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/*
* 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/modules/include/module_common_types.h"
#include "webrtc/modules/utility/include/audio_frame_operations.h"
namespace webrtc {
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;
}
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->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] = (src_audio[2 * i] + src_audio[2 * i + 1]) >> 1;
}
}
int AudioFrameOperations::StereoToMono(AudioFrame* frame) {
if (frame->num_channels_ != 2) {
return -1;
}
StereoToMono(frame->data_, frame->samples_per_channel_, frame->data_);
frame->num_channels_ = 1;
return 0;
}
void AudioFrameOperations::SwapStereoChannels(AudioFrame* frame) {
if (frame->num_channels_ != 2) return;
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) {
memset(frame.data_, 0, sizeof(int16_t) *
frame.samples_per_channel_ * frame.num_channels_);
}
int AudioFrameOperations::Scale(float left, float right, AudioFrame& frame) {
if (frame.num_channels_ != 2) {
return -1;
}
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) {
int32_t temp_data = 0;
// Ensure that the output result is saturated [-32768, +32767].
for (size_t i = 0; i < frame.samples_per_channel_ * frame.num_channels_;
i++) {
temp_data = static_cast<int32_t>(scale * frame.data_[i]);
if (temp_data < -32768) {
frame.data_[i] = -32768;
} else if (temp_data > 32767) {
frame.data_[i] = 32767;
} else {
frame.data_[i] = static_cast<int16_t>(temp_data);
}
}
return 0;
}
} // namespace webrtc
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