Add mute state field to AudioFrame.

webrtc/modules/audio_coding/neteq/neteq_impl.cc

 /*
  *  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.
  */
 
 #include "webrtc/modules/audio_coding/neteq/neteq_impl.h"
 
 #include <assert.h>
-#include <memory.h>  // memset
 
 #include <algorithm>
 #include <utility>
 #include <vector>
 
 #include "webrtc/api/audio_codecs/audio_decoder.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/safe_conversions.h"
 #include "webrtc/base/sanitizer.h"
@@ skipping 1031 matching lines @@
     RTC_DCHECK_GE(sync_buffer_->next_index(), missing_lookahead_samples);
     sync_buffer_->set_next_index(sync_buffer_->next_index() -
                                  missing_lookahead_samples);
   }
   if (audio_frame->samples_per_channel_ != output_size_samples_) {
     LOG(LS_ERROR) << "audio_frame->samples_per_channel_ ("
                   << audio_frame->samples_per_channel_
                   << ") != output_size_samples_ (" << output_size_samples_
                   << ")";
     // TODO(minyue): treatment of under-run, filling zeros
-    memset(audio_frame->data_, 0, num_output_samples * sizeof(int16_t));
+    audio_frame->Mute();
     return kSampleUnderrun;
   }
 
   // Should always have overlap samples left in the |sync_buffer_|.
   RTC_DCHECK_GE(sync_buffer_->FutureLength(), expand_->overlap_length());
 
+  // TODO(yujo): For muted frames, this can be a copy rather than an addition.
   if (play_dtmf) {
-    return_value =
-        DtmfOverdub(dtmf_event, sync_buffer_->Channels(), audio_frame->data_);
+    return_value = DtmfOverdub(dtmf_event, sync_buffer_->Channels(),
+                               audio_frame->mutable_data());
   }
 
   // Update the background noise parameters if last operation wrote data
   // straight from the decoder to the |sync_buffer_|. That is, none of the
   // operations that modify the signal can be followed by a parameter update.
   if ((last_mode_ == kModeNormal) ||
       (last_mode_ == kModeAccelerateFail) ||
       (last_mode_ == kModePreemptiveExpandFail) ||
       (last_mode_ == kModeRfc3389Cng) ||
       (last_mode_ == kModeCodecInternalCng)) {
@@ skipping 1094 matching lines @@
   }
 }
 
 void NetEqImpl::CreateDecisionLogic() {
   decision_logic_.reset(DecisionLogic::Create(
       fs_hz_, output_size_samples_, playout_mode_, decoder_database_.get(),
       *packet_buffer_.get(), delay_manager_.get(), buffer_level_filter_.get(),
       tick_timer_.get()));
 }
 }  // namespace webrtc