ff761fba82
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
110 lines
3.3 KiB
C++
110 lines
3.3 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 "webrtc/modules/include/module_common_types.h"
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#include "webrtc/modules/utility/include/audio_frame_operations.h"
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namespace webrtc {
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void AudioFrameOperations::MonoToStereo(const int16_t* src_audio,
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size_t samples_per_channel,
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int16_t* dst_audio) {
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for (size_t 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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size_t samples_per_channel,
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int16_t* dst_audio) {
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for (size_t 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 (size_t 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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}
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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 (size_t 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 (size_t 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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