a7384a1126
Now there is only one API to get the data or the channels (one const and one no const) merged or by band. The band is passed in as a parameter, instead of calling different methods. BUG=webrtc:3146 R=andrew@webrtc.org, bjornv@webrtc.org, kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/27249004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7790 4adac7df-926f-26a2-2b94-8c16560cd09d
133 lines
4.7 KiB
C++
133 lines
4.7 KiB
C++
/*
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* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_BUFFER_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_BUFFER_H_
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#include <vector>
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#include "webrtc/common_audio/include/audio_util.h"
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#include "webrtc/modules/audio_processing/channel_buffer.h"
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#include "webrtc/modules/audio_processing/include/audio_processing.h"
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#include "webrtc/modules/audio_processing/splitting_filter.h"
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#include "webrtc/modules/interface/module_common_types.h"
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#include "webrtc/system_wrappers/interface/scoped_ptr.h"
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#include "webrtc/system_wrappers/interface/scoped_vector.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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class PushSincResampler;
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class IFChannelBuffer;
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enum Band {
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kBand0To8kHz = 0,
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kBand8To16kHz = 1,
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kBand16To24kHz = 2
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};
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class AudioBuffer {
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public:
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// TODO(ajm): Switch to take ChannelLayouts.
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AudioBuffer(int input_samples_per_channel,
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int num_input_channels,
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int process_samples_per_channel,
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int num_process_channels,
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int output_samples_per_channel);
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virtual ~AudioBuffer();
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int num_channels() const;
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int samples_per_channel() const;
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int samples_per_split_channel() const;
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int samples_per_keyboard_channel() const;
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// Sample array accessors. Channels are guaranteed to be stored contiguously
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// in memory. Prefer to use the const variants of each accessor when
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// possible, since they incur less float<->int16 conversion overhead.
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int16_t* data(int channel);
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const int16_t* data_const(int channel) const;
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int16_t* const* channels();
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const int16_t* const* channels_const() const;
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int16_t* split_data(int channel, Band band);
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const int16_t* split_data_const(int channel, Band band) const;
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int16_t* const* split_channels(Band band);
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const int16_t* const* split_channels_const(Band band) const;
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// Returns a pointer to the low-pass data downmixed to mono. If this data
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// isn't already available it re-calculates it.
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const int16_t* mixed_low_pass_data();
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const int16_t* low_pass_reference(int channel) const;
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// Float versions of the accessors, with automatic conversion back and forth
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// as necessary. The range of the numbers are the same as for int16_t.
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float* data_f(int channel);
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const float* data_const_f(int channel) const;
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float* const* channels_f();
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const float* const* channels_const_f() const;
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float* split_data_f(int channel, Band band);
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const float* split_data_const_f(int channel, Band band) const;
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float* const* split_channels_f(Band band);
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const float* const* split_channels_const_f(Band band) const;
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const float* keyboard_data() const;
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void set_activity(AudioFrame::VADActivity activity);
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AudioFrame::VADActivity activity() const;
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// Use for int16 interleaved data.
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void DeinterleaveFrom(AudioFrame* audioFrame);
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// If |data_changed| is false, only the non-audio data members will be copied
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// to |frame|.
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void InterleaveTo(AudioFrame* frame, bool data_changed) const;
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// Use for float deinterleaved data.
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void CopyFrom(const float* const* data,
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int samples_per_channel,
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AudioProcessing::ChannelLayout layout);
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void CopyTo(int samples_per_channel,
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AudioProcessing::ChannelLayout layout,
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float* const* data);
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void CopyLowPassToReference();
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// Splits the signal into different bands.
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void SplitIntoFrequencyBands();
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// Recombine the different bands into one signal.
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void MergeFrequencyBands();
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private:
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// Called from DeinterleaveFrom() and CopyFrom().
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void InitForNewData();
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const int input_samples_per_channel_;
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const int num_input_channels_;
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const int proc_samples_per_channel_;
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const int num_proc_channels_;
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const int output_samples_per_channel_;
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int samples_per_split_channel_;
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bool mixed_low_pass_valid_;
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bool reference_copied_;
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AudioFrame::VADActivity activity_;
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const float* keyboard_data_;
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scoped_ptr<IFChannelBuffer> channels_;
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ScopedVector<IFChannelBuffer> split_channels_;
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scoped_ptr<SplittingFilter> splitting_filter_;
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scoped_ptr<ChannelBuffer<int16_t> > mixed_low_pass_channels_;
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scoped_ptr<ChannelBuffer<int16_t> > low_pass_reference_channels_;
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scoped_ptr<ChannelBuffer<float> > input_buffer_;
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scoped_ptr<ChannelBuffer<float> > process_buffer_;
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ScopedVector<PushSincResampler> input_resamplers_;
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ScopedVector<PushSincResampler> output_resamplers_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_BUFFER_H_
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