6321b49a0d
This CL is in preparation to move the AudioFrame into webrtc/api. The AudioFrame is a POD type used for representing 10ms of audio. It appears as a parameter and return value of interfaces being migrated to webrtc/api, in particular AudioMixer. Here, methods operator+=, operator>>=, Mute are moved into a new target webrtc/audio/utility/audio_frame_operations, and dependencies are changed to use the new versions. The old AudioFrame methods are marked deprecated. The audio frame utilities in webrtc/modules/utility:audio_frame_operations are also moved to the new location. TBR=kjellander@webrtc.org BUG=webrtc:6548 NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2424173003 Cr-Commit-Position: refs/heads/master@{#15413}
84 lines
3.3 KiB
C++
84 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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#ifndef WEBRTC_AUDIO_UTILITY_AUDIO_FRAME_OPERATIONS_H_
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#define WEBRTC_AUDIO_UTILITY_AUDIO_FRAME_OPERATIONS_H_
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#include <stddef.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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// Add samples in |frame_to_add| with samples in |result_frame|
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// putting the results in |results_frame|. The fields
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// |vad_activity_| and |speech_type_| of the result frame are
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// updated. If |result_frame| is empty (|samples_per_channel_|==0),
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// the samples in |frame_to_add| are added to it. The number of
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// channels and number of samples per channel must match except when
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// |result_frame| is empty.
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static void Add(const AudioFrame& frame_to_add, AudioFrame* result_frame);
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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,
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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,
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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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// Conditionally zero out contents of |frame| for implementing audio mute:
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// |previous_frame_muted| && |current_frame_muted| - Zero out whole frame.
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// |previous_frame_muted| && !|current_frame_muted| - Fade-in at frame start.
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// !|previous_frame_muted| && |current_frame_muted| - Fade-out at frame end.
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// !|previous_frame_muted| && !|current_frame_muted| - Leave frame untouched.
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static void Mute(AudioFrame* frame,
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bool previous_frame_muted,
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bool current_frame_muted);
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// Zero out contents of frame.
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static void Mute(AudioFrame* frame);
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// Halve samples in |frame|.
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static void ApplyHalfGain(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 // WEBRTC_AUDIO_UTILITY_AUDIO_FRAME_OPERATIONS_H_
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