webrtc_m130/webrtc/audio/utility/audio_frame_operations.cc
yujo 36b1a5fcec Add mute state field to AudioFrame and switch some callers to use it. Also make AudioFrame::data_ private and instead provide:
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}
2017-06-12 19:45:32 +00:00

331 lines
10 KiB
C++

/*
* 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/audio/utility/audio_frame_operations.h"
#include <algorithm>
#include "webrtc/base/checks.h"
#include "webrtc/base/safe_conversions.h"
#include "webrtc/modules/include/module_common_types.h"
namespace webrtc {
namespace {
// 2.7ms @ 48kHz, 4ms @ 32kHz, 8ms @ 16kHz.
const size_t kMuteFadeFrames = 128;
const float kMuteFadeInc = 1.0f / kMuteFadeFrames;
} // namespace
void AudioFrameOperations::Add(const AudioFrame& frame_to_add,
AudioFrame* result_frame) {
// Sanity check.
RTC_DCHECK(result_frame);
RTC_DCHECK_GT(result_frame->num_channels_, 0);
RTC_DCHECK_EQ(result_frame->num_channels_, frame_to_add.num_channels_);
bool no_previous_data = result_frame->muted();
if (result_frame->samples_per_channel_ != frame_to_add.samples_per_channel_) {
// Special case we have no data to start with.
RTC_DCHECK_EQ(result_frame->samples_per_channel_, 0);
result_frame->samples_per_channel_ = frame_to_add.samples_per_channel_;
no_previous_data = true;
}
if (result_frame->vad_activity_ == AudioFrame::kVadActive ||
frame_to_add.vad_activity_ == AudioFrame::kVadActive) {
result_frame->vad_activity_ = AudioFrame::kVadActive;
} else if (result_frame->vad_activity_ == AudioFrame::kVadUnknown ||
frame_to_add.vad_activity_ == AudioFrame::kVadUnknown) {
result_frame->vad_activity_ = AudioFrame::kVadUnknown;
}
if (result_frame->speech_type_ != frame_to_add.speech_type_)
result_frame->speech_type_ = AudioFrame::kUndefined;
if (!frame_to_add.muted()) {
const int16_t* in_data = frame_to_add.data();
int16_t* out_data = result_frame->mutable_data();
size_t length =
frame_to_add.samples_per_channel_ * frame_to_add.num_channels_;
if (no_previous_data) {
std::copy(in_data, in_data + length, out_data);
} else {
for (size_t i = 0; i < length; i++) {
const int32_t wrap_guard = static_cast<int32_t>(out_data[i]) +
static_cast<int32_t>(in_data[i]);
out_data[i] = rtc::saturated_cast<int16_t>(wrap_guard);
}
}
}
}
void AudioFrameOperations::MonoToStereo(const int16_t* src_audio,
size_t samples_per_channel,
int16_t* dst_audio) {
for (size_t i = 0; i < samples_per_channel; i++) {
dst_audio[2 * i] = src_audio[i];
dst_audio[2 * i + 1] = src_audio[i];
}
}
int AudioFrameOperations::MonoToStereo(AudioFrame* frame) {
if (frame->num_channels_ != 1) {
return -1;
}
if ((frame->samples_per_channel_ * 2) >= AudioFrame::kMaxDataSizeSamples) {
// Not enough memory to expand from mono to stereo.
return -1;
}
if (!frame->muted()) {
// TODO(yujo): this operation can be done in place.
int16_t data_copy[AudioFrame::kMaxDataSizeSamples];
memcpy(data_copy, frame->data(),
sizeof(int16_t) * frame->samples_per_channel_);
MonoToStereo(data_copy, frame->samples_per_channel_, frame->mutable_data());
}
frame->num_channels_ = 2;
return 0;
}
void AudioFrameOperations::StereoToMono(const int16_t* src_audio,
size_t samples_per_channel,
int16_t* dst_audio) {
for (size_t i = 0; i < samples_per_channel; i++) {
dst_audio[i] =
(static_cast<int32_t>(src_audio[2 * i]) + src_audio[2 * i + 1]) >> 1;
}
}
int AudioFrameOperations::StereoToMono(AudioFrame* frame) {
if (frame->num_channels_ != 2) {
return -1;
}
RTC_DCHECK_LE(frame->samples_per_channel_ * 2,
AudioFrame::kMaxDataSizeSamples);
if (!frame->muted()) {
StereoToMono(frame->data(), frame->samples_per_channel_,
frame->mutable_data());
}
frame->num_channels_ = 1;
return 0;
}
void AudioFrameOperations::QuadToStereo(const int16_t* src_audio,
size_t samples_per_channel,
int16_t* dst_audio) {
for (size_t i = 0; i < samples_per_channel; i++) {
dst_audio[i * 2] =
(static_cast<int32_t>(src_audio[4 * i]) + src_audio[4 * i + 1]) >> 1;
dst_audio[i * 2 + 1] =
(static_cast<int32_t>(src_audio[4 * i + 2]) + src_audio[4 * i + 3]) >>
1;
}
}
int AudioFrameOperations::QuadToStereo(AudioFrame* frame) {
if (frame->num_channels_ != 4) {
return -1;
}
RTC_DCHECK_LE(frame->samples_per_channel_ * 4,
AudioFrame::kMaxDataSizeSamples);
if (!frame->muted()) {
QuadToStereo(frame->data(), frame->samples_per_channel_,
frame->mutable_data());
}
frame->num_channels_ = 2;
return 0;
}
void AudioFrameOperations::QuadToMono(const int16_t* src_audio,
size_t samples_per_channel,
int16_t* dst_audio) {
for (size_t i = 0; i < samples_per_channel; i++) {
dst_audio[i] =
(static_cast<int32_t>(src_audio[4 * i]) + src_audio[4 * i + 1] +
src_audio[4 * i + 2] + src_audio[4 * i + 3]) >> 2;
}
}
int AudioFrameOperations::QuadToMono(AudioFrame* frame) {
if (frame->num_channels_ != 4) {
return -1;
}
RTC_DCHECK_LE(frame->samples_per_channel_ * 4,
AudioFrame::kMaxDataSizeSamples);
if (!frame->muted()) {
QuadToMono(frame->data(), frame->samples_per_channel_,
frame->mutable_data());
}
frame->num_channels_ = 1;
return 0;
}
void AudioFrameOperations::DownmixChannels(const int16_t* src_audio,
size_t src_channels,
size_t samples_per_channel,
size_t dst_channels,
int16_t* dst_audio) {
if (src_channels == 2 && dst_channels == 1) {
StereoToMono(src_audio, samples_per_channel, dst_audio);
return;
} else if (src_channels == 4 && dst_channels == 2) {
QuadToStereo(src_audio, samples_per_channel, dst_audio);
return;
} else if (src_channels == 4 && dst_channels == 1) {
QuadToMono(src_audio, samples_per_channel, dst_audio);
return;
}
RTC_NOTREACHED() << "src_channels: " << src_channels
<< ", dst_channels: " << dst_channels;
}
int AudioFrameOperations::DownmixChannels(size_t dst_channels,
AudioFrame* frame) {
if (frame->num_channels_ == 2 && dst_channels == 1) {
return StereoToMono(frame);
} else if (frame->num_channels_ == 4 && dst_channels == 2) {
return QuadToStereo(frame);
} else if (frame->num_channels_ == 4 && dst_channels == 1) {
return QuadToMono(frame);
}
return -1;
}
void AudioFrameOperations::SwapStereoChannels(AudioFrame* frame) {
RTC_DCHECK(frame);
if (frame->num_channels_ != 2 || frame->muted()) {
return;
}
int16_t* frame_data = frame->mutable_data();
for (size_t i = 0; i < frame->samples_per_channel_ * 2; i += 2) {
int16_t temp_data = frame_data[i];
frame_data[i] = frame_data[i + 1];
frame_data[i + 1] = temp_data;
}
}
void AudioFrameOperations::Mute(AudioFrame* frame,
bool previous_frame_muted,
bool current_frame_muted) {
RTC_DCHECK(frame);
if (!previous_frame_muted && !current_frame_muted) {
// Not muted, don't touch.
} else if (previous_frame_muted && current_frame_muted) {
// Frame fully muted.
size_t total_samples = frame->samples_per_channel_ * frame->num_channels_;
RTC_DCHECK_GE(AudioFrame::kMaxDataSizeSamples, total_samples);
frame->Mute();
} else {
// Fade is a no-op on a muted frame.
if (frame->muted()) {
return;
}
// Limit number of samples to fade, if frame isn't long enough.
size_t count = kMuteFadeFrames;
float inc = kMuteFadeInc;
if (frame->samples_per_channel_ < kMuteFadeFrames) {
count = frame->samples_per_channel_;
if (count > 0) {
inc = 1.0f / count;
}
}
size_t start = 0;
size_t end = count;
float start_g = 0.0f;
if (current_frame_muted) {
// Fade out the last |count| samples of frame.
RTC_DCHECK(!previous_frame_muted);
start = frame->samples_per_channel_ - count;
end = frame->samples_per_channel_;
start_g = 1.0f;
inc = -inc;
} else {
// Fade in the first |count| samples of frame.
RTC_DCHECK(previous_frame_muted);
}
// Perform fade.
int16_t* frame_data = frame->mutable_data();
size_t channels = frame->num_channels_;
for (size_t j = 0; j < channels; ++j) {
float g = start_g;
for (size_t i = start * channels; i < end * channels; i += channels) {
g += inc;
frame_data[i + j] *= g;
}
}
}
}
void AudioFrameOperations::Mute(AudioFrame* frame) {
Mute(frame, true, true);
}
void AudioFrameOperations::ApplyHalfGain(AudioFrame* frame) {
RTC_DCHECK(frame);
RTC_DCHECK_GT(frame->num_channels_, 0);
if (frame->num_channels_ < 1 || frame->muted()) {
return;
}
int16_t* frame_data = frame->mutable_data();
for (size_t i = 0; i < frame->samples_per_channel_ * frame->num_channels_;
i++) {
frame_data[i] = frame_data[i] >> 1;
}
}
int AudioFrameOperations::Scale(float left, float right, AudioFrame* frame) {
if (frame->num_channels_ != 2) {
return -1;
} else if (frame->muted()) {
return 0;
}
int16_t* frame_data = frame->mutable_data();
for (size_t i = 0; i < frame->samples_per_channel_; i++) {
frame_data[2 * i] = static_cast<int16_t>(left * frame_data[2 * i]);
frame_data[2 * i + 1] = static_cast<int16_t>(right * frame_data[2 * i + 1]);
}
return 0;
}
int AudioFrameOperations::ScaleWithSat(float scale, AudioFrame* frame) {
if (frame->muted()) {
return 0;
}
int16_t* frame_data = frame->mutable_data();
for (size_t i = 0; i < frame->samples_per_channel_ * frame->num_channels_;
i++) {
frame_data[i] = rtc::saturated_cast<int16_t>(scale * frame_data[i]);
}
return 0;
}
} // namespace webrtc