Fix Channel::GetSendCodec when used together with SetEncoder.

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 b0bae4e4ceb0e05cd1ca421efdbc5943cf640e19..0a9e9fce34b8a3a3217eef8dbc4bac09b17182b3 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -1278,23 +1278,34 @@ 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;
-
-  if (_rtpRtcpModule->RegisterSendPayload(lies) != 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;
+
+  // 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");
@@ -1343,18 +1354,16 @@ int32_t Channel::DeRegisterVoiceEngineObserver() {
 }
 
 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;
 }
 
@@ -1383,6 +1392,8 @@ int32_t Channel::SetSendCodec(const CodecInst& codec) {
     }
   }
 
+  cached_send_codec_.reset();
+
   return 0;
 }