Add mute state field to AudioFrame.

webrtc/voice_engine/file_recorder.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/file_recorder.h"
 
 #include <list>
 
+#include "webrtc/audio/utility/audio_frame_operations.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/platform_thread.h"
 #include "webrtc/common_audio/resampler/include/resampler.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/media_file/media_file.h"
 #include "webrtc/modules/media_file/media_file_defines.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/typedefs.h"
 #include "webrtc/voice_engine/coder.h"
@@ skipping 127 matching lines @@
     return -1;
   }
   AudioFrame tempAudioFrame;
   tempAudioFrame.samples_per_channel_ = 0;
   if (incomingAudioFrame.num_channels_ == 2 && !_moduleFile->IsStereo()) {
     // Recording mono but incoming audio is (interleaved) stereo.
     tempAudioFrame.num_channels_ = 1;
     tempAudioFrame.sample_rate_hz_ = incomingAudioFrame.sample_rate_hz_;
     tempAudioFrame.samples_per_channel_ =
         incomingAudioFrame.samples_per_channel_;
-    for (size_t i = 0; i < (incomingAudioFrame.samples_per_channel_); i++) {
-      // Sample value is the average of left and right buffer rounded to
-      // closest integer value. Note samples can be either 1 or 2 byte.
-      tempAudioFrame.data_[i] = ((incomingAudioFrame.data_[2 * i] +
-                                  incomingAudioFrame.data_[(2 * i) + 1] + 1) >>
-                                 1);
+    if (!incomingAudioFrame.muted()) {
+      AudioFrameOperations::StereoToMono(
+          incomingAudioFrame.data(), incomingAudioFrame.samples_per_channel_,
+          tempAudioFrame.mutable_data());
     }
   } else if (incomingAudioFrame.num_channels_ == 1 && _moduleFile->IsStereo()) {
     // Recording stereo but incoming audio is mono.
     tempAudioFrame.num_channels_ = 2;
     tempAudioFrame.sample_rate_hz_ = incomingAudioFrame.sample_rate_hz_;
     tempAudioFrame.samples_per_channel_ =
         incomingAudioFrame.samples_per_channel_;
-    for (size_t i = 0; i < (incomingAudioFrame.samples_per_channel_); i++) {
-      // Duplicate sample to both channels
-      tempAudioFrame.data_[2 * i] = incomingAudioFrame.data_[i];
-      tempAudioFrame.data_[2 * i + 1] = incomingAudioFrame.data_[i];
+    if (!incomingAudioFrame.muted()) {
+      AudioFrameOperations::MonoToStereo(
+          incomingAudioFrame.data(), incomingAudioFrame.samples_per_channel_,
+          tempAudioFrame.mutable_data());
     }
   }
 
   const AudioFrame* ptrAudioFrame = &incomingAudioFrame;
   if (tempAudioFrame.samples_per_channel_ != 0) {
     // If ptrAudioFrame is not empty it contains the audio to be recorded.
     ptrAudioFrame = &tempAudioFrame;
   }
 
   // Encode the audio data before writing to file. Don't encode if the codec
   // is PCM.
   // NOTE: stereo recording is only supported for WAV files.
   // TODO(hellner): WAV expect PCM in little endian byte order. Not
   // "encoding" with PCM coder should be a problem for big endian systems.
   size_t encodedLenInBytes = 0;
   if (_fileFormat == kFileFormatPreencodedFile ||
       STR_CASE_CMP(codec_info_.plname, "L16") != 0) {
     if (_audioEncoder.Encode(*ptrAudioFrame, _audioBuffer,
                              &encodedLenInBytes) == -1) {
       LOG(LS_WARNING) << "RecordAudioToFile() codec " << codec_info_.plname
                       << " not supported or failed to encode stream.";
       return -1;
     }
   } else {
     size_t outLen = 0;
     _audioResampler.ResetIfNeeded(ptrAudioFrame->sample_rate_hz_,
                                   codec_info_.plfreq,
                                   ptrAudioFrame->num_channels_);
+    // TODO(yujo): skip resample if frame is muted.
     _audioResampler.Push(
-        ptrAudioFrame->data_,
+        ptrAudioFrame->data(),
         ptrAudioFrame->samples_per_channel_ * ptrAudioFrame->num_channels_,
         reinterpret_cast<int16_t*>(_audioBuffer), MAX_AUDIO_BUFFER_IN_BYTES,
         outLen);
     encodedLenInBytes = outLen * sizeof(int16_t);
   }
 
   // Codec may not be operating at a frame rate of 10 ms. Whenever enough
   // 10 ms chunks of data has been pushed to the encoder an encoded frame
   // will be available. Wait until then.
   if (encodedLenInBytes) {
@@ skipping 32 matching lines @@
 }  // namespace
 
 std::unique_ptr<FileRecorder> FileRecorder::CreateFileRecorder(
     uint32_t instanceID,
     FileFormats fileFormat) {
   return std::unique_ptr<FileRecorder>(
       new FileRecorderImpl(instanceID, fileFormat));
 }
 
 }  // namespace webrtc