17e40641b3
This is mainly to support the native audio format in Chrome. Although this implementation just moves the float->int conversion under the hood, we will transition AudioProcessing towards supporting this format throughout. - Add a test which verifies we get identical output with the float and int interfaces. - The float and int wrappers are tasked with conversion to the AudioBuffer format. A new shared Process/Analyze method does most of the work. - Add a new field to the debug.proto to hold deinterleaved data. - Add helpers to audio_utils.cc, and start using numeric_limits. - Note that there was no performance difference between numeric_limits and a literal value when measured on Linux using gcc or clang. BUG=2894 R=aluebs@webrtc.org, bjornv@webrtc.org, henrikg@webrtc.org, tommi@webrtc.org, turaj@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9179004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5641 4adac7df-926f-26a2-2b94-8c16560cd09d
94 lines
3.1 KiB
C++
94 lines
3.1 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_MAIN_SOURCE_AUDIO_BUFFER_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_BUFFER_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/typedefs.h"
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namespace webrtc {
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struct AudioChannel;
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struct SplitAudioChannel;
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class AudioBuffer {
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public:
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AudioBuffer(int max_num_channels, int 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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int32_t* analysis_filter_state1(int channel) const;
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int32_t* analysis_filter_state2(int channel) const;
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int32_t* synthesis_filter_state1(int channel) const;
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int32_t* synthesis_filter_state2(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, int samples_per_channel,
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int num_channels);
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void CopyTo(int samples_per_channel, int num_channels,
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float* const* data) const;
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void Mix(int num_mixed_channels);
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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(int num_channels);
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const int max_num_channels_;
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int num_channels_;
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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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const int samples_per_channel_;
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int samples_per_split_channel_;
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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_array<AudioChannel> channels_;
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scoped_array<SplitAudioChannel> split_channels_;
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scoped_array<AudioChannel> mixed_channels_;
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// TODO(andrew): improve this, we don't need the full 32 kHz space here.
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scoped_array<AudioChannel> mixed_low_pass_channels_;
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scoped_array<AudioChannel> low_pass_reference_channels_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_BUFFER_H_
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