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}
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_MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_
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#define WEBRTC_MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_
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#include "webrtc/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, size_t 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, size_t 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_MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_
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