Add mute state field to AudioFrame.

webrtc/modules/audio_coding/acm2/audio_coding_module.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.
  */
 
@@ skipping 317 matching lines @@
   return (STR_CASE_CMP(codec.plname, "CN") == 0);
 }
 
 // Stereo-to-mono can be used as in-place.
 int DownMix(const AudioFrame& frame,
             size_t length_out_buff,
             int16_t* out_buff) {
   if (length_out_buff < frame.samples_per_channel_) {
     return -1;
   }
-  for (size_t n = 0; n < frame.samples_per_channel_; ++n)
-    out_buff[n] = (frame.data_[2 * n] + frame.data_[2 * n + 1]) >> 1;
+  if (!frame.muted()) {
+    const int16_t* frame_data = frame.data();
+    for (size_t n = 0; n < frame.samples_per_channel_; ++n)
+      out_buff[n] = (frame_data[2 * n] + frame_data[2 * n + 1]) >> 1;
+  } else {
+    memset(out_buff, 0, 2 * frame.samples_per_channel_);
+  }
   return 0;
 }
 
 // Mono-to-stereo can be used as in-place.
 int UpMix(const AudioFrame& frame, size_t length_out_buff, int16_t* out_buff) {

[hlundin-webrtc, 2017/03/16 14:47:48]

Add special case for frame.muted() here too, or drop it in DownMix. I think in
practice we will never have a muted frame in the send chain, so optimization
here is not likely to be worth the extra code.

[yujo, 2017/03/16 23:37:21]

Done. Also noticed my bug above...

   if (length_out_buff < frame.samples_per_channel_) {
     return -1;
   }
+  const int16_t* frame_data = frame.data();
   for (size_t n = frame.samples_per_channel_; n != 0; --n) {
     size_t i = n - 1;
-    int16_t sample = frame.data_[i];
+    int16_t sample = frame_data[i];
     out_buff[2 * i + 1] = sample;
     out_buff[2 * i] = sample;
   }
   return 0;
 }
 
 void ConvertEncodedInfoToFragmentationHeader(
     const AudioEncoder::EncodedInfo& info,
     RTPFragmentationHeader* frag) {
   if (info.redundant.empty()) {
@@ skipping 364 matching lines @@
       if (UpMix(*ptr_frame, WEBRTC_10MS_PCM_AUDIO, input_data->buffer) < 0)
         return -1;
     } else {
       if (DownMix(*ptr_frame, WEBRTC_10MS_PCM_AUDIO, input_data->buffer) < 0)
         return -1;
     }
   }
 
   // When adding data to encoders this pointer is pointing to an audio buffer
   // with correct number of channels.
-  const int16_t* ptr_audio = ptr_frame->data_;
+  const int16_t* ptr_audio = ptr_frame->data();
 
   // For pushing data to primary, point the |ptr_audio| to correct buffer.
   if (!same_num_channels)
     ptr_audio = input_data->buffer;
 
+  // TODO(yujo): Skip encode of muted frames.

[hlundin-webrtc, 2017/03/16 14:47:48]

See above: I don't think we will have muted input frames, right?

[yujo, 2017/03/16 23:37:21]

Muted input is actually my motivation for this whole change:
https://cs.chromium.org/chromium/src/third_party/webrtc/voice_engine/channel....

[hlundin-webrtc, 2017/03/17 14:29:38]

Oh! Then I see. I though you were only in it to shave off complexity for muted
output channels.

   input_data->input_timestamp = ptr_frame->timestamp_;
   input_data->audio = ptr_audio;
   input_data->length_per_channel = ptr_frame->samples_per_channel_;
   input_data->audio_channel = current_num_channels;
 
   return 0;
 }
 
 // Perform a resampling and down-mix if required. We down-mix only if
 // encoder is mono and input is stereo. In case of dual-streaming, both
 // encoders has to be mono for down-mix to take place.
 // |*ptr_out| will point to the pre-processed audio-frame. If no pre-processing
 // is required, |*ptr_out| points to |in_frame|.
+// TODO(yujo): Make this more efficient for muted frames.
 int AudioCodingModuleImpl::PreprocessToAddData(const AudioFrame& in_frame,
                                                const AudioFrame** ptr_out) {
   const bool resample =
       in_frame.sample_rate_hz_ != encoder_stack_->SampleRateHz();
 
   // This variable is true if primary codec and secondary codec (if exists)
   // are both mono and input is stereo.
   // TODO(henrik.lundin): This condition should probably be
   //   in_frame.num_channels_ > encoder_stack_->NumChannels()
   const bool down_mix =
@@ skipping 29 matching lines @@
     }
 
     expected_in_ts_ += static_cast<uint32_t>(in_frame.samples_per_channel_);
     expected_codec_ts_ += static_cast<uint32_t>(in_frame.samples_per_channel_);
     return 0;
   }
 
   *ptr_out = &preprocess_frame_;
   preprocess_frame_.num_channels_ = in_frame.num_channels_;
   int16_t audio[WEBRTC_10MS_PCM_AUDIO];
-  const int16_t* src_ptr_audio = in_frame.data_;
-  int16_t* dest_ptr_audio = preprocess_frame_.data_;
+  const int16_t* src_ptr_audio = in_frame.data();
   if (down_mix) {
     // If a resampling is required the output of a down-mix is written into a
     // local buffer, otherwise, it will be written to the output frame.
-    if (resample)
-      dest_ptr_audio = audio;
+    int16_t* dest_ptr_audio = resample ?
+        audio : preprocess_frame_.mutable_data();
     if (DownMix(in_frame, WEBRTC_10MS_PCM_AUDIO, dest_ptr_audio) < 0)
       return -1;
     preprocess_frame_.num_channels_ = 1;
     // Set the input of the resampler is the down-mixed signal.
     src_ptr_audio = audio;
   }
 
   preprocess_frame_.timestamp_ = expected_codec_ts_;
   preprocess_frame_.samples_per_channel_ = in_frame.samples_per_channel_;
   preprocess_frame_.sample_rate_hz_ = in_frame.sample_rate_hz_;
   // If it is required, we have to do a resampling.
   if (resample) {
     // The result of the resampler is written to output frame.
-    dest_ptr_audio = preprocess_frame_.data_;
+    int16_t* dest_ptr_audio = preprocess_frame_.mutable_data();
 
     int samples_per_channel = resampler_.Resample10Msec(
         src_ptr_audio, in_frame.sample_rate_hz_, encoder_stack_->SampleRateHz(),
         preprocess_frame_.num_channels_, AudioFrame::kMaxDataSizeSamples,
         dest_ptr_audio);
 
     if (samples_per_channel < 0) {
       WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
                    "Cannot add 10 ms audio, resampling failed");
       return -1;
@@ skipping 542 matching lines @@
 // Checks the validity of the parameters of the given codec
 bool AudioCodingModule::IsCodecValid(const CodecInst& codec) {
   bool valid = acm2::RentACodec::IsCodecValid(codec);
   if (!valid)
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, -1,
                  "Invalid codec setting");
   return valid;
 }
 
 }  // namespace webrtc