Update audio_processing module
Corresponds to upstream commit 524e9b043e7e86fd72353b987c9d5f6a1ebf83e1 Update notes: * Pull in third party license file * Replace .gypi files with BUILD.gn to keep track of what changes upstream * Bunch of new filse pulled in as dependencies * Won't build yet due to changes needed on top of these
This commit is contained in:
52
webrtc/common_audio/resampler/include/push_resampler.h
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webrtc/common_audio/resampler/include/push_resampler.h
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/*
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* Copyright (c) 2013 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_COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_
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#define WEBRTC_COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_
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#include "webrtc/base/scoped_ptr.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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// Wraps PushSincResampler to provide stereo support.
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// TODO(ajm): add support for an arbitrary number of channels.
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template <typename T>
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class PushResampler {
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public:
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PushResampler();
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virtual ~PushResampler();
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// Must be called whenever the parameters change. Free to be called at any
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// time as it is a no-op if parameters have not changed since the last call.
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int InitializeIfNeeded(int src_sample_rate_hz, int dst_sample_rate_hz,
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int num_channels);
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// Returns the total number of samples provided in destination (e.g. 32 kHz,
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// 2 channel audio gives 640 samples).
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int Resample(const T* src, size_t src_length, T* dst, size_t dst_capacity);
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private:
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rtc::scoped_ptr<PushSincResampler> sinc_resampler_;
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rtc::scoped_ptr<PushSincResampler> sinc_resampler_right_;
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int src_sample_rate_hz_;
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int dst_sample_rate_hz_;
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int num_channels_;
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rtc::scoped_ptr<T[]> src_left_;
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rtc::scoped_ptr<T[]> src_right_;
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rtc::scoped_ptr<T[]> dst_left_;
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rtc::scoped_ptr<T[]> dst_right_;
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};
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} // namespace webrtc
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#endif // WEBRTC_COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_
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webrtc/common_audio/resampler/include/resampler.h
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webrtc/common_audio/resampler/include/resampler.h
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/*
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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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/*
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* A wrapper for resampling a numerous amount of sampling combinations.
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*/
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#ifndef WEBRTC_RESAMPLER_RESAMPLER_H_
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#define WEBRTC_RESAMPLER_RESAMPLER_H_
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#include <stddef.h>
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#include "webrtc/typedefs.h"
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namespace webrtc {
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// All methods return 0 on success and -1 on failure.
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class Resampler
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{
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public:
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Resampler();
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Resampler(int inFreq, int outFreq, int num_channels);
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~Resampler();
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// Reset all states
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int Reset(int inFreq, int outFreq, int num_channels);
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// Reset all states if any parameter has changed
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int ResetIfNeeded(int inFreq, int outFreq, int num_channels);
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// Resample samplesIn to samplesOut.
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int Push(const int16_t* samplesIn, size_t lengthIn, int16_t* samplesOut,
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size_t maxLen, size_t &outLen);
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private:
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enum ResamplerMode
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{
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kResamplerMode1To1,
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kResamplerMode1To2,
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kResamplerMode1To3,
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kResamplerMode1To4,
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kResamplerMode1To6,
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kResamplerMode1To12,
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kResamplerMode2To3,
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kResamplerMode2To11,
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kResamplerMode4To11,
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kResamplerMode8To11,
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kResamplerMode11To16,
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kResamplerMode11To32,
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kResamplerMode2To1,
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kResamplerMode3To1,
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kResamplerMode4To1,
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kResamplerMode6To1,
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kResamplerMode12To1,
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kResamplerMode3To2,
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kResamplerMode11To2,
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kResamplerMode11To4,
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kResamplerMode11To8
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};
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// Generic pointers since we don't know what states we'll need
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void* state1_;
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void* state2_;
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void* state3_;
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// Storage if needed
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int16_t* in_buffer_;
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int16_t* out_buffer_;
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size_t in_buffer_size_;
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size_t out_buffer_size_;
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size_t in_buffer_size_max_;
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size_t out_buffer_size_max_;
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int my_in_frequency_khz_;
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int my_out_frequency_khz_;
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ResamplerMode my_mode_;
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int num_channels_;
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// Extra instance for stereo
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Resampler* slave_left_;
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Resampler* slave_right_;
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};
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} // namespace webrtc
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#endif // WEBRTC_RESAMPLER_RESAMPLER_H_
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