| Index: webrtc/modules/utility/source/audio_frame_operations.cc
|
| diff --git a/webrtc/modules/utility/source/audio_frame_operations.cc b/webrtc/modules/utility/source/audio_frame_operations.cc
|
| deleted file mode 100644
|
| index 102407d0f0f0fd223f6eaf9836dfa2b18278a5f7..0000000000000000000000000000000000000000
|
| --- a/webrtc/modules/utility/source/audio_frame_operations.cc
|
| +++ /dev/null
|
| @@ -1,160 +0,0 @@
|
| -/*
|
| - * 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"
|
| -#include "webrtc/base/checks.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::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, 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);
|
| - memset(frame->data_, 0, sizeof(frame->data_[0]) * total_samples);
|
| - } else {
|
| - // 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.
|
| - 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;
|
| - }
|
| - }
|
| - }
|
| -}
|
| -
|
| -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
|
|
|