21299d4e00
We want to remove energy_ entirely as we've seen that carrying around this potentially invalid value is dangerous. Results in the removal of AudioBuffer::is_muted(). This wasn't used in practice any longer, after the level calculation moved directly to channel.cc Instead, now use ProcessMuted() in channel.cc, to shortcut the level computation when the signal is muted. BUG=3315 TESTED=Muting the channel in voe_cmd_test results in rms=127. R=bjornv@webrtc.org, kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/12529004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6159 4adac7df-926f-26a2-2b94-8c16560cd09d
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/interface/module_common_types.h"
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#include "webrtc/modules/utility/interface/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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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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}
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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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