blob: acf5753cbb275e1fa9ba14e2d5075f3cca63f097 [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/modules/audio_processing/common.h"
#include "webrtc/modules/audio_processing/include/audio_processing.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;
struct SplitFilterStates {
SplitFilterStates() {
memset(analysis_filter_state1, 0, sizeof(analysis_filter_state1));
memset(analysis_filter_state2, 0, sizeof(analysis_filter_state2));
memset(synthesis_filter_state1, 0, sizeof(synthesis_filter_state1));
memset(synthesis_filter_state2, 0, sizeof(synthesis_filter_state2));
static const int kStateSize = 6;
int analysis_filter_state1[kStateSize];
int analysis_filter_state2[kStateSize];
int synthesis_filter_state1[kStateSize];
int synthesis_filter_state2[kStateSize];
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;
int samples_per_channel() const;
int samples_per_split_channel() const;
int samples_per_keyboard_channel() 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(int channel) const;
int16_t* low_pass_split_data(int channel);
const int16_t* low_pass_split_data(int channel) const;
int16_t* high_pass_split_data(int channel);
const int16_t* high_pass_split_data(int channel) 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_f(int channel) const;
float* low_pass_split_data_f(int channel);
const float* low_pass_split_data_f(int channel) const;
float* high_pass_split_data_f(int channel);
const float* high_pass_split_data_f(int channel) const;
const float* keyboard_data() const;
SplitFilterStates* filter_states(int channel);
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();
// Called from DeinterleaveFrom() and CopyFrom().
void InitForNewData();
const int input_samples_per_channel_;
const int num_input_channels_;
const int proc_samples_per_channel_;
const int num_proc_channels_;
const int output_samples_per_channel_;
int samples_per_split_channel_;
bool mixed_low_pass_valid_;
bool reference_copied_;
AudioFrame::VADActivity activity_;
const float* keyboard_data_;
scoped_ptr<IFChannelBuffer> channels_;
scoped_ptr<IFChannelBuffer> split_channels_low_;
scoped_ptr<IFChannelBuffer> split_channels_high_;
scoped_ptr<SplitFilterStates[]> filter_states_;
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