Moves channel-dependent audio input processing to separate encoder task queue

webrtc/voice_engine/channel.cc

Index: webrtc/voice_engine/channel.cc
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index 2791f7f8b4ed555ed9da3e8d3f45ab6b9fa18171..8a80dffda3e77f93164336f00b000e9418b81297 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -20,6 +20,7 @@
 #include "webrtc/base/format_macros.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/rate_limiter.h"
+#include "webrtc/base/task_queue.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/config.h"
@@ -386,12 +387,31 @@ class VoERtcpObserver : public RtcpBandwidthObserver {
   std::map<uint32_t, uint32_t> extended_max_sequence_number_;
 };
 
+class Channel::ProcessAndEncodeAudioTask : public rtc::QueuedTask {
+ public:
+  ProcessAndEncodeAudioTask(std::unique_ptr<AudioFrame> audio_input,
+                            Channel* channel)
+      : audio_input_(std::move(audio_input)), channel_(channel) {}
+
+ private:
+  bool Run() override {
+    RTC_DCHECK_RUN_ON(channel_->encoder_queue_);
+    channel_->ProcessAndEncodeAudioOnTaskQueue(std::move(audio_input_));
+    return true;
+  }
+
+  std::unique_ptr<AudioFrame> audio_input_;
+  Channel* const channel_;
+};
+
 int32_t Channel::SendData(FrameType frameType,
                           uint8_t payloadType,
                           uint32_t timeStamp,
                           const uint8_t* payloadData,
                           size_t payloadSize,
                           const RTPFragmentationHeader* fragmentation) {
+  RTC_DCHECK_RUN_ON(encoder_queue_);
+  // LOG(INFO) << "SendData: " << payloadSize;
   WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::SendData(frameType=%u, payloadType=%u, timeStamp=%u,"
                " payloadSize=%" PRIuS ", fragmentation=0x%x)",
@@ -421,7 +441,7 @@ int32_t Channel::SendData(FrameType frameType,
 
   _lastLocalTimeStamp = timeStamp;
   _lastPayloadType = payloadType;
-
+  // LOG(INFO) << "__SendData";
   return 0;
 }
 
@@ -910,6 +930,7 @@ Channel::Channel(int32_t channelId,
       _audioDeviceModulePtr(NULL),
       _voiceEngineObserverPtr(NULL),
       _callbackCritSectPtr(NULL),
+      encoder_queue_(nullptr),
       _transportPtr(NULL),
       _sendFrameType(0),
       _externalMixing(false),
@@ -1131,9 +1152,11 @@ int32_t Channel::SetEngineInformation(Statistics& engineStatistics,
                                       ProcessThread& moduleProcessThread,
                                       AudioDeviceModule& audioDeviceModule,
                                       VoiceEngineObserver* voiceEngineObserver,
-                                      rtc::CriticalSection* callbackCritSect) {
+                                      rtc::CriticalSection* callbackCritSect,
+                                      rtc::TaskQueue* encoder_queue) {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::SetEngineInformation()");
+  RTC_DCHECK(encoder_queue);
   _engineStatisticsPtr = &engineStatistics;
   _outputMixerPtr = &outputMixer;
   _transmitMixerPtr = &transmitMixer,
@@ -1141,11 +1164,7 @@ int32_t Channel::SetEngineInformation(Statistics& engineStatistics,
   _audioDeviceModulePtr = &audioDeviceModule;
   _voiceEngineObserverPtr = voiceEngineObserver;
   _callbackCritSectPtr = callbackCritSect;
-  return 0;
-}
-
-int32_t Channel::UpdateLocalTimeStamp() {
-  _timeStamp += static_cast<uint32_t>(_audioFrame.samples_per_channel_);
+  encoder_queue_ = encoder_queue;
   return 0;
 }
 
@@ -2763,100 +2782,97 @@ int Channel::ResendPackets(const uint16_t* sequence_numbers, int length) {
   return _rtpRtcpModule->SendNACK(sequence_numbers, length);
 }
 
-uint32_t Channel::Demultiplex(const AudioFrame& audioFrame) {
-  WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
-               "Channel::Demultiplex()");
-  _audioFrame.CopyFrom(audioFrame);
-  _audioFrame.id_ = _channelId;
-  return 0;
+void Channel::ProcessAndEncodeAudio(const AudioFrame& audio_input) {
+  RTC_DCHECK(encoder_queue_);
+  std::unique_ptr<AudioFrame> audio_source(new AudioFrame());
+  audio_source->CopyFrom(audio_input);
+  audio_source->id_ = _channelId;
+  encoder_queue_->PostTask(std::unique_ptr<rtc::QueuedTask>(
+      new ProcessAndEncodeAudioTask(std::move(audio_source), this)));
 }
 
-void Channel::Demultiplex(const int16_t* audio_data,
-                          int sample_rate,
-                          size_t number_of_frames,
-                          size_t number_of_channels) {
+void Channel::ProcessAndEncodeAudio(const int16_t* audio_data,
+                                    int sample_rate,
+                                    size_t number_of_frames,
+                                    size_t number_of_channels) {
+  RTC_DCHECK(encoder_queue_);
   CodecInst codec;
   GetSendCodec(codec);
-
-  // Never upsample or upmix the capture signal here. This should be done at the
-  // end of the send chain.
-  _audioFrame.sample_rate_hz_ = std::min(codec.plfreq, sample_rate);
-  _audioFrame.num_channels_ = std::min(number_of_channels, codec.channels);
+  std::unique_ptr<AudioFrame> audio_source(new AudioFrame());
+  audio_source->sample_rate_hz_ = std::min(codec.plfreq, sample_rate);
+  audio_source->num_channels_ = std::min(number_of_channels, codec.channels);
   RemixAndResample(audio_data, number_of_frames, number_of_channels,
-                   sample_rate, &input_resampler_, &_audioFrame);
+                   sample_rate, &input_resampler_, audio_source.get());
+  audio_source->id_ = _channelId;
+  encoder_queue_->PostTask(std::unique_ptr<rtc::QueuedTask>(
+      new ProcessAndEncodeAudioTask(std::move(audio_source), this)));
 }
 
-uint32_t Channel::PrepareEncodeAndSend(int mixingFrequency) {
-  WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
-               "Channel::PrepareEncodeAndSend()");
+void Channel::ProcessAndEncodeAudioOnTaskQueue(
+    std::unique_ptr<AudioFrame> audio_input) {
+  RTC_DCHECK_RUN_ON(encoder_queue_);
+  PrepareEncodeAndSend(audio_input.get());
+  EncodeAndSend(audio_input.get());
+}
 
-  if (_audioFrame.samples_per_channel_ == 0) {
-    WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
-                 "Channel::PrepareEncodeAndSend() invalid audio frame");
-    return 0xFFFFFFFF;
-  }
+uint32_t Channel::PrepareEncodeAndSend(AudioFrame* audio_input) {
+  RTC_DCHECK_RUN_ON(encoder_queue_);
+  RTC_DCHECK(audio_input->samples_per_channel_);
 
   if (channel_state_.Get().input_file_playing) {
-    MixOrReplaceAudioWithFile(mixingFrequency);
+    MixOrReplaceAudioWithFile(audio_input);
   }
 
   bool is_muted = InputMute();  // Cache locally as InputMute() takes a lock.
-  AudioFrameOperations::Mute(&_audioFrame, previous_frame_muted_, is_muted);
+  AudioFrameOperations::Mute(audio_input, previous_frame_muted_, is_muted);
 
   if (channel_state_.Get().input_external_media) {
     rtc::CritScope cs(&_callbackCritSect);
-    const bool isStereo = (_audioFrame.num_channels_ == 2);
+    const bool isStereo = (audio_input->num_channels_ == 2);
     if (_inputExternalMediaCallbackPtr) {
       _inputExternalMediaCallbackPtr->Process(
-          _channelId, kRecordingPerChannel, (int16_t*)_audioFrame.data_,
-          _audioFrame.samples_per_channel_, _audioFrame.sample_rate_hz_,
+          _channelId, kRecordingPerChannel, (int16_t*)audio_input->data_,
+          audio_input->samples_per_channel_, audio_input->sample_rate_hz_,
           isStereo);
     }
   }
 
   if (_includeAudioLevelIndication) {
     size_t length =
-        _audioFrame.samples_per_channel_ * _audioFrame.num_channels_;
-    RTC_CHECK_LE(length, sizeof(_audioFrame.data_));
+        audio_input->samples_per_channel_ * audio_input->num_channels_;
+    RTC_CHECK_LE(length, sizeof(audio_input->data_));
     if (is_muted && previous_frame_muted_) {
       rms_level_.AnalyzeMuted(length);
     } else {
       rms_level_.Analyze(
-          rtc::ArrayView<const int16_t>(_audioFrame.data_, length));
+          rtc::ArrayView<const int16_t>(audio_input->data_, length));
     }
   }
   previous_frame_muted_ = is_muted;
-
   return 0;
 }
 
-uint32_t Channel::EncodeAndSend() {
-  WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
-               "Channel::EncodeAndSend()");
+uint32_t Channel::EncodeAndSend(AudioFrame* audio_input) {
+  RTC_DCHECK_RUN_ON(encoder_queue_);
+  RTC_DCHECK_LE(audio_input->num_channels_, 2);
+  RTC_DCHECK(audio_input->samples_per_channel_);
 
-  assert(_audioFrame.num_channels_ <= 2);
-  if (_audioFrame.samples_per_channel_ == 0) {
-    WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
-                 "Channel::EncodeAndSend() invalid audio frame");
-    return 0xFFFFFFFF;
-  }
-
-  _audioFrame.id_ = _channelId;
+  audio_input->id_ = _channelId;
 
   // --- Add 10ms of raw (PCM) audio data to the encoder @ 32kHz.
 
   // The ACM resamples internally.
-  _audioFrame.timestamp_ = _timeStamp;
+  audio_input->timestamp_ = _timeStamp;
   // This call will trigger AudioPacketizationCallback::SendData if encoding
   // is done and payload is ready for packetization and transmission.
   // Otherwise, it will return without invoking the callback.
-  if (audio_coding_->Add10MsData((AudioFrame&)_audioFrame) < 0) {
-    WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, _channelId),
-                 "Channel::EncodeAndSend() ACM encoding failed");
+  if (audio_coding_->Add10MsData(*audio_input) < 0) {
+    LOG(LS_ERROR) << "ACM::Add10MsData() failed for channel " << _channelId;
     return 0xFFFFFFFF;
   }
 
-  _timeStamp += static_cast<uint32_t>(_audioFrame.samples_per_channel_);
+  _timeStamp += static_cast<uint32_t>(audio_input->samples_per_channel_);
+  // LOG(INFO) << "___EncodeAndSend";
   return 0;
 }
 
@@ -3079,46 +3095,44 @@ int Channel::GetRtpRtcp(RtpRtcp** rtpRtcpModule,
 
 // TODO(andrew): refactor Mix functions here and in transmit_mixer.cc to use
 // a shared helper.
-int32_t Channel::MixOrReplaceAudioWithFile(int mixingFrequency) {
+int32_t Channel::MixOrReplaceAudioWithFile(AudioFrame* audio_input) {
+  RTC_DCHECK_RUN_ON(encoder_queue_);
   std::unique_ptr<int16_t[]> fileBuffer(new int16_t[640]);
   size_t fileSamples(0);
+  const int mixingFrequency = audio_input->sample_rate_hz_;
 
-  {
-    rtc::CritScope cs(&_fileCritSect);
-
-    if (!input_file_player_) {
-      WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
-                   "Channel::MixOrReplaceAudioWithFile() fileplayer"
-                   " doesnt exist");
-      return -1;
-    }
+  if (!input_file_player_) {
+    WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
+                 "Channel::MixOrReplaceAudioWithFile() fileplayer"
+                 " doesnt exist");
+    return -1;
+  }
 
-    if (input_file_player_->Get10msAudioFromFile(fileBuffer.get(), &fileSamples,
-                                                 mixingFrequency) == -1) {
-      WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
-                   "Channel::MixOrReplaceAudioWithFile() file mixing "
-                   "failed");
-      return -1;
-    }
-    if (fileSamples == 0) {
-      WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
-                   "Channel::MixOrReplaceAudioWithFile() file is ended");
-      return 0;
-    }
+  if (input_file_player_->Get10msAudioFromFile(fileBuffer.get(), &fileSamples,
+                                               mixingFrequency) == -1) {
+    WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
+                 "Channel::MixOrReplaceAudioWithFile() file mixing "
+                 "failed");
+    return -1;
+  }
+  if (fileSamples == 0) {
+    WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
+                 "Channel::MixOrReplaceAudioWithFile() file is ended");
+    return 0;
   }
 
-  assert(_audioFrame.samples_per_channel_ == fileSamples);
+  assert(audio_input->samples_per_channel_ == fileSamples);
 
   if (_mixFileWithMicrophone) {
     // Currently file stream is always mono.
     // TODO(xians): Change the code when FilePlayer supports real stereo.
-    MixWithSat(_audioFrame.data_, _audioFrame.num_channels_, fileBuffer.get(),
+    MixWithSat(audio_input->data_, audio_input->num_channels_, fileBuffer.get(),
                1, fileSamples);
   } else {
     // Replace ACM audio with file.
     // Currently file stream is always mono.
     // TODO(xians): Change the code when FilePlayer supports real stereo.
-    _audioFrame.UpdateFrame(
+    audio_input->UpdateFrame(
         _channelId, 0xFFFFFFFF, fileBuffer.get(), fileSamples, mixingFrequency,
         AudioFrame::kNormalSpeech, AudioFrame::kVadUnknown, 1);
   }