Voe::Channel: Turned GetPlayoutFrequency into GetRtpTimestampRateHz.

webrtc/modules/audio_coding/include/audio_coding_module.h

 /*
  *  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 524 matching lines @@
   //   -curr_receive_codec : parameters of the codec associated with the last
   //                         received payload, c.f. common_types.h for
   //                         the definition of CodecInst.
   //
   // Return value:
   //   -1 if failed to retrieve the codec,
   //    0 if the codec is successfully retrieved.
   //
   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.
   //
   // Inputs:
   //   -incoming_payload   : received payload.
   //   -payload_len_bytes  : the length of payload in bytes.
   //   -rtp_info           : the relevant information retrieved from RTP
   //                         header.
   //
@@ skipping 239 matching lines @@
   virtual std::vector<uint16_t> GetNackList(
       int64_t round_trip_time_ms) const = 0;
 
   virtual void GetDecodingCallStatistics(
       AudioDecodingCallStats* call_stats) const = 0;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_