2b6bc8d84f
It's just a container for two IFChannelBuffers, and doesn't earn its keep. The main problem is that the number of methods it needs that just forward calls to either of its two IFChannelBuffers was already large, and was about to grow. R=aluebs@webrtc.org, minyue@webrtc.org Review URL: https://webrtc-codereview.appspot.com/16919004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6717 4adac7df-926f-26a2-2b94-8c16560cd09d
131 lines
4.7 KiB
C++
131 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/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/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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struct SplitFilterStates {
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SplitFilterStates() {
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memset(analysis_filter_state1, 0, sizeof(analysis_filter_state1));
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memset(analysis_filter_state2, 0, sizeof(analysis_filter_state2));
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memset(synthesis_filter_state1, 0, sizeof(synthesis_filter_state1));
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memset(synthesis_filter_state2, 0, sizeof(synthesis_filter_state2));
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}
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static const int kStateSize = 6;
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int analysis_filter_state1[kStateSize];
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int analysis_filter_state2[kStateSize];
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int synthesis_filter_state1[kStateSize];
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int synthesis_filter_state2[kStateSize];
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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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// It can be assumed that channels are stored contiguously.
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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* 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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// 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* 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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const float* keyboard_data() const;
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SplitFilterStates* filter_states(int channel);
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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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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<SplitFilterStates[]> filter_states_;
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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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