Remove VoEVideoSync interface.

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 a9cf2b26dc8edea51bb8d56b57d13a52275e58b2..f279ec0ac3fd7d418035a628ed5c79a86167257b 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -1227,16 +1227,16 @@ int32_t Channel::StopPlayout() {
 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(
@@ -3013,23 +3013,9 @@ void Channel::GetDecodingCallStatistics(AudioDecodingCallStats* stats) const {
   audio_coding_->GetDecodingCallStatistics(stats);
 }
 
-bool Channel::GetDelayEstimate(int* jitter_buffer_delay_ms,
-                               int* playout_buffer_delay_ms) 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();
+  rtc::CritScope lock(&video_sync_lock_);
+  return audio_coding_->FilteredCurrentDelayMs() + playout_delay_ms_;
 }
 
 int Channel::SetMinimumPlayoutDelay(int delayMs) {
@@ -3067,30 +3053,6 @@ int Channel::GetPlayoutTimestamp(unsigned int& timestamp) {
   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();