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 525 matching lines @@
   //                         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;
 
   ///////////////////////////////////////////////////////////////////////////
+  // rtc::Optional<SdpAudioFormat> ReceiveFormat()
+  // Get the format associated with last received payload.
+  //
+  // Return value:
+  //    An SdpAudioFormat describing the format associated with the last
+  //    received payload.
+  //    An empty Optional if no payload has yet been received.
+  //
+  virtual rtc::Optional<SdpAudioFormat> ReceiveFormat() const = 0;
+
+  ///////////////////////////////////////////////////////////////////////////
   // 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.
   //
   // Return value:
@@ skipping 238 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_