Voe::Channel: Turned GetPlayoutFrequency into GetRtpTimestampRateHz.

webrtc/modules/audio_coding/acm2/acm_receiver.cc

Index: webrtc/modules/audio_coding/acm2/acm_receiver.cc
diff --git a/webrtc/modules/audio_coding/acm2/acm_receiver.cc b/webrtc/modules/audio_coding/acm2/acm_receiver.cc
index 89eee00fc37edc10d715f1720ff193688d863b8d..2589bb8d25c2c1d0c247d3b8ee930d22a4a3114a 100644
--- a/webrtc/modules/audio_coding/acm2/acm_receiver.cc
+++ b/webrtc/modules/audio_coding/acm2/acm_receiver.cc
@@ -98,6 +98,8 @@ int AcmReceiver::InsertPacket(const WebRtcRTPHeader& rtp_header,
       }
     } else {
       last_audio_decoder_ = ci;
+      last_audio_format_ = neteq_->GetDecoderFormat(ci->pltype);
+      RTC_DCHECK(last_audio_format_);

[kwiberg-webrtc, 2016/10/07 12:05:23]

Are you sure this will always be the case? I seem to recall that if you use one
of the legacy ways of setting the decoders in NetEq, you get no SdpAudioFormat.

[ossu, 2016/10/07 12:21:59]

That is no longer true, however you can avoid getting one if there's no current
decoder, but since RtpHeaderToDecoder() got one, above, we should be fine.

However, now that you mention it, it is quite possible to get a zero clockrate
in that SdpAudioFormat (only when using an external decoder for a format we
don't support internally), so I'll have to deal with that in
GetRtpClockrateHz(). Thanks!

       last_packet_sample_rate_hz_ = rtc::Optional<int>(ci->plfreq);
     }
 
@@ -238,6 +240,7 @@ void AcmReceiver::RemoveAllCodecs() {
   rtc::CritScope lock(&crit_sect_);
   neteq_->RemoveAllPayloadTypes();
   last_audio_decoder_ = rtc::Optional<CodecInst>();
+  last_audio_format_ = rtc::Optional<SdpAudioFormat>();
   last_packet_sample_rate_hz_ = rtc::Optional<int>();
 }
 
@@ -250,6 +253,7 @@ int AcmReceiver::RemoveCodec(uint8_t payload_type) {
   }
   if (last_audio_decoder_ && payload_type == last_audio_decoder_->pltype) {
     last_audio_decoder_ = rtc::Optional<CodecInst>();
+    last_audio_format_ = rtc::Optional<SdpAudioFormat>();
     last_packet_sample_rate_hz_ = rtc::Optional<int>();
   }
   return 0;
@@ -272,6 +276,11 @@ int AcmReceiver::LastAudioCodec(CodecInst* codec) const {
   return 0;
 }
 
+rtc::Optional<SdpAudioFormat> AcmReceiver::LastAudioFormat() const {
+  rtc::CritScope lock(&crit_sect_);
+  return last_audio_format_;
+}
+
 void AcmReceiver::GetNetworkStatistics(NetworkStatistics* acm_stat) {
   NetEqNetworkStatistics neteq_stat;
   // NetEq function always returns zero, so we don't check the return value.