modules/audio_mixer/frame_combiner.cc - src - Git at Google

 /*
  *  Copyright (c) 2017 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_mixer/frame_combiner.h"

 #include <algorithm>
 #include <array>
 #include <functional>

 #include "api/array_view.h"
 #include "audio/utility/audio_frame_operations.h"
 #include "common_audio/include/audio_util.h"
 #include "modules/audio_mixer/audio_frame_manipulator.h"
 #include "modules/audio_mixer/audio_mixer_impl.h"
 #include "modules/audio_processing/logging/apm_data_dumper.h"
 #include "rtc_base/arraysize.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
 #include "system_wrappers/include/metrics.h"

 namespace webrtc {
 namespace {

 // Stereo, 48 kHz, 10 ms.
 constexpr int kMaximumAmountOfChannels = 2;
 constexpr int kMaximumChannelSize = 48 * AudioMixerImpl::kFrameDurationInMs;

 using OneChannelBuffer = std::array<float, kMaximumChannelSize>;

 std::unique_ptr<AudioProcessing> CreateLimiter() {
   Config config;
   config.Set<ExperimentalAgc>(new ExperimentalAgc(false));

   std::unique_ptr<AudioProcessing> limiter(
       AudioProcessingBuilder().Create(config));
   RTC_DCHECK(limiter);
   webrtc::AudioProcessing::Config apm_config;
   apm_config.residual_echo_detector.enabled = false;
   limiter->ApplyConfig(apm_config);

   const auto check_no_error = [](int x) {
     RTC_DCHECK_EQ(x, AudioProcessing::kNoError);
   };
   auto* const gain_control = limiter->gain_control();
   check_no_error(gain_control->set_mode(GainControl::kFixedDigital));

   // We smoothly limit the mixed frame to -7 dbFS. -6 would correspond to the
   // divide-by-2 but -7 is used instead to give a bit of headroom since the
   // AGC is not a hard limiter.
   check_no_error(gain_control->set_target_level_dbfs(7));

   check_no_error(gain_control->set_compression_gain_db(0));
   check_no_error(gain_control->enable_limiter(true));
   check_no_error(gain_control->Enable(true));

   return limiter;
 }

 void SetAudioFrameFields(const std::vector<AudioFrame*>& mix_list,
                          size_t number_of_channels,
                          int sample_rate,
                          size_t number_of_streams,
                          AudioFrame* audio_frame_for_mixing) {
   const size_t samples_per_channel = static_cast<size_t>(
       (sample_rate * webrtc::AudioMixerImpl::kFrameDurationInMs) / 1000);

   // TODO(minyue): Issue bugs.webrtc.org/3390.
   // Audio frame timestamp. The 'timestamp_' field is set to dummy
   // value '0', because it is only supported in the one channel case and
   // is then updated in the helper functions.
   audio_frame_for_mixing->UpdateFrame(
       0, nullptr, samples_per_channel, sample_rate, AudioFrame::kUndefined,
       AudioFrame::kVadUnknown, number_of_channels);

   if (mix_list.empty()) {
     audio_frame_for_mixing->elapsed_time_ms_ = -1;
   } else if (mix_list.size() == 1) {
     audio_frame_for_mixing->timestamp_ = mix_list[0]->timestamp_;
     audio_frame_for_mixing->elapsed_time_ms_ = mix_list[0]->elapsed_time_ms_;
   }
 }

 void MixFewFramesWithNoLimiter(const std::vector<AudioFrame*>& mix_list,
                                AudioFrame* audio_frame_for_mixing) {
   if (mix_list.empty()) {
     audio_frame_for_mixing->Mute();
     return;
   }
   RTC_DCHECK_LE(mix_list.size(), 1);
   std::copy(mix_list[0]->data(),
             mix_list[0]->data() +
                 mix_list[0]->num_channels_ * mix_list[0]->samples_per_channel_,
             audio_frame_for_mixing->mutable_data());
 }

 std::array<OneChannelBuffer, kMaximumAmountOfChannels> MixToFloatFrame(
     const std::vector<AudioFrame*>& mix_list,
     size_t samples_per_channel,
     size_t number_of_channels) {
   // Convert to FloatS16 and mix.
   using OneChannelBuffer = std::array<float, kMaximumChannelSize>;
   std::array<OneChannelBuffer, kMaximumAmountOfChannels> mixing_buffer{};

   for (size_t i = 0; i < mix_list.size(); ++i) {
     const AudioFrame* const frame = mix_list[i];
     for (size_t j = 0; j < number_of_channels; ++j) {
       for (size_t k = 0; k < samples_per_channel; ++k) {
         mixing_buffer[j][k] += frame->data()[number_of_channels * k + j];
       }
     }
   }
   return mixing_buffer;
 }

 void RunApmAgcLimiter(AudioFrameView<float> mixing_buffer_view,
                       AudioProcessing* apm_agc_limiter) {
   // Halve all samples to avoid saturation before limiting. The input
   // format of APM is Float. Convert the samples from FloatS16 to
   // Float.
   for (size_t i = 0; i < mixing_buffer_view.num_channels(); ++i) {
     std::transform(mixing_buffer_view.channel(i).begin(),
                    mixing_buffer_view.channel(i).end(),
                    mixing_buffer_view.channel(i).begin(),
                    [](float a) { return FloatS16ToFloat(a / 2); });
   }

   const int sample_rate =
       static_cast<int>(mixing_buffer_view.samples_per_channel()) * 1000 /
       AudioMixerImpl::kFrameDurationInMs;
   StreamConfig processing_config(sample_rate,
                                  mixing_buffer_view.num_channels());

   // Smoothly limit the audio.
   apm_agc_limiter->ProcessStream(mixing_buffer_view.data(), processing_config,
                                  processing_config, mixing_buffer_view.data());

   // And now we can safely restore the level. This procedure results in
   // some loss of resolution, deemed acceptable.
   //
   // It's possible to apply the gain in the AGC (with a target level of 0 dbFS
   // and compression gain of 6 dB). However, in the transition frame when this
   // is enabled (moving from one to two audio sources) it has the potential to
   // create discontinuities in the mixed frame.
   //
   // Instead we double the samples in the frame..
   // Also convert the samples back to FloatS16.
   for (size_t i = 0; i < mixing_buffer_view.num_channels(); ++i) {
     std::transform(mixing_buffer_view.channel(i).begin(),
                    mixing_buffer_view.channel(i).end(),
                    mixing_buffer_view.channel(i).begin(),
                    [](float a) { return FloatToFloatS16(a * 2); });
   }
 }

 void RunApmAgc2Limiter(AudioFrameView<float> mixing_buffer_view,
                        FixedGainController* apm_agc2_limiter) {
   const size_t sample_rate = mixing_buffer_view.samples_per_channel() * 1000 /
                              AudioMixerImpl::kFrameDurationInMs;
   apm_agc2_limiter->SetSampleRate(sample_rate);
   apm_agc2_limiter->Process(mixing_buffer_view);
 }

 // Both interleaves and rounds.
 void InterleaveToAudioFrame(AudioFrameView<const float> mixing_buffer_view,
                             AudioFrame* audio_frame_for_mixing) {
   const size_t number_of_channels = mixing_buffer_view.num_channels();
   const size_t samples_per_channel = mixing_buffer_view.samples_per_channel();
   // Put data in the result frame.
   for (size_t i = 0; i < number_of_channels; ++i) {
     for (size_t j = 0; j < samples_per_channel; ++j) {
       audio_frame_for_mixing->mutable_data()[number_of_channels * j + i] =
           FloatS16ToS16(mixing_buffer_view.channel(i)[j]);
     }
   }
 }
 }  // namespace

 FrameCombiner::FrameCombiner(LimiterType limiter_type)
     : limiter_type_(limiter_type),
       apm_agc_limiter_(limiter_type_ == LimiterType::kApmAgcLimiter
                            ? CreateLimiter()
                            : nullptr),
       data_dumper_(new ApmDataDumper(0)),
       apm_agc2_limiter_(data_dumper_.get()) {
   apm_agc2_limiter_.SetGain(0.f);
 }

 FrameCombiner::FrameCombiner(bool use_limiter)
     : FrameCombiner(use_limiter ? LimiterType::kApmAgcLimiter
                                 : LimiterType::kNoLimiter) {}

 FrameCombiner::~FrameCombiner() = default;

 void FrameCombiner::SetLimiterType(LimiterType limiter_type) {
   // TODO(aleloi): remove this method and make limiter_type_ const
   // when we have finished moved to APM-AGC2.
   limiter_type_ = limiter_type;
   if (limiter_type_ == LimiterType::kApmAgcLimiter &&
       apm_agc_limiter_ == nullptr) {
     apm_agc_limiter_ = CreateLimiter();
   }
 }

 void FrameCombiner::Combine(const std::vector<AudioFrame*>& mix_list,
                             size_t number_of_channels,
                             int sample_rate,
                             size_t number_of_streams,
                             AudioFrame* audio_frame_for_mixing) {
   RTC_DCHECK(audio_frame_for_mixing);

   LogMixingStats(mix_list, sample_rate, number_of_streams);

   SetAudioFrameFields(mix_list, number_of_channels, sample_rate,
                       number_of_streams, audio_frame_for_mixing);

   const size_t samples_per_channel = static_cast<size_t>(
       (sample_rate * webrtc::AudioMixerImpl::kFrameDurationInMs) / 1000);

   for (const auto* frame : mix_list) {
     RTC_DCHECK_EQ(samples_per_channel, frame->samples_per_channel_);
     RTC_DCHECK_EQ(sample_rate, frame->sample_rate_hz_);
   }

   // The 'num_channels_' field of frames in 'mix_list' could be
   // different from 'number_of_channels'.
   for (auto* frame : mix_list) {
     RemixFrame(number_of_channels, frame);
   }

   if (number_of_streams <= 1) {
     MixFewFramesWithNoLimiter(mix_list, audio_frame_for_mixing);
     return;
   }

   std::array<OneChannelBuffer, kMaximumAmountOfChannels> mixing_buffer =
       MixToFloatFrame(mix_list, samples_per_channel, number_of_channels);

   // Put float data in an AudioFrameView.
   std::array<float*, kMaximumAmountOfChannels> channel_pointers{};
   for (size_t i = 0; i < number_of_channels; ++i) {
     channel_pointers[i] = &mixing_buffer[i][0];
   }
   AudioFrameView<float> mixing_buffer_view(
       &channel_pointers[0], number_of_channels, samples_per_channel);

   if (limiter_type_ == LimiterType::kApmAgcLimiter) {
     RunApmAgcLimiter(mixing_buffer_view, apm_agc_limiter_.get());
   } else if (limiter_type_ == LimiterType::kApmAgc2Limiter) {
     RunApmAgc2Limiter(mixing_buffer_view, &apm_agc2_limiter_);
   }

   InterleaveToAudioFrame(mixing_buffer_view, audio_frame_for_mixing);
 }

 void FrameCombiner::LogMixingStats(const std::vector<AudioFrame*>& mix_list,
                                    int sample_rate,
                                    size_t number_of_streams) const {
   // Log every second.
   uma_logging_counter_++;
   if (uma_logging_counter_ > 1000 / AudioMixerImpl::kFrameDurationInMs) {
     uma_logging_counter_ = 0;
     RTC_HISTOGRAM_COUNTS_100("WebRTC.Audio.AudioMixer.NumIncomingStreams",
                              static_cast<int>(number_of_streams));
     RTC_HISTOGRAM_ENUMERATION(
         "WebRTC.Audio.AudioMixer.NumIncomingActiveStreams",
         static_cast<int>(mix_list.size()),
         AudioMixerImpl::kMaximumAmountOfMixedAudioSources);

     using NativeRate = AudioProcessing::NativeRate;
     static constexpr NativeRate native_rates[] = {
         NativeRate::kSampleRate8kHz, NativeRate::kSampleRate16kHz,
         NativeRate::kSampleRate32kHz, NativeRate::kSampleRate48kHz};
     const auto* rate_position = std::lower_bound(
         std::begin(native_rates), std::end(native_rates), sample_rate);

     RTC_HISTOGRAM_ENUMERATION(
         "WebRTC.Audio.AudioMixer.MixingRate",
         std::distance(std::begin(native_rates), rate_position),
         arraysize(native_rates));
   }
 }

 }  // namespace webrtc
	/*
	* Copyright (c) 2017 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_mixer/frame_combiner.h"

	#include <algorithm>
	#include <array>
	#include <functional>

	#include "api/array_view.h"
	#include "audio/utility/audio_frame_operations.h"
	#include "common_audio/include/audio_util.h"
	#include "modules/audio_mixer/audio_frame_manipulator.h"
	#include "modules/audio_mixer/audio_mixer_impl.h"
	#include "modules/audio_processing/logging/apm_data_dumper.h"
	#include "rtc_base/arraysize.h"
	#include "rtc_base/checks.h"
	#include "rtc_base/logging.h"
	#include "system_wrappers/include/metrics.h"

	namespace webrtc {
	namespace {

	// Stereo, 48 kHz, 10 ms.
	constexpr int kMaximumAmountOfChannels = 2;
	constexpr int kMaximumChannelSize = 48 * AudioMixerImpl::kFrameDurationInMs;

	using OneChannelBuffer = std::array<float, kMaximumChannelSize>;

	std::unique_ptr<AudioProcessing> CreateLimiter() {
	Config config;
	config.Set<ExperimentalAgc>(new ExperimentalAgc(false));

	std::unique_ptr<AudioProcessing> limiter(
	AudioProcessingBuilder().Create(config));
	RTC_DCHECK(limiter);
	webrtc::AudioProcessing::Config apm_config;
	apm_config.residual_echo_detector.enabled = false;
	limiter->ApplyConfig(apm_config);

	const auto check_no_error = [](int x) {
	RTC_DCHECK_EQ(x, AudioProcessing::kNoError);
	};
	auto* const gain_control = limiter->gain_control();
	check_no_error(gain_control->set_mode(GainControl::kFixedDigital));

	// We smoothly limit the mixed frame to -7 dbFS. -6 would correspond to the
	// divide-by-2 but -7 is used instead to give a bit of headroom since the
	// AGC is not a hard limiter.
	check_no_error(gain_control->set_target_level_dbfs(7));

	check_no_error(gain_control->set_compression_gain_db(0));
	check_no_error(gain_control->enable_limiter(true));
	check_no_error(gain_control->Enable(true));

	return limiter;
	}

	void SetAudioFrameFields(const std::vector<AudioFrame*>& mix_list,
	size_t number_of_channels,
	int sample_rate,
	size_t number_of_streams,
	AudioFrame* audio_frame_for_mixing) {
	const size_t samples_per_channel = static_cast<size_t>(
	(sample_rate * webrtc::AudioMixerImpl::kFrameDurationInMs) / 1000);

	// TODO(minyue): Issue bugs.webrtc.org/3390.
	// Audio frame timestamp. The 'timestamp_' field is set to dummy
	// value '0', because it is only supported in the one channel case and
	// is then updated in the helper functions.
	audio_frame_for_mixing->UpdateFrame(
	0, nullptr, samples_per_channel, sample_rate, AudioFrame::kUndefined,
	AudioFrame::kVadUnknown, number_of_channels);

	if (mix_list.empty()) {
	audio_frame_for_mixing->elapsed_time_ms_ = -1;
	} else if (mix_list.size() == 1) {
	audio_frame_for_mixing->timestamp_ = mix_list[0]->timestamp_;
	audio_frame_for_mixing->elapsed_time_ms_ = mix_list[0]->elapsed_time_ms_;
	}
	}

	void MixFewFramesWithNoLimiter(const std::vector<AudioFrame*>& mix_list,
	AudioFrame* audio_frame_for_mixing) {
	if (mix_list.empty()) {
	audio_frame_for_mixing->Mute();
	return;
	}
	RTC_DCHECK_LE(mix_list.size(), 1);
	std::copy(mix_list[0]->data(),
	mix_list[0]->data() +
	mix_list[0]->num_channels_ * mix_list[0]->samples_per_channel_,
	audio_frame_for_mixing->mutable_data());
	}

	std::array<OneChannelBuffer, kMaximumAmountOfChannels> MixToFloatFrame(
	const std::vector<AudioFrame*>& mix_list,
	size_t samples_per_channel,
	size_t number_of_channels) {
	// Convert to FloatS16 and mix.
	using OneChannelBuffer = std::array<float, kMaximumChannelSize>;
	std::array<OneChannelBuffer, kMaximumAmountOfChannels> mixing_buffer{};

	for (size_t i = 0; i < mix_list.size(); ++i) {
	const AudioFrame* const frame = mix_list[i];
	for (size_t j = 0; j < number_of_channels; ++j) {
	for (size_t k = 0; k < samples_per_channel; ++k) {
	mixing_buffer[j][k] += frame->data()[number_of_channels * k + j];
	}
	}
	}
	return mixing_buffer;
	}

	void RunApmAgcLimiter(AudioFrameView<float> mixing_buffer_view,
	AudioProcessing* apm_agc_limiter) {
	// Halve all samples to avoid saturation before limiting. The input
	// format of APM is Float. Convert the samples from FloatS16 to
	// Float.
	for (size_t i = 0; i < mixing_buffer_view.num_channels(); ++i) {
	std::transform(mixing_buffer_view.channel(i).begin(),
	mixing_buffer_view.channel(i).end(),
	mixing_buffer_view.channel(i).begin(),
	[](float a) { return FloatS16ToFloat(a / 2); });
	}

	const int sample_rate =
	static_cast<int>(mixing_buffer_view.samples_per_channel()) * 1000 /
	AudioMixerImpl::kFrameDurationInMs;
	StreamConfig processing_config(sample_rate,
	mixing_buffer_view.num_channels());

	// Smoothly limit the audio.
	apm_agc_limiter->ProcessStream(mixing_buffer_view.data(), processing_config,
	processing_config, mixing_buffer_view.data());

	// And now we can safely restore the level. This procedure results in
	// some loss of resolution, deemed acceptable.
	//
	// It's possible to apply the gain in the AGC (with a target level of 0 dbFS
	// and compression gain of 6 dB). However, in the transition frame when this
	// is enabled (moving from one to two audio sources) it has the potential to
	// create discontinuities in the mixed frame.
	//
	// Instead we double the samples in the frame..
	// Also convert the samples back to FloatS16.
	for (size_t i = 0; i < mixing_buffer_view.num_channels(); ++i) {
	std::transform(mixing_buffer_view.channel(i).begin(),
	mixing_buffer_view.channel(i).end(),
	mixing_buffer_view.channel(i).begin(),
	[](float a) { return FloatToFloatS16(a * 2); });
	}
	}

	void RunApmAgc2Limiter(AudioFrameView<float> mixing_buffer_view,
	FixedGainController* apm_agc2_limiter) {
	const size_t sample_rate = mixing_buffer_view.samples_per_channel() * 1000 /
	AudioMixerImpl::kFrameDurationInMs;
	apm_agc2_limiter->SetSampleRate(sample_rate);
	apm_agc2_limiter->Process(mixing_buffer_view);
	}

	// Both interleaves and rounds.
	void InterleaveToAudioFrame(AudioFrameView<const float> mixing_buffer_view,
	AudioFrame* audio_frame_for_mixing) {
	const size_t number_of_channels = mixing_buffer_view.num_channels();
	const size_t samples_per_channel = mixing_buffer_view.samples_per_channel();
	// Put data in the result frame.
	for (size_t i = 0; i < number_of_channels; ++i) {
	for (size_t j = 0; j < samples_per_channel; ++j) {
	audio_frame_for_mixing->mutable_data()[number_of_channels * j + i] =
	FloatS16ToS16(mixing_buffer_view.channel(i)[j]);
	}
	}
	}
	} // namespace

	FrameCombiner::FrameCombiner(LimiterType limiter_type)
	: limiter_type_(limiter_type),
	apm_agc_limiter_(limiter_type_ == LimiterType::kApmAgcLimiter
	? CreateLimiter()
	: nullptr),
	data_dumper_(new ApmDataDumper(0)),
	apm_agc2_limiter_(data_dumper_.get()) {
	apm_agc2_limiter_.SetGain(0.f);
	}

	FrameCombiner::FrameCombiner(bool use_limiter)
	: FrameCombiner(use_limiter ? LimiterType::kApmAgcLimiter
	: LimiterType::kNoLimiter) {}

	FrameCombiner::~FrameCombiner() = default;

	void FrameCombiner::SetLimiterType(LimiterType limiter_type) {
	// TODO(aleloi): remove this method and make limiter_type_ const
	// when we have finished moved to APM-AGC2.
	limiter_type_ = limiter_type;
	if (limiter_type_ == LimiterType::kApmAgcLimiter &&
	apm_agc_limiter_ == nullptr) {
	apm_agc_limiter_ = CreateLimiter();
	}
	}

	void FrameCombiner::Combine(const std::vector<AudioFrame*>& mix_list,
	size_t number_of_channels,
	int sample_rate,
	size_t number_of_streams,
	AudioFrame* audio_frame_for_mixing) {
	RTC_DCHECK(audio_frame_for_mixing);

	LogMixingStats(mix_list, sample_rate, number_of_streams);

	SetAudioFrameFields(mix_list, number_of_channels, sample_rate,
	number_of_streams, audio_frame_for_mixing);

	const size_t samples_per_channel = static_cast<size_t>(
	(sample_rate * webrtc::AudioMixerImpl::kFrameDurationInMs) / 1000);

	for (const auto* frame : mix_list) {
	RTC_DCHECK_EQ(samples_per_channel, frame->samples_per_channel_);
	RTC_DCHECK_EQ(sample_rate, frame->sample_rate_hz_);
	}

	// The 'num_channels_' field of frames in 'mix_list' could be
	// different from 'number_of_channels'.
	for (auto* frame : mix_list) {
	RemixFrame(number_of_channels, frame);
	}

	if (number_of_streams <= 1) {
	MixFewFramesWithNoLimiter(mix_list, audio_frame_for_mixing);
	return;
	}

	std::array<OneChannelBuffer, kMaximumAmountOfChannels> mixing_buffer =
	MixToFloatFrame(mix_list, samples_per_channel, number_of_channels);

	// Put float data in an AudioFrameView.
	std::array<float*, kMaximumAmountOfChannels> channel_pointers{};
	for (size_t i = 0; i < number_of_channels; ++i) {
	channel_pointers[i] = &mixing_buffer[i][0];
	}
	AudioFrameView<float> mixing_buffer_view(
	&channel_pointers[0], number_of_channels, samples_per_channel);

	if (limiter_type_ == LimiterType::kApmAgcLimiter) {
	RunApmAgcLimiter(mixing_buffer_view, apm_agc_limiter_.get());
	} else if (limiter_type_ == LimiterType::kApmAgc2Limiter) {
	RunApmAgc2Limiter(mixing_buffer_view, &apm_agc2_limiter_);
	}

	InterleaveToAudioFrame(mixing_buffer_view, audio_frame_for_mixing);
	}

	void FrameCombiner::LogMixingStats(const std::vector<AudioFrame*>& mix_list,
	int sample_rate,
	size_t number_of_streams) const {
	// Log every second.
	uma_logging_counter_++;
	if (uma_logging_counter_ > 1000 / AudioMixerImpl::kFrameDurationInMs) {
	uma_logging_counter_ = 0;
	RTC_HISTOGRAM_COUNTS_100("WebRTC.Audio.AudioMixer.NumIncomingStreams",
	static_cast<int>(number_of_streams));
	RTC_HISTOGRAM_ENUMERATION(
	"WebRTC.Audio.AudioMixer.NumIncomingActiveStreams",
	static_cast<int>(mix_list.size()),
	AudioMixerImpl::kMaximumAmountOfMixedAudioSources);

	using NativeRate = AudioProcessing::NativeRate;
	static constexpr NativeRate native_rates[] = {
	NativeRate::kSampleRate8kHz, NativeRate::kSampleRate16kHz,
	NativeRate::kSampleRate32kHz, NativeRate::kSampleRate48kHz};
	const auto* rate_position = std::lower_bound(
	std::begin(native_rates), std::end(native_rates), sample_rate);

	RTC_HISTOGRAM_ENUMERATION(
	"WebRTC.Audio.AudioMixer.MixingRate",
	std::distance(std::begin(native_rates), rate_position),
	arraysize(native_rates));
	}
	}

	} // namespace webrtc