blob: 1a8232b02b3ae4bed9b2dcf21e8c1361eb1cd801 [file] [log] [blame]
/*
* 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 "audio/utility/audio_frame_operations.h"
#include <string.h>
#include <algorithm>
#include <cstdint>
#include "rtc_base/checks.h"
#include "rtc_base/numerics/safe_conversions.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