audio/remix_resample_unittest.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 "audio/remix_resample.h"

 #include <cmath>

 #include "common_audio/resampler/include/push_resampler.h"
 #include "rtc_base/arraysize.h"
 #include "rtc_base/checks.h"
 #include "test/gtest.h"

 namespace webrtc {
 namespace voe {
 namespace {

 class UtilityTest : public ::testing::Test {
  protected:
   UtilityTest() {
     src_frame_.sample_rate_hz_ = 16000;
     src_frame_.samples_per_channel_ = src_frame_.sample_rate_hz_ / 100;
     src_frame_.num_channels_ = 1;
     dst_frame_.CopyFrom(src_frame_);
     golden_frame_.CopyFrom(src_frame_);
   }

   void RunResampleTest(int src_channels,
                        int src_sample_rate_hz,
                        int dst_channels,
                        int dst_sample_rate_hz);

   PushResampler<int16_t> resampler_;
   AudioFrame src_frame_;
   AudioFrame dst_frame_;
   AudioFrame golden_frame_;
 };

 // Sets the signal value to increase by `data` with every sample. Floats are
 // used so non-integer values result in rounding error, but not an accumulating
 // error.
 void SetMonoFrame(float data, int sample_rate_hz, AudioFrame* frame) {
   frame->Mute();
   frame->num_channels_ = 1;
   frame->sample_rate_hz_ = sample_rate_hz;
   frame->samples_per_channel_ = rtc::CheckedDivExact(sample_rate_hz, 100);
   int16_t* frame_data = frame->mutable_data();
   for (size_t i = 0; i < frame->samples_per_channel_; i++) {
     frame_data[i] = static_cast<int16_t>(data * i);
   }
 }

 // Keep the existing sample rate.
 void SetMonoFrame(float data, AudioFrame* frame) {
   SetMonoFrame(data, frame->sample_rate_hz_, frame);
 }

 // Sets the signal value to increase by `left` and `right` with every sample in
 // each channel respectively.
 void SetStereoFrame(float left,
                     float right,
                     int sample_rate_hz,
                     AudioFrame* frame) {
   frame->Mute();
   frame->num_channels_ = 2;
   frame->sample_rate_hz_ = sample_rate_hz;
   frame->samples_per_channel_ = rtc::CheckedDivExact(sample_rate_hz, 100);
   int16_t* frame_data = frame->mutable_data();
   for (size_t i = 0; i < frame->samples_per_channel_; i++) {
     frame_data[i * 2] = static_cast<int16_t>(left * i);
     frame_data[i * 2 + 1] = static_cast<int16_t>(right * i);
   }
 }

 // Keep the existing sample rate.
 void SetStereoFrame(float left, float right, AudioFrame* frame) {
   SetStereoFrame(left, right, frame->sample_rate_hz_, frame);
 }

 // Sets the signal value to increase by `ch1`, `ch2`, `ch3`, `ch4` with every
 // sample in each channel respectively.
 void SetQuadFrame(float ch1,
                   float ch2,
                   float ch3,
                   float ch4,
                   int sample_rate_hz,
                   AudioFrame* frame) {
   frame->Mute();
   frame->num_channels_ = 4;
   frame->sample_rate_hz_ = sample_rate_hz;
   frame->samples_per_channel_ = rtc::CheckedDivExact(sample_rate_hz, 100);
   int16_t* frame_data = frame->mutable_data();
   for (size_t i = 0; i < frame->samples_per_channel_; i++) {
     frame_data[i * 4] = static_cast<int16_t>(ch1 * i);
     frame_data[i * 4 + 1] = static_cast<int16_t>(ch2 * i);
     frame_data[i * 4 + 2] = static_cast<int16_t>(ch3 * i);
     frame_data[i * 4 + 3] = static_cast<int16_t>(ch4 * i);
   }
 }

 void VerifyParams(const AudioFrame& ref_frame, const AudioFrame& test_frame) {
   EXPECT_EQ(ref_frame.num_channels_, test_frame.num_channels_);
   EXPECT_EQ(ref_frame.samples_per_channel_, test_frame.samples_per_channel_);
   EXPECT_EQ(ref_frame.sample_rate_hz_, test_frame.sample_rate_hz_);
 }

 // Computes the best SNR based on the error between `ref_frame` and
 // `test_frame`. It allows for up to a `max_delay` in samples between the
 // signals to compensate for the resampling delay.
 float ComputeSNR(const AudioFrame& ref_frame,
                  const AudioFrame& test_frame,
                  size_t max_delay) {
   VerifyParams(ref_frame, test_frame);
   float best_snr = 0;
   size_t best_delay = 0;
   for (size_t delay = 0; delay <= max_delay; delay++) {
     float mse = 0;
     float variance = 0;
     const int16_t* ref_frame_data = ref_frame.data();
     const int16_t* test_frame_data = test_frame.data();
     for (size_t i = 0;
          i < ref_frame.samples_per_channel_ * ref_frame.num_channels_ - delay;
          i++) {
       int error = ref_frame_data[i] - test_frame_data[i + delay];
       mse += error * error;
       variance += ref_frame_data[i] * ref_frame_data[i];
     }
     float snr = 100;  // We assign 100 dB to the zero-error case.
     if (mse > 0)
       snr = 10 * std::log10(variance / mse);
     if (snr > best_snr) {
       best_snr = snr;
       best_delay = delay;
     }
   }
   printf("SNR=%.1f dB at delay=%zu\n", best_snr, best_delay);
   return best_snr;
 }

 void VerifyFramesAreEqual(const AudioFrame& ref_frame,
                           const AudioFrame& test_frame) {
   VerifyParams(ref_frame, test_frame);
   const int16_t* ref_frame_data = ref_frame.data();
   const int16_t* test_frame_data = test_frame.data();
   for (size_t i = 0;
        i < ref_frame.samples_per_channel_ * ref_frame.num_channels_; i++) {
     EXPECT_EQ(ref_frame_data[i], test_frame_data[i]);
   }
 }

 void UtilityTest::RunResampleTest(int src_channels,
                                   int src_sample_rate_hz,
                                   int dst_channels,
                                   int dst_sample_rate_hz) {
   PushResampler<int16_t> resampler;  // Create a new one with every test.
   const int16_t kSrcCh1 = 30;  // Shouldn't overflow for any used sample rate.
   const int16_t kSrcCh2 = 15;
   const int16_t kSrcCh3 = 22;
   const int16_t kSrcCh4 = 8;
   const float resampling_factor =
       (1.0 * src_sample_rate_hz) / dst_sample_rate_hz;
   const float dst_ch1 = resampling_factor * kSrcCh1;
   const float dst_ch2 = resampling_factor * kSrcCh2;
   const float dst_ch3 = resampling_factor * kSrcCh3;
   const float dst_ch4 = resampling_factor * kSrcCh4;
   const float dst_stereo_to_mono = (dst_ch1 + dst_ch2) / 2;
   const float dst_quad_to_mono = (dst_ch1 + dst_ch2 + dst_ch3 + dst_ch4) / 4;
   const float dst_quad_to_stereo_ch1 = (dst_ch1 + dst_ch2) / 2;
   const float dst_quad_to_stereo_ch2 = (dst_ch3 + dst_ch4) / 2;
   if (src_channels == 1)
     SetMonoFrame(kSrcCh1, src_sample_rate_hz, &src_frame_);
   else if (src_channels == 2)
     SetStereoFrame(kSrcCh1, kSrcCh2, src_sample_rate_hz, &src_frame_);
   else
     SetQuadFrame(kSrcCh1, kSrcCh2, kSrcCh3, kSrcCh4, src_sample_rate_hz,
                  &src_frame_);

   if (dst_channels == 1) {
     SetMonoFrame(0, dst_sample_rate_hz, &dst_frame_);
     if (src_channels == 1)
       SetMonoFrame(dst_ch1, dst_sample_rate_hz, &golden_frame_);
     else if (src_channels == 2)
       SetMonoFrame(dst_stereo_to_mono, dst_sample_rate_hz, &golden_frame_);
     else
       SetMonoFrame(dst_quad_to_mono, dst_sample_rate_hz, &golden_frame_);
   } else {
     SetStereoFrame(0, 0, dst_sample_rate_hz, &dst_frame_);
     if (src_channels == 1)
       SetStereoFrame(dst_ch1, dst_ch1, dst_sample_rate_hz, &golden_frame_);
     else if (src_channels == 2)
       SetStereoFrame(dst_ch1, dst_ch2, dst_sample_rate_hz, &golden_frame_);
     else
       SetStereoFrame(dst_quad_to_stereo_ch1, dst_quad_to_stereo_ch2,
                      dst_sample_rate_hz, &golden_frame_);
   }

   // The sinc resampler has a known delay, which we compute here. Multiplying by
   // two gives us a crude maximum for any resampling, as the old resampler
   // typically (but not always) has lower delay.
   static const size_t kInputKernelDelaySamples = 16;
   const size_t max_delay = static_cast<size_t>(
       static_cast<double>(dst_sample_rate_hz) / src_sample_rate_hz *
       kInputKernelDelaySamples * dst_channels * 2);
   printf("(%d, %d Hz) -> (%d, %d Hz) ",  // SNR reported on the same line later.
          src_channels, src_sample_rate_hz, dst_channels, dst_sample_rate_hz);
   RemixAndResample(src_frame_, &resampler, &dst_frame_);

   if (src_sample_rate_hz == 96000 && dst_sample_rate_hz == 8000) {
     // The sinc resampler gives poor SNR at this extreme conversion, but we
     // expect to see this rarely in practice.
     EXPECT_GT(ComputeSNR(golden_frame_, dst_frame_, max_delay), 14.0f);
   } else {
     EXPECT_GT(ComputeSNR(golden_frame_, dst_frame_, max_delay), 46.0f);
   }
 }

 TEST_F(UtilityTest, RemixAndResampleCopyFrameSucceeds) {
   // Stereo -> stereo.
   SetStereoFrame(10, 10, &src_frame_);
   SetStereoFrame(0, 0, &dst_frame_);
   RemixAndResample(src_frame_, &resampler_, &dst_frame_);
   VerifyFramesAreEqual(src_frame_, dst_frame_);

   // Mono -> mono.
   SetMonoFrame(20, &src_frame_);
   SetMonoFrame(0, &dst_frame_);
   RemixAndResample(src_frame_, &resampler_, &dst_frame_);
   VerifyFramesAreEqual(src_frame_, dst_frame_);
 }

 TEST_F(UtilityTest, RemixAndResampleMixingOnlySucceeds) {
   // Stereo -> mono.
   SetStereoFrame(0, 0, &dst_frame_);
   SetMonoFrame(10, &src_frame_);
   SetStereoFrame(10, 10, &golden_frame_);
   RemixAndResample(src_frame_, &resampler_, &dst_frame_);
   VerifyFramesAreEqual(dst_frame_, golden_frame_);

   // Mono -> stereo.
   SetMonoFrame(0, &dst_frame_);
   SetStereoFrame(10, 20, &src_frame_);
   SetMonoFrame(15, &golden_frame_);
   RemixAndResample(src_frame_, &resampler_, &dst_frame_);
   VerifyFramesAreEqual(golden_frame_, dst_frame_);
 }

 TEST_F(UtilityTest, RemixAndResampleSucceeds) {
   const int kSampleRates[] = {8000, 16000, 32000, 44100, 48000, 96000};
   const int kSampleRatesSize = arraysize(kSampleRates);
   const int kSrcChannels[] = {1, 2, 4};
   const int kSrcChannelsSize = arraysize(kSrcChannels);
   const int kDstChannels[] = {1, 2};
   const int kDstChannelsSize = arraysize(kDstChannels);

   for (int src_rate = 0; src_rate < kSampleRatesSize; src_rate++) {
     for (int dst_rate = 0; dst_rate < kSampleRatesSize; dst_rate++) {
       for (int src_channel = 0; src_channel < kSrcChannelsSize; src_channel++) {
         for (int dst_channel = 0; dst_channel < kDstChannelsSize;
              dst_channel++) {
           RunResampleTest(kSrcChannels[src_channel], kSampleRates[src_rate],
                           kDstChannels[dst_channel], kSampleRates[dst_rate]);
         }
       }
     }
   }
 }

 }  // namespace
 }  // namespace voe
 }  // 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 "audio/remix_resample.h"

	#include <cmath>

	#include "common_audio/resampler/include/push_resampler.h"
	#include "rtc_base/arraysize.h"
	#include "rtc_base/checks.h"
	#include "test/gtest.h"

	namespace webrtc {
	namespace voe {
	namespace {

	class UtilityTest : public ::testing::Test {
	protected:
	UtilityTest() {
	src_frame_.sample_rate_hz_ = 16000;
	src_frame_.samples_per_channel_ = src_frame_.sample_rate_hz_ / 100;
	src_frame_.num_channels_ = 1;
	dst_frame_.CopyFrom(src_frame_);
	golden_frame_.CopyFrom(src_frame_);
	}

	void RunResampleTest(int src_channels,
	int src_sample_rate_hz,
	int dst_channels,
	int dst_sample_rate_hz);

	PushResampler<int16_t> resampler_;
	AudioFrame src_frame_;
	AudioFrame dst_frame_;
	AudioFrame golden_frame_;
	};

	// Sets the signal value to increase by `data` with every sample. Floats are
	// used so non-integer values result in rounding error, but not an accumulating
	// error.
	void SetMonoFrame(float data, int sample_rate_hz, AudioFrame* frame) {
	frame->Mute();
	frame->num_channels_ = 1;
	frame->sample_rate_hz_ = sample_rate_hz;
	frame->samples_per_channel_ = rtc::CheckedDivExact(sample_rate_hz, 100);
	int16_t* frame_data = frame->mutable_data();
	for (size_t i = 0; i < frame->samples_per_channel_; i++) {
	frame_data[i] = static_cast<int16_t>(data * i);
	}
	}

	// Keep the existing sample rate.
	void SetMonoFrame(float data, AudioFrame* frame) {
	SetMonoFrame(data, frame->sample_rate_hz_, frame);
	}

	// Sets the signal value to increase by `left` and `right` with every sample in
	// each channel respectively.
	void SetStereoFrame(float left,
	float right,
	int sample_rate_hz,
	AudioFrame* frame) {
	frame->Mute();
	frame->num_channels_ = 2;
	frame->sample_rate_hz_ = sample_rate_hz;
	frame->samples_per_channel_ = rtc::CheckedDivExact(sample_rate_hz, 100);
	int16_t* frame_data = frame->mutable_data();
	for (size_t i = 0; i < frame->samples_per_channel_; i++) {
	frame_data[i * 2] = static_cast<int16_t>(left * i);
	frame_data[i * 2 + 1] = static_cast<int16_t>(right * i);
	}
	}

	// Keep the existing sample rate.
	void SetStereoFrame(float left, float right, AudioFrame* frame) {
	SetStereoFrame(left, right, frame->sample_rate_hz_, frame);
	}

	// Sets the signal value to increase by `ch1`, `ch2`, `ch3`, `ch4` with every
	// sample in each channel respectively.
	void SetQuadFrame(float ch1,
	float ch2,
	float ch3,
	float ch4,
	int sample_rate_hz,
	AudioFrame* frame) {
	frame->Mute();
	frame->num_channels_ = 4;
	frame->sample_rate_hz_ = sample_rate_hz;
	frame->samples_per_channel_ = rtc::CheckedDivExact(sample_rate_hz, 100);
	int16_t* frame_data = frame->mutable_data();
	for (size_t i = 0; i < frame->samples_per_channel_; i++) {
	frame_data[i * 4] = static_cast<int16_t>(ch1 * i);
	frame_data[i * 4 + 1] = static_cast<int16_t>(ch2 * i);
	frame_data[i * 4 + 2] = static_cast<int16_t>(ch3 * i);
	frame_data[i * 4 + 3] = static_cast<int16_t>(ch4 * i);
	}
	}

	void VerifyParams(const AudioFrame& ref_frame, const AudioFrame& test_frame) {
	EXPECT_EQ(ref_frame.num_channels_, test_frame.num_channels_);
	EXPECT_EQ(ref_frame.samples_per_channel_, test_frame.samples_per_channel_);
	EXPECT_EQ(ref_frame.sample_rate_hz_, test_frame.sample_rate_hz_);
	}

	// Computes the best SNR based on the error between `ref_frame` and
	// `test_frame`. It allows for up to a `max_delay` in samples between the
	// signals to compensate for the resampling delay.
	float ComputeSNR(const AudioFrame& ref_frame,
	const AudioFrame& test_frame,
	size_t max_delay) {
	VerifyParams(ref_frame, test_frame);
	float best_snr = 0;
	size_t best_delay = 0;
	for (size_t delay = 0; delay <= max_delay; delay++) {
	float mse = 0;
	float variance = 0;
	const int16_t* ref_frame_data = ref_frame.data();
	const int16_t* test_frame_data = test_frame.data();
	for (size_t i = 0;
	i < ref_frame.samples_per_channel_ * ref_frame.num_channels_ - delay;
	i++) {
	int error = ref_frame_data[i] - test_frame_data[i + delay];
	mse += error * error;
	variance += ref_frame_data[i] * ref_frame_data[i];
	}
	float snr = 100; // We assign 100 dB to the zero-error case.
	if (mse > 0)
	snr = 10 * std::log10(variance / mse);
	if (snr > best_snr) {
	best_snr = snr;
	best_delay = delay;
	}
	}
	printf("SNR=%.1f dB at delay=%zu\n", best_snr, best_delay);
	return best_snr;
	}

	void VerifyFramesAreEqual(const AudioFrame& ref_frame,
	const AudioFrame& test_frame) {
	VerifyParams(ref_frame, test_frame);
	const int16_t* ref_frame_data = ref_frame.data();
	const int16_t* test_frame_data = test_frame.data();
	for (size_t i = 0;
	i < ref_frame.samples_per_channel_ * ref_frame.num_channels_; i++) {
	EXPECT_EQ(ref_frame_data[i], test_frame_data[i]);
	}
	}

	void UtilityTest::RunResampleTest(int src_channels,
	int src_sample_rate_hz,
	int dst_channels,
	int dst_sample_rate_hz) {
	PushResampler<int16_t> resampler; // Create a new one with every test.
	const int16_t kSrcCh1 = 30; // Shouldn't overflow for any used sample rate.
	const int16_t kSrcCh2 = 15;
	const int16_t kSrcCh3 = 22;
	const int16_t kSrcCh4 = 8;
	const float resampling_factor =
	(1.0 * src_sample_rate_hz) / dst_sample_rate_hz;
	const float dst_ch1 = resampling_factor * kSrcCh1;
	const float dst_ch2 = resampling_factor * kSrcCh2;
	const float dst_ch3 = resampling_factor * kSrcCh3;
	const float dst_ch4 = resampling_factor * kSrcCh4;
	const float dst_stereo_to_mono = (dst_ch1 + dst_ch2) / 2;
	const float dst_quad_to_mono = (dst_ch1 + dst_ch2 + dst_ch3 + dst_ch4) / 4;
	const float dst_quad_to_stereo_ch1 = (dst_ch1 + dst_ch2) / 2;
	const float dst_quad_to_stereo_ch2 = (dst_ch3 + dst_ch4) / 2;
	if (src_channels == 1)
	SetMonoFrame(kSrcCh1, src_sample_rate_hz, &src_frame_);
	else if (src_channels == 2)
	SetStereoFrame(kSrcCh1, kSrcCh2, src_sample_rate_hz, &src_frame_);
	else
	SetQuadFrame(kSrcCh1, kSrcCh2, kSrcCh3, kSrcCh4, src_sample_rate_hz,
	&src_frame_);

	if (dst_channels == 1) {
	SetMonoFrame(0, dst_sample_rate_hz, &dst_frame_);
	if (src_channels == 1)
	SetMonoFrame(dst_ch1, dst_sample_rate_hz, &golden_frame_);
	else if (src_channels == 2)
	SetMonoFrame(dst_stereo_to_mono, dst_sample_rate_hz, &golden_frame_);
	else
	SetMonoFrame(dst_quad_to_mono, dst_sample_rate_hz, &golden_frame_);
	} else {
	SetStereoFrame(0, 0, dst_sample_rate_hz, &dst_frame_);
	if (src_channels == 1)
	SetStereoFrame(dst_ch1, dst_ch1, dst_sample_rate_hz, &golden_frame_);
	else if (src_channels == 2)
	SetStereoFrame(dst_ch1, dst_ch2, dst_sample_rate_hz, &golden_frame_);
	else
	SetStereoFrame(dst_quad_to_stereo_ch1, dst_quad_to_stereo_ch2,
	dst_sample_rate_hz, &golden_frame_);
	}

	// The sinc resampler has a known delay, which we compute here. Multiplying by
	// two gives us a crude maximum for any resampling, as the old resampler
	// typically (but not always) has lower delay.
	static const size_t kInputKernelDelaySamples = 16;
	const size_t max_delay = static_cast<size_t>(
	static_cast<double>(dst_sample_rate_hz) / src_sample_rate_hz *
	kInputKernelDelaySamples * dst_channels * 2);
	printf("(%d, %d Hz) -> (%d, %d Hz) ", // SNR reported on the same line later.
	src_channels, src_sample_rate_hz, dst_channels, dst_sample_rate_hz);
	RemixAndResample(src_frame_, &resampler, &dst_frame_);

	if (src_sample_rate_hz == 96000 && dst_sample_rate_hz == 8000) {
	// The sinc resampler gives poor SNR at this extreme conversion, but we
	// expect to see this rarely in practice.
	EXPECT_GT(ComputeSNR(golden_frame_, dst_frame_, max_delay), 14.0f);
	} else {
	EXPECT_GT(ComputeSNR(golden_frame_, dst_frame_, max_delay), 46.0f);
	}
	}

	TEST_F(UtilityTest, RemixAndResampleCopyFrameSucceeds) {
	// Stereo -> stereo.
	SetStereoFrame(10, 10, &src_frame_);
	SetStereoFrame(0, 0, &dst_frame_);
	RemixAndResample(src_frame_, &resampler_, &dst_frame_);
	VerifyFramesAreEqual(src_frame_, dst_frame_);

	// Mono -> mono.
	SetMonoFrame(20, &src_frame_);
	SetMonoFrame(0, &dst_frame_);
	RemixAndResample(src_frame_, &resampler_, &dst_frame_);
	VerifyFramesAreEqual(src_frame_, dst_frame_);
	}

	TEST_F(UtilityTest, RemixAndResampleMixingOnlySucceeds) {
	// Stereo -> mono.
	SetStereoFrame(0, 0, &dst_frame_);
	SetMonoFrame(10, &src_frame_);
	SetStereoFrame(10, 10, &golden_frame_);
	RemixAndResample(src_frame_, &resampler_, &dst_frame_);
	VerifyFramesAreEqual(dst_frame_, golden_frame_);

	// Mono -> stereo.
	SetMonoFrame(0, &dst_frame_);
	SetStereoFrame(10, 20, &src_frame_);
	SetMonoFrame(15, &golden_frame_);
	RemixAndResample(src_frame_, &resampler_, &dst_frame_);
	VerifyFramesAreEqual(golden_frame_, dst_frame_);
	}

	TEST_F(UtilityTest, RemixAndResampleSucceeds) {
	const int kSampleRates[] = {8000, 16000, 32000, 44100, 48000, 96000};
	const int kSampleRatesSize = arraysize(kSampleRates);
	const int kSrcChannels[] = {1, 2, 4};
	const int kSrcChannelsSize = arraysize(kSrcChannels);
	const int kDstChannels[] = {1, 2};
	const int kDstChannelsSize = arraysize(kDstChannels);

	for (int src_rate = 0; src_rate < kSampleRatesSize; src_rate++) {
	for (int dst_rate = 0; dst_rate < kSampleRatesSize; dst_rate++) {
	for (int src_channel = 0; src_channel < kSrcChannelsSize; src_channel++) {
	for (int dst_channel = 0; dst_channel < kDstChannelsSize;
	dst_channel++) {
	RunResampleTest(kSrcChannels[src_channel], kSampleRates[src_rate],
	kDstChannels[dst_channel], kSampleRates[dst_rate]);
	}
	}
	}
	}
	}

	} // namespace
	} // namespace voe
	} // namespace webrtc