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Update to current webrtc library

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This is from the upstream library commit id
3326535126e435f1ba647885ce43a8f0f3d317eb, corresponding to Chromium
88.0.4290.1.
This commit is contained in:
Arun Raghavan
2020-10-12 18:08:02 -04:00
parent b1b02581d3
commit bcec8b0b21
859 changed files with 76187 additions and 49580 deletions
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225
webrtc/modules/audio_processing/audio_buffer.h
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@ -8,156 +8,171 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_BUFFER_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_BUFFER_H_
#ifndef MODULES_AUDIO_PROCESSING_AUDIO_BUFFER_H_
#define MODULES_AUDIO_PROCESSING_AUDIO_BUFFER_H_
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_audio/channel_buffer.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/modules/audio_processing/splitting_filter.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/system_wrappers/include/scoped_vector.h"
#include "webrtc/typedefs.h"
#include <stddef.h>
#include <stdint.h>
#include <memory>
#include <vector>
#include "common_audio/channel_buffer.h"
#include "modules/audio_processing/include/audio_processing.h"
namespace webrtc {
class PushSincResampler;
class IFChannelBuffer;
class SplittingFilter;
enum Band {
kBand0To8kHz = 0,
kBand8To16kHz = 1,
kBand16To24kHz = 2
};
enum Band { kBand0To8kHz = 0, kBand8To16kHz = 1, kBand16To24kHz = 2 };
// Stores any audio data in a way that allows the audio processing module to
// operate on it in a controlled manner.
class AudioBuffer {
public:
// TODO(ajm): Switch to take ChannelLayouts.
static const int kSplitBandSize = 160;
static const size_t kMaxSampleRate = 384000;
AudioBuffer(size_t input_rate,
size_t input_num_channels,
size_t buffer_rate,
size_t buffer_num_channels,
size_t output_rate,
size_t output_num_channels);
// The constructor below will be deprecated.
AudioBuffer(size_t input_num_frames,
int num_input_channels,
size_t process_num_frames,
int num_process_channels,
size_t input_num_channels,
size_t buffer_num_frames,
size_t buffer_num_channels,
size_t output_num_frames);
virtual ~AudioBuffer();
int num_channels() const;
void set_num_channels(int num_channels);
size_t num_frames() const;
size_t num_frames_per_band() const;
size_t num_keyboard_frames() const;
size_t num_bands() const;
AudioBuffer(const AudioBuffer&) = delete;
AudioBuffer& operator=(const AudioBuffer&) = delete;
// Returns a pointer array to the full-band channels.
// Specify that downmixing should be done by selecting a single channel.
void set_downmixing_to_specific_channel(size_t channel);
// Specify that downmixing should be done by averaging all channels,.
void set_downmixing_by_averaging();
// Set the number of channels in the buffer. The specified number of channels
// cannot be larger than the specified buffer_num_channels. The number is also
// reset at each call to CopyFrom or InterleaveFrom.
void set_num_channels(size_t num_channels);
size_t num_channels() const { return num_channels_; }
size_t num_frames() const { return buffer_num_frames_; }
size_t num_frames_per_band() const { return num_split_frames_; }
size_t num_bands() const { return num_bands_; }
// Returns pointer arrays to the full-band channels.
// Usage:
// channels()[channel][sample].
// Where:
// 0 <= channel < |num_proc_channels_|
// 0 <= sample < |proc_num_frames_|
int16_t* const* channels();
const int16_t* const* channels_const() const;
float* const* channels_f();
const float* const* channels_const_f() const;
// 0 <= channel < |buffer_num_channels_|
// 0 <= sample < |buffer_num_frames_|
float* const* channels() { return data_->channels(); }
const float* const* channels_const() const { return data_->channels(); }
// Returns a pointer array to the bands for a specific channel.
// Returns pointer arrays to the bands for a specific channel.
// Usage:
// split_bands(channel)[band][sample].
// Where:
// 0 <= channel < |num_proc_channels_|
// 0 <= channel < |buffer_num_channels_|
// 0 <= band < |num_bands_|
// 0 <= sample < |num_split_frames_|
int16_t* const* split_bands(int channel);
const int16_t* const* split_bands_const(int channel) const;
float* const* split_bands_f(int channel);
const float* const* split_bands_const_f(int channel) const;
const float* const* split_bands_const(size_t channel) const {
return split_data_.get() ? split_data_->bands(channel)
: data_->bands(channel);
}
float* const* split_bands(size_t channel) {
return split_data_.get() ? split_data_->bands(channel)
: data_->bands(channel);
}
// Returns a pointer array to the channels for a specific band.
// Usage:
// split_channels(band)[channel][sample].
// Where:
// 0 <= band < |num_bands_|
// 0 <= channel < |num_proc_channels_|
// 0 <= channel < |buffer_num_channels_|
// 0 <= sample < |num_split_frames_|
int16_t* const* split_channels(Band band);
const int16_t* const* split_channels_const(Band band) const;
float* const* split_channels_f(Band band);
const float* const* split_channels_const_f(Band band) const;
const float* const* split_channels_const(Band band) const {
if (split_data_.get()) {
return split_data_->channels(band);
} else {
return band == kBand0To8kHz ? data_->channels() : nullptr;
}
}
// Returns a pointer to the ChannelBuffer that encapsulates the full-band
// data.
ChannelBuffer<int16_t>* data();
const ChannelBuffer<int16_t>* data() const;
ChannelBuffer<float>* data_f();
const ChannelBuffer<float>* data_f() const;
// Copies data into the buffer.
void CopyFrom(const int16_t* const interleaved_data,
const StreamConfig& stream_config);
void CopyFrom(const float* const* stacked_data,
const StreamConfig& stream_config);
// Returns a pointer to the ChannelBuffer that encapsulates the split data.
ChannelBuffer<int16_t>* split_data();
const ChannelBuffer<int16_t>* split_data() const;
ChannelBuffer<float>* split_data_f();
const ChannelBuffer<float>* split_data_f() const;
// Copies data from the buffer.
void CopyTo(const StreamConfig& stream_config,
int16_t* const interleaved_data);
void CopyTo(const StreamConfig& stream_config, float* const* stacked_data);
void CopyTo(AudioBuffer* buffer) const;
// Returns a pointer to the low-pass data downmixed to mono. If this data
// isn't already available it re-calculates it.
const int16_t* mixed_low_pass_data();
const int16_t* low_pass_reference(int channel) const;
const float* keyboard_data() const;
void set_activity(AudioFrame::VADActivity activity);
AudioFrame::VADActivity activity() const;
// Use for int16 interleaved data.
void DeinterleaveFrom(AudioFrame* audioFrame);
// If |data_changed| is false, only the non-audio data members will be copied
// to |frame|.
void InterleaveTo(AudioFrame* frame, bool data_changed);
// Use for float deinterleaved data.
void CopyFrom(const float* const* data, const StreamConfig& stream_config);
void CopyTo(const StreamConfig& stream_config, float* const* data);
void CopyLowPassToReference();
// Splits the signal into different bands.
// Splits the buffer data into frequency bands.
void SplitIntoFrequencyBands();
// Recombine the different bands into one signal.
// Recombines the frequency bands into a full-band signal.
void MergeFrequencyBands();
// Copies the split bands data into the integer two-dimensional array.
void ExportSplitChannelData(size_t channel,
int16_t* const* split_band_data) const;
// Copies the data in the integer two-dimensional array into the split_bands
// data.
void ImportSplitChannelData(size_t channel,
const int16_t* const* split_band_data);
static const size_t kMaxSplitFrameLength = 160;
static const size_t kMaxNumBands = 3;
// Deprecated methods, will be removed soon.
float* const* channels_f() { return channels(); }
const float* const* channels_const_f() const { return channels_const(); }
const float* const* split_bands_const_f(size_t channel) const {
return split_bands_const(channel);
}
float* const* split_bands_f(size_t channel) { return split_bands(channel); }
const float* const* split_channels_const_f(Band band) const {
return split_channels_const(band);
}
private:
// Called from DeinterleaveFrom() and CopyFrom().
void InitForNewData();
FRIEND_TEST_ALL_PREFIXES(AudioBufferTest,
SetNumChannelsSetsChannelBuffersNumChannels);
void RestoreNumChannels();
// The audio is passed into DeinterleaveFrom() or CopyFrom() with input
// format (samples per channel and number of channels).
const size_t input_num_frames_;
const int num_input_channels_;
// The audio is stored by DeinterleaveFrom() or CopyFrom() with processing
// format.
const size_t proc_num_frames_;
const int num_proc_channels_;
// The audio is returned by InterleaveTo() and CopyTo() with output samples
// per channels and the current number of channels. This last one can be
// changed at any time using set_num_channels().
const size_t input_num_channels_;
const size_t buffer_num_frames_;
const size_t buffer_num_channels_;
const size_t output_num_frames_;
int num_channels_;
const size_t output_num_channels_;
size_t num_channels_;
size_t num_bands_;
size_t num_split_frames_;
bool mixed_low_pass_valid_;
bool reference_copied_;
AudioFrame::VADActivity activity_;
const float* keyboard_data_;
rtc::scoped_ptr<IFChannelBuffer> data_;
rtc::scoped_ptr<IFChannelBuffer> split_data_;
rtc::scoped_ptr<SplittingFilter> splitting_filter_;
rtc::scoped_ptr<ChannelBuffer<int16_t> > mixed_low_pass_channels_;
rtc::scoped_ptr<ChannelBuffer<int16_t> > low_pass_reference_channels_;
rtc::scoped_ptr<IFChannelBuffer> input_buffer_;
rtc::scoped_ptr<IFChannelBuffer> output_buffer_;
rtc::scoped_ptr<ChannelBuffer<float> > process_buffer_;
ScopedVector<PushSincResampler> input_resamplers_;
ScopedVector<PushSincResampler> output_resamplers_;
std::unique_ptr<ChannelBuffer<float>> data_;
std::unique_ptr<ChannelBuffer<float>> split_data_;
std::unique_ptr<SplittingFilter> splitting_filter_;
std::vector<std::unique_ptr<PushSincResampler>> input_resamplers_;
std::vector<std::unique_ptr<PushSincResampler>> output_resamplers_;
bool downmix_by_averaging_ = true;
size_t channel_for_downmixing_ = 0;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_BUFFER_H_
#endif // MODULES_AUDIO_PROCESSING_AUDIO_BUFFER_H_