blob: 2a5f29f4f53e2fc53dff1d77280b9a2f3aa52c8b [file] [log] [blame]
/*
* 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 AUDIO_UTILITY_AUDIO_FRAME_OPERATIONS_H_
#define AUDIO_UTILITY_AUDIO_FRAME_OPERATIONS_H_
#include <stddef.h>
#include <stdint.h>
#include "absl/base/attributes.h"
#include "api/audio/audio_frame.h"
namespace webrtc {
// 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);
// `frame.num_channels_` will be updated. This version checks for sufficient
// buffer size and that `num_channels_` is mono. Use UpmixChannels
// instead. TODO(bugs.webrtc.org/8649): remove.
ABSL_DEPRECATED("bugs.webrtc.org/8649")
static int MonoToStereo(AudioFrame* frame);
// `frame.num_channels_` will be updated. This version checks that
// `num_channels_` is stereo. Use DownmixChannels
// instead. TODO(bugs.webrtc.org/8649): remove.
ABSL_DEPRECATED("bugs.webrtc.org/8649")
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 `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 N channels to Mono, and Quad to Stereo.
static void DownmixChannels(size_t dst_channels, AudioFrame* frame);
// `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 Mono to N
// channels. The single channel is replicated.
static void UpmixChannels(size_t target_number_of_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 ScaleWithSat(float scale, AudioFrame* frame);
};
} // namespace webrtc
#endif // AUDIO_UTILITY_AUDIO_FRAME_OPERATIONS_H_