934a265a47
Each audio processing step is given a pointer to an AudioBuffer, where it can read and write int data. This patch adds corresponding AudioBuffer methods to read and write float data; the buffer will automatically convert the stored data between int and float as necessary. This patch also modifies the echo cancellation step to make use of the new methods (it was already using floats internally; now it doesn't have to convert from and to ints anymore). (The reference data to the ApmTest.Process test had to be modified slightly; this is because the echo canceller no longer unnecessarily converts float data to int and then immediately back to float for each iteration in the loop in EchoCancellationImpl::ProcessCaptureAudio.) BUG= R=aluebs@webrtc.org, andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/18399005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6138 4adac7df-926f-26a2-2b94-8c16560cd09d
140 lines
4.9 KiB
C++
140 lines
4.9 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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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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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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const int16_t* mixed_data(int channel) const;
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const int16_t* mixed_low_pass_data(int channel) const;
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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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float* low_pass_split_data_f(int channel);
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float* high_pass_split_data_f(int channel);
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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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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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bool reference_copied_;
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AudioFrame::VADActivity activity_;
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bool is_muted_;
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// If non-null, use this instead of channels_->channel(0). This is an
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// optimization for the case num_proc_channels_ == 1 that allows us to point
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// to the data instead of copying it.
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int16_t* data_;
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const float* keyboard_data_;
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scoped_ptr<IFChannelBuffer> 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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