voice_engine/output_mixer_unittest.cc - src/webrtc - 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 <math.h>

 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/voice_engine/output_mixer.h"
 #include "webrtc/voice_engine/output_mixer_internal.h"

 namespace webrtc {
 namespace voe {
 namespace {

 class OutputMixerTest : public ::testing::Test {
  protected:
   OutputMixerTest() {
     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 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(AudioFrame* frame, float data, int sample_rate_hz) {
   memset(frame->data_, 0, sizeof(frame->data_));
   frame->num_channels_ = 1;
   frame->sample_rate_hz_ = sample_rate_hz;
   frame->samples_per_channel_ = sample_rate_hz / 100;
   for (int i = 0; i < frame->samples_per_channel_; i++) {
     frame->data_[i] = data * i;
   }
 }

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

 // Sets the signal value to increase by |left| and |right| with every sample in
 // each channel respectively.
 void SetStereoFrame(AudioFrame* frame, float left, float right,
                     int sample_rate_hz) {
   memset(frame->data_, 0, sizeof(frame->data_));
   frame->num_channels_ = 2;
   frame->sample_rate_hz_ = sample_rate_hz;
   frame->samples_per_channel_ = sample_rate_hz / 100;
   for (int i = 0; i < frame->samples_per_channel_; i++) {
     frame->data_[i * 2] = left * i;
     frame->data_[i * 2 + 1] = right * i;
   }
 }

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

 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,
                  int max_delay) {
   VerifyParams(ref_frame, test_frame);
   float best_snr = 0;
   int best_delay = 0;
   for (int delay = 0; delay <= max_delay; delay++) {
     float mse = 0;
     float variance = 0;
     for (int 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 * log10(variance / mse);
     if (snr > best_snr) {
       best_snr = snr;
       best_delay = delay;
     }
   }
   printf("SNR=%.1f dB at delay=%d\n", best_snr, best_delay);
   return best_snr;
 }

 void VerifyFramesAreEqual(const AudioFrame& ref_frame,
                           const AudioFrame& test_frame) {
   VerifyParams(ref_frame, test_frame);
   for (int i = 0; i < ref_frame.samples_per_channel_ * ref_frame.num_channels_;
       i++) {
     EXPECT_EQ(ref_frame.data_[i], test_frame.data_[i]);
   }
 }

 void OutputMixerTest::RunResampleTest(int src_channels,
                                       int src_sample_rate_hz,
                                       int dst_channels,
                                       int dst_sample_rate_hz) {
   PushResampler resampler;  // Create a new one with every test.
   const int16_t kSrcLeft = 30;  // Shouldn't overflow for any used sample rate.
   const int16_t kSrcRight = 15;
   const float resampling_factor = (1.0 * src_sample_rate_hz) /
       dst_sample_rate_hz;
   const float dst_left = resampling_factor * kSrcLeft;
   const float dst_right = resampling_factor * kSrcRight;
   const float dst_mono = (dst_left + dst_right) / 2;
   if (src_channels == 1)
     SetMonoFrame(&src_frame_, kSrcLeft, src_sample_rate_hz);
   else
     SetStereoFrame(&src_frame_, kSrcLeft, kSrcRight, src_sample_rate_hz);

   if (dst_channels == 1) {
     SetMonoFrame(&dst_frame_, 0, dst_sample_rate_hz);
     if (src_channels == 1)
       SetMonoFrame(&golden_frame_, dst_left, dst_sample_rate_hz);
     else
       SetMonoFrame(&golden_frame_, dst_mono, dst_sample_rate_hz);
   } else {
     SetStereoFrame(&dst_frame_, 0, 0, dst_sample_rate_hz);
     if (src_channels == 1)
       SetStereoFrame(&golden_frame_, dst_left, dst_left, dst_sample_rate_hz);
     else
       SetStereoFrame(&golden_frame_, dst_left, dst_right, dst_sample_rate_hz);
   }

   // 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 int kInputKernelDelaySamples = 16;
   const int max_delay = 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);
   EXPECT_EQ(0, 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(OutputMixerTest, RemixAndResampleCopyFrameSucceeds) {
   // Stereo -> stereo.
   SetStereoFrame(&src_frame_, 10, 10);
   SetStereoFrame(&dst_frame_, 0, 0);
   EXPECT_EQ(0, RemixAndResample(src_frame_, &resampler_, &dst_frame_));
   VerifyFramesAreEqual(src_frame_, dst_frame_);

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

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

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

 TEST_F(OutputMixerTest, RemixAndResampleSucceeds) {
   // TODO(ajm): convert this to the parameterized TEST_P style used in
   // sinc_resampler_unittest.cc. We can then easily add tighter SNR thresholds.
   const int kSampleRates[] = {8000, 16000, 32000, 44100, 48000, 96000};
   const int kSampleRatesSize = sizeof(kSampleRates) / sizeof(*kSampleRates);
   const int kChannels[] = {1, 2};
   const int kChannelsSize = sizeof(kChannels) / sizeof(*kChannels);
   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 < kChannelsSize; src_channel++) {
         for (int dst_channel = 0; dst_channel < kChannelsSize; dst_channel++) {
           RunResampleTest(kChannels[src_channel], kSampleRates[src_rate],
                           kChannels[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 <math.h>

	#include "testing/gtest/include/gtest/gtest.h"
	#include "webrtc/voice_engine/output_mixer.h"
	#include "webrtc/voice_engine/output_mixer_internal.h"

	namespace webrtc {
	namespace voe {
	namespace {

	class OutputMixerTest : public ::testing::Test {
	protected:
	OutputMixerTest() {
	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 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(AudioFrame* frame, float data, int sample_rate_hz) {
	memset(frame->data_, 0, sizeof(frame->data_));
	frame->num_channels_ = 1;
	frame->sample_rate_hz_ = sample_rate_hz;
	frame->samples_per_channel_ = sample_rate_hz / 100;
	for (int i = 0; i < frame->samples_per_channel_; i++) {
	frame->data_[i] = data * i;
	}
	}

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

	// Sets the signal value to increase by \|left\| and \|right\| with every sample in
	// each channel respectively.
	void SetStereoFrame(AudioFrame* frame, float left, float right,
	int sample_rate_hz) {
	memset(frame->data_, 0, sizeof(frame->data_));
	frame->num_channels_ = 2;
	frame->sample_rate_hz_ = sample_rate_hz;
	frame->samples_per_channel_ = sample_rate_hz / 100;
	for (int i = 0; i < frame->samples_per_channel_; i++) {
	frame->data_[i * 2] = left * i;
	frame->data_[i * 2 + 1] = right * i;
	}
	}

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

	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,
	int max_delay) {
	VerifyParams(ref_frame, test_frame);
	float best_snr = 0;
	int best_delay = 0;
	for (int delay = 0; delay <= max_delay; delay++) {
	float mse = 0;
	float variance = 0;
	for (int 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 * log10(variance / mse);
	if (snr > best_snr) {
	best_snr = snr;
	best_delay = delay;
	}
	}
	printf("SNR=%.1f dB at delay=%d\n", best_snr, best_delay);
	return best_snr;
	}

	void VerifyFramesAreEqual(const AudioFrame& ref_frame,
	const AudioFrame& test_frame) {
	VerifyParams(ref_frame, test_frame);
	for (int i = 0; i < ref_frame.samples_per_channel_ * ref_frame.num_channels_;
	i++) {
	EXPECT_EQ(ref_frame.data_[i], test_frame.data_[i]);
	}
	}

	void OutputMixerTest::RunResampleTest(int src_channels,
	int src_sample_rate_hz,
	int dst_channels,
	int dst_sample_rate_hz) {
	PushResampler resampler; // Create a new one with every test.
	const int16_t kSrcLeft = 30; // Shouldn't overflow for any used sample rate.
	const int16_t kSrcRight = 15;
	const float resampling_factor = (1.0 * src_sample_rate_hz) /
	dst_sample_rate_hz;
	const float dst_left = resampling_factor * kSrcLeft;
	const float dst_right = resampling_factor * kSrcRight;
	const float dst_mono = (dst_left + dst_right) / 2;
	if (src_channels == 1)
	SetMonoFrame(&src_frame_, kSrcLeft, src_sample_rate_hz);
	else
	SetStereoFrame(&src_frame_, kSrcLeft, kSrcRight, src_sample_rate_hz);

	if (dst_channels == 1) {
	SetMonoFrame(&dst_frame_, 0, dst_sample_rate_hz);
	if (src_channels == 1)
	SetMonoFrame(&golden_frame_, dst_left, dst_sample_rate_hz);
	else
	SetMonoFrame(&golden_frame_, dst_mono, dst_sample_rate_hz);
	} else {
	SetStereoFrame(&dst_frame_, 0, 0, dst_sample_rate_hz);
	if (src_channels == 1)
	SetStereoFrame(&golden_frame_, dst_left, dst_left, dst_sample_rate_hz);
	else
	SetStereoFrame(&golden_frame_, dst_left, dst_right, dst_sample_rate_hz);
	}

	// 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 int kInputKernelDelaySamples = 16;
	const int max_delay = 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);
	EXPECT_EQ(0, 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(OutputMixerTest, RemixAndResampleCopyFrameSucceeds) {
	// Stereo -> stereo.
	SetStereoFrame(&src_frame_, 10, 10);
	SetStereoFrame(&dst_frame_, 0, 0);
	EXPECT_EQ(0, RemixAndResample(src_frame_, &resampler_, &dst_frame_));
	VerifyFramesAreEqual(src_frame_, dst_frame_);

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

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

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

	TEST_F(OutputMixerTest, RemixAndResampleSucceeds) {
	// TODO(ajm): convert this to the parameterized TEST_P style used in
	// sinc_resampler_unittest.cc. We can then easily add tighter SNR thresholds.
	const int kSampleRates[] = {8000, 16000, 32000, 44100, 48000, 96000};
	const int kSampleRatesSize = sizeof(kSampleRates) / sizeof(*kSampleRates);
	const int kChannels[] = {1, 2};
	const int kChannelsSize = sizeof(kChannels) / sizeof(*kChannels);
	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 < kChannelsSize; src_channel++) {
	for (int dst_channel = 0; dst_channel < kChannelsSize; dst_channel++) {
	RunResampleTest(kChannels[src_channel], kSampleRates[src_rate],
	kChannels[dst_channel], kSampleRates[dst_rate]);
	}
	}
	}
	}
	}

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