Fix Channel::GetSendCodec when used together with SetEncoder.

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 1260 matching lines @@
         VE_RTP_RTCP_MODULE_ERROR, kTraceWarning,
         "StartSend() RTP/RTCP failed to stop sending");
   }
   _rtpRtcpModule->SetSendingMediaStatus(false);
 }
 
 bool Channel::SetEncoder(int payload_type,
                          std::unique_ptr<AudioEncoder> encoder) {
   RTC_DCHECK_GE(payload_type, 0);
   RTC_DCHECK_LE(payload_type, 127);
-  // TODO(ossu): Make a CodecInst up for now. It seems like very little of this
-  // information is actually used, possibly only payload type and clock rate.
-  CodecInst lies;
-  lies.pltype = payload_type;
-  strncpy(lies.plname, "audio", sizeof(lies.plname));
-  lies.plname[sizeof(lies.plname) - 1] = 0;
+  // TODO(ossu): Make CodecInsts up, for now: one for the RTP/RTCP module and
+  // one for for us to keep track of sample rate and number of channels, etc.
+
+  // The RTP/RTCP module needs to know the RTP timestamp rate (i.e. clockrate)
+  // as well as some other things, so we collect this info and send it along.
+  CodecInst rtp_codec;
+  rtp_codec.pltype = payload_type;
+  strncpy(rtp_codec.plname, "audio", sizeof(rtp_codec.plname));
+  rtp_codec.plname[sizeof(rtp_codec.plname) - 1] = 0;
   // Seems unclear if it should be clock rate or sample rate. CodecInst
   // supposedly carries the sample rate, but only clock rate seems sensible to
   // send to the RTP/RTCP module.
-  lies.plfreq = encoder->RtpTimestampRateHz();
-  lies.pacsize = 0;
-  lies.channels = encoder->NumChannels();
-  lies.rate = 0;
+  rtp_codec.plfreq = encoder->RtpTimestampRateHz();
+  rtp_codec.pacsize = rtc::CheckedDivExact(
+      static_cast<int>(encoder->Max10MsFramesInAPacket() * rtp_codec.plfreq),
+      100);
+  rtp_codec.channels = encoder->NumChannels();
+  rtp_codec.rate = 0;
 
-  if (_rtpRtcpModule->RegisterSendPayload(lies) != 0) {
+  // For audio encoding we need, instead, the actual sample rate of the codec.
+  // The rest of the information should be the same.
+  CodecInst send_codec = rtp_codec;
+  send_codec.plfreq = encoder->SampleRateHz();
+  cached_send_codec_.emplace(send_codec);

[the sun, 2017/06/13 09:25:07]

Though this may work in practice - won't the TSAN bots complain at some point?
Writing on one thread, reading on another.

[ossu, 2017/06/13 09:39:15]

I was a bit worried about that as well, but having looked into it a bit, I can't
find that any of the previous code enforces thread safety either. There are
checkers in CodecManager and ChannelProxy, but no locks. C.f.
CodecManager::GetCodecInst (just returns a value in an Optional) and
CodecManager::RegisterEncoder (RTC_DCHECK(thread_checker...)).

+
+  if (_rtpRtcpModule->RegisterSendPayload(rtp_codec) != 0) {
     _rtpRtcpModule->DeRegisterSendPayload(payload_type);
-    if (_rtpRtcpModule->RegisterSendPayload(lies) != 0) {
+    if (_rtpRtcpModule->RegisterSendPayload(rtp_codec) != 0) {
       WEBRTC_TRACE(
           kTraceError, kTraceVoice, VoEId(_instanceId, _channelId),
           "SetEncoder() failed to register codec to RTP/RTCP module");
       return false;
     }
   }
 
   audio_coding_->SetEncoder(std::move(encoder));
   codec_manager_.UnsetCodecInst();
   return true;
@@ skipping 28 matching lines @@
     _engineStatisticsPtr->SetLastError(
         VE_INVALID_OPERATION, kTraceWarning,
         "DeRegisterVoiceEngineObserver() observer already disabled");
     return 0;
   }
   _voiceEngineObserverPtr = NULL;
   return 0;
 }
 
 int32_t Channel::GetSendCodec(CodecInst& codec) {
-  {
+  if (cached_send_codec_) {
+    codec = *cached_send_codec_;
+    return 0;
+  } else {
     const CodecInst* send_codec = codec_manager_.GetCodecInst();
     if (send_codec) {
       codec = *send_codec;
       return 0;
     }
   }
-  rtc::Optional<CodecInst> acm_send_codec = audio_coding_->SendCodec();
-  if (acm_send_codec) {
-    codec = *acm_send_codec;
-    return 0;
-  }
   return -1;
 }
 
 int32_t Channel::GetRecCodec(CodecInst& codec) {
   return (audio_coding_->ReceiveCodec(&codec));
 }
 
 int32_t Channel::SetSendCodec(const CodecInst& codec) {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::SetSendCodec()");
 
   if (!codec_manager_.RegisterEncoder(codec) ||
       !codec_manager_.MakeEncoder(&rent_a_codec_, audio_coding_.get())) {
     WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, _channelId),
                  "SetSendCodec() failed to register codec to ACM");
     return -1;
   }
 
   if (_rtpRtcpModule->RegisterSendPayload(codec) != 0) {
     _rtpRtcpModule->DeRegisterSendPayload(codec.pltype);
     if (_rtpRtcpModule->RegisterSendPayload(codec) != 0) {
       WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, _channelId),
                    "SetSendCodec() failed to register codec to"
                    " RTP/RTCP module");
       return -1;
     }
   }
 
+  cached_send_codec_.reset();
+
   return 0;
 }
 
 void Channel::SetBitRate(int bitrate_bps, int64_t probing_interval_ms) {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::SetBitRate(bitrate_bps=%d)", bitrate_bps);
   audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
     if (*encoder) {
       (*encoder)->OnReceivedUplinkBandwidth(
           bitrate_bps, rtc::Optional<int64_t>(probing_interval_ms));
@@ skipping 1733 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