Add muted_output parameter to ACM

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 669 matching lines @@
   // Input:
   //   -desired_freq_hz    : the desired sampling frequency, in Hertz, of the
   //                         output audio. If set to -1, the function returns
   //                         the audio at the current sampling frequency.
   //
   // Output:
   //   -audio_frame        : output audio frame which contains raw audio data
   //                         and other relevant parameters, c.f.
   //                         module_common_types.h for the definition of
   //                         AudioFrame.
+  //   -muted              : if true, the sample data in audio_frame is not
+  //                         populated, and must be interpreted as all zero.
   //
   // Return value:
   //   -1 if the function fails,
   //    0 if the function succeeds.
   //
   virtual int32_t PlayoutData10Ms(int32_t desired_freq_hz,
-                                        AudioFrame* audio_frame) = 0;
+                                  AudioFrame* audio_frame,
+                                  bool* muted) = 0;
+
+  /////////////////////////////////////////////////////////////////////////////
+  // Same as above, but without the muted parameter. This methods should not be
+  // used if enable_fast_accelerate was set to true in NetEq::Config.
+  // TODO(henrik.lundin) Remove this method when downstream dependencies are
+  // ready.
+  virtual int32_t PlayoutData10Ms(int32_t desired_freq_hz,
+                                  AudioFrame* audio_frame) = 0;
 
   ///////////////////////////////////////////////////////////////////////////
   //   Codec specific
   //
 
   ///////////////////////////////////////////////////////////////////////////
   // int SetOpusApplication()
   // Sets the intended application if current send codec is Opus. Opus uses this
   // to optimize the encoding for applications like VOIP and music. Currently,
   // two modes are supported: kVoip and kAudio.
@@ skipping 92 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_