4ecea3e105
We previously had an error when a mono capture device was used with a stereo codec. This is prevented by avoiding any remixing in AudioProcessing. Instead, capture side downmixing is now done before resampling. Upmixing can now be handled properly by AudioCoding, since the AudioProcessing error condition has been removed. On the render side, downmixing now occurs before resampling. Ideally this would be handled still earlier in the chain. Similarly, downmixing for the AudioProcessing reference data occurs before resampling. This code has been refactored into RemixAndResample, with a comprehensive unittest added in output_mixer_unittest.cc. BUG=issue624 TEST=manually through voe_cmd_test, by using mono and stereo capture and render devices with mono and stereo codecs. voice_engine_unittest, voe_auto_test. Review URL: https://webrtc-codereview.appspot.com/676004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2448 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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