ddbb8a2c24
Select "processing" rates based on the input and output sampling rates. Resample the input streams to those rates, and if necessary to the output rate. - Remove deprecated stream format APIs. - Remove deprecated device sample rate APIs. - Add a ChannelBuffer class to help manage deinterleaved channels. - Clean up the splitting filter state. - Add a unit test which verifies the output against known-working native format output. BUG=2894 R=aluebs@webrtc.org, bjornv@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9919004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5959 4adac7df-926f-26a2-2b94-8c16560cd09d
124 lines
4.2 KiB
C++
124 lines
4.2 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 SplitChannelBuffer;
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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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int16_t* data(int channel) const;
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int16_t* low_pass_split_data(int channel) const;
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int16_t* high_pass_split_data(int channel) const;
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int16_t* mixed_data(int channel) const;
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int16_t* mixed_low_pass_data(int channel) const;
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int16_t* low_pass_reference(int channel) const;
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SplitFilterStates* filter_states(int channel) const;
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void set_activity(AudioFrame::VADActivity activity);
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AudioFrame::VADActivity activity() const;
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bool is_muted() const;
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// Use for int16 interleaved data.
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void DeinterleaveFrom(AudioFrame* audioFrame);
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void InterleaveTo(AudioFrame* audioFrame) const;
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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 CopyAndMix(int num_mixed_channels);
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void CopyAndMixLowPass(int num_mixed_channels);
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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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int num_mixed_channels_;
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int num_mixed_low_pass_channels_;
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// Whether the original data was replaced with mixed data.
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bool data_was_mixed_;
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bool reference_copied_;
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AudioFrame::VADActivity activity_;
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bool is_muted_;
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int16_t* data_;
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scoped_ptr<ChannelBuffer<int16_t> > channels_;
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scoped_ptr<SplitChannelBuffer> split_channels_;
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scoped_ptr<SplitFilterStates[]> filter_states_;
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scoped_ptr<ChannelBuffer<int16_t> > mixed_channels_;
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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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