Remove VoEVideoSync interface.

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.
  */
 
@@ skipping 1209 matching lines @@
 
   channel_state_.SetPlaying(false);
   _outputAudioLevel.Clear();
 
   return 0;
 }
 
 int32_t Channel::StartSend() {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::StartSend()");
-  // Resume the previous sequence number which was reset by StopSend().
-  // This needs to be done before |sending| is set to true.
-  if (send_sequence_number_)
-    SetInitSequenceNumber(send_sequence_number_);
-
   if (channel_state_.Get().sending) {
     return 0;
   }
   channel_state_.SetSending(true);
 
+  // Resume the previous sequence number which was reset by StopSend().
+  // This needs to be done before |sending| is set to true.

[kwiberg-webrtc, 2017/02/09 10:16:40]

Is this comment still good? It looks like you're doing it afterwards now...

[the sun, 2017/02/11 12:05:51]

Updated the comment to be more clear. AFAICT it still applies since
RTPSender::SetSendingStatus() may otherwise randomize the sequence number.

[kwiberg-webrtc, 2017/02/11 21:25:25]

Oh, OK, it was referring to the _rtpRtcpModule->SetSendingStatus(true) call
below, not the channel_state_.SetSending(true) call.

+  if (send_sequence_number_) {
+    _rtpRtcpModule->SetSequenceNumber(send_sequence_number_);
+  }
   _rtpRtcpModule->SetSendingMediaStatus(true);
   if (_rtpRtcpModule->SetSendingStatus(true) != 0) {
     _engineStatisticsPtr->SetLastError(
         VE_RTP_RTCP_MODULE_ERROR, kTraceError,
         "StartSend() RTP/RTCP failed to start sending");
     _rtpRtcpModule->SetSendingMediaStatus(false);
     rtc::CritScope cs(&_callbackCritSect);
     channel_state_.SetSending(false);
     return -1;
   }
@@ skipping 1756 matching lines @@
 }
 
 int Channel::GetNetworkStatistics(NetworkStatistics& stats) {
   return audio_coding_->GetNetworkStatistics(&stats);
 }
 
 void Channel::GetDecodingCallStatistics(AudioDecodingCallStats* stats) const {
   audio_coding_->GetDecodingCallStatistics(stats);
 }
 
-bool Channel::GetDelayEstimate(int* jitter_buffer_delay_ms,
-                               int* playout_buffer_delay_ms) const {
+uint32_t Channel::GetDelayEstimate() const {
   rtc::CritScope lock(&video_sync_lock_);
-  *jitter_buffer_delay_ms = audio_coding_->FilteredCurrentDelayMs();
-  *playout_buffer_delay_ms = playout_delay_ms_;
-  return true;
-}
-
-uint32_t Channel::GetDelayEstimate() const {
-  int jitter_buffer_delay_ms = 0;
-  int playout_buffer_delay_ms = 0;
-  GetDelayEstimate(&jitter_buffer_delay_ms, &playout_buffer_delay_ms);
-  return jitter_buffer_delay_ms + playout_buffer_delay_ms;
-}
-
-int Channel::LeastRequiredDelayMs() const {
-  return audio_coding_->LeastRequiredDelayMs();
+  return audio_coding_->FilteredCurrentDelayMs() + playout_delay_ms_;
 }
 
 int Channel::SetMinimumPlayoutDelay(int delayMs) {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::SetMinimumPlayoutDelay()");
   if ((delayMs < kVoiceEngineMinMinPlayoutDelayMs) ||
       (delayMs > kVoiceEngineMaxMinPlayoutDelayMs)) {
     _engineStatisticsPtr->SetLastError(
         VE_INVALID_ARGUMENT, kTraceError,
         "SetMinimumPlayoutDelay() invalid min delay");
@@ skipping 17 matching lines @@
   if (playout_timestamp_rtp == 0) {
     _engineStatisticsPtr->SetLastError(
         VE_CANNOT_RETRIEVE_VALUE, kTraceStateInfo,
         "GetPlayoutTimestamp() failed to retrieve timestamp");
     return -1;
   }
   timestamp = playout_timestamp_rtp;
   return 0;
 }
 
-int Channel::SetInitTimestamp(unsigned int timestamp) {
-  WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
-               "Channel::SetInitTimestamp()");
-  if (channel_state_.Get().sending) {
-    _engineStatisticsPtr->SetLastError(VE_SENDING, kTraceError,
-                                       "SetInitTimestamp() already sending");
-    return -1;
-  }
-  _rtpRtcpModule->SetStartTimestamp(timestamp);
-  return 0;
-}
-
-int Channel::SetInitSequenceNumber(short sequenceNumber) {
-  WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
-               "Channel::SetInitSequenceNumber()");
-  if (channel_state_.Get().sending) {
-    _engineStatisticsPtr->SetLastError(
-        VE_SENDING, kTraceError, "SetInitSequenceNumber() already sending");
-    return -1;
-  }
-  _rtpRtcpModule->SetSequenceNumber(sequenceNumber);
-  return 0;
-}
-
 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) {
@@ skipping 220 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