webrtc/modules/audio_processing/audio_buffer.cc - src - Git at Google

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */

 #include "webrtc/modules/audio_processing/audio_buffer.h"

 #include "webrtc/common_audio/include/audio_util.h"
 #include "webrtc/common_audio/resampler/push_sinc_resampler.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/common_audio/channel_buffer.h"
 #include "webrtc/modules/audio_processing/common.h"

 namespace webrtc {
 namespace {

 const size_t kSamplesPer16kHzChannel = 160;
 const size_t kSamplesPer32kHzChannel = 320;
 const size_t kSamplesPer48kHzChannel = 480;

 int KeyboardChannelIndex(const StreamConfig& stream_config) {
   if (!stream_config.has_keyboard()) {
     assert(false);
     return 0;
   }

   return stream_config.num_channels();
 }

 size_t NumBandsFromSamplesPerChannel(size_t num_frames) {
   size_t num_bands = 1;
   if (num_frames == kSamplesPer32kHzChannel ||
       num_frames == kSamplesPer48kHzChannel) {
     num_bands = rtc::CheckedDivExact(num_frames, kSamplesPer16kHzChannel);
   }
   return num_bands;
 }

 }  // namespace

 AudioBuffer::AudioBuffer(size_t input_num_frames,
                          size_t num_input_channels,
                          size_t process_num_frames,
                          size_t num_process_channels,
                          size_t output_num_frames)
   : input_num_frames_(input_num_frames),
     num_input_channels_(num_input_channels),
     proc_num_frames_(process_num_frames),
     num_proc_channels_(num_process_channels),
     output_num_frames_(output_num_frames),
     num_channels_(num_process_channels),
     num_bands_(NumBandsFromSamplesPerChannel(proc_num_frames_)),
     num_split_frames_(rtc::CheckedDivExact(proc_num_frames_, num_bands_)),
     mixed_low_pass_valid_(false),
     reference_copied_(false),
     activity_(AudioFrame::kVadUnknown),
     keyboard_data_(NULL),
     data_(new IFChannelBuffer(proc_num_frames_, num_proc_channels_)) {
   assert(input_num_frames_ > 0);
   assert(proc_num_frames_ > 0);
   assert(output_num_frames_ > 0);
   assert(num_input_channels_ > 0);
   assert(num_proc_channels_ > 0 && num_proc_channels_ <= num_input_channels_);

   if (input_num_frames_ != proc_num_frames_ ||
       output_num_frames_ != proc_num_frames_) {
     // Create an intermediate buffer for resampling.
     process_buffer_.reset(new ChannelBuffer<float>(proc_num_frames_,
                                                    num_proc_channels_));

     if (input_num_frames_ != proc_num_frames_) {
       for (size_t i = 0; i < num_proc_channels_; ++i) {
         input_resamplers_.push_back(std::unique_ptr<PushSincResampler>(
             new PushSincResampler(input_num_frames_, proc_num_frames_)));
       }
     }

     if (output_num_frames_ != proc_num_frames_) {
       for (size_t i = 0; i < num_proc_channels_; ++i) {
         output_resamplers_.push_back(std::unique_ptr<PushSincResampler>(
             new PushSincResampler(proc_num_frames_, output_num_frames_)));
       }
     }
   }

   if (num_bands_ > 1) {
     split_data_.reset(new IFChannelBuffer(proc_num_frames_,
                                           num_proc_channels_,
                                           num_bands_));
     splitting_filter_.reset(new SplittingFilter(num_proc_channels_,
                                                 num_bands_,
                                                 proc_num_frames_));
   }
 }

 AudioBuffer::~AudioBuffer() {}

 void AudioBuffer::CopyFrom(const float* const* data,
                            const StreamConfig& stream_config) {
   assert(stream_config.num_frames() == input_num_frames_);
   assert(stream_config.num_channels() == num_input_channels_);
   InitForNewData();
   // Initialized lazily because there's a different condition in
   // DeinterleaveFrom.
   const bool need_to_downmix =
       num_input_channels_ > 1 && num_proc_channels_ == 1;
   if (need_to_downmix && !input_buffer_) {
     input_buffer_.reset(
         new IFChannelBuffer(input_num_frames_, num_proc_channels_));
   }

   if (stream_config.has_keyboard()) {
     keyboard_data_ = data[KeyboardChannelIndex(stream_config)];
   }

   // Downmix.
   const float* const* data_ptr = data;
   if (need_to_downmix) {
     DownmixToMono<float, float>(data, input_num_frames_, num_input_channels_,
                                 input_buffer_->fbuf()->channels()[0]);
     data_ptr = input_buffer_->fbuf_const()->channels();
   }

   // Resample.
   if (input_num_frames_ != proc_num_frames_) {
     for (size_t i = 0; i < num_proc_channels_; ++i) {
       input_resamplers_[i]->Resample(data_ptr[i],
                                      input_num_frames_,
                                      process_buffer_->channels()[i],
                                      proc_num_frames_);
     }
     data_ptr = process_buffer_->channels();
   }

   // Convert to the S16 range.
   for (size_t i = 0; i < num_proc_channels_; ++i) {
     FloatToFloatS16(data_ptr[i],
                     proc_num_frames_,
                     data_->fbuf()->channels()[i]);
   }
 }

 void AudioBuffer::CopyTo(const StreamConfig& stream_config,
                          float* const* data) {
   assert(stream_config.num_frames() == output_num_frames_);
   assert(stream_config.num_channels() == num_channels_ || num_channels_ == 1);

   // Convert to the float range.
   float* const* data_ptr = data;
   if (output_num_frames_ != proc_num_frames_) {
     // Convert to an intermediate buffer for subsequent resampling.
     data_ptr = process_buffer_->channels();
   }
   for (size_t i = 0; i < num_channels_; ++i) {
     FloatS16ToFloat(data_->fbuf()->channels()[i],
                     proc_num_frames_,
                     data_ptr[i]);
   }

   // Resample.
   if (output_num_frames_ != proc_num_frames_) {
     for (size_t i = 0; i < num_channels_; ++i) {
       output_resamplers_[i]->Resample(data_ptr[i],
                                       proc_num_frames_,
                                       data[i],
                                       output_num_frames_);
     }
   }

   // Upmix.
   for (size_t i = num_channels_; i < stream_config.num_channels(); ++i) {
     memcpy(data[i], data[0], output_num_frames_ * sizeof(**data));
   }
 }

 void AudioBuffer::InitForNewData() {
   keyboard_data_ = NULL;
   mixed_low_pass_valid_ = false;
   reference_copied_ = false;
   activity_ = AudioFrame::kVadUnknown;
   num_channels_ = num_proc_channels_;
 }

 const int16_t* const* AudioBuffer::channels_const() const {
   return data_->ibuf_const()->channels();
 }

 int16_t* const* AudioBuffer::channels() {
   mixed_low_pass_valid_ = false;
   return data_->ibuf()->channels();
 }

 const int16_t* const* AudioBuffer::split_bands_const(size_t channel) const {
   return split_data_.get() ?
          split_data_->ibuf_const()->bands(channel) :
          data_->ibuf_const()->bands(channel);
 }

 int16_t* const* AudioBuffer::split_bands(size_t channel) {
   mixed_low_pass_valid_ = false;
   return split_data_.get() ?
          split_data_->ibuf()->bands(channel) :
          data_->ibuf()->bands(channel);
 }

 const int16_t* const* AudioBuffer::split_channels_const(Band band) const {
   if (split_data_.get()) {
     return split_data_->ibuf_const()->channels(band);
   } else {
     return band == kBand0To8kHz ? data_->ibuf_const()->channels() : nullptr;
   }
 }

 int16_t* const* AudioBuffer::split_channels(Band band) {
   mixed_low_pass_valid_ = false;
   if (split_data_.get()) {
     return split_data_->ibuf()->channels(band);
   } else {
     return band == kBand0To8kHz ? data_->ibuf()->channels() : nullptr;
   }
 }

 ChannelBuffer<int16_t>* AudioBuffer::data() {
   mixed_low_pass_valid_ = false;
   return data_->ibuf();
 }

 const ChannelBuffer<int16_t>* AudioBuffer::data() const {
   return data_->ibuf_const();
 }

 ChannelBuffer<int16_t>* AudioBuffer::split_data() {
   mixed_low_pass_valid_ = false;
   return split_data_.get() ? split_data_->ibuf() : data_->ibuf();
 }

 const ChannelBuffer<int16_t>* AudioBuffer::split_data() const {
   return split_data_.get() ? split_data_->ibuf_const() : data_->ibuf_const();
 }

 const float* const* AudioBuffer::channels_const_f() const {
   return data_->fbuf_const()->channels();
 }

 float* const* AudioBuffer::channels_f() {
   mixed_low_pass_valid_ = false;
   return data_->fbuf()->channels();
 }

 const float* const* AudioBuffer::split_bands_const_f(size_t channel) const {
   return split_data_.get() ?
          split_data_->fbuf_const()->bands(channel) :
          data_->fbuf_const()->bands(channel);
 }

 float* const* AudioBuffer::split_bands_f(size_t channel) {
   mixed_low_pass_valid_ = false;
   return split_data_.get() ?
          split_data_->fbuf()->bands(channel) :
          data_->fbuf()->bands(channel);
 }

 const float* const* AudioBuffer::split_channels_const_f(Band band) const {
   if (split_data_.get()) {
     return split_data_->fbuf_const()->channels(band);
   } else {
     return band == kBand0To8kHz ? data_->fbuf_const()->channels() : nullptr;
   }
 }

 float* const* AudioBuffer::split_channels_f(Band band) {
   mixed_low_pass_valid_ = false;
   if (split_data_.get()) {
     return split_data_->fbuf()->channels(band);
   } else {
     return band == kBand0To8kHz ? data_->fbuf()->channels() : nullptr;
   }
 }

 ChannelBuffer<float>* AudioBuffer::data_f() {
   mixed_low_pass_valid_ = false;
   return data_->fbuf();
 }

 const ChannelBuffer<float>* AudioBuffer::data_f() const {
   return data_->fbuf_const();
 }

 ChannelBuffer<float>* AudioBuffer::split_data_f() {
   mixed_low_pass_valid_ = false;
   return split_data_.get() ? split_data_->fbuf() : data_->fbuf();
 }

 const ChannelBuffer<float>* AudioBuffer::split_data_f() const {
   return split_data_.get() ? split_data_->fbuf_const() : data_->fbuf_const();
 }

 const int16_t* AudioBuffer::mixed_low_pass_data() {
   if (num_proc_channels_ == 1) {
     return split_bands_const(0)[kBand0To8kHz];
   }

   if (!mixed_low_pass_valid_) {
     if (!mixed_low_pass_channels_.get()) {
       mixed_low_pass_channels_.reset(
           new ChannelBuffer<int16_t>(num_split_frames_, 1));
     }

     DownmixToMono<int16_t, int32_t>(split_channels_const(kBand0To8kHz),
                                     num_split_frames_, num_channels_,
                                     mixed_low_pass_channels_->channels()[0]);
     mixed_low_pass_valid_ = true;
   }
   return mixed_low_pass_channels_->channels()[0];
 }

 const int16_t* AudioBuffer::low_pass_reference(int channel) const {
   if (!reference_copied_) {
     return NULL;
   }

   return low_pass_reference_channels_->channels()[channel];
 }

 const float* AudioBuffer::keyboard_data() const {
   return keyboard_data_;
 }

 void AudioBuffer::set_activity(AudioFrame::VADActivity activity) {
   activity_ = activity;
 }

 AudioFrame::VADActivity AudioBuffer::activity() const {
   return activity_;
 }

 size_t AudioBuffer::num_channels() const {
   return num_channels_;
 }

 void AudioBuffer::set_num_channels(size_t num_channels) {
   num_channels_ = num_channels;
 }

 size_t AudioBuffer::num_frames() const {
   return proc_num_frames_;
 }

 size_t AudioBuffer::num_frames_per_band() const {
   return num_split_frames_;
 }

 size_t AudioBuffer::num_keyboard_frames() const {
   // We don't resample the keyboard channel.
   return input_num_frames_;
 }

 size_t AudioBuffer::num_bands() const {
   return num_bands_;
 }

 // The resampler is only for supporting 48kHz to 16kHz in the reverse stream.
 void AudioBuffer::DeinterleaveFrom(AudioFrame* frame) {
   assert(frame->num_channels_ == num_input_channels_);
   assert(frame->samples_per_channel_ == input_num_frames_);
   InitForNewData();
   // Initialized lazily because there's a different condition in CopyFrom.
   if ((input_num_frames_ != proc_num_frames_) && !input_buffer_) {
     input_buffer_.reset(
         new IFChannelBuffer(input_num_frames_, num_proc_channels_));
   }
   activity_ = frame->vad_activity_;

   int16_t* const* deinterleaved;
   if (input_num_frames_ == proc_num_frames_) {
     deinterleaved = data_->ibuf()->channels();
   } else {
     deinterleaved = input_buffer_->ibuf()->channels();
   }
   if (num_proc_channels_ == 1) {
     // Downmix and deinterleave simultaneously.
     DownmixInterleavedToMono(frame->data_, input_num_frames_,
                              num_input_channels_, deinterleaved[0]);
   } else {
     assert(num_proc_channels_ == num_input_channels_);
     Deinterleave(frame->data_,
                  input_num_frames_,
                  num_proc_channels_,
                  deinterleaved);
   }

   // Resample.
   if (input_num_frames_ != proc_num_frames_) {
     for (size_t i = 0; i < num_proc_channels_; ++i) {
       input_resamplers_[i]->Resample(input_buffer_->fbuf_const()->channels()[i],
                                      input_num_frames_,
                                      data_->fbuf()->channels()[i],
                                      proc_num_frames_);
     }
   }
 }

 void AudioBuffer::InterleaveTo(AudioFrame* frame, bool data_changed) {
   frame->vad_activity_ = activity_;
   if (!data_changed) {
     return;
   }

   assert(frame->num_channels_ == num_channels_ || num_channels_ == 1);
   assert(frame->samples_per_channel_ == output_num_frames_);

   // Resample if necessary.
   IFChannelBuffer* data_ptr = data_.get();
   if (proc_num_frames_ != output_num_frames_) {
     if (!output_buffer_) {
       output_buffer_.reset(
           new IFChannelBuffer(output_num_frames_, num_channels_));
     }
     for (size_t i = 0; i < num_channels_; ++i) {
       output_resamplers_[i]->Resample(
           data_->fbuf()->channels()[i], proc_num_frames_,
           output_buffer_->fbuf()->channels()[i], output_num_frames_);
     }
     data_ptr = output_buffer_.get();
   }

   if (frame->num_channels_ == num_channels_) {
     Interleave(data_ptr->ibuf()->channels(), output_num_frames_, num_channels_,
                frame->data_);
   } else {
     UpmixMonoToInterleaved(data_ptr->ibuf()->channels()[0], output_num_frames_,
                            frame->num_channels_, frame->data_);
   }
 }

 void AudioBuffer::CopyLowPassToReference() {
   reference_copied_ = true;
   if (!low_pass_reference_channels_.get() ||
       low_pass_reference_channels_->num_channels() != num_channels_) {
     low_pass_reference_channels_.reset(
         new ChannelBuffer<int16_t>(num_split_frames_,
                                    num_proc_channels_));
   }
   for (size_t i = 0; i < num_proc_channels_; i++) {
     memcpy(low_pass_reference_channels_->channels()[i],
            split_bands_const(i)[kBand0To8kHz],
            low_pass_reference_channels_->num_frames_per_band() *
                sizeof(split_bands_const(i)[kBand0To8kHz][0]));
   }
 }

 void AudioBuffer::SplitIntoFrequencyBands() {
   splitting_filter_->Analysis(data_.get(), split_data_.get());
 }

 void AudioBuffer::MergeFrequencyBands() {
   splitting_filter_->Synthesis(split_data_.get(), data_.get());
 }

 }  // namespace webrtc
	/*
	* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
	*
	* Use of this source code is governed by a BSD-style license
	* that can be found in the LICENSE file in the root of the source
	* tree. An additional intellectual property rights grant can be found
	* in the file PATENTS. All contributing project authors may
	* be found in the AUTHORS file in the root of the source tree.
	*/

	#include "webrtc/modules/audio_processing/audio_buffer.h"

	#include "webrtc/common_audio/include/audio_util.h"
	#include "webrtc/common_audio/resampler/push_sinc_resampler.h"
	#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
	#include "webrtc/common_audio/channel_buffer.h"
	#include "webrtc/modules/audio_processing/common.h"

	namespace webrtc {
	namespace {

	const size_t kSamplesPer16kHzChannel = 160;
	const size_t kSamplesPer32kHzChannel = 320;
	const size_t kSamplesPer48kHzChannel = 480;

	int KeyboardChannelIndex(const StreamConfig& stream_config) {
	if (!stream_config.has_keyboard()) {
	assert(false);
	return 0;
	}

	return stream_config.num_channels();
	}

	size_t NumBandsFromSamplesPerChannel(size_t num_frames) {
	size_t num_bands = 1;
	if (num_frames == kSamplesPer32kHzChannel \|\|
	num_frames == kSamplesPer48kHzChannel) {
	num_bands = rtc::CheckedDivExact(num_frames, kSamplesPer16kHzChannel);
	}
	return num_bands;
	}

	} // namespace

	AudioBuffer::AudioBuffer(size_t input_num_frames,
	size_t num_input_channels,
	size_t process_num_frames,
	size_t num_process_channels,
	size_t output_num_frames)
	: input_num_frames_(input_num_frames),
	num_input_channels_(num_input_channels),
	proc_num_frames_(process_num_frames),
	num_proc_channels_(num_process_channels),
	output_num_frames_(output_num_frames),
	num_channels_(num_process_channels),
	num_bands_(NumBandsFromSamplesPerChannel(proc_num_frames_)),
	num_split_frames_(rtc::CheckedDivExact(proc_num_frames_, num_bands_)),
	mixed_low_pass_valid_(false),
	reference_copied_(false),
	activity_(AudioFrame::kVadUnknown),
	keyboard_data_(NULL),
	data_(new IFChannelBuffer(proc_num_frames_, num_proc_channels_)) {
	assert(input_num_frames_ > 0);
	assert(proc_num_frames_ > 0);
	assert(output_num_frames_ > 0);
	assert(num_input_channels_ > 0);
	assert(num_proc_channels_ > 0 && num_proc_channels_ <= num_input_channels_);

	if (input_num_frames_ != proc_num_frames_ \|\|
	output_num_frames_ != proc_num_frames_) {
	// Create an intermediate buffer for resampling.
	process_buffer_.reset(new ChannelBuffer<float>(proc_num_frames_,
	num_proc_channels_));

	if (input_num_frames_ != proc_num_frames_) {
	for (size_t i = 0; i < num_proc_channels_; ++i) {
	input_resamplers_.push_back(std::unique_ptr<PushSincResampler>(
	new PushSincResampler(input_num_frames_, proc_num_frames_)));
	}
	}

	if (output_num_frames_ != proc_num_frames_) {
	for (size_t i = 0; i < num_proc_channels_; ++i) {
	output_resamplers_.push_back(std::unique_ptr<PushSincResampler>(
	new PushSincResampler(proc_num_frames_, output_num_frames_)));
	}
	}
	}

	if (num_bands_ > 1) {
	split_data_.reset(new IFChannelBuffer(proc_num_frames_,
	num_proc_channels_,
	num_bands_));
	splitting_filter_.reset(new SplittingFilter(num_proc_channels_,
	num_bands_,
	proc_num_frames_));
	}
	}

	AudioBuffer::~AudioBuffer() {}

	void AudioBuffer::CopyFrom(const float* const* data,
	const StreamConfig& stream_config) {
	assert(stream_config.num_frames() == input_num_frames_);
	assert(stream_config.num_channels() == num_input_channels_);
	InitForNewData();
	// Initialized lazily because there's a different condition in
	// DeinterleaveFrom.
	const bool need_to_downmix =
	num_input_channels_ > 1 && num_proc_channels_ == 1;
	if (need_to_downmix && !input_buffer_) {
	input_buffer_.reset(
	new IFChannelBuffer(input_num_frames_, num_proc_channels_));
	}

	if (stream_config.has_keyboard()) {
	keyboard_data_ = data[KeyboardChannelIndex(stream_config)];
	}

	// Downmix.
	const float* const* data_ptr = data;
	if (need_to_downmix) {
	DownmixToMono<float, float>(data, input_num_frames_, num_input_channels_,
	input_buffer_->fbuf()->channels()[0]);
	data_ptr = input_buffer_->fbuf_const()->channels();
	}

	// Resample.
	if (input_num_frames_ != proc_num_frames_) {
	for (size_t i = 0; i < num_proc_channels_; ++i) {
	input_resamplers_[i]->Resample(data_ptr[i],
	input_num_frames_,
	process_buffer_->channels()[i],
	proc_num_frames_);
	}
	data_ptr = process_buffer_->channels();
	}

	// Convert to the S16 range.
	for (size_t i = 0; i < num_proc_channels_; ++i) {
	FloatToFloatS16(data_ptr[i],
	proc_num_frames_,
	data_->fbuf()->channels()[i]);
	}
	}

	void AudioBuffer::CopyTo(const StreamConfig& stream_config,
	float* const* data) {
	assert(stream_config.num_frames() == output_num_frames_);
	assert(stream_config.num_channels() == num_channels_ \|\| num_channels_ == 1);

	// Convert to the float range.
	float* const* data_ptr = data;
	if (output_num_frames_ != proc_num_frames_) {
	// Convert to an intermediate buffer for subsequent resampling.
	data_ptr = process_buffer_->channels();
	}
	for (size_t i = 0; i < num_channels_; ++i) {
	FloatS16ToFloat(data_->fbuf()->channels()[i],
	proc_num_frames_,
	data_ptr[i]);
	}

	// Resample.
	if (output_num_frames_ != proc_num_frames_) {
	for (size_t i = 0; i < num_channels_; ++i) {
	output_resamplers_[i]->Resample(data_ptr[i],
	proc_num_frames_,
	data[i],
	output_num_frames_);
	}
	}

	// Upmix.
	for (size_t i = num_channels_; i < stream_config.num_channels(); ++i) {
	memcpy(data[i], data[0], output_num_frames_ * sizeof(**data));
	}
	}

	void AudioBuffer::InitForNewData() {
	keyboard_data_ = NULL;
	mixed_low_pass_valid_ = false;
	reference_copied_ = false;
	activity_ = AudioFrame::kVadUnknown;
	num_channels_ = num_proc_channels_;
	}

	const int16_t* const* AudioBuffer::channels_const() const {
	return data_->ibuf_const()->channels();
	}

	int16_t* const* AudioBuffer::channels() {
	mixed_low_pass_valid_ = false;
	return data_->ibuf()->channels();
	}

	const int16_t* const* AudioBuffer::split_bands_const(size_t channel) const {
	return split_data_.get() ?
	split_data_->ibuf_const()->bands(channel) :
	data_->ibuf_const()->bands(channel);
	}

	int16_t* const* AudioBuffer::split_bands(size_t channel) {
	mixed_low_pass_valid_ = false;
	return split_data_.get() ?
	split_data_->ibuf()->bands(channel) :
	data_->ibuf()->bands(channel);
	}

	const int16_t* const* AudioBuffer::split_channels_const(Band band) const {
	if (split_data_.get()) {
	return split_data_->ibuf_const()->channels(band);
	} else {
	return band == kBand0To8kHz ? data_->ibuf_const()->channels() : nullptr;
	}
	}

	int16_t* const* AudioBuffer::split_channels(Band band) {
	mixed_low_pass_valid_ = false;
	if (split_data_.get()) {
	return split_data_->ibuf()->channels(band);
	} else {
	return band == kBand0To8kHz ? data_->ibuf()->channels() : nullptr;
	}
	}

	ChannelBuffer<int16_t>* AudioBuffer::data() {
	mixed_low_pass_valid_ = false;
	return data_->ibuf();
	}

	const ChannelBuffer<int16_t>* AudioBuffer::data() const {
	return data_->ibuf_const();
	}

	ChannelBuffer<int16_t>* AudioBuffer::split_data() {
	mixed_low_pass_valid_ = false;
	return split_data_.get() ? split_data_->ibuf() : data_->ibuf();
	}

	const ChannelBuffer<int16_t>* AudioBuffer::split_data() const {
	return split_data_.get() ? split_data_->ibuf_const() : data_->ibuf_const();
	}

	const float* const* AudioBuffer::channels_const_f() const {
	return data_->fbuf_const()->channels();
	}

	float* const* AudioBuffer::channels_f() {
	mixed_low_pass_valid_ = false;
	return data_->fbuf()->channels();
	}

	const float* const* AudioBuffer::split_bands_const_f(size_t channel) const {
	return split_data_.get() ?
	split_data_->fbuf_const()->bands(channel) :
	data_->fbuf_const()->bands(channel);
	}

	float* const* AudioBuffer::split_bands_f(size_t channel) {
	mixed_low_pass_valid_ = false;
	return split_data_.get() ?
	split_data_->fbuf()->bands(channel) :
	data_->fbuf()->bands(channel);
	}

	const float* const* AudioBuffer::split_channels_const_f(Band band) const {
	if (split_data_.get()) {
	return split_data_->fbuf_const()->channels(band);
	} else {
	return band == kBand0To8kHz ? data_->fbuf_const()->channels() : nullptr;
	}
	}

	float* const* AudioBuffer::split_channels_f(Band band) {
	mixed_low_pass_valid_ = false;
	if (split_data_.get()) {
	return split_data_->fbuf()->channels(band);
	} else {
	return band == kBand0To8kHz ? data_->fbuf()->channels() : nullptr;
	}
	}

	ChannelBuffer<float>* AudioBuffer::data_f() {
	mixed_low_pass_valid_ = false;
	return data_->fbuf();
	}

	const ChannelBuffer<float>* AudioBuffer::data_f() const {
	return data_->fbuf_const();
	}

	ChannelBuffer<float>* AudioBuffer::split_data_f() {
	mixed_low_pass_valid_ = false;
	return split_data_.get() ? split_data_->fbuf() : data_->fbuf();
	}

	const ChannelBuffer<float>* AudioBuffer::split_data_f() const {
	return split_data_.get() ? split_data_->fbuf_const() : data_->fbuf_const();
	}

	const int16_t* AudioBuffer::mixed_low_pass_data() {
	if (num_proc_channels_ == 1) {
	return split_bands_const(0)[kBand0To8kHz];
	}

	if (!mixed_low_pass_valid_) {
	if (!mixed_low_pass_channels_.get()) {
	mixed_low_pass_channels_.reset(
	new ChannelBuffer<int16_t>(num_split_frames_, 1));
	}

	DownmixToMono<int16_t, int32_t>(split_channels_const(kBand0To8kHz),
	num_split_frames_, num_channels_,
	mixed_low_pass_channels_->channels()[0]);
	mixed_low_pass_valid_ = true;
	}
	return mixed_low_pass_channels_->channels()[0];
	}

	const int16_t* AudioBuffer::low_pass_reference(int channel) const {
	if (!reference_copied_) {
	return NULL;
	}

	return low_pass_reference_channels_->channels()[channel];
	}

	const float* AudioBuffer::keyboard_data() const {
	return keyboard_data_;
	}

	void AudioBuffer::set_activity(AudioFrame::VADActivity activity) {
	activity_ = activity;
	}

	AudioFrame::VADActivity AudioBuffer::activity() const {
	return activity_;
	}

	size_t AudioBuffer::num_channels() const {
	return num_channels_;
	}

	void AudioBuffer::set_num_channels(size_t num_channels) {
	num_channels_ = num_channels;
	}

	size_t AudioBuffer::num_frames() const {
	return proc_num_frames_;
	}

	size_t AudioBuffer::num_frames_per_band() const {
	return num_split_frames_;
	}

	size_t AudioBuffer::num_keyboard_frames() const {
	// We don't resample the keyboard channel.
	return input_num_frames_;
	}

	size_t AudioBuffer::num_bands() const {
	return num_bands_;
	}

	// The resampler is only for supporting 48kHz to 16kHz in the reverse stream.
	void AudioBuffer::DeinterleaveFrom(AudioFrame* frame) {
	assert(frame->num_channels_ == num_input_channels_);
	assert(frame->samples_per_channel_ == input_num_frames_);
	InitForNewData();
	// Initialized lazily because there's a different condition in CopyFrom.
	if ((input_num_frames_ != proc_num_frames_) && !input_buffer_) {
	input_buffer_.reset(
	new IFChannelBuffer(input_num_frames_, num_proc_channels_));
	}
	activity_ = frame->vad_activity_;

	int16_t* const* deinterleaved;
	if (input_num_frames_ == proc_num_frames_) {
	deinterleaved = data_->ibuf()->channels();
	} else {
	deinterleaved = input_buffer_->ibuf()->channels();
	}
	if (num_proc_channels_ == 1) {
	// Downmix and deinterleave simultaneously.
	DownmixInterleavedToMono(frame->data_, input_num_frames_,
	num_input_channels_, deinterleaved[0]);
	} else {
	assert(num_proc_channels_ == num_input_channels_);
	Deinterleave(frame->data_,
	input_num_frames_,
	num_proc_channels_,
	deinterleaved);
	}

	// Resample.
	if (input_num_frames_ != proc_num_frames_) {
	for (size_t i = 0; i < num_proc_channels_; ++i) {
	input_resamplers_[i]->Resample(input_buffer_->fbuf_const()->channels()[i],
	input_num_frames_,
	data_->fbuf()->channels()[i],
	proc_num_frames_);
	}
	}
	}

	void AudioBuffer::InterleaveTo(AudioFrame* frame, bool data_changed) {
	frame->vad_activity_ = activity_;
	if (!data_changed) {
	return;
	}

	assert(frame->num_channels_ == num_channels_ \|\| num_channels_ == 1);
	assert(frame->samples_per_channel_ == output_num_frames_);

	// Resample if necessary.
	IFChannelBuffer* data_ptr = data_.get();
	if (proc_num_frames_ != output_num_frames_) {
	if (!output_buffer_) {
	output_buffer_.reset(
	new IFChannelBuffer(output_num_frames_, num_channels_));
	}
	for (size_t i = 0; i < num_channels_; ++i) {
	output_resamplers_[i]->Resample(
	data_->fbuf()->channels()[i], proc_num_frames_,
	output_buffer_->fbuf()->channels()[i], output_num_frames_);
	}
	data_ptr = output_buffer_.get();
	}

	if (frame->num_channels_ == num_channels_) {
	Interleave(data_ptr->ibuf()->channels(), output_num_frames_, num_channels_,
	frame->data_);
	} else {
	UpmixMonoToInterleaved(data_ptr->ibuf()->channels()[0], output_num_frames_,
	frame->num_channels_, frame->data_);
	}
	}

	void AudioBuffer::CopyLowPassToReference() {
	reference_copied_ = true;
	if (!low_pass_reference_channels_.get() \|\|
	low_pass_reference_channels_->num_channels() != num_channels_) {
	low_pass_reference_channels_.reset(
	new ChannelBuffer<int16_t>(num_split_frames_,
	num_proc_channels_));
	}
	for (size_t i = 0; i < num_proc_channels_; i++) {
	memcpy(low_pass_reference_channels_->channels()[i],
	split_bands_const(i)[kBand0To8kHz],
	low_pass_reference_channels_->num_frames_per_band() *
	sizeof(split_bands_const(i)[kBand0To8kHz][0]));
	}
	}

	void AudioBuffer::SplitIntoFrequencyBands() {
	splitting_filter_->Analysis(data_.get(), split_data_.get());
	}

	void AudioBuffer::MergeFrequencyBands() {
	splitting_filter_->Synthesis(split_data_.get(), data_.get());
	}

	} // namespace webrtc