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
59 lines
2.2 KiB
C++
59 lines
2.2 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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#ifndef WEBRTC_VOICE_ENGINE_AUDIO_FRAME_OPERATIONS_H_
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#define WEBRTC_VOICE_ENGINE_AUDIO_FRAME_OPERATIONS_H_
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#include "typedefs.h"
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namespace webrtc {
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class AudioFrame;
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// TODO(andrew): consolidate this with utility.h and audio_frame_manipulator.h.
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// Change reference parameters to pointers. Consider using a namespace rather
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// than a class.
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class AudioFrameOperations {
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public:
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// Upmixes mono |src_audio| to stereo |dst_audio|. This is an out-of-place
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// operation, meaning src_audio and dst_audio must point to different
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// buffers. It is the caller's responsibility to ensure that |dst_audio| is
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// sufficiently large.
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static void MonoToStereo(const int16_t* src_audio, int samples_per_channel,
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int16_t* dst_audio);
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// |frame.num_channels_| will be updated. This version checks for sufficient
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// buffer size and that |num_channels_| is mono.
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static int MonoToStereo(AudioFrame* frame);
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// Downmixes stereo |src_audio| to mono |dst_audio|. This is an in-place
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// operation, meaning |src_audio| and |dst_audio| may point to the same
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// buffer.
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static void StereoToMono(const int16_t* src_audio, int samples_per_channel,
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int16_t* dst_audio);
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// |frame.num_channels_| will be updated. This version checks that
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// |num_channels_| is stereo.
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static int StereoToMono(AudioFrame* frame);
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// Swap the left and right channels of |frame|. Fails silently if |frame| is
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// not stereo.
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static void SwapStereoChannels(AudioFrame* frame);
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// Zeros out the audio and sets |frame.energy| to zero.
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static void Mute(AudioFrame& frame);
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static int Scale(float left, float right, AudioFrame& frame);
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static int ScaleWithSat(float scale, AudioFrame& frame);
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};
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} // namespace webrtc
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#endif // #ifndef WEBRTC_VOICE_ENGINE_AUDIO_FRAME_OPERATIONS_H_
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