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) {
+    RTC_DCHECK(channel_);
+  }
+
+ private:
+  bool Run() override {
+    RTC_DCHECK_RUN_ON(channel_->encoder_queue_);
+    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);

[tommi, 2017/03/31 09:44:10]

should we also add:

RTC_DCHECK(!encoder_queue_);
(note: that's making sure the member variable is null)

[henrika_webrtc, 2017/03/31 11:42:46]

Wow, why not actually ;-)

   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);

[tommi, 2017/03/31 09:44:10]

It's too bad that these two methods, Get() and SetSending() both grab the same
lock, copy state, only for the sake of setting a bool and even so, they're prone
to a race since when we call SetSending(false), we don't know if |sending| is
still true[1], which indicates a bug.

The four lock/unlock operations, checking the value and setting it, could all be
accomplished in a single atomic operation:

if (rtc::AtomicOps::CompareAndSwap(&channel_state_.sending_, 0) == 0)
  return;

RTC_DCHECK(encoder_queue_);
...


[1]: That is assuming that the lock is actually needed on this thread, in the
first place.  It could very well be that all modification to the |sending| flag,
is done on the current thread, and if so, the current thread does not need to
hold a lock to examine its value.  However, there aren't thread checks or
documentation that explain and verify how this works, so at the moment, it
doesn't look like we really know.

[the sun, 2017/03/31 10:19:37]

I totally agree that the current construct is... terrible - I don't find any
diplomatic way to put it - but resorting to a one-off solution with atomics
seems like adding too much complexity for something which should be very simple.
I'd use that in an inner loop, which none of the functions in voe::Channel
really is (that's why we've been getting away with taking ~5 locks in
GetAudioFrameWithMuted(), for instance).

This CL is not making the situation worse (and StopSend() is not called that
often), so I suggest we leave such cleanup for a later time.
This class is riddled with questionable design choices. That's one of the
reasons Audio[Send|Receive]Stream access it via ChannelProxy - where we have
added thread checks and assertions on inputs/outputs - in an attempt to lock
down how voe::Channel is actually used. Once the legacy VoE APIs are gone, we'll
be able to strip out everything from voe::Channel which falls outside of those
assumptions. Then cleaning up and splitting the class' send/receive duties (it
is duplex right now!) into the stream classes will be much easier.

[tommi, 2017/03/31 10:58:52]

Sounds good.  My preference as far as the flag goes, would be that we clarify
the threading model with thread checks and then take action. I suspect that a
lock or atomic ops might not be needed and we can start removing much of the
extra complexity.

[the sun, 2017/03/31 11:04:10]

I believe your suspicion is correct...

 
+  // 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_);

[tommi, 2017/03/31 09:44:10]

we don't DCHECK pointers that we dereference anyway

[henrika_webrtc, 2017/03/31 11:42:46]

Acknowledged.

+  RTC_DCHECK(channel_state_.Get().sending);
+  std::unique_ptr<AudioFrame> audio_frame(new AudioFrame());
+  audio_frame->CopyFrom(audio_input);

[tommi, 2017/03/31 09:44:10]

It feels like we should be able to avoid this. Can you add a TODO here so that
we don't have to do this allocation+copy for every buffer?

[the sun, 2017/03/31 10:19:37]

The allocation can easily be avoided by aggregating the frame in the task
object.

AFAICT the copy is needed, unless we do some major surgery and bless the ADM
with a pool of buffers it can pass ownership of (something akin to how SwapQueue
works - we'd need to send a prealloc:d buffer back for each one we receive).

[tommi, 2017/03/31 10:58:52]

yes, something like that should work. A TODO or something for now, just so that
we won't forget, works for me.

[henrika_webrtc, 2017/03/31 11:42:46]

Done.

+  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_);

[tommi, 2017/03/31 09:44:10]

this pointer is used anyway below, so no need to dcheck

[henrika_webrtc, 2017/03/31 11:42:47]

Done.

+  RTC_DCHECK(channel_state_.Get().sending);
   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);
 
     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;
     }
   }
 
-  assert(_audioFrame.samples_per_channel_ == fileSamples);
+  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