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

webrtc/voice_engine/channel.cc

 /*
  *  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 "webrtc/voice_engine/channel.h"
 
 #include <algorithm>
 #include <utility>
 
 #include "webrtc/audio/utility/audio_frame_operations.h"
 #include "webrtc/base/array_view.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/format_macros.h"
 #include "webrtc/base/location.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/call/rtp_transport_controller_send.h"
 #include "webrtc/config.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/pacing/packet_router.h"
 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
@@ skipping 368 matching lines @@
   }
 
  private:
   Channel* owner_;
   // Maps remote side ssrc to extended highest sequence number received.
   std::map<uint32_t, uint32_t> extended_max_sequence_number_;
   rtc::CriticalSection crit_;
   RtcpBandwidthObserver* bandwidth_observer_ GUARDED_BY(crit_);
 };
 
+class Channel::ProcessAndEncodeAudioTask : public rtc::QueuedTask {
+ public:
+  ProcessAndEncodeAudioTask(std::unique_ptr<AudioFrame> audio_frame,
+                            Channel* channel)
+      : audio_frame_(std::move(audio_frame)), channel_(channel) {}

[the sun, 2017/03/30 10:13:32]

super nit: you should DCHECK(channel) here instead of before the dereference; it
is part of the invariant.

[henrika_webrtc, 2017/03/30 11:16:31]

Done.

+
+ private:
+  bool Run() override {
+    RTC_DCHECK_RUN_ON(channel_->encoder_queue_);
+    RTC_DCHECK(channel_);
+    channel_->ProcessAndEncodeAudioOnTaskQueue(audio_frame_.get());
+    return true;
+  }
+
+  std::unique_ptr<AudioFrame> audio_frame_;
+  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_);
   WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::SendData(frameType=%u, payloadType=%u, timeStamp=%u,"
                " payloadSize=%" PRIuS ", fragmentation=0x%x)",
                frameType, payloadType, timeStamp, payloadSize, fragmentation);
 
   if (_includeAudioLevelIndication) {
     // Store current audio level in the RTP/RTCP module.
     // The level will be used in combination with voice-activity state
     // (frameType) to add an RTP header extension
     _rtpRtcpModule->SetAudioLevel(rms_level_.Average());
   }
 
   // Push data from ACM to RTP/RTCP-module to deliver audio frame for
   // packetization.
   // This call will trigger Transport::SendPacket() from the RTP/RTCP module.
   if (!_rtpRtcpModule->SendOutgoingData(
           (FrameType&)frameType, payloadType, timeStamp,
           // Leaving the time when this frame was
           // received from the capture device as
           // undefined for voice for now.
           -1, payloadData, payloadSize, fragmentation, nullptr, nullptr)) {
     _engineStatisticsPtr->SetLastError(
         VE_RTP_RTCP_MODULE_ERROR, kTraceWarning,
         "Channel::SendData() failed to send data to RTP/RTCP module");
     return -1;
   }
 
-  _lastLocalTimeStamp = timeStamp;
-  _lastPayloadType = payloadType;
-
   return 0;
 }
 
 bool Channel::SendRtp(const uint8_t* data,
                       size_t len,
                       const PacketOptions& options) {
   WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::SendPacket(channel=%d, len=%" PRIuS ")", len);
 
   rtc::CritScope cs(&_callbackCritSect);
@@ skipping 314 matching lines @@
     if (output_file_player_) {
       if (output_file_player_->Frequency() > highestNeeded) {
         highestNeeded = output_file_player_->Frequency();
       }
     }
   }
 
   return (highestNeeded);
 }
 
-int32_t Channel::CreateChannel(
-    Channel*& channel,
-    int32_t channelId,
-    uint32_t instanceId,
-    const VoEBase::ChannelConfig& config) {
+int32_t Channel::CreateChannel(Channel*& channel,
+                               int32_t channelId,
+                               uint32_t instanceId,
+                               const VoEBase::ChannelConfig& config) {
   WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(instanceId, channelId),
                "Channel::CreateChannel(channelId=%d, instanceId=%d)", channelId,
                instanceId);
 
   channel = new Channel(channelId, instanceId, config);
   if (channel == NULL) {
     WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(instanceId, channelId),
                  "Channel::CreateChannel() unable to allocate memory for"
                  " channel");
     return -1;
@@ skipping 86 matching lines @@
       _outputMixerPtr(NULL),
       _moduleProcessThreadPtr(NULL),
       _audioDeviceModulePtr(NULL),
       _voiceEngineObserverPtr(NULL),
       _callbackCritSectPtr(NULL),
       _transportPtr(NULL),
       input_mute_(false),
       previous_frame_muted_(false),
       _outputGain(1.0f),
       _mixFileWithMicrophone(false),
-      _lastLocalTimeStamp(0),
-      _lastPayloadType(0),
       _includeAudioLevelIndication(false),
       transport_overhead_per_packet_(0),
       rtp_overhead_per_packet_(0),
       _outputSpeechType(AudioFrame::kNormalSpeech),
       restored_packet_in_use_(false),
       rtcp_observer_(new VoERtcpObserver(this)),
       associate_send_channel_(ChannelOwner(nullptr)),
       pacing_enabled_(config.enable_voice_pacing),
       feedback_observer_proxy_(new TransportFeedbackProxy()),
       seq_num_allocator_proxy_(new TransportSequenceNumberProxy()),
@@ skipping 213 matching lines @@
     _moduleProcessThreadPtr->DeRegisterModule(_rtpRtcpModule.get());
 
   // End of modules shutdown
 }
 
 int32_t Channel::SetEngineInformation(Statistics& engineStatistics,
                                       OutputMixer& outputMixer,
                                       ProcessThread& moduleProcessThread,
                                       AudioDeviceModule& audioDeviceModule,
                                       VoiceEngineObserver* voiceEngineObserver,
-                                      rtc::CriticalSection* callbackCritSect) {
+                                      rtc::CriticalSection* callbackCritSect,
+                                      rtc::TaskQueue* encoder_queue) {
+  RTC_DCHECK(encoder_queue);
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::SetEngineInformation()");
   _engineStatisticsPtr = &engineStatistics;
   _outputMixerPtr = &outputMixer;
   _moduleProcessThreadPtr = &moduleProcessThread;
   _audioDeviceModulePtr = &audioDeviceModule;
   _voiceEngineObserverPtr = voiceEngineObserver;
   _callbackCritSectPtr = callbackCritSect;
+  encoder_queue_ = encoder_queue;
   return 0;
 }
 
-int32_t Channel::UpdateLocalTimeStamp() {
-  _timeStamp += static_cast<uint32_t>(_audioFrame.samples_per_channel_);
-  return 0;
-}
-
 void Channel::SetSink(std::unique_ptr<AudioSinkInterface> sink) {
   rtc::CritScope cs(&_callbackCritSect);
   audio_sink_ = std::move(sink);
 }
 
 const rtc::scoped_refptr<AudioDecoderFactory>&
 Channel::GetAudioDecoderFactory() const {
   return decoder_factory_;
 }
 
@@ skipping 60 matching lines @@
         "StartSend() RTP/RTCP failed to start sending");
     _rtpRtcpModule->SetSendingMediaStatus(false);
     rtc::CritScope cs(&_callbackCritSect);
     channel_state_.SetSending(false);
     return -1;
   }
 
   return 0;
 }
 
-int32_t Channel::StopSend() {
+void Channel::StopSend() {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::StopSend()");
   if (!channel_state_.Get().sending) {
-    return 0;
+    return;
   }
   channel_state_.SetSending(false);
 
+  // Post a task to the encoder thread which sets an event when the task is
+  // executed. We know that no more encoding tasks will be added to the task
+  // queue for this channel since sending is now deactivated. It means that,
+  // if we wait for the event to bet set, we know that no more pending tasks
+  // exists and it is therfore guaranteed that the task queue will never try
+  // to acccess and invalid channel object.
+  RTC_DCHECK(encoder_queue_);
+  rtc::Event flush(false, false);
+  encoder_queue_->PostTask([&flush](){ flush.Set(); });
+  flush.Wait(rtc::Event::kForever);
+
   // Store the sequence number to be able to pick up the same sequence for
   // the next StartSend(). This is needed for restarting device, otherwise
   // it might cause libSRTP to complain about packets being replayed.
   // TODO(xians): Remove this workaround after RtpRtcpModule's refactoring
   // CL is landed. See issue
   // https://code.google.com/p/webrtc/issues/detail?id=2111 .
   send_sequence_number_ = _rtpRtcpModule->SequenceNumber();
 
   // Reset sending SSRC and sequence number and triggers direct transmission
   // of RTCP BYE
   if (_rtpRtcpModule->SetSendingStatus(false) == -1) {
     _engineStatisticsPtr->SetLastError(
         VE_RTP_RTCP_MODULE_ERROR, kTraceWarning,
         "StartSend() RTP/RTCP failed to stop sending");
   }
   _rtpRtcpModule->SetSendingMediaStatus(false);
-
-  return 0;
 }
 
 int32_t Channel::RegisterVoiceEngineObserver(VoiceEngineObserver& observer) {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::RegisterVoiceEngineObserver()");
   rtc::CritScope cs(&_callbackCritSect);
 
   if (_voiceEngineObserverPtr) {
     _engineStatisticsPtr->SetLastError(
         VE_INVALID_OPERATION, kTraceError,
@@ skipping 1380 matching lines @@
     audio_coding_->EnableNack(maxNumberOfPackets);
   else
     audio_coding_->DisableNack();
 }
 
 // Called when we are missing one or more packets.
 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_);

[the sun, 2017/03/30 10:13:33]

Also
RTC_DCHECK(channel_state_.Get().sending);

[henrika_webrtc, 2017/03/30 11:16:31]

Done.

+  std::unique_ptr<AudioFrame> audio_frame(new AudioFrame());
+  audio_frame->CopyFrom(audio_input);
+  audio_frame->id_ = ChannelId();
+  encoder_queue_->PostTask(std::unique_ptr<rtc::QueuedTask>(
+      new ProcessAndEncodeAudioTask(std::move(audio_frame), 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_);

[the sun, 2017/03/30 10:13:33]

Here too:
RTC_DCHECK(channel_state_.Get().sending);

[henrika_webrtc, 2017/03/30 11:16:31]

Done.

   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_frame(new AudioFrame());
+  audio_frame->id_ = ChannelId();
+  audio_frame->sample_rate_hz_ = std::min(codec.plfreq, sample_rate);
+  audio_frame->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_frame.get());
+  encoder_queue_->PostTask(std::unique_ptr<rtc::QueuedTask>(
+      new ProcessAndEncodeAudioTask(std::move(audio_frame), this)));
 }
 
-uint32_t Channel::PrepareEncodeAndSend(int mixingFrequency) {
-  WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
-               "Channel::PrepareEncodeAndSend()");
+void Channel::ProcessAndEncodeAudioOnTaskQueue(AudioFrame* audio_input) {
+  RTC_DCHECK_RUN_ON(encoder_queue_);
+  RTC_DCHECK_GT(audio_input->samples_per_channel_, 0);
+  RTC_DCHECK_LE(audio_input->num_channels_, 2);
+  RTC_DCHECK_EQ(audio_input->id_, ChannelId());
 
-  if (_audioFrame.samples_per_channel_ == 0) {
-    WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
-                 "Channel::PrepareEncodeAndSend() invalid audio frame");
-    return 0xFFFFFFFF;
+  if (channel_state_.Get().input_file_playing) {
+    MixOrReplaceAudioWithFile(audio_input);
   }
 
-  if (channel_state_.Get().input_file_playing) {
-    MixOrReplaceAudioWithFile(mixingFrequency);
-  }
-
-  bool is_muted = InputMute();  // Cache locally as InputMute() takes a lock.
-  AudioFrameOperations::Mute(&_audioFrame, previous_frame_muted_, is_muted);
+  bool is_muted = InputMute();
+  AudioFrameOperations::Mute(audio_input, previous_frame_muted_, is_muted);
 
   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()");
-
-  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;
-
-  // --- Add 10ms of raw (PCM) audio data to the encoder @ 32kHz.
+  // 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");
-    return 0xFFFFFFFF;
+  if (audio_coding_->Add10MsData(*audio_input) < 0) {
+    LOG(LS_ERROR) << "ACM::Add10MsData() failed for channel " << _channelId;
+    return;
   }
 
-  _timeStamp += static_cast<uint32_t>(_audioFrame.samples_per_channel_);
-  return 0;
+  _timeStamp += static_cast<uint32_t>(audio_input->samples_per_channel_);
 }
 
 void Channel::set_associate_send_channel(const ChannelOwner& channel) {
   RTC_DCHECK(!channel.channel() ||
              channel.channel()->ChannelId() != _channelId);
   rtc::CritScope lock(&assoc_send_channel_lock_);
   associate_send_channel_ = channel;
 }
 
 void Channel::DisassociateSendChannel(int channel_id) {
@@ skipping 88 matching lines @@
 
 int Channel::GetRtpRtcp(RtpRtcp** rtpRtcpModule,
                         RtpReceiver** rtp_receiver) const {
   *rtpRtcpModule = _rtpRtcpModule.get();
   *rtp_receiver = rtp_receiver_.get();
   return 0;
 }
 
 // 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);
+  rtc::CritScope cs(&_fileCritSect);

[the sun, 2017/03/30 10:13:33]

fyi: you're holding the lock for longer now. don't know if it matters

[henrika_webrtc, 2017/03/30 11:16:31]

Let me fix that.

 
-    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_) {
+    WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
+                 "Channel::MixOrReplaceAudioWithFile() fileplayer"
+                 " doesnt exist");
+    return -1;
   }
 
-  assert(_audioFrame.samples_per_channel_ == fileSamples);
+  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;
+  }
+
+  RTC_DCHECK_EQ(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);
   }
   return 0;
 }
 
 int32_t Channel::MixAudioWithFile(AudioFrame& audioFrame, int mixingFrequency) {
   assert(mixingFrequency <= 48000);
 
   std::unique_ptr<int16_t[]> fileBuffer(new int16_t[960]);
@@ skipping 169 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
       0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc