Add muted_output parameter to ACM

webrtc/modules/audio_coding/acm2/acm_receiver.h

 /*
  *  Copyright (c) 2013 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 64 matching lines @@
   //
   // Input:
   //   -desired_freq_hz       : specifies the sampling rate [Hz] of the output
   //                            audio. If set -1 indicates to resampling is
   //                            is required and the audio returned at the
   //                            sampling rate of the decoder.
   //
   // Output:
   //   -audio_frame           : an audio frame were output data and
   //                            associated parameters are written to.
+  //   -muted                 : if true, the sample data in audio_frame is not
+  //                            populated, and must be interpreted as all zero.
   //
   // Return value             : 0 if OK.
   //                           -1 if NetEq returned an error.
   //
-  int GetAudio(int desired_freq_hz, AudioFrame* audio_frame);
+  int GetAudio(int desired_freq_hz, AudioFrame* audio_frame, bool* muted);
 
   //
   // Adds a new codec to the NetEq codec database.
   //
   // Input:
   //   - acm_codec_id        : ACM codec ID; -1 means external decoder.
   //   - payload_type        : payload type.
   //   - sample_rate_hz      : sample rate.
   //   - audio_decoder       : pointer to a decoder object. If it's null, then
   //                           NetEq will internally create a decoder object
@@ skipping 170 matching lines @@
   Clock* clock_;  // TODO(henrik.lundin) Make const if possible.
   bool resampled_last_output_frame_ GUARDED_BY(crit_sect_);
   rtc::Optional<int> last_packet_sample_rate_hz_ GUARDED_BY(crit_sect_);
 };
 
 }  // namespace acm2
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_