Voe::Channel: Turned GetPlayoutFrequency into GetRtpTimestampRateHz.

webrtc/voice_engine/channel.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 422 matching lines @@
   int ResendPackets(const uint16_t* sequence_numbers, int length);
   int32_t MixOrReplaceAudioWithFile(int mixingFrequency);
   int32_t MixAudioWithFile(AudioFrame& audioFrame, int mixingFrequency);
   void UpdatePlayoutTimestamp(bool rtcp);
   void RegisterReceiveCodecsToRTPModule();
 
   int SetSendRtpHeaderExtension(bool enable,
                                 RTPExtensionType type,
                                 unsigned char id);
 
-  int32_t GetPlayoutFrequency() const;
+  int32_t GetPayloadFrequency() const;

[kwiberg-webrtc, 2016/09/20 15:34:54]

Consider adding a comment that explains what this method does. :-)

Also, in the ACM we're trying to standardize on the names "SampleRateHz" and
"RtpTimestampRateHz", which IMHO are more descriptive. So consider calling it
"GetPayloadSampleRateHz".

Also, the return type should probably be plain int.

[ossu, 2016/09/20 15:43:51]

I don't know!
Alright.
Agreed.

   int64_t GetRTT(bool allow_associate_channel) const;
 
   rtc::CriticalSection _fileCritSect;
   rtc::CriticalSection _callbackCritSect;
   rtc::CriticalSection volume_settings_critsect_;
   uint32_t _instanceId;
   int32_t _channelId;
 
   ChannelState channel_state_;
 
@@ skipping 92 matching lines @@
   std::unique_ptr<RateLimiter> retransmission_rate_limiter_;
 
   // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed.
   rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
 };
 
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_CHANNEL_H_