webrtc_m130/webrtc/audio/utility/audio_frame_operations.h
oprypin 67fdb80837 Reland of Enable cpplint and fix cpplint errors in webrtc/*audio (patchset #1 id:1 of https://codereview.webrtc.org/2739143002/ )
Reason for revert:
Can reland it if backwards compatible API is kept.

Original issue's description:
> Revert of Enable cpplint and fix cpplint errors in webrtc/*audio (patchset #4 id:180001 of https://codereview.webrtc.org/2683033004/ )
>
> Reason for revert:
> The API change in audio/utility/audio_frame_operations.h caused breakage. Need to keep backward-compatible API.
>
> Original issue's description:
> > Enable cpplint and fix cpplint errors in webrtc/*audio
> >
> > Change usage accordingly throughout the codebase
> >
> > BUG=webrtc:5268
> >
> > TESTED=Fixed issues reported by:
> > find webrtc/*audio -type f -name *.cc -o -name *.h | xargs cpplint.py
> >
> > Review-Url: https://codereview.webrtc.org/2683033004
> > Cr-Commit-Position: refs/heads/master@{#17133}
> > Committed: aebe55ca6c
>
> TBR=henrika@webrtc.org,henrik.lundin@webrtc.org,kwiberg@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:5268
>
> Review-Url: https://codereview.webrtc.org/2739143002
> Cr-Commit-Position: refs/heads/master@{#17138}
> Committed: e47c1d3ca1

TBR=henrika@webrtc.org,henrik.lundin@webrtc.org,kwiberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
BUG=webrtc:5268

Review-Url: https://codereview.webrtc.org/2739073003
Cr-Commit-Position: refs/heads/master@{#17144}
2017-03-09 14:25:06 +00:00

134 lines
5.4 KiB
C++

/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_AUDIO_UTILITY_AUDIO_FRAME_OPERATIONS_H_
#define WEBRTC_AUDIO_UTILITY_AUDIO_FRAME_OPERATIONS_H_
#include <stddef.h>
#include "webrtc/typedefs.h"
namespace webrtc {
class AudioFrame;
// TODO(andrew): consolidate this with utility.h and audio_frame_manipulator.h.
// Change reference parameters to pointers. Consider using a namespace rather
// than a class.
class AudioFrameOperations {
public:
// Add samples in |frame_to_add| with samples in |result_frame|
// putting the results in |results_frame|. The fields
// |vad_activity_| and |speech_type_| of the result frame are
// updated. If |result_frame| is empty (|samples_per_channel_|==0),
// the samples in |frame_to_add| are added to it. The number of
// channels and number of samples per channel must match except when
// |result_frame| is empty.
static void Add(const AudioFrame& frame_to_add, AudioFrame* result_frame);
// Upmixes mono |src_audio| to stereo |dst_audio|. This is an out-of-place
// operation, meaning src_audio and dst_audio must point to different
// buffers. It is the caller's responsibility to ensure that |dst_audio| is
// sufficiently large.
static void MonoToStereo(const int16_t* src_audio,
size_t samples_per_channel,
int16_t* dst_audio);
// |frame.num_channels_| will be updated. This version checks for sufficient
// buffer size and that |num_channels_| is mono.
static int MonoToStereo(AudioFrame* frame);
// Downmixes stereo |src_audio| to mono |dst_audio|. This is an in-place
// operation, meaning |src_audio| and |dst_audio| may point to the same
// buffer.
static void StereoToMono(const int16_t* src_audio,
size_t samples_per_channel,
int16_t* dst_audio);
// |frame.num_channels_| will be updated. This version checks that
// |num_channels_| is stereo.
static int StereoToMono(AudioFrame* frame);
// Downmixes 4 channels |src_audio| to stereo |dst_audio|. This is an in-place
// operation, meaning |src_audio| and |dst_audio| may point to the same
// buffer.
static void QuadToStereo(const int16_t* src_audio,
size_t samples_per_channel,
int16_t* dst_audio);
// |frame.num_channels_| will be updated. This version checks that
// |num_channels_| is 4 channels.
static int QuadToStereo(AudioFrame* frame);
// Downmixes 4 channels |src_audio| to mono |dst_audio|. This is an in-place
// operation, meaning |src_audio| and |dst_audio| may point to the same
// buffer.
static void QuadToMono(const int16_t* src_audio,
size_t samples_per_channel,
int16_t* dst_audio);
// |frame.num_channels_| will be updated. This version checks that
// |num_channels_| is 4 channels.
static int QuadToMono(AudioFrame* frame);
// Downmixes |src_channels| |src_audio| to |dst_channels| |dst_audio|.
// This is an in-place operation, meaning |src_audio| and |dst_audio|
// may point to the same buffer. Supported channel combinations are
// Stereo to Mono, Quad to Mono, and Quad to Stereo.
static void DownmixChannels(const int16_t* src_audio,
size_t src_channels,
size_t samples_per_channel,
size_t dst_channels,
int16_t* dst_audio);
// |frame.num_channels_| will be updated. This version checks that
// |num_channels_| and |dst_channels| are valid and performs relevant
// downmix. Supported channel combinations are Stereo to Mono, Quad to Mono,
// and Quad to Stereo.
static int DownmixChannels(size_t dst_channels, AudioFrame* frame);
// Swap the left and right channels of |frame|. Fails silently if |frame| is
// not stereo.
static void SwapStereoChannels(AudioFrame* frame);
// Conditionally zero out contents of |frame| for implementing audio mute:
// |previous_frame_muted| && |current_frame_muted| - Zero out whole frame.
// |previous_frame_muted| && !|current_frame_muted| - Fade-in at frame start.
// !|previous_frame_muted| && |current_frame_muted| - Fade-out at frame end.
// !|previous_frame_muted| && !|current_frame_muted| - Leave frame untouched.
static void Mute(AudioFrame* frame,
bool previous_frame_muted,
bool current_frame_muted);
// Zero out contents of frame.
static void Mute(AudioFrame* frame);
// Halve samples in |frame|.
static void ApplyHalfGain(AudioFrame* frame);
static int Scale(float left, float right, AudioFrame* frame);
static int Scale(float left, float right, AudioFrame& frame) { // NOLINT
// TODO(oprypin): drop this method
return Scale(left, right, &frame);
}
static int ScaleWithSat(float scale, AudioFrame* frame);
static int ScaleWithSat(float scale, AudioFrame& frame) { // NOLINT
// TODO(oprypin): drop this method
return ScaleWithSat(scale, &frame);
}
};
} // namespace webrtc
#endif // WEBRTC_AUDIO_UTILITY_AUDIO_FRAME_OPERATIONS_H_