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 "modules/audio_processing/audio_buffer.h"

 #include <string.h>

 #include <cstdint>

 #include "common_audio/channel_buffer.h"
 #include "common_audio/include/audio_util.h"
 #include "common_audio/resampler/push_sinc_resampler.h"
 #include "modules/audio_processing/splitting_filter.h"
 #include "rtc_base/checks.h"

 namespace webrtc {
 namespace {

 constexpr size_t kSamplesPer32kHzChannel = 320;
 constexpr size_t kSamplesPer48kHzChannel = 480;
 constexpr size_t kMaxSamplesPerChannel = AudioBuffer::kMaxSampleRate / 100;

 size_t NumBandsFromFramesPerChannel(size_t num_frames) {
   if (num_frames == kSamplesPer32kHzChannel) {
     return 2;
   }
   if (num_frames == kSamplesPer48kHzChannel) {
     return 3;
   }
   return 1;
 }

 }  // namespace

 AudioBuffer::AudioBuffer(size_t input_rate,
                          size_t input_num_channels,
                          size_t buffer_rate,
                          size_t buffer_num_channels,
                          size_t output_rate,
                          size_t output_num_channels)
     : AudioBuffer(static_cast<int>(input_rate) / 100,
                   input_num_channels,
                   static_cast<int>(buffer_rate) / 100,
                   buffer_num_channels,
                   static_cast<int>(output_rate) / 100) {}

 AudioBuffer::AudioBuffer(size_t input_num_frames,
                          size_t input_num_channels,
                          size_t buffer_num_frames,
                          size_t buffer_num_channels,
                          size_t output_num_frames)
     : input_num_frames_(input_num_frames),
       input_num_channels_(input_num_channels),
       buffer_num_frames_(buffer_num_frames),
       buffer_num_channels_(buffer_num_channels),
       output_num_frames_(output_num_frames),
       output_num_channels_(0),
       num_channels_(buffer_num_channels),
       num_bands_(NumBandsFromFramesPerChannel(buffer_num_frames_)),
       num_split_frames_(rtc::CheckedDivExact(buffer_num_frames_, num_bands_)),
       data_(new ChannelBuffer<float>(buffer_num_frames_, buffer_num_channels_)),
       output_buffer_(
           new ChannelBuffer<float>(output_num_frames_, num_channels_)) {
   RTC_DCHECK_GT(input_num_frames_, 0);
   RTC_DCHECK_GT(buffer_num_frames_, 0);
   RTC_DCHECK_GT(output_num_frames_, 0);
   RTC_DCHECK_GT(input_num_channels_, 0);
   RTC_DCHECK_GT(buffer_num_channels_, 0);
   RTC_DCHECK_LE(buffer_num_channels_, input_num_channels_);

   const bool input_resampling_needed = input_num_frames_ != buffer_num_frames_;
   const bool output_resampling_needed =
       output_num_frames_ != buffer_num_frames_;
   if (input_resampling_needed) {
     for (size_t i = 0; i < buffer_num_channels_; ++i) {
       input_resamplers_.push_back(std::unique_ptr<PushSincResampler>(
           new PushSincResampler(input_num_frames_, buffer_num_frames_)));
     }
   }

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

   if (num_bands_ > 1) {
     split_data_.reset(new ChannelBuffer<float>(
         buffer_num_frames_, buffer_num_channels_, num_bands_));
     splitting_filter_.reset(new SplittingFilter(
         buffer_num_channels_, num_bands_, buffer_num_frames_));
   }
 }

 AudioBuffer::~AudioBuffer() {}

 void AudioBuffer::set_downmixing_to_specific_channel(size_t channel) {
   downmix_by_averaging_ = false;
   RTC_DCHECK_GT(input_num_channels_, channel);
   channel_for_downmixing_ = std::min(channel, input_num_channels_ - 1);
 }

 void AudioBuffer::set_downmixing_by_averaging() {
   downmix_by_averaging_ = true;
 }

 void AudioBuffer::CopyFrom(const float* const* data,
                            const StreamConfig& stream_config) {
   RTC_DCHECK_EQ(stream_config.num_frames(), input_num_frames_);
   RTC_DCHECK_EQ(stream_config.num_channels(), input_num_channels_);
   RestoreNumChannels();
   const bool downmix_needed = input_num_channels_ > 1 && num_channels_ == 1;

   const bool resampling_needed = input_num_frames_ != buffer_num_frames_;

   if (downmix_needed) {
     RTC_DCHECK_GE(kMaxSamplesPerChannel, input_num_frames_);

     std::array<float, kMaxSamplesPerChannel> downmix;
     if (downmix_by_averaging_) {
       const float kOneByNumChannels = 1.f / input_num_channels_;
       for (size_t i = 0; i < input_num_frames_; ++i) {
         float value = data[0][i];
         for (size_t j = 1; j < input_num_channels_; ++j) {
           value += data[j][i];
         }
         downmix[i] = value * kOneByNumChannels;
       }
     }
     const float* downmixed_data =
         downmix_by_averaging_ ? downmix.data() : data[channel_for_downmixing_];

     if (resampling_needed) {
       input_resamplers_[0]->Resample(downmixed_data, input_num_frames_,
                                      data_->channels()[0], buffer_num_frames_);
     }
     const float* data_to_convert =
         resampling_needed ? data_->channels()[0] : downmixed_data;
     FloatToFloatS16(data_to_convert, buffer_num_frames_, data_->channels()[0]);
   } else {
     if (resampling_needed) {
       for (size_t i = 0; i < num_channels_; ++i) {
         input_resamplers_[i]->Resample(data[i], input_num_frames_,
                                        data_->channels()[i],
                                        buffer_num_frames_);
         FloatToFloatS16(data_->channels()[i], buffer_num_frames_,
                         data_->channels()[i]);
       }
     } else {
       for (size_t i = 0; i < num_channels_; ++i) {
         FloatToFloatS16(data[i], buffer_num_frames_, data_->channels()[i]);
       }
     }
   }
 }

 void AudioBuffer::CopyTo(const StreamConfig& stream_config,
                          float* const* data) {
   RTC_DCHECK_EQ(stream_config.num_frames(), output_num_frames_);

   const bool resampling_needed = output_num_frames_ != buffer_num_frames_;
   if (resampling_needed) {
     for (size_t i = 0; i < num_channels_; ++i) {
       FloatS16ToFloat(data_->channels()[i], buffer_num_frames_,
                       data_->channels()[i]);
       output_resamplers_[i]->Resample(data_->channels()[i], buffer_num_frames_,
                                       data[i], output_num_frames_);
     }
   } else {
     for (size_t i = 0; i < num_channels_; ++i) {
       FloatS16ToFloat(data_->channels()[i], buffer_num_frames_, data[i]);
     }
   }

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

 void AudioBuffer::RestoreNumChannels() {
   num_channels_ = buffer_num_channels_;
   data_->set_num_channels(buffer_num_channels_);
   if (split_data_.get()) {
     split_data_->set_num_channels(buffer_num_channels_);
   }
 }

 void AudioBuffer::set_num_channels(size_t num_channels) {
   RTC_DCHECK_GE(buffer_num_channels_, num_channels);
   num_channels_ = num_channels;
   data_->set_num_channels(num_channels);
   if (split_data_.get()) {
     split_data_->set_num_channels(num_channels);
   }
 }

 // The resampler is only for supporting 48kHz to 16kHz in the reverse stream.
 void AudioBuffer::CopyFrom(const AudioFrame* frame) {
   RTC_DCHECK_EQ(frame->num_channels_, input_num_channels_);
   RTC_DCHECK_EQ(frame->samples_per_channel_, input_num_frames_);
   RestoreNumChannels();

   const bool resampling_required = input_num_frames_ != buffer_num_frames_;

   const int16_t* interleaved = frame->data();
   if (num_channels_ == 1) {
     if (input_num_channels_ == 1) {
       if (resampling_required) {
         std::array<float, kMaxSamplesPerChannel> float_buffer;
         S16ToFloatS16(interleaved, input_num_frames_, float_buffer.data());
         input_resamplers_[0]->Resample(float_buffer.data(), input_num_frames_,
                                        data_->channels()[0],
                                        buffer_num_frames_);
       } else {
         S16ToFloatS16(interleaved, input_num_frames_, data_->channels()[0]);
       }
     } else {
       std::array<float, kMaxSamplesPerChannel> float_buffer;
       float* downmixed_data =
           resampling_required ? float_buffer.data() : data_->channels()[0];
       if (downmix_by_averaging_) {
         for (size_t j = 0, k = 0; j < input_num_frames_; ++j) {
           int32_t sum = 0;
           for (size_t i = 0; i < input_num_channels_; ++i, ++k) {
             sum += interleaved[k];
           }
           downmixed_data[j] = sum / static_cast<int16_t>(input_num_channels_);
         }
       } else {
         for (size_t j = 0, k = channel_for_downmixing_; j < input_num_frames_;
              ++j, k += input_num_channels_) {
           downmixed_data[j] = interleaved[k];
         }
       }

       if (resampling_required) {
         input_resamplers_[0]->Resample(downmixed_data, input_num_frames_,
                                        data_->channels()[0],
                                        buffer_num_frames_);
       }
     }
   } else {
     auto deinterleave_channel = [](size_t channel, size_t num_channels,
                                    size_t samples_per_channel, const int16_t* x,
                                    float* y) {
       for (size_t j = 0, k = channel; j < samples_per_channel;
            ++j, k += num_channels) {
         y[j] = x[k];
       }
     };

     if (resampling_required) {
       std::array<float, kMaxSamplesPerChannel> float_buffer;
       for (size_t i = 0; i < num_channels_; ++i) {
         deinterleave_channel(i, num_channels_, input_num_frames_, interleaved,
                              float_buffer.data());
         input_resamplers_[i]->Resample(float_buffer.data(), input_num_frames_,
                                        data_->channels()[i],
                                        buffer_num_frames_);
       }
     } else {
       for (size_t i = 0; i < num_channels_; ++i) {
         deinterleave_channel(i, num_channels_, input_num_frames_, interleaved,
                              data_->channels()[i]);
       }
     }
   }
 }

 void AudioBuffer::CopyTo(AudioFrame* frame) const {
   RTC_DCHECK(frame->num_channels_ == num_channels_ || num_channels_ == 1);
   RTC_DCHECK_EQ(frame->samples_per_channel_, output_num_frames_);

   const bool resampling_required = buffer_num_frames_ != output_num_frames_;

   int16_t* interleaved = frame->mutable_data();
   if (num_channels_ == 1) {
     std::array<float, kMaxSamplesPerChannel> float_buffer;

     if (resampling_required) {
       output_resamplers_[0]->Resample(data_->channels()[0], buffer_num_frames_,
                                       float_buffer.data(), output_num_frames_);
     }
     const float* deinterleaved =
         resampling_required ? float_buffer.data() : data_->channels()[0];

     if (frame->num_channels_ == 1) {
       for (size_t j = 0; j < output_num_frames_; ++j) {
         interleaved[j] = FloatS16ToS16(deinterleaved[j]);
       }
     } else {
       for (size_t i = 0, k = 0; i < output_num_frames_; ++i) {
         float tmp = FloatS16ToS16(deinterleaved[i]);
         for (size_t j = 0; j < frame->num_channels_; ++j, ++k) {
           interleaved[k] = tmp;
         }
       }
     }
   } else {
     auto interleave_channel = [](size_t channel, size_t num_channels,
                                  size_t samples_per_channel, const float* x,
                                  int16_t* y) {
       for (size_t k = 0, j = channel; k < samples_per_channel;
            ++k, j += num_channels) {
         y[j] = FloatS16ToS16(x[k]);
       }
     };

     if (resampling_required) {
       for (size_t i = 0; i < num_channels_; ++i) {
         std::array<float, kMaxSamplesPerChannel> float_buffer;
         output_resamplers_[i]->Resample(data_->channels()[i],
                                         buffer_num_frames_, float_buffer.data(),
                                         output_num_frames_);
         interleave_channel(i, frame->num_channels_, output_num_frames_,
                            float_buffer.data(), interleaved);
       }
     } else {
       for (size_t i = 0; i < num_channels_; ++i) {
         interleave_channel(i, frame->num_channels_, output_num_frames_,
                            data_->channels()[i], interleaved);
       }
     }

     for (size_t i = num_channels_; i < frame->num_channels_; ++i) {
       for (size_t j = 0, k = i, n = num_channels_; j < output_num_frames_;
            ++j, k += frame->num_channels_, n += frame->num_channels_) {
         interleaved[k] = interleaved[n];
       }
     }
   }
 }

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

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

 void AudioBuffer::ExportSplitChannelData(size_t channel,
                                          int16_t* const* split_band_data) {
   for (size_t k = 0; k < num_bands(); ++k) {
     const float* band_data = split_bands(channel)[k];

     RTC_DCHECK(split_band_data[k]);
     RTC_DCHECK(band_data);
     for (size_t i = 0; i < num_frames_per_band(); ++i) {
       split_band_data[k][i] = FloatS16ToS16(band_data[i]);
     }
   }
 }

 void AudioBuffer::ImportSplitChannelData(
     size_t channel,
     const int16_t* const* split_band_data) {
   for (size_t k = 0; k < num_bands(); ++k) {
     float* band_data = split_bands(channel)[k];
     RTC_DCHECK(split_band_data[k]);
     RTC_DCHECK(band_data);
     for (size_t i = 0; i < num_frames_per_band(); ++i) {
       band_data[i] = split_band_data[k][i];
     }
   }
 }

 }  // 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 "modules/audio_processing/audio_buffer.h"

	#include <string.h>

	#include <cstdint>

	#include "common_audio/channel_buffer.h"
	#include "common_audio/include/audio_util.h"
	#include "common_audio/resampler/push_sinc_resampler.h"
	#include "modules/audio_processing/splitting_filter.h"
	#include "rtc_base/checks.h"

	namespace webrtc {
	namespace {

	constexpr size_t kSamplesPer32kHzChannel = 320;
	constexpr size_t kSamplesPer48kHzChannel = 480;
	constexpr size_t kMaxSamplesPerChannel = AudioBuffer::kMaxSampleRate / 100;

	size_t NumBandsFromFramesPerChannel(size_t num_frames) {
	if (num_frames == kSamplesPer32kHzChannel) {
	return 2;
	}
	if (num_frames == kSamplesPer48kHzChannel) {
	return 3;
	}
	return 1;
	}

	} // namespace

	AudioBuffer::AudioBuffer(size_t input_rate,
	size_t input_num_channels,
	size_t buffer_rate,
	size_t buffer_num_channels,
	size_t output_rate,
	size_t output_num_channels)
	: AudioBuffer(static_cast<int>(input_rate) / 100,
	input_num_channels,
	static_cast<int>(buffer_rate) / 100,
	buffer_num_channels,
	static_cast<int>(output_rate) / 100) {}

	AudioBuffer::AudioBuffer(size_t input_num_frames,
	size_t input_num_channels,
	size_t buffer_num_frames,
	size_t buffer_num_channels,
	size_t output_num_frames)
	: input_num_frames_(input_num_frames),
	input_num_channels_(input_num_channels),
	buffer_num_frames_(buffer_num_frames),
	buffer_num_channels_(buffer_num_channels),
	output_num_frames_(output_num_frames),
	output_num_channels_(0),
	num_channels_(buffer_num_channels),
	num_bands_(NumBandsFromFramesPerChannel(buffer_num_frames_)),
	num_split_frames_(rtc::CheckedDivExact(buffer_num_frames_, num_bands_)),
	data_(new ChannelBuffer<float>(buffer_num_frames_, buffer_num_channels_)),
	output_buffer_(
	new ChannelBuffer<float>(output_num_frames_, num_channels_)) {
	RTC_DCHECK_GT(input_num_frames_, 0);
	RTC_DCHECK_GT(buffer_num_frames_, 0);
	RTC_DCHECK_GT(output_num_frames_, 0);
	RTC_DCHECK_GT(input_num_channels_, 0);
	RTC_DCHECK_GT(buffer_num_channels_, 0);
	RTC_DCHECK_LE(buffer_num_channels_, input_num_channels_);

	const bool input_resampling_needed = input_num_frames_ != buffer_num_frames_;
	const bool output_resampling_needed =
	output_num_frames_ != buffer_num_frames_;
	if (input_resampling_needed) {
	for (size_t i = 0; i < buffer_num_channels_; ++i) {
	input_resamplers_.push_back(std::unique_ptr<PushSincResampler>(
	new PushSincResampler(input_num_frames_, buffer_num_frames_)));
	}
	}

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

	if (num_bands_ > 1) {
	split_data_.reset(new ChannelBuffer<float>(
	buffer_num_frames_, buffer_num_channels_, num_bands_));
	splitting_filter_.reset(new SplittingFilter(
	buffer_num_channels_, num_bands_, buffer_num_frames_));
	}
	}

	AudioBuffer::~AudioBuffer() {}

	void AudioBuffer::set_downmixing_to_specific_channel(size_t channel) {
	downmix_by_averaging_ = false;
	RTC_DCHECK_GT(input_num_channels_, channel);
	channel_for_downmixing_ = std::min(channel, input_num_channels_ - 1);
	}

	void AudioBuffer::set_downmixing_by_averaging() {
	downmix_by_averaging_ = true;
	}

	void AudioBuffer::CopyFrom(const float* const* data,
	const StreamConfig& stream_config) {
	RTC_DCHECK_EQ(stream_config.num_frames(), input_num_frames_);
	RTC_DCHECK_EQ(stream_config.num_channels(), input_num_channels_);
	RestoreNumChannels();
	const bool downmix_needed = input_num_channels_ > 1 && num_channels_ == 1;

	const bool resampling_needed = input_num_frames_ != buffer_num_frames_;

	if (downmix_needed) {
	RTC_DCHECK_GE(kMaxSamplesPerChannel, input_num_frames_);

	std::array<float, kMaxSamplesPerChannel> downmix;
	if (downmix_by_averaging_) {
	const float kOneByNumChannels = 1.f / input_num_channels_;
	for (size_t i = 0; i < input_num_frames_; ++i) {
	float value = data[0][i];
	for (size_t j = 1; j < input_num_channels_; ++j) {
	value += data[j][i];
	}
	downmix[i] = value * kOneByNumChannels;
	}
	}
	const float* downmixed_data =
	downmix_by_averaging_ ? downmix.data() : data[channel_for_downmixing_];

	if (resampling_needed) {
	input_resamplers_[0]->Resample(downmixed_data, input_num_frames_,
	data_->channels()[0], buffer_num_frames_);
	}
	const float* data_to_convert =
	resampling_needed ? data_->channels()[0] : downmixed_data;
	FloatToFloatS16(data_to_convert, buffer_num_frames_, data_->channels()[0]);
	} else {
	if (resampling_needed) {
	for (size_t i = 0; i < num_channels_; ++i) {
	input_resamplers_[i]->Resample(data[i], input_num_frames_,
	data_->channels()[i],
	buffer_num_frames_);
	FloatToFloatS16(data_->channels()[i], buffer_num_frames_,
	data_->channels()[i]);
	}
	} else {
	for (size_t i = 0; i < num_channels_; ++i) {
	FloatToFloatS16(data[i], buffer_num_frames_, data_->channels()[i]);
	}
	}
	}
	}

	void AudioBuffer::CopyTo(const StreamConfig& stream_config,
	float* const* data) {
	RTC_DCHECK_EQ(stream_config.num_frames(), output_num_frames_);

	const bool resampling_needed = output_num_frames_ != buffer_num_frames_;
	if (resampling_needed) {
	for (size_t i = 0; i < num_channels_; ++i) {
	FloatS16ToFloat(data_->channels()[i], buffer_num_frames_,
	data_->channels()[i]);
	output_resamplers_[i]->Resample(data_->channels()[i], buffer_num_frames_,
	data[i], output_num_frames_);
	}
	} else {
	for (size_t i = 0; i < num_channels_; ++i) {
	FloatS16ToFloat(data_->channels()[i], buffer_num_frames_, data[i]);
	}
	}

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

	void AudioBuffer::RestoreNumChannels() {
	num_channels_ = buffer_num_channels_;
	data_->set_num_channels(buffer_num_channels_);
	if (split_data_.get()) {
	split_data_->set_num_channels(buffer_num_channels_);
	}
	}

	void AudioBuffer::set_num_channels(size_t num_channels) {
	RTC_DCHECK_GE(buffer_num_channels_, num_channels);
	num_channels_ = num_channels;
	data_->set_num_channels(num_channels);
	if (split_data_.get()) {
	split_data_->set_num_channels(num_channels);
	}
	}

	// The resampler is only for supporting 48kHz to 16kHz in the reverse stream.
	void AudioBuffer::CopyFrom(const AudioFrame* frame) {
	RTC_DCHECK_EQ(frame->num_channels_, input_num_channels_);
	RTC_DCHECK_EQ(frame->samples_per_channel_, input_num_frames_);
	RestoreNumChannels();

	const bool resampling_required = input_num_frames_ != buffer_num_frames_;

	const int16_t* interleaved = frame->data();
	if (num_channels_ == 1) {
	if (input_num_channels_ == 1) {
	if (resampling_required) {
	std::array<float, kMaxSamplesPerChannel> float_buffer;
	S16ToFloatS16(interleaved, input_num_frames_, float_buffer.data());
	input_resamplers_[0]->Resample(float_buffer.data(), input_num_frames_,
	data_->channels()[0],
	buffer_num_frames_);
	} else {
	S16ToFloatS16(interleaved, input_num_frames_, data_->channels()[0]);
	}
	} else {
	std::array<float, kMaxSamplesPerChannel> float_buffer;
	float* downmixed_data =
	resampling_required ? float_buffer.data() : data_->channels()[0];
	if (downmix_by_averaging_) {
	for (size_t j = 0, k = 0; j < input_num_frames_; ++j) {
	int32_t sum = 0;
	for (size_t i = 0; i < input_num_channels_; ++i, ++k) {
	sum += interleaved[k];
	}
	downmixed_data[j] = sum / static_cast<int16_t>(input_num_channels_);
	}
	} else {
	for (size_t j = 0, k = channel_for_downmixing_; j < input_num_frames_;
	++j, k += input_num_channels_) {
	downmixed_data[j] = interleaved[k];
	}
	}

	if (resampling_required) {
	input_resamplers_[0]->Resample(downmixed_data, input_num_frames_,
	data_->channels()[0],
	buffer_num_frames_);
	}
	}
	} else {
	auto deinterleave_channel = [](size_t channel, size_t num_channels,
	size_t samples_per_channel, const int16_t* x,
	float* y) {
	for (size_t j = 0, k = channel; j < samples_per_channel;
	++j, k += num_channels) {
	y[j] = x[k];
	}
	};

	if (resampling_required) {
	std::array<float, kMaxSamplesPerChannel> float_buffer;
	for (size_t i = 0; i < num_channels_; ++i) {
	deinterleave_channel(i, num_channels_, input_num_frames_, interleaved,
	float_buffer.data());
	input_resamplers_[i]->Resample(float_buffer.data(), input_num_frames_,
	data_->channels()[i],
	buffer_num_frames_);
	}
	} else {
	for (size_t i = 0; i < num_channels_; ++i) {
	deinterleave_channel(i, num_channels_, input_num_frames_, interleaved,
	data_->channels()[i]);
	}
	}
	}
	}

	void AudioBuffer::CopyTo(AudioFrame* frame) const {
	RTC_DCHECK(frame->num_channels_ == num_channels_ \|\| num_channels_ == 1);
	RTC_DCHECK_EQ(frame->samples_per_channel_, output_num_frames_);

	const bool resampling_required = buffer_num_frames_ != output_num_frames_;

	int16_t* interleaved = frame->mutable_data();
	if (num_channels_ == 1) {
	std::array<float, kMaxSamplesPerChannel> float_buffer;

	if (resampling_required) {
	output_resamplers_[0]->Resample(data_->channels()[0], buffer_num_frames_,
	float_buffer.data(), output_num_frames_);
	}
	const float* deinterleaved =
	resampling_required ? float_buffer.data() : data_->channels()[0];

	if (frame->num_channels_ == 1) {
	for (size_t j = 0; j < output_num_frames_; ++j) {
	interleaved[j] = FloatS16ToS16(deinterleaved[j]);
	}
	} else {
	for (size_t i = 0, k = 0; i < output_num_frames_; ++i) {
	float tmp = FloatS16ToS16(deinterleaved[i]);
	for (size_t j = 0; j < frame->num_channels_; ++j, ++k) {
	interleaved[k] = tmp;
	}
	}
	}
	} else {
	auto interleave_channel = [](size_t channel, size_t num_channels,
	size_t samples_per_channel, const float* x,
	int16_t* y) {
	for (size_t k = 0, j = channel; k < samples_per_channel;
	++k, j += num_channels) {
	y[j] = FloatS16ToS16(x[k]);
	}
	};

	if (resampling_required) {
	for (size_t i = 0; i < num_channels_; ++i) {
	std::array<float, kMaxSamplesPerChannel> float_buffer;
	output_resamplers_[i]->Resample(data_->channels()[i],
	buffer_num_frames_, float_buffer.data(),
	output_num_frames_);
	interleave_channel(i, frame->num_channels_, output_num_frames_,
	float_buffer.data(), interleaved);
	}
	} else {
	for (size_t i = 0; i < num_channels_; ++i) {
	interleave_channel(i, frame->num_channels_, output_num_frames_,
	data_->channels()[i], interleaved);
	}
	}

	for (size_t i = num_channels_; i < frame->num_channels_; ++i) {
	for (size_t j = 0, k = i, n = num_channels_; j < output_num_frames_;
	++j, k += frame->num_channels_, n += frame->num_channels_) {
	interleaved[k] = interleaved[n];
	}
	}
	}
	}

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

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

	void AudioBuffer::ExportSplitChannelData(size_t channel,
	int16_t* const* split_band_data) {
	for (size_t k = 0; k < num_bands(); ++k) {
	const float* band_data = split_bands(channel)[k];

	RTC_DCHECK(split_band_data[k]);
	RTC_DCHECK(band_data);
	for (size_t i = 0; i < num_frames_per_band(); ++i) {
	split_band_data[k][i] = FloatS16ToS16(band_data[i]);
	}
	}
	}

	void AudioBuffer::ImportSplitChannelData(
	size_t channel,
	const int16_t* const* split_band_data) {
	for (size_t k = 0; k < num_bands(); ++k) {
	float* band_data = split_bands(channel)[k];
	RTC_DCHECK(split_band_data[k]);
	RTC_DCHECK(band_data);
	for (size_t i = 0; i < num_frames_per_band(); ++i) {
	band_data[i] = split_band_data[k][i];
	}
	}
	}

	} // namespace webrtc