be05c74ec8
This doesn't change the behavior at all. The logic behind this is having one class which manages all the splitting filters, because in the future we plan to add a 3 band one for 48kHz support. It also breaks the dependency of the AudioBuffer with the filter states of these filters (which are going to be different for the 3 band one). The AudioBuffer is complicated enough and is going to need changes to support 3 bands in the future, so any simplification is a good idea. On top of that it eliminates repeated code in the APM (now only iterating over channels, but then also deciding in how many bands to split). This should be managed by the AudioBuffer directly. BUG=webrtc:3146 R=bjornv@webrtc.org, kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/32469004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7705 4adac7df-926f-26a2-2b94-8c16560cd09d
137 lines
5.0 KiB
C++
137 lines
5.0 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/modules/audio_processing/common.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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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(int channel) const;
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int16_t* const* channels();
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const int16_t* const* channels() const;
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int16_t* low_pass_split_data(int channel);
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const int16_t* low_pass_split_data(int channel) const;
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int16_t* high_pass_split_data(int channel);
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const int16_t* high_pass_split_data(int channel) const;
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int16_t* const* low_pass_split_channels();
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const int16_t* const* low_pass_split_channels() const;
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int16_t* const* high_pass_split_channels();
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const int16_t* const* high_pass_split_channels() 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_f(int channel) const;
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float* const* channels_f();
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const float* const* channels_f() const;
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float* low_pass_split_data_f(int channel);
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const float* low_pass_split_data_f(int channel) const;
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float* high_pass_split_data_f(int channel);
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const float* high_pass_split_data_f(int channel) const;
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float* const* low_pass_split_channels_f();
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const float* const* low_pass_split_channels_f() const;
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float* const* high_pass_split_channels_f();
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const float* const* high_pass_split_channels_f() 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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scoped_ptr<IFChannelBuffer> split_channels_low_;
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scoped_ptr<IFChannelBuffer> split_channels_high_;
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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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