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Add mute state field to AudioFrame and switch some callers to use it. Also make AudioFrame::data_ private and instead provide:

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const int16_t* data() const;
int16_t* mutable_data();

- data() returns a zeroed static buffer on muted frames (to avoid unnecessary zeroing of the member buffer) and directly returns AudioFrame::data_ on unmuted frames.
- mutable_data(), lazily zeroes AudioFrame::data_ if the frame is currently muted, sets muted=false, and returns AudioFrame::data_.

These accessors serve to "force" callers to be aware of the mute state field, i.e. lazy zeroing is not the primary motivation.

This change only optimizes handling of muted frames where it is somewhat trivial to do so. Other improvements requiring more significant structural changes will come later.

BUG=webrtc:7343
TBR=henrika

Review-Url: https://codereview.webrtc.org/2750783004
Cr-Commit-Position: refs/heads/master@{#18543}
This commit is contained in:
yujo
2017-06-12 12:45:32 -07:00
committed by Commit Bot
parent 0703856b53
commit 36b1a5fcec
55 changed files with 658 additions and 316 deletions
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50
webrtc/modules/audio_coding/acm2/audio_coding_module.cc
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@ -325,24 +325,37 @@ void UpdateCodecTypeHistogram(size_t codec_type) {
int DownMix(const AudioFrame& frame,
size_t length_out_buff,
int16_t* out_buff) {
if (length_out_buff < frame.samples_per_channel_) {
return -1;
RTC_DCHECK_EQ(frame.num_channels_, 2);
RTC_DCHECK_GE(length_out_buff, frame.samples_per_channel_);
if (!frame.muted()) {
const int16_t* frame_data = frame.data();
for (size_t n = 0; n < frame.samples_per_channel_; ++n) {
out_buff[n] = static_cast<int16_t>(
(static_cast<int32_t>(frame_data[2 * n]) +
static_cast<int32_t>(frame_data[2 * n + 1])) >> 1);
}
} else {
memset(out_buff, 0, frame.samples_per_channel_);
}
for (size_t n = 0; n < frame.samples_per_channel_; ++n)
out_buff[n] = (frame.data_[2 * n] + frame.data_[2 * n + 1]) >> 1;
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) {
if (length_out_buff < frame.samples_per_channel_) {
return -1;
}
for (size_t n = frame.samples_per_channel_; n != 0; --n) {
size_t i = n - 1;
int16_t sample = frame.data_[i];
out_buff[2 * i + 1] = sample;
out_buff[2 * i] = sample;
RTC_DCHECK_EQ(frame.num_channels_, 1);
RTC_DCHECK_GE(length_out_buff, 2 * frame.samples_per_channel_);
if (!frame.muted()) {
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];
out_buff[2 * i + 1] = sample;
out_buff[2 * i] = sample;
}
} else {
memset(out_buff, 0, 2 * frame.samples_per_channel_);
}
return 0;
}
@ -725,12 +738,13 @@ int AudioCodingModuleImpl::Add10MsDataInternal(const AudioFrame& audio_frame,
// 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.
input_data->input_timestamp = ptr_frame->timestamp_;
input_data->audio = ptr_audio;
input_data->length_per_channel = ptr_frame->samples_per_channel_;
@ -744,6 +758,7 @@ int AudioCodingModuleImpl::Add10MsDataInternal(const AudioFrame& audio_frame,
// 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 =
@ -793,13 +808,12 @@ int AudioCodingModuleImpl::PreprocessToAddData(const AudioFrame& in_frame,
*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;
@ -813,7 +827,7 @@ int AudioCodingModuleImpl::PreprocessToAddData(const AudioFrame& in_frame,
// 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(),