bool setupFFmpeg_l() {
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#if (LIBAVCODEC_VERSION_MAJOR < 58)
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avcodec_register_all();
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#endif
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InfoL << "libavcodec " << LIBAVCODEC_VERSION_MAJOR << "." << LIBAVCODEC_VERSION_MINOR;
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//InfoL << "libavformat " << LIBAVFORMAT_VERSION_MAJOR << "." << LIBAVFORMAT_VERSION_MINOR;
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InfoL << "libavutil " << LIBAVUTIL_VERSION_MAJOR << "." << LIBAVUTIL_VERSION_MAJOR;
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return true;
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}
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@ -411,21 +421,23 @@ FFmpegDecoder::FFmpegDecoder(const Track::Ptr &track, int thread_num, const std:
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_context->flags |= AV_CODEC_FLAG_LOW_DELAY;
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_context->flags2 |= AV_CODEC_FLAG2_FAST;
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if (track->getTrackType() == TrackVideo) {
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_context->width = static_pointer_cast<VideoTrack>(track)->getVideoWidth();
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_context->height = static_pointer_cast<VideoTrack>(track)->getVideoHeight();
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}
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auto video = static_pointer_cast<VideoTrack>(track);
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_context->width = video->getVideoWidth();
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_context->height = video->getVideoHeight();
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InfoL << "decode video " << video->getCodecName() << " " << _context->width << "x" << _context->height;
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} else {
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auto audio = static_pointer_cast<AudioTrack>(track);
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InfoL << "decode audio " << audio->getCodecName() << " " << audio->getAudioSampleRate() << "x" << audio->getAudioChannel();
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switch (track->getCodecId()) {
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case CodecG711A:
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case CodecG711U: {
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AudioTrack::Ptr audio = static_pointer_cast<AudioTrack>(track);
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_context->channels = audio->getAudioChannel();
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_context->sample_rate = audio->getAudioSampleRate();
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_context->channel_layout = av_get_default_channel_layout(_context->channels);
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break;
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}
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default:
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break;
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default: break;
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}
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}
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AVDictionary *dict = nullptr;
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if (thread_num <= 0) {
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@ -491,10 +503,6 @@ void FFmpegDecoder::flush() {
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}
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}
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const AVCodecContext *FFmpegDecoder::getContext() const {
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return _context.get();
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}
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bool FFmpegDecoder::inputFrame_l(const Frame::Ptr &frame, bool live, bool enable_merge) {
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if (_do_merger && enable_merge) {
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return _merger.inputFrame(frame, [this, live](uint64_t dts, uint64_t pts, const Buffer::Ptr &buffer, bool have_idr) {
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@ -575,6 +583,82 @@ void FFmpegDecoder::onDecode(const FFmpegFrame::Ptr &frame) {
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////////////////////////////////////////////////////////////////////////////////////////////////////////////
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FFmpegAudioFifo::~FFmpegAudioFifo() {
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if (_fifo) {
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av_audio_fifo_free(_fifo);
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_fifo = nullptr;
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}
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}
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int FFmpegAudioFifo::size() const {
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return _fifo ? av_audio_fifo_size(_fifo) : 0;
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}
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bool FFmpegAudioFifo::Write(const AVFrame *frame) {
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_format = (AVSampleFormat)frame->format;
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if (!_fifo) {
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_fifo = av_audio_fifo_alloc(_format, frame->channels, frame->nb_samples);
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if (!_fifo) {
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WarnL << "av_audio_fifo_alloc " << frame->channels << "x" << frame->nb_samples << "error";
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return false;
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}
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}
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_channels = frame->channels;
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if (_samplerate != frame->sample_rate) {
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_samplerate = frame->sample_rate;
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// 假定传入frame的时间戳是以ms为单位的
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_timebase = 1000.0 / _samplerate;
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}
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if (frame->pts != AV_NOPTS_VALUE) {
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// 计算fifo audio第一个采样的时间戳
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double tsp = frame->pts - _timebase * av_audio_fifo_size(_fifo);
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// flv.js和webrtc对音频时间戳增量有要求, rtc要求更加严格!
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// 得尽量保证时间戳是按照sample_size累加,否则容易出现破音或杂音等问题
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if (fabs(_tsp) < DBL_EPSILON || fabs(tsp - _tsp) > 200) {
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InfoL << "reset base_tsp " << (int64_t)_tsp << "->" << (int64_t)tsp;
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_tsp = tsp;
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}
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} else {
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_tsp = 0;
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}
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av_audio_fifo_write(_fifo, (void **)frame->data, frame->nb_samples);
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return true;
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}
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bool FFmpegAudioFifo::Read(AVFrame *frame, int sample_size) {
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assert(_fifo);
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int fifo_size = av_audio_fifo_size(_fifo);
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if (fifo_size < sample_size)
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return false;
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// fill linedata
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av_samples_get_buffer_size(frame->linesize, _channels, sample_size, _format, 0);
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frame->nb_samples = sample_size;
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frame->format = _format;
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frame->channel_layout = av_get_default_channel_layout(_channels);
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frame->sample_rate = _samplerate;
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if (fabs(_tsp) > DBL_EPSILON) {
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frame->pts = _tsp;
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// advance tsp by sample_size
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_tsp += sample_size * _timebase;
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}
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else {
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frame->pts = AV_NOPTS_VALUE;
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}
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int ret = av_frame_get_buffer(frame, 0);
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if (ret < 0) {
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WarnL << "av_frame_get_buffer error " << ffmpeg_err(ret);
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return false;
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}
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av_audio_fifo_read(_fifo, (void **)frame->data, sample_size);
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return true;
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}
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////////////////////////////////////////////////////////////////////////////////////////////////////////////
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FFmpegSwr::FFmpegSwr(AVSampleFormat output, int channel, int channel_layout, int samplerate) {
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_target_format = output;
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_target_channels = channel;
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@ -694,5 +778,334 @@ FFmpegFrame::Ptr FFmpegSws::inputFrame(const FFmpegFrame::Ptr &frame, int &ret,
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return nullptr;
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}
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///////////////////////////////////////////////////////////////////////////////////////////////////////
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void setupContext(AVCodecContext *_context, int bitrate) {
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//保存AVFrame的引用
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#ifdef FF_API_OLD_ENCDEC
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_context->refcounted_frames = 1;
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#endif
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_context->flags |= AV_CODEC_FLAG_LOW_DELAY;
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_context->flags2 |= AV_CODEC_FLAG2_FAST;
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_context->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
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_context->time_base.num = 1;
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_context->time_base.den = 1000; // {1, 1000}
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_context->bit_rate = bitrate;
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}
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FFmpegEncoder::FFmpegEncoder(const Track::Ptr &track, int thread_num) {
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setupFFmpeg();
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const AVCodec *codec = nullptr;
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const AVCodec *codec_default = nullptr;
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_codecId = track->getCodecId();
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switch (_codecId) {
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case CodecH264:
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codec_default = getCodec<false>({ AV_CODEC_ID_H264 });
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if (checkIfSupportedNvidia()) {
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codec = getCodec<false>({ { "libopenh264" },
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{ AV_CODEC_ID_H264 },
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{ "h264_qsv" },
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{ "h264_videotoolbox" },
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{ "h264_cuvid" },
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{ "h264_nvmpi" } });
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} else {
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codec = getCodec<false>({ { "libopenh264" }, { AV_CODEC_ID_H264 }, { "h264_qsv" }, { "h264_videotoolbox" }, { "h264_nvmpi" } });
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}
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break;
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case CodecH265:
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codec_default = getCodec<false>({ AV_CODEC_ID_HEVC });
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if (checkIfSupportedNvidia()) {
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codec = getCodec<false>({ { AV_CODEC_ID_HEVC }, { "hevc_qsv" }, { "hevc_videotoolbox" }, { "hevc_cuvid" }, { "hevc_nvmpi" } });
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} else {
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codec = getCodec<false>({ { AV_CODEC_ID_HEVC }, { "hevc_qsv" }, { "hevc_videotoolbox" }, { "hevc_nvmpi" } });
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}
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break;
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case CodecAAC:
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codec = getCodec<false>({ AV_CODEC_ID_AAC });
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break;
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case CodecG711A:
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codec = getCodec<false>({ AV_CODEC_ID_PCM_ALAW });
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break;
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case CodecG711U:
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codec = getCodec<false>({ AV_CODEC_ID_PCM_MULAW });
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break;
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case CodecOpus:
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codec = getCodec<false>({ AV_CODEC_ID_OPUS });
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break;
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case CodecVP8:
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codec = getCodec<false>({ AV_CODEC_ID_VP8 });
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break;
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case CodecVP9:
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codec = getCodec<false>({ AV_CODEC_ID_VP9 });
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break;
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default:
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break;
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}
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if (!codec) {
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throw std::runtime_error("未找到编码器");
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}
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if (thread_num <= 0) {
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av_dict_set(&_dict, "threads", "auto", 0);
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} else {
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av_dict_set(&_dict, "threads", to_string(MIN(thread_num, thread::hardware_concurrency())).data(), 0);
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}
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av_dict_set(&_dict, "zerolatency", "1", 0);
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if (strcmp(codec->name, "libx264") == 0 || strcmp(codec->name, "libx265") == 0) {
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av_dict_set(&_dict, "preset", "ultrafast", 0);
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}
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while (true) {
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bool ret = false;
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if (getTrackType() == TrackVideo) {
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// 不设置时,仅第一个I帧前存一次sps和pps
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// 设置后,I帧钱不会存储sps和pps, 但_context->extradata会有数据,需要手动处理
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// _context->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
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VideoTrack::Ptr video = static_pointer_cast<VideoTrack>(track);
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ret = openVideoCodec(video->getVideoWidth(), video->getVideoHeight(), track->getBitRate(), codec);
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} else {
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AudioTrack::Ptr audio = static_pointer_cast<AudioTrack>(track);
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ret = openAudioCodec(audio->getAudioSampleRate(), audio->getAudioChannel(), track->getBitRate(), codec);
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}
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if (ret) {
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_codec = codec;
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//成功
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InfoL << "打开编码器成功:" << codec->name << ", frameSize " << _context->frame_size;
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// we do not send complete frames, check this
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if (getTrackType() == TrackAudio) {
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var_frame_size = codec->capabilities & AV_CODEC_CAP_VARIABLE_FRAME_SIZE;
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if (var_frame_size) {
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InfoL << codec->name << " support var frame_size";
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}
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}
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break;
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}
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if (codec_default && codec_default != codec) {
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//硬件编解码器打开失败,尝试软件的
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WarnL << "打开编码器" << codec->name << "失败,原因是:" << ffmpeg_err(ret) << ", 再尝试打开编码器"
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<< codec_default->name;
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codec = codec_default;
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continue;
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}
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throw std::runtime_error(StrPrinter << "打开编码器" << codec->name << "失败:" << ffmpeg_err(ret));
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}
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}
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FFmpegEncoder::~FFmpegEncoder() {
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stopThread(true);
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flush();
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av_dict_free(&_dict);
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}
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bool FFmpegEncoder::openVideoCodec(int width, int height, int bitrate, const AVCodec *codec) {
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_context.reset(avcodec_alloc_context3(codec), [](AVCodecContext *ctx) { avcodec_free_context(&ctx); });
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if (_context) {
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setupContext(_context.get(), bitrate);
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// 不设置时,仅第一个I帧前存一次sps和pps
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// 设置后,I帧钱不会存储sps和pps, 但_context->extradata会有数据,需要手动处理
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// _context->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
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_context->width = width;
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_context->height = height;
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// gop
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_context->gop_size = 200;
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// 禁用b帧
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_context->max_b_frames = 0;
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_context->has_b_frames = 0;
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InfoL << "openVideoCodec " << codec->name << " " << _context->width << "x" << _context->height;
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_context->pix_fmt = AV_PIX_FMT_YUV420P; // codec->pix_fmts[0];
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// sws_.reset(new FFmpegSws(_context->pix_fmt, _context->width, _context->height));
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return avcodec_open2(_context.get(), codec, &_dict) >= 0;
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}
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return false;
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}
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bool FFmpegEncoder::openAudioCodec(int samplerate, int channel, int bitrate, const AVCodec *codec) {
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_context.reset(avcodec_alloc_context3(codec), [](AVCodecContext *ctx) { avcodec_free_context(&ctx); });
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if (_context) {
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setupContext(_context.get(), bitrate);
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_context->sample_fmt = codec->sample_fmts[0];
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_context->sample_rate = samplerate;
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_context->channels = channel;
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_context->channel_layout = av_get_default_channel_layout(_context->channels);
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if (getCodecId() == CodecOpus)
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_context->compression_level = 1;
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//_sample_bytes = av_get_bytes_per_sample(_context->sample_fmt) * _context->channels;
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_swr.reset(
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new FFmpegSwr(_context->sample_fmt, _context->channels, _context->channel_layout, _context->sample_rate));
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InfoL << "openAudioCodec " << codec->name << " " << _context->sample_rate << "x" << _context->channels;
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return avcodec_open2(_context.get(), codec, &_dict) >= 0;
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}
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return false;
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}
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void FFmpegEncoder::flush() {
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while (true) {
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auto packet = alloc_av_packet();
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auto ret = avcodec_receive_packet(_context.get(), packet.get());
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if (ret == AVERROR(EAGAIN)) {
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avcodec_send_frame(_context.get(), nullptr);
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continue;
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}
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if (ret == AVERROR_EOF) {
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break;
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}
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if (ret < 0) {
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WarnL << "avcodec_receive_frame failed:" << ffmpeg_err(ret);
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break;
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}
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onEncode(packet.get());
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}
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}
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bool FFmpegEncoder::inputFrame(const FFmpegFrame::Ptr &frame, bool async) {
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if (async && !TaskManager::isEnabled() && getContext()->codec_type == AVMEDIA_TYPE_VIDEO) {
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//开启异步编码,且为视频,尝试启动异步解码线程
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startThread("encoder thread");
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}
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if (!async || !TaskManager::isEnabled()) {
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return inputFrame_l(frame);
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}
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return addEncodeTask([this, frame]() { inputFrame_l(frame); });
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}
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bool FFmpegEncoder::inputFrame_l(FFmpegFrame::Ptr input) {
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AVFrame *frame = input->get();
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AVCodecContext *context = _context.get();
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if (getTrackType() == TrackAudio) {
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if (_swr) {
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// 转成同样采样率和通道
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input = _swr->inputFrame(input);
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frame = input->get();
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// 保证每次塞给解码器的都是一帧音频
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if (!var_frame_size && _context->frame_size && frame->nb_samples != _context->frame_size) {
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// add this frame to _audio_buffer
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if (!_fifo)
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_fifo.reset(new FFmpegAudioFifo());
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// TraceL << "in " << frame->pts << ",samples " << frame->nb_samples;
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_fifo->Write(frame);
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while (1) {
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FFmpegFrame audio_frame;
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if (!_fifo->Read(audio_frame.get(), _context->frame_size)){
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break;
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}
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if (!encodeFrame(audio_frame.get())) {
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break;
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}
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}
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return true;
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}
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}
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} else {
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if (frame->format != context->pix_fmt || frame->width != context->width || frame->height != context->height) {
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if (_sws) {
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input = _sws->inputFrame(input);
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frame = input->get();
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} else {
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// @todo reopen videocodec?
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openVideoCodec(frame->width, frame->height, 512000, _codec);
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}
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}
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}
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return encodeFrame(frame);
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}
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bool FFmpegEncoder::encodeFrame(AVFrame *frame) {
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// TraceL << "enc " << frame->pts;
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int ret = avcodec_send_frame(_context.get(), frame);
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if (ret < 0) {
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WarnL << "Error sending a frame " << frame->pts << " to the encoder: " << ffmpeg_err(ret);
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return false;
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}
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while (ret >= 0) {
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auto packet = alloc_av_packet();
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ret = avcodec_receive_packet(_context.get(), packet.get());
|
||||
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
|
||||
break;
|
||||
else if (ret < 0) {
|
||||
WarnL << "Error encoding a frame: " << ffmpeg_err(ret);
|
||||
return false;
|
||||
}
|
||||
// TraceL << "out " << packet->pts << "," << packet->dts << ", size: " << packet->size;
|
||||
onEncode(packet.get());
|
||||
}
|
||||
return true;
|
||||
}
|
||||
|
||||
void FFmpegEncoder::onEncode(AVPacket *packet) {
|
||||
// process frame
|
||||
if (!_cb)
|
||||
return;
|
||||
switch (_codecId) {
|
||||
case CodecH264: {
|
||||
auto frame = FrameImp::create<H264Frame>();
|
||||
frame->_dts = packet->dts;
|
||||
frame->_pts = packet->pts;
|
||||
frame->_buffer.assign((const char *)packet->data, packet->size);
|
||||
frame->_prefix_size = prefixSize((const char *)packet->data, packet->size);
|
||||
_cb(frame);
|
||||
break;
|
||||
}
|
||||
case CodecH265: {
|
||||
auto frame = FrameImp::create<H265Frame>();
|
||||
frame->_dts = packet->dts;
|
||||
frame->_pts = packet->pts;
|
||||
frame->_buffer.assign((const char *)packet->data, packet->size);
|
||||
frame->_prefix_size = prefixSize((const char *)packet->data, packet->size);
|
||||
_cb(frame);
|
||||
break;
|
||||
}
|
||||
case CodecAAC: {
|
||||
auto frame = FrameImp::create<>();
|
||||
frame->_codec_id = _codecId;
|
||||
frame->_dts = packet->dts;
|
||||
frame->_pts = packet->pts;
|
||||
frame->_buffer.reserve(ADTS_HEADER_LEN + packet->size);
|
||||
if (_context && _context->extradata && _context->extradata_size) {
|
||||
uint8_t adts[ADTS_HEADER_LEN];
|
||||
auto cfg = std::string((const char *)_context->extradata, _context->extradata_size);
|
||||
dumpAacConfig(cfg, packet->size, adts, ADTS_HEADER_LEN);
|
||||
frame->_prefix_size = ADTS_HEADER_LEN;
|
||||
frame->_buffer.append((char*)adts, ADTS_HEADER_LEN);
|
||||
}
|
||||
frame->_buffer.append((const char *)packet->data, packet->size);
|
||||
_cb(frame);
|
||||
break;
|
||||
}
|
||||
case CodecOpus:
|
||||
case CodecG711A:
|
||||
case CodecG711U: {
|
||||
auto frame = FrameImp::create<>();
|
||||
frame->_codec_id = _codecId;
|
||||
frame->_dts = packet->dts;
|
||||
frame->_pts = packet->pts;
|
||||
frame->_buffer.assign((const char *)packet->data, packet->size);
|
||||
_cb(frame);
|
||||
break;
|
||||
}
|
||||
case CodecVP8:
|
||||
case CodecVP9: {
|
||||
auto frame = FrameImp::create<>();
|
||||
frame->_codec_id = _codecId;
|
||||
frame->_dts = packet->dts;
|
||||
frame->_pts = packet->pts;
|
||||
frame->_buffer.assign((const char *)packet->data, packet->size);
|
||||
_cb(frame);
|
||||
break;
|
||||
}
|
||||
default: break;
|
||||
}
|
||||
}
|
||||
|
||||
} //namespace mediakit
|
||||
#endif//ENABLE_FFMPEG
|
||||
|
@ -63,6 +63,24 @@ private:
|
||||
SwrContext *_ctx = nullptr;
|
||||
};
|
||||
|
||||
class FFmpegAudioFifo {
|
||||
public:
|
||||
FFmpegAudioFifo() = default;
|
||||
~FFmpegAudioFifo();
|
||||
|
||||
bool Write(const AVFrame *frame);
|
||||
bool Read(AVFrame *frame, int sample_size);
|
||||
int size() const;
|
||||
|
||||
private:
|
||||
int _channels = 0;
|
||||
int _samplerate = 0;
|
||||
double _tsp = 0;
|
||||
double _timebase = 0;
|
||||
AVAudioFifo *_fifo = nullptr;
|
||||
AVSampleFormat _format = AV_SAMPLE_FMT_NONE;
|
||||
};
|
||||
|
||||
class TaskManager {
|
||||
public:
|
||||
TaskManager() = default;
|
||||
@ -108,7 +126,7 @@ public:
|
||||
bool inputFrame(const Frame::Ptr &frame, bool live, bool async, bool enable_merge = true);
|
||||
void setOnDecode(onDec cb);
|
||||
void flush();
|
||||
const AVCodecContext *getContext() const;
|
||||
const AVCodecContext *getContext() const { return _context.get(); }
|
||||
|
||||
private:
|
||||
void onDecode(const FFmpegFrame::Ptr &frame);
|
||||
@ -145,6 +163,41 @@ private:
|
||||
AVPixelFormat _target_format = AV_PIX_FMT_NONE;
|
||||
};
|
||||
|
||||
class FFmpegEncoder : public TaskManager, public CodecInfo {
|
||||
public:
|
||||
using Ptr = std::shared_ptr<FFmpegEncoder>;
|
||||
using onEnc = std::function<void(const Frame::Ptr &)>;
|
||||
|
||||
FFmpegEncoder(const Track::Ptr &track, int thread_num = 2);
|
||||
~FFmpegEncoder() override;
|
||||
|
||||
void flush();
|
||||
CodecId getCodecId() const override { return _codecId; }
|
||||
const AVCodecContext *getContext() const { return _context.get(); }
|
||||
|
||||
void setOnEncode(onEnc cb) { _cb = std::move(cb); }
|
||||
bool inputFrame(const FFmpegFrame::Ptr &frame, bool async);
|
||||
|
||||
private:
|
||||
bool inputFrame_l(FFmpegFrame::Ptr frame);
|
||||
bool encodeFrame(AVFrame *frame);
|
||||
void onEncode(AVPacket *packet);
|
||||
bool openVideoCodec(int width, int height, int bitrate, const AVCodec *codec);
|
||||
bool openAudioCodec(int samplerate, int channel, int bitrate, const AVCodec *codec);
|
||||
|
||||
private:
|
||||
onEnc _cb;
|
||||
CodecId _codecId;
|
||||
const AVCodec *_codec = nullptr;
|
||||
AVDictionary *_dict = nullptr;
|
||||
std::shared_ptr<AVCodecContext> _context;
|
||||
|
||||
std::unique_ptr<FFmpegSws> _sws;
|
||||
std::unique_ptr<FFmpegSwr> _swr;
|
||||
std::unique_ptr<FFmpegAudioFifo> _fifo;
|
||||
bool var_frame_size = false;
|
||||
};
|
||||
|
||||
}//namespace mediakit
|
||||
#endif// ENABLE_FFMPEG
|
||||
#endif //ZLMEDIAKIT_TRANSCODE_H
|
||||
|
@ -33,7 +33,14 @@ foreach(TEST_SRC ${TEST_SRC_LIST})
|
||||
endif()
|
||||
endif()
|
||||
|
||||
message(STATUS "add test: ${TEST_EXE_NAME}")
|
||||
if (NOT ENABLE_FFMPEG)
|
||||
# 暂时过滤掉依赖 WebRTC 的测试模块
|
||||
if ("${TEST_EXE_NAME}" MATCHES "test_audio_transcode")
|
||||
continue()
|
||||
endif ()
|
||||
endif ()
|
||||
|
||||
message(STATUS "add test:${TEST_EXE_NAME}")
|
||||
add_executable(${TEST_EXE_NAME} ${TEST_SRC})
|
||||
target_compile_options(${TEST_EXE_NAME}
|
||||
PRIVATE ${COMPILE_OPTIONS_DEFAULT})
|
||||
|
118
tests/test_audio_transcode.cpp
Normal file
118
tests/test_audio_transcode.cpp
Normal file
@ -0,0 +1,118 @@
|
||||
#include "Codec/Transcode.h"
|
||||
#include "Record/MP4Demuxer.h"
|
||||
#include "Record/MP4Muxer.h"
|
||||
#include "Extension/AAC.h"
|
||||
#include "Extension/Opus.h"
|
||||
#include "Extension/G711.h"
|
||||
#include "Util/logger.h"
|
||||
using namespace mediakit;
|
||||
struct TransCtx {
|
||||
using Ptr = std::shared_ptr<TransCtx>;
|
||||
TransCtx(const char *prefix, CodecId codec) {
|
||||
char path[256];
|
||||
Track::Ptr track;
|
||||
switch (codec) {
|
||||
case CodecAAC:
|
||||
track.reset(new AACTrack(44100, 1));
|
||||
sprintf(path, "%s_aac.mp4", prefix);
|
||||
break;
|
||||
case CodecOpus:
|
||||
track.reset(new OpusTrack());
|
||||
sprintf(path, "%s_opus.mp4", prefix);
|
||||
break;
|
||||
case CodecG711A:
|
||||
case CodecG711U:
|
||||
track.reset(new G711Track(codec, 8000, 1, 16));
|
||||
sprintf(path, "%s_711%c.mp4", prefix, codec == CodecG711A ? 'A' : 'U');
|
||||
break;
|
||||
default:
|
||||
return;
|
||||
break;
|
||||
}
|
||||
file.reset(new MP4Muxer());
|
||||
file->openMP4(path);
|
||||
file->addTrack(track);
|
||||
enc.reset(new FFmpegEncoder(track));
|
||||
enc->setOnEncode([this](const Frame::Ptr &frame) { file->inputFrame(frame); });
|
||||
}
|
||||
~TransCtx() {
|
||||
enc = nullptr;
|
||||
file = nullptr;
|
||||
}
|
||||
void inputFrame(const FFmpegFrame::Ptr &frame) { enc->inputFrame(frame, false); }
|
||||
FFmpegEncoder::Ptr enc;
|
||||
std::shared_ptr<MP4Muxer> file;
|
||||
};
|
||||
|
||||
int TranscodeAudio(const char *srcPath, const char *dstPath) {
|
||||
MP4Demuxer srcMp4;
|
||||
srcMp4.openMP4(srcPath);
|
||||
|
||||
auto srcTrack = srcMp4.getTrack(TrackAudio);
|
||||
if (!srcTrack) {
|
||||
printf("unable to find audioTrack %s\n", srcPath);
|
||||
return -1;
|
||||
}
|
||||
std::vector<TransCtx::Ptr> trans;
|
||||
FFmpegDecoder audioDec(srcTrack);
|
||||
auto dstCodec = getCodecId(dstPath);
|
||||
if (!strcasecmp(dstPath, "aac"))
|
||||
dstCodec = CodecAAC;
|
||||
if (dstCodec != CodecInvalid) {
|
||||
std::string dstFile(srcPath);
|
||||
auto pos = dstFile.rfind('_');
|
||||
if (pos == dstFile.npos)
|
||||
pos = dstFile.rfind('.');
|
||||
if (pos != dstFile.npos)
|
||||
dstFile = dstFile.substr(0, pos);
|
||||
if (dstCodec == srcTrack->getCodecId()) {
|
||||
printf("same codec %s, skip transcode\n", dstPath);
|
||||
return 0;
|
||||
}
|
||||
trans.push_back(std::make_shared<TransCtx>(dstFile.c_str(), dstCodec));
|
||||
}
|
||||
else {
|
||||
for (auto codec : { CodecAAC, CodecOpus, CodecG711A, CodecG711U }) {
|
||||
if (codec == srcTrack->getCodecId())
|
||||
continue;
|
||||
trans.push_back(std::make_shared<TransCtx>(dstPath, codec));
|
||||
}
|
||||
}
|
||||
// srcTrack -> audioDec
|
||||
srcTrack->addDelegate([&](const Frame::Ptr &frame) -> bool {
|
||||
audioDec.inputFrame(frame, true, false, false);
|
||||
return true;
|
||||
});
|
||||
// audioDec -> audioEnc
|
||||
audioDec.setOnDecode([&](const FFmpegFrame::Ptr &frame) {
|
||||
for (TransCtx::Ptr p : trans)
|
||||
p->inputFrame(frame);
|
||||
});
|
||||
toolkit::Ticker tick;
|
||||
printf("startReadMp4 %" PRIu64 "ms\n", srcMp4.getDurationMS());
|
||||
bool key, eof;
|
||||
Frame::Ptr frame;
|
||||
while (true) {
|
||||
// srcMp4->srcTrack
|
||||
frame = srcMp4.readFrame(key, eof);
|
||||
if (eof) {
|
||||
printf("eof break loop, it tooks %" PRIu64 " ms\n", tick.elapsedTime());
|
||||
break;
|
||||
}
|
||||
}
|
||||
return 0;
|
||||
}
|
||||
|
||||
int main(int argc, char *argv[]) {
|
||||
if (argc != 3) {
|
||||
printf("usage src.mp4 dst_prefix/codecName\n");
|
||||
return 0;
|
||||
}
|
||||
toolkit::Logger::Instance().add(std::make_shared<toolkit::ConsoleChannel>());
|
||||
try {
|
||||
return TranscodeAudio(argv[1], argv[2]);
|
||||
} catch (std::exception e) {
|
||||
printf("exception: %s\n", e.what());
|
||||
return -1;
|
||||
}
|
||||
}
|
Loading…
Reference in New Issue
Block a user