Voe::Channel: Turned GetPlayoutFrequency into GetRtpTimestampRateHz.

webrtc/modules/audio_coding/include/audio_coding_module.h

Index: webrtc/modules/audio_coding/include/audio_coding_module.h
diff --git a/webrtc/modules/audio_coding/include/audio_coding_module.h b/webrtc/modules/audio_coding/include/audio_coding_module.h
index fc8ae1ed513cb5098c2406f09a4d63a2e1f9720a..f7fed0467e66b7fcbcc2bbb06254a66b5d0baa2d 100644
--- a/webrtc/modules/audio_coding/include/audio_coding_module.h
+++ b/webrtc/modules/audio_coding/include/audio_coding_module.h
@@ -542,6 +542,8 @@ class AudioCodingModule {
   //
   virtual int32_t ReceiveCodec(CodecInst* curr_receive_codec) const = 0;
 
+  virtual rtc::Optional<SdpAudioFormat> ReceiveFormat() const = 0;

[hlundin-webrtc, 2016/10/07 12:32:19]

A bit of a description wouldn't go amiss. In particular, this information
relates to the last incoming packet (not the last output audio), and under what
circumstances is it empty.

[ossu, 2016/10/07 12:41:32]

That's right! I remember thinking: "I'll comment this later". :)

[kwiberg-webrtc, 2016/10/07 12:53:52]

Then you're all set! It's the thought that counts.

+
   ///////////////////////////////////////////////////////////////////////////
   // int32_t IncomingPacket()
   // Call this function to insert a parsed RTP packet into ACM.