14b43beb7c
TBR=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/915006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
112 lines
3.2 KiB
C++
112 lines
3.2 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "audio_frame_operations.h"
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#include "module_common_types.h"
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namespace webrtc {
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void AudioFrameOperations::MonoToStereo(const int16_t* src_audio,
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int samples_per_channel,
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int16_t* dst_audio) {
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for (int i = 0; i < samples_per_channel; i++) {
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dst_audio[2 * i] = src_audio[i];
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dst_audio[2 * i + 1] = src_audio[i];
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}
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}
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int AudioFrameOperations::MonoToStereo(AudioFrame* frame) {
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if (frame->num_channels_ != 1) {
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return -1;
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}
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if ((frame->samples_per_channel_ * 2) >= AudioFrame::kMaxDataSizeSamples) {
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// Not enough memory to expand from mono to stereo.
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return -1;
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}
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int16_t data_copy[AudioFrame::kMaxDataSizeSamples];
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memcpy(data_copy, frame->data_,
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sizeof(int16_t) * frame->samples_per_channel_);
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MonoToStereo(data_copy, frame->samples_per_channel_, frame->data_);
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frame->num_channels_ = 2;
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return 0;
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}
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void AudioFrameOperations::StereoToMono(const int16_t* src_audio,
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int samples_per_channel,
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int16_t* dst_audio) {
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for (int i = 0; i < samples_per_channel; i++) {
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dst_audio[i] = (src_audio[2 * i] + src_audio[2 * i + 1]) >> 1;
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}
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}
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int AudioFrameOperations::StereoToMono(AudioFrame* frame) {
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if (frame->num_channels_ != 2) {
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return -1;
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}
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StereoToMono(frame->data_, frame->samples_per_channel_, frame->data_);
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frame->num_channels_ = 1;
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return 0;
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}
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void AudioFrameOperations::SwapStereoChannels(AudioFrame* frame) {
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if (frame->num_channels_ != 2) return;
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for (int i = 0; i < frame->samples_per_channel_ * 2; i += 2) {
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int16_t temp_data = frame->data_[i];
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frame->data_[i] = frame->data_[i + 1];
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frame->data_[i + 1] = temp_data;
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}
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}
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void AudioFrameOperations::Mute(AudioFrame& frame) {
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memset(frame.data_, 0, sizeof(int16_t) *
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frame.samples_per_channel_ * frame.num_channels_);
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frame.energy_ = 0;
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}
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int AudioFrameOperations::Scale(float left, float right, AudioFrame& frame) {
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if (frame.num_channels_ != 2) {
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return -1;
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}
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for (int i = 0; i < frame.samples_per_channel_; i++) {
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frame.data_[2 * i] =
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static_cast<int16_t>(left * frame.data_[2 * i]);
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frame.data_[2 * i + 1] =
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static_cast<int16_t>(right * frame.data_[2 * i + 1]);
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}
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return 0;
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}
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int AudioFrameOperations::ScaleWithSat(float scale, AudioFrame& frame) {
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int32_t temp_data = 0;
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// Ensure that the output result is saturated [-32768, +32767].
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for (int i = 0; i < frame.samples_per_channel_ * frame.num_channels_;
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i++) {
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temp_data = static_cast<int32_t>(scale * frame.data_[i]);
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if (temp_data < -32768) {
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frame.data_[i] = -32768;
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} else if (temp_data > 32767) {
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frame.data_[i] = 32767;
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} else {
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frame.data_[i] = static_cast<int16_t>(temp_data);
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}
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}
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return 0;
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}
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} // namespace webrtc
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