重构 audio_transcode 代码：

- 独立出 RtcMediaSource，并只对rtc开放 - 增加Rtc g711转码开关 - 修改说明
2026-06-27 04:22:20 +08:00 · 2022-11-29 20:38:59 +08:00
parent 2adc12c4ab
commit f4b2fd9c05
8 changed files with 281 additions and 135 deletions
--- a/webrtc/RtcMediaSource.cpp
+++ b/webrtc/RtcMediaSource.cpp
@@ -0,0 +1,199 @@
+#include "RtcMediaSource.h"
+#include "Common/config.h"
+#include "Codec/Transcode.h"
+#include "Extension/AAC.h"
+#include "Extension/Opus.h"
+#include "Extension/G711.h"
+// for RTC configure
+#include "WebRtcTransport.h"
+namespace mediakit {
+
+bool needTransToOpus(CodecId codec) {
+  GET_CONFIG(int, transG711, Rtc::kTranscodeG711);
+  switch (codec)
+  {
+  case CodecG711U:
+  case CodecG711A:
+      return transG711;
+  case CodecAAC:
+      return true;
+  default:
+      return false;
+  }
+}
+
+bool needTransToAac(CodecId codec) {
+    GET_CONFIG(int, transG711, Rtc::kTranscodeG711);
+    switch (codec)
+    {
+    case CodecG711U:
+    case CodecG711A:
+        return transG711;
+    case CodecOpus:
+        return true;
+    default:
+        return false;
+    }
+}
+
+RtcMediaSourceMuxer::RtcMediaSourceMuxer(const MediaTuple& tuple, const ProtocolOption &option, const TitleSdp::Ptr &title) 
+  : RtspMediaSourceMuxer(tuple, option, title, RTC_SCHEMA)
+{
+  if (_option.audio_transcode) {
+#ifndef ENABLE_FFMPEG
+      WarnL << "without ffmpeg, skip transcode setting";
+      _option.audio_transcode = false;
+#endif
+  }
+}
+
+RtspMediaSource::Ptr RtcMediaSourceImp::clone(const std::string &stream) {
+    auto tuple = _tuple;
+    tuple.stream = stream;
+    auto src_imp = std::make_shared<RtcMediaSourceImp>(tuple);
+    src_imp->setSdp(getSdp());
+    src_imp->setProtocolOption(getProtocolOption());
+    return src_imp;
+}
+
+bool RtcMediaSourceMuxer::inputFrame(const Frame::Ptr &frame)
+{
+  if (_clear_cache && _on_demand) {
+    _clear_cache = false;
+    _media_src->clearCache();
+  }
+  if (_enabled || !_on_demand) {
+#if defined(ENABLE_FFMPEG)
+    if (_option.audio_transcode && needTransToOpus(frame->getCodecId())) {
+      if (!_audio_dec) { // addTrack可能没调, 这边根据情况再调一次
+        Track::Ptr track;
+        switch (frame->getCodecId())
+        {
+        case CodecAAC:
+          if (frame->prefixSize()) {
+            std::string cfg = makeAacConfig((uint8_t *)(frame->data()), frame->prefixSize());
+            track = std::make_shared<AACTrack>(cfg);
+          }
+          else {
+            track = std::make_shared<AACTrack>(44100, 2);
+          }
+          break;
+        case CodecG711A:
+        case CodecG711U:
+          track.reset(new G711Track(frame->getCodecId()));
+          break;
+        default:
+          break;
+        }
+        if (track)
+          addTrack(track);
+        if (!_audio_dec) return false;
+      }
+      if (readerCount()) {
+        _audio_dec->inputFrame(frame, true, false);
+        if (!_count)
+          InfoL << "start transcode " << frame->getCodecName() << "," << frame->pts() << "->Opus";
+        _count++;
+      }
+      else if (_count) {
+        InfoL << "stop transcode with " << _count << " items";
+        _count = 0;
+      }
+      return true;
+    }
+#endif
+    return RtspMuxer::inputFrame(frame);
+  }
+  return false;
+}
+
+#if defined(ENABLE_FFMPEG)
+
+bool RtcMediaSourceMuxer::addTrack(const Track::Ptr & track)
+{
+  Track::Ptr newTrack = track;
+  if (_option.audio_transcode && needTransToOpus(track->getCodecId())) {
+    newTrack = std::make_shared<OpusTrack>();
+    GET_CONFIG(int, bitrate, General::kOpusBitrate);
+    newTrack->setBitRate(bitrate);
+    _audio_dec.reset(new FFmpegDecoder(track));
+    _audio_enc.reset(new FFmpegEncoder(newTrack));
+    // aac to opus
+    _audio_dec->setOnDecode([this](const FFmpegFrame::Ptr & frame) {
+      _audio_enc->inputFrame(frame, false);
+    });
+    _audio_enc->setOnEncode([this](const Frame::Ptr& frame) {
+        RtspMuxer::inputFrame(frame);
+    });
+  }
+  return RtspMuxer::addTrack(newTrack);
+}
+
+
+void RtcMediaSourceMuxer::resetTracks()
+{
+  RtspMuxer::resetTracks();
+  _audio_dec = nullptr;
+  _audio_enc = nullptr;
+  if (_count) {
+    InfoL << "stop transcode with " << _count << " items";
+    _count = 0;
+  }
+}
+
+bool RtcMediaSourceImp::addTrack(const Track::Ptr &track)
+{
+  if (_muxer) {
+    Track::Ptr newTrack = track;
+    if (_option.audio_transcode && needTransToAac(track->getCodecId())) {
+      newTrack.reset(new AACTrack(44100, 2));
+      GET_CONFIG(int, bitrate, General::kAacBitrate);
+      newTrack->setBitRate(bitrate);
+      _audio_dec.reset(new FFmpegDecoder(track));
+      _audio_enc.reset(new FFmpegEncoder(newTrack));
+      // hook data to newTack
+      track->addDelegate([this](const Frame::Ptr &frame) -> bool {
+        if (_all_track_ready && 0 == _muxer->totalReaderCount()) {
+          if (_count) {
+            InfoL << "stop transcode with " << _count << " items";
+            _count = 0;
+          }
+          return true;
+        }
+        if (_audio_dec) {
+          if (!_count)
+            InfoL << "start transcode " << frame->getCodecName() << "," << frame->pts() << "->AAC";
+          _count++;
+          _audio_dec->inputFrame(frame, true, false);
+        }
+        return true;
+      });
+      _audio_dec->setOnDecode([this](const FFmpegFrame::Ptr & frame) {
+        _audio_enc->inputFrame(frame, false);
+      });
+      _audio_enc->setOnEncode([newTrack](const Frame::Ptr& frame) {
+        newTrack->inputFrame(frame);
+      });
+    }
+
+    if (_muxer->addTrack(newTrack)) {
+      newTrack->addDelegate(_muxer);
+      return true;
+    }
+  }
+  return false;
+}
+
+void RtcMediaSourceImp::resetTracks()
+{
+  RtspMediaSourceImp::resetTracks();
+  _audio_dec = nullptr;
+  _audio_enc = nullptr;
+  if (_count) {
+    InfoL << "stop transcode with " << _count << " items";
+    _count = 0;
+  }
+}
+
+#endif
+}