Add mute state field to AudioFrame.

webrtc/voice_engine/utility.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/voice_engine/utility.h"
 
 #include "webrtc/audio/utility/audio_frame_operations.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/voice_engine/voice_engine_defines.h"
 
 namespace webrtc {
 namespace voe {
 
 void RemixAndResample(const AudioFrame& src_frame,
                       PushResampler<int16_t>* resampler,
                       AudioFrame* dst_frame) {
-  RemixAndResample(src_frame.data_, src_frame.samples_per_channel_,
+  RemixAndResample(src_frame.data(), src_frame.samples_per_channel_,
                    src_frame.num_channels_, src_frame.sample_rate_hz_,
                    resampler, dst_frame);
   dst_frame->timestamp_ = src_frame.timestamp_;
   dst_frame->elapsed_time_ms_ = src_frame.elapsed_time_ms_;
   dst_frame->ntp_time_ms_ = src_frame.ntp_time_ms_;
 }
 
 void RemixAndResample(const int16_t* src_data,
                       size_t samples_per_channel,
                       size_t num_channels,
@@ skipping 18 matching lines @@
     audio_ptr_num_channels = dst_frame->num_channels_;
   }
 
   if (resampler->InitializeIfNeeded(sample_rate_hz, dst_frame->sample_rate_hz_,
                                     audio_ptr_num_channels) == -1) {
     FATAL() << "InitializeIfNeeded failed: sample_rate_hz = " << sample_rate_hz
             << ", dst_frame->sample_rate_hz_ = " << dst_frame->sample_rate_hz_
             << ", audio_ptr_num_channels = " << audio_ptr_num_channels;
   }
 
+  // TODO(yujo): for muted input frames, don't resample. Either 1) allow
+  // resampler to return output length without doing the resample, so we know
+  // how much to zero here; or 2) make resampler accept a hint that the input is
+  // zeroed.
   const size_t src_length = samples_per_channel * audio_ptr_num_channels;
-  int out_length = resampler->Resample(audio_ptr, src_length, dst_frame->data_,
+  int out_length = resampler->Resample(audio_ptr, src_length,
+                                       dst_frame->mutable_data(),
                                        AudioFrame::kMaxDataSizeSamples);
   if (out_length == -1) {
     FATAL() << "Resample failed: audio_ptr = " << audio_ptr
             << ", src_length = " << src_length
-            << ", dst_frame->data_ = " << dst_frame->data_;
+            << ", dst_frame->mutable_data() = " << dst_frame->mutable_data();
   }
   dst_frame->samples_per_channel_ = out_length / audio_ptr_num_channels;
 
   // Upmix after resampling.
   if (num_channels == 1 && dst_frame->num_channels_ == 2) {
     // The audio in dst_frame really is mono at this point; MonoToStereo will
     // set this back to stereo.
     dst_frame->num_channels_ = 1;
     AudioFrameOperations::MonoToStereo(dst_frame);
   }
@@ skipping 30 matching lines @@
     int32_t temp = 0;
     for (size_t i = 0; i < source_len; ++i) {
       temp = source[i] + target[i];
       target[i] = WebRtcSpl_SatW32ToW16(temp);
     }
   }
 }
 
 }  // namespace voe
 }  // namespace webrtc