blob: a526ca07c86baa354f251e2c0bb60a89986bba91 [file] [log] [blame]
* Copyright (c) 2011 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.
#include <vector>
#include "webrtc/common_audio/include/audio_util.h"
#include "webrtc/modules/audio_processing/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/interface/scoped_ptr.h"
#include "webrtc/system_wrappers/interface/scoped_vector.h"
#include "webrtc/typedefs.h"
namespace webrtc {
class PushSincResampler;
class IFChannelBuffer;
static const int kMaxNumBands = 3;
enum Band {
kBand0To8kHz = 0,
kBand8To16kHz = 1,
kBand16To24kHz = 2
class AudioBuffer {
// TODO(ajm): Switch to take ChannelLayouts.
AudioBuffer(int input_samples_per_channel,
int num_input_channels,
int process_samples_per_channel,
int num_process_channels,
int output_samples_per_channel);
virtual ~AudioBuffer();
int num_channels() const;
void set_num_channels(int num_channels);
int samples_per_channel() const;
int samples_per_split_channel() const;
int samples_per_keyboard_channel() const;
int num_bands() const;
// Sample array accessors. Channels are guaranteed to be stored contiguously
// in memory. Prefer to use the const variants of each accessor when
// possible, since they incur less float<->int16 conversion overhead.
int16_t* data(int channel);
const int16_t* data_const(int channel) const;
int16_t* const* channels();
const int16_t* const* channels_const() const;
int16_t* const* split_bands(int channel);
const int16_t* const* split_bands_const(int channel) const;
int16_t* const* split_channels(Band band);
const int16_t* const* split_channels_const(Band band) 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;
// Float versions of the accessors, with automatic conversion back and forth
// as necessary. The range of the numbers are the same as for int16_t.
float* data_f(int channel);
const float* data_const_f(int channel) const;
float* const* channels_f();
const float* const* channels_const_f() const;
float* const* split_bands_f(int channel);
const float* const* split_bands_const_f(int channel) const;
float* const* split_channels_f(Band band);
const float* const* split_channels_const_f(Band band) 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) const;
// Use for float deinterleaved data.
void CopyFrom(const float* const* data,
int samples_per_channel,
AudioProcessing::ChannelLayout layout);
void CopyTo(int samples_per_channel,
AudioProcessing::ChannelLayout layout,
float* const* data);
void CopyLowPassToReference();
// Splits the signal into different bands.
void SplitIntoFrequencyBands();
// Recombine the different bands into one signal.
void MergeFrequencyBands();
// Called from DeinterleaveFrom() and CopyFrom().
void InitForNewData();
// The audio is passed into DeinterleaveFrom() or CopyFrom() with input
// format (samples per channel and number of channels).
const int input_samples_per_channel_;
const int num_input_channels_;
// The audio is stored by DeinterleaveFrom() or CopyFrom() with processing
// format.
const int proc_samples_per_channel_;
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 int output_samples_per_channel_;
int num_channels_;
int num_bands_;
int samples_per_split_channel_;
bool mixed_low_pass_valid_;
bool reference_copied_;
AudioFrame::VADActivity activity_;
const float* keyboard_data_;
scoped_ptr<IFChannelBuffer> channels_;
ScopedVector<IFChannelBuffer> split_channels_;
scoped_ptr<int16_t*[]> bands_;
scoped_ptr<float*[]> bands_f_;
scoped_ptr<SplittingFilter> splitting_filter_;
scoped_ptr<ChannelBuffer<int16_t> > mixed_low_pass_channels_;
scoped_ptr<ChannelBuffer<int16_t> > low_pass_reference_channels_;
scoped_ptr<ChannelBuffer<float> > input_buffer_;
scoped_ptr<ChannelBuffer<float> > process_buffer_;
ScopedVector<PushSincResampler> input_resamplers_;
ScopedVector<PushSincResampler> output_resamplers_;
} // namespace webrtc