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 0f0831c8b1f8cbd737a7b919e4a277b04cb87ff1..c1cfa6fdd9959c2b8c11f73dffe7fecb76c28bc8 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -21,6 +21,8 @@
 #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/config.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
@@ -48,6 +50,10 @@ namespace {
 
 constexpr int64_t kMaxRetransmissionWindowMs = 1000;
 constexpr int64_t kMinRetransmissionWindowMs = 30;
+// Number of preallocated audio frames in the pool of audio frames.
+// Local tests on Android devices have shown that we never reduce the size of
+// the pool below 5.
+constexpr size_t kAudioFramePoolSize = 10;
 
 }  // namespace
 
@@ -406,12 +412,41 @@ class VoERtcpObserver : public RtcpBandwidthObserver {
   RtcpBandwidthObserver* bandwidth_observer_ GUARDED_BY(crit_);
 };
 
+class Channel::ProcessAndEncodeAudioTask : public rtc::QueuedTask {
+ public:
+  ProcessAndEncodeAudioTask(std::unique_ptr<AudioFrame> audio_input,
+                            Channel* channel,
+                            AudioFramePool* audio_frame_pool)
+      : audio_input_(std::move(audio_input)),
+        channel_(channel),
+        audio_frame_pool_(audio_frame_pool) {}
+
+  ~ProcessAndEncodeAudioTask() override {
+    RTC_DCHECK_RUN_ON(channel_->encoder_queue_);
+    // Return the utilized audio frame to the pool so it can be used again.
+    audio_frame_pool_->Push(std::move(audio_input_));
+  }
+
+ private:
+  bool Run() override {
+    RTC_DCHECK_RUN_ON(channel_->encoder_queue_);
+    RTC_DCHECK(channel_);
+    channel_->ProcessAndEncodeAudioOnTaskQueue(audio_input_.get());
+    return true;
+  }
+
+  std::unique_ptr<AudioFrame> audio_input_;
+  Channel* const channel_;
+  AudioFramePool* audio_frame_pool_;
+};
+
 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)",
@@ -441,7 +476,6 @@ int32_t Channel::SendData(FrameType frameType,
 
   _lastLocalTimeStamp = timeStamp;
   _lastPayloadType = payloadType;
-
   return 0;
 }
 
@@ -882,6 +916,7 @@ Channel::Channel(int32_t channelId,
       _audioDeviceModulePtr(NULL),
       _voiceEngineObserverPtr(NULL),
       _callbackCritSectPtr(NULL),
+      encoder_queue_(nullptr),
       _transportPtr(NULL),
       input_mute_(false),
       previous_frame_muted_(false),
@@ -902,6 +937,7 @@ Channel::Channel(int32_t channelId,
       rtp_packet_sender_proxy_(new RtpPacketSenderProxy()),
       retransmission_rate_limiter_(new RateLimiter(Clock::GetRealTimeClock(),
                                                    kMaxRetransmissionWindowMs)),
+      audio_frame_pool_(kAudioFramePoolSize),
       decoder_factory_(config.acm_config.decoder_factory) {
   WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::Channel() - ctor");
@@ -1097,20 +1133,18 @@ 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;
   _moduleProcessThreadPtr = &moduleProcessThread;
   _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;
 }
 
@@ -2589,89 +2623,92 @@ 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_frame = audio_frame_pool_.Pop();
+  RTC_DCHECK(audio_frame) << "Pool of audio frames is empty";
+  if (audio_frame) {
+    audio_frame->CopyFrom(audio_input);
+    audio_frame->id_ = _channelId;
+    encoder_queue_->PostTask(
+        std::unique_ptr<rtc::QueuedTask>(new ProcessAndEncodeAudioTask(
+            std::move(audio_frame), this, &audio_frame_pool_)));
+  }
 }
 
-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);
+  std::unique_ptr<AudioFrame> audio_frame = audio_frame_pool_.Pop();
+  RTC_DCHECK(audio_frame) << "Pool of audio frames is empty";
+  if (audio_frame) {
+    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_, audio_frame.get());
+    audio_frame->id_ = _channelId;
+    encoder_queue_->PostTask(
+        std::unique_ptr<rtc::QueuedTask>(new ProcessAndEncodeAudioTask(
+            std::move(audio_frame), this, &audio_frame_pool_)));
+  }
+}
 
-  // 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);
-  RemixAndResample(audio_data, number_of_frames, number_of_channels,
-                   sample_rate, &input_resampler_, &_audioFrame);
+void Channel::ProcessAndEncodeAudioOnTaskQueue(AudioFrame* audio_input) {
+  RTC_DCHECK_RUN_ON(encoder_queue_);
+  PrepareEncodeAndSend(audio_input);
+  EncodeAndSend(audio_input);
 }
 
-uint32_t Channel::PrepareEncodeAndSend(int mixingFrequency) {
-  WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
-               "Channel::PrepareEncodeAndSend()");
-
-  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 (_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_);
   return 0;
 }
 
@@ -2781,46 +2818,44 @@ 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);
   }