third_party/opus/tests/opus_benchmark.cc - cobalt - Git at Google

 // Copyright 2019 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.

 #include <cmath>
 #include <cstdio>
 #include <cstdlib>
 #include <ctime>
 #include <limits>

 #include "base/files/file_path.h"
 #include "base/path_service.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "third_party/opus/src/include/opus.h"

 // Set to 1 to generate golden values instead of checking against them.
 // Used to ease updating when new references are better or acceptable.
 #define REGENERATE_REFERENCES 0

 namespace {

 constexpr int RETRIES = 3;
 constexpr int MAX_FRAME_SAMP = 5760;  // 120ms * 48kHz

 base::FilePath GetDataPath(const std::string& filename) {
   base::FilePath path;
   DCHECK(base::PathService::Get(base::DIR_SOURCE_ROOT, &path));
   return path.AppendASCII("third_party")
       .AppendASCII("opus")
       .AppendASCII("tests")
       .AppendASCII("resources")
       .AppendASCII(filename);
 }

 void opus_run(int frame_size_ms,
               int sampling_khz,
               int channels,
               int bit_rate,
               int complexity,
               size_t audio_duration_sec,
               float* encoding_rtf,
               float* decoding_rtf) {
   const int frame_size = frame_size_ms * sampling_khz;
   const int application =
       channels == 1 ? OPUS_APPLICATION_VOIP : OPUS_APPLICATION_AUDIO;

   /* Read PCM */
   auto in_filename = GetDataPath("speech_mono_32_48kHz.pcm").value();
   FILE* fp = fopen(in_filename.c_str(), "rb");

   // TODO(b/1002973): Review error handling. ASSERT_* just return from function,
   //                  meaning calling code is executed.
   // We might want to factorize encoder creation anyway (read pcm once instead
   //                            of re-opening it for each bitrate / complexity).
   ASSERT_NE(fp, nullptr) << "Could not open: " << in_filename;

   fseek(fp, 0, SEEK_END);
   const size_t total_samp = ftell(fp) / sizeof(int16_t);
   rewind(fp);

   int16_t* in_data =
       (int16_t*)malloc((total_samp + frame_size * channels) * sizeof(*in_data));
   const size_t check_total_samp =
       fread(&in_data[0], sizeof(int16_t), total_samp, fp);
   ASSERT_EQ(check_total_samp, total_samp) << "Error reading input pcm file.";
   fclose(fp);

   const opus_int32 max_bytes = frame_size * channels * sizeof(int16_t);
   int16_t* out_data =
       (int16_t*)malloc((frame_size * channels) * sizeof(int16_t));
   uint8_t* bit_stream = (uint8_t*)malloc(max_bytes * sizeof(*bit_stream));

   /* Create an encoder */
   opus_int32 res;
   OpusEncoder* enc =
       opus_encoder_create(sampling_khz * 1000, channels, application, &res);
   ASSERT_TRUE(res == OPUS_OK && enc != nullptr)
       << "Could not instantiate an Opus encoder. Error code: " << res;

   opus_encoder_ctl(enc, OPUS_SET_BITRATE(bit_rate));
   opus_encoder_ctl(enc, OPUS_SET_COMPLEXITY(complexity));

   /* Create a decoder */
   OpusDecoder* dec = opus_decoder_create(sampling_khz * 1000, channels, &res);
   ASSERT_TRUE(res == OPUS_OK && dec != nullptr)
       << "Could not instantiate an Opus decoder. Error code: " << res;

   /* Transcode up to |audio_duration_sec| only */
   size_t data_pointer = 0;
   size_t decoded_ms = 0;
   float encoding_sec = 0;
   float decoding_sec = 0;
   while (decoded_ms < audio_duration_sec * 1000) {
     /* Encode */
     clock_t clocks = clock();
     res = opus_encode(enc, &in_data[data_pointer], frame_size, &bit_stream[0],
                       max_bytes);
     clocks = clock() - clocks;
     EXPECT_GT(res, 0) << "No bytes were encoded successfully.";

     encoding_sec += 1. * clocks / CLOCKS_PER_SEC;

     /* Decode */
     clocks = clock();
     res =
         opus_decode(dec, &bit_stream[0], res, &out_data[0], MAX_FRAME_SAMP, 0);
     clocks = clock() - clocks;
     EXPECT_EQ(res, frame_size) << "Wrong number of samples returned by decoder";

     decoding_sec += 1. * clocks / CLOCKS_PER_SEC;

     /* Update data pointer and time tracker */
     data_pointer += frame_size * channels;
     decoded_ms += frame_size_ms;
   }

   *encoding_rtf = encoding_sec / audio_duration_sec;
   *decoding_rtf = decoding_sec / audio_duration_sec;

   /* Clean up */
   opus_encoder_destroy(enc);
   opus_decoder_destroy(dec);
   free(in_data);
   free(out_data);
   free(bit_stream);
 }
 }  // namespace

 TEST(OpusBenchmark, SpeechMono48kHzNexus5) {
   const int sampling_khz = 48;
   const int channels = 1;
   size_t audio_duration_sec = 240;
   const int frame_size_ms = 20;
   constexpr int nr_bit_rates = 8;
   int32_t bit_rates[nr_bit_rates] = {64000, 32000, 24000, 16000,
                                      12000, 10000, 8000,  6000};
   constexpr int nr_complexities = 11;
   int32_t complexities[nr_complexities] = {0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10};

 #if REGENERATE_REFERENCES
   fprintf(stderr, "bitrate,complexity,encoding_rtf,decoding_rtf\n");
 #else
   // Real-time factors, scaled *1000 for cosmetic purpose.
   // Lower is best, any value <= 1000 is real-time.

   // This reference numbers were generated on a NEXUS 5.
   // This means they are irrelevant if you run the test on a linux workstation.
   // TODO(b/1002973): Have a set of references by architecture?
   //              Or at least, disable the tests if not run on expected target.
   float encoding_reference[nr_bit_rates][nr_complexities] = {
       // NB: RTF is dropping for complexities 9 and 10 at bit_rate 64000.
       // On linux x64, it drops for complexities 7, 8, 9, 10.
       // That seems ok, sanitizers don't report anything.
       // It looks to be operating more often in CELT-only mode compared to
       // Hybrid mode for this particular audio file (possibly because of
       // the presence of the background music).
       {18.0, 20.2, 22.8, 24.1, 29.3, 29.2, 31.5, 31.5, 35.6, 15.5, 21.6},
       {16.9, 19.2, 21.7, 23.0, 28.1, 28.1, 30.1, 30.1, 34.2, 34.2, 32.6},
       {16.6, 18.8, 21.3, 22.6, 27.8, 27.8, 29.7, 29.7, 33.7, 33.7, 39.9},
       {12.7, 13.6, 20.9, 22.2, 27.4, 27.3, 29.2, 29.2, 33.1, 33.0, 39.1},
       {8.4, 9.1, 16.0, 17.3, 22.4, 22.4, 24.2, 24.3, 27.7, 27.7, 33.8},
       {8.4, 9.1, 10.3, 11.2, 13.8, 13.8, 14.7, 14.7, 16.6, 16.6, 26.6},
       {8.4, 9.0, 10.3, 11.1, 13.7, 13.7, 14.6, 14.6, 16.5, 16.5, 22.6},
       {8.3, 9.0, 10.2, 11.1, 13.7, 13.7, 14.6, 14.6, 16.4, 16.4, 22.5}};

   float decoding_reference[nr_bit_rates][nr_complexities] = {
       {7.1, 7.5, 7.5, 7.5, 7.5, 7.5, 7.5, 7.5, 7.5, 5.8, 5.8},
       {6.5, 6.8, 6.8, 6.8, 6.9, 6.9, 6.9, 6.9, 6.9, 6.9, 6.2},
       {6.3, 6.6, 6.6, 6.6, 6.7, 6.7, 6.7, 6.7, 6.7, 6.7, 6.7},
       {3.2, 3.2, 6.4, 6.4, 6.5, 6.4, 6.4, 6.5, 6.5, 6.4, 6.5},
       {2.2, 2.2, 3.1, 3.1, 3.2, 3.2, 3.2, 3.2, 3.2, 3.2, 3.2},
       {2.1, 2.1, 2.1, 2.1, 2.2, 2.2, 2.2, 2.2, 2.2, 1.9, 2.5},
       {2.1, 2.1, 2.1, 2.1, 2.1, 2.1, 2.1, 2.1, 2.1, 2.1, 2.1},
       {2.1, 2.1, 2.1, 2.1, 2.1, 2.1, 2.1, 2.1, 2.1, 2.1, 2.1}};
 #endif

   for (int i = 0; i < nr_bit_rates; i++) {
     for (int j = 0; j < nr_complexities; j++) {
       // To reduce brittleness, each test is retried independently.
 #if REGENERATE_REFERENCES
       float encoding_best = std::numeric_limits<float>::infinity();
       float decoding_best = std::numeric_limits<float>::infinity();
 #endif
       for (int k = 0; k < RETRIES; k++) {
         float encoding_rtf;
         float decoding_rtf;
         opus_run(frame_size_ms, sampling_khz, channels, bit_rates[i],
                  complexities[j], audio_duration_sec, &encoding_rtf,
                  &decoding_rtf);
         // Scale to match reference unit.
         encoding_rtf *= 1000;
         decoding_rtf *= 1000;

 #if REGENERATE_REFERENCES
         encoding_best = std::min(encoding_best, encoding_rtf);
         decoding_best = std::min(decoding_best, decoding_rtf);
 #else
         // Allow 5% relative error.
         auto encoding_ref = encoding_reference[i][j];
         auto decoding_ref = decoding_reference[i][j];
         const float encoding_abs_error = .05 * encoding_ref;
         const float decoding_abs_error = .05 * decoding_ref;
         if (std::abs(encoding_rtf - encoding_ref) < encoding_abs_error) {
           SUCCEED();
           break;  // No need for retry.
         } else if (k == RETRIES - 1) {
           // Nb: If returned value is consistently lower (better),
           //     one just have to update the reference.
           EXPECT_NEAR(encoding_rtf, encoding_ref, encoding_abs_error)
               << "Mismatch for bitrate " << bit_rates[i] << " and complexity "
               << complexities[j];
           EXPECT_NEAR(decoding_rtf, decoding_ref, decoding_abs_error)
               << "Mismatch for bitrate " << bit_rates[i] << " and complexity "
               << complexities[j];
         }
 #endif
       }
 #if REGENERATE_REFERENCES
       fprintf(stderr, "%d, %d, %.2f, %.2f\n", bit_rates[i], complexities[j],
               encoding_best, decoding_best);
 #endif
     }
   }
 }
	// Copyright 2019 The Chromium Authors. All rights reserved.
	// Use of this source code is governed by a BSD-style license that can be
	// found in the LICENSE file.

	#include <cmath>
	#include <cstdio>
	#include <cstdlib>
	#include <ctime>
	#include <limits>

	#include "base/files/file_path.h"
	#include "base/path_service.h"
	#include "testing/gtest/include/gtest/gtest.h"
	#include "third_party/opus/src/include/opus.h"

	// Set to 1 to generate golden values instead of checking against them.
	// Used to ease updating when new references are better or acceptable.
	#define REGENERATE_REFERENCES 0

	namespace {

	constexpr int RETRIES = 3;
	constexpr int MAX_FRAME_SAMP = 5760; // 120ms * 48kHz

	base::FilePath GetDataPath(const std::string& filename) {
	base::FilePath path;
	DCHECK(base::PathService::Get(base::DIR_SOURCE_ROOT, &path));
	return path.AppendASCII("third_party")
	.AppendASCII("opus")
	.AppendASCII("tests")
	.AppendASCII("resources")
	.AppendASCII(filename);
	}

	void opus_run(int frame_size_ms,
	int sampling_khz,
	int channels,
	int bit_rate,
	int complexity,
	size_t audio_duration_sec,
	float* encoding_rtf,
	float* decoding_rtf) {
	const int frame_size = frame_size_ms * sampling_khz;
	const int application =
	channels == 1 ? OPUS_APPLICATION_VOIP : OPUS_APPLICATION_AUDIO;

	/* Read PCM */
	auto in_filename = GetDataPath("speech_mono_32_48kHz.pcm").value();
	FILE* fp = fopen(in_filename.c_str(), "rb");

	// TODO(b/1002973): Review error handling. ASSERT_* just return from function,
	// meaning calling code is executed.
	// We might want to factorize encoder creation anyway (read pcm once instead
	// of re-opening it for each bitrate / complexity).
	ASSERT_NE(fp, nullptr) << "Could not open: " << in_filename;

	fseek(fp, 0, SEEK_END);
	const size_t total_samp = ftell(fp) / sizeof(int16_t);
	rewind(fp);

	int16_t* in_data =
	(int16_t)malloc((total_samp + frame_size channels) * sizeof(*in_data));
	const size_t check_total_samp =
	fread(&in_data[0], sizeof(int16_t), total_samp, fp);
	ASSERT_EQ(check_total_samp, total_samp) << "Error reading input pcm file.";
	fclose(fp);

	const opus_int32 max_bytes = frame_size * channels * sizeof(int16_t);
	int16_t* out_data =
	(int16_t)malloc((frame_size channels) * sizeof(int16_t));
	uint8_t* bit_stream = (uint8_t)malloc(max_bytes sizeof(*bit_stream));

	/* Create an encoder */
	opus_int32 res;
	OpusEncoder* enc =
	opus_encoder_create(sampling_khz * 1000, channels, application, &res);
	ASSERT_TRUE(res == OPUS_OK && enc != nullptr)
	<< "Could not instantiate an Opus encoder. Error code: " << res;

	opus_encoder_ctl(enc, OPUS_SET_BITRATE(bit_rate));
	opus_encoder_ctl(enc, OPUS_SET_COMPLEXITY(complexity));

	/* Create a decoder */
	OpusDecoder* dec = opus_decoder_create(sampling_khz * 1000, channels, &res);
	ASSERT_TRUE(res == OPUS_OK && dec != nullptr)
	<< "Could not instantiate an Opus decoder. Error code: " << res;

	/* Transcode up to \|audio_duration_sec\| only */
	size_t data_pointer = 0;
	size_t decoded_ms = 0;
	float encoding_sec = 0;
	float decoding_sec = 0;
	while (decoded_ms < audio_duration_sec * 1000) {
	/* Encode */
	clock_t clocks = clock();
	res = opus_encode(enc, &in_data[data_pointer], frame_size, &bit_stream[0],
	max_bytes);
	clocks = clock() - clocks;
	EXPECT_GT(res, 0) << "No bytes were encoded successfully.";

	encoding_sec += 1. * clocks / CLOCKS_PER_SEC;

	/* Decode */
	clocks = clock();
	res =
	opus_decode(dec, &bit_stream[0], res, &out_data[0], MAX_FRAME_SAMP, 0);
	clocks = clock() - clocks;
	EXPECT_EQ(res, frame_size) << "Wrong number of samples returned by decoder";

	decoding_sec += 1. * clocks / CLOCKS_PER_SEC;

	/* Update data pointer and time tracker */
	data_pointer += frame_size * channels;
	decoded_ms += frame_size_ms;
	}

	*encoding_rtf = encoding_sec / audio_duration_sec;
	*decoding_rtf = decoding_sec / audio_duration_sec;

	/* Clean up */
	opus_encoder_destroy(enc);
	opus_decoder_destroy(dec);
	free(in_data);
	free(out_data);
	free(bit_stream);
	}
	} // namespace

	TEST(OpusBenchmark, SpeechMono48kHzNexus5) {
	const int sampling_khz = 48;
	const int channels = 1;
	size_t audio_duration_sec = 240;
	const int frame_size_ms = 20;
	constexpr int nr_bit_rates = 8;
	int32_t bit_rates[nr_bit_rates] = {64000, 32000, 24000, 16000,
	12000, 10000, 8000, 6000};
	constexpr int nr_complexities = 11;
	int32_t complexities[nr_complexities] = {0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10};

	#if REGENERATE_REFERENCES
	fprintf(stderr, "bitrate,complexity,encoding_rtf,decoding_rtf\n");
	#else
	// Real-time factors, scaled *1000 for cosmetic purpose.
	// Lower is best, any value <= 1000 is real-time.

	// This reference numbers were generated on a NEXUS 5.
	// This means they are irrelevant if you run the test on a linux workstation.
	// TODO(b/1002973): Have a set of references by architecture?
	// Or at least, disable the tests if not run on expected target.
	float encoding_reference[nr_bit_rates][nr_complexities] = {
	// NB: RTF is dropping for complexities 9 and 10 at bit_rate 64000.
	// On linux x64, it drops for complexities 7, 8, 9, 10.
	// That seems ok, sanitizers don't report anything.
	// It looks to be operating more often in CELT-only mode compared to
	// Hybrid mode for this particular audio file (possibly because of
	// the presence of the background music).
	{18.0, 20.2, 22.8, 24.1, 29.3, 29.2, 31.5, 31.5, 35.6, 15.5, 21.6},
	{16.9, 19.2, 21.7, 23.0, 28.1, 28.1, 30.1, 30.1, 34.2, 34.2, 32.6},
	{16.6, 18.8, 21.3, 22.6, 27.8, 27.8, 29.7, 29.7, 33.7, 33.7, 39.9},
	{12.7, 13.6, 20.9, 22.2, 27.4, 27.3, 29.2, 29.2, 33.1, 33.0, 39.1},
	{8.4, 9.1, 16.0, 17.3, 22.4, 22.4, 24.2, 24.3, 27.7, 27.7, 33.8},
	{8.4, 9.1, 10.3, 11.2, 13.8, 13.8, 14.7, 14.7, 16.6, 16.6, 26.6},
	{8.4, 9.0, 10.3, 11.1, 13.7, 13.7, 14.6, 14.6, 16.5, 16.5, 22.6},
	{8.3, 9.0, 10.2, 11.1, 13.7, 13.7, 14.6, 14.6, 16.4, 16.4, 22.5}};

	float decoding_reference[nr_bit_rates][nr_complexities] = {
	{7.1, 7.5, 7.5, 7.5, 7.5, 7.5, 7.5, 7.5, 7.5, 5.8, 5.8},
	{6.5, 6.8, 6.8, 6.8, 6.9, 6.9, 6.9, 6.9, 6.9, 6.9, 6.2},
	{6.3, 6.6, 6.6, 6.6, 6.7, 6.7, 6.7, 6.7, 6.7, 6.7, 6.7},
	{3.2, 3.2, 6.4, 6.4, 6.5, 6.4, 6.4, 6.5, 6.5, 6.4, 6.5},
	{2.2, 2.2, 3.1, 3.1, 3.2, 3.2, 3.2, 3.2, 3.2, 3.2, 3.2},
	{2.1, 2.1, 2.1, 2.1, 2.2, 2.2, 2.2, 2.2, 2.2, 1.9, 2.5},
	{2.1, 2.1, 2.1, 2.1, 2.1, 2.1, 2.1, 2.1, 2.1, 2.1, 2.1},
	{2.1, 2.1, 2.1, 2.1, 2.1, 2.1, 2.1, 2.1, 2.1, 2.1, 2.1}};
	#endif

	for (int i = 0; i < nr_bit_rates; i++) {
	for (int j = 0; j < nr_complexities; j++) {
	// To reduce brittleness, each test is retried independently.
	#if REGENERATE_REFERENCES
	float encoding_best = std::numeric_limits<float>::infinity();
	float decoding_best = std::numeric_limits<float>::infinity();
	#endif
	for (int k = 0; k < RETRIES; k++) {
	float encoding_rtf;
	float decoding_rtf;
	opus_run(frame_size_ms, sampling_khz, channels, bit_rates[i],
	complexities[j], audio_duration_sec, &encoding_rtf,
	&decoding_rtf);
	// Scale to match reference unit.
	encoding_rtf *= 1000;
	decoding_rtf *= 1000;

	#if REGENERATE_REFERENCES
	encoding_best = std::min(encoding_best, encoding_rtf);
	decoding_best = std::min(decoding_best, decoding_rtf);
	#else
	// Allow 5% relative error.
	auto encoding_ref = encoding_reference[i][j];
	auto decoding_ref = decoding_reference[i][j];
	const float encoding_abs_error = .05 * encoding_ref;
	const float decoding_abs_error = .05 * decoding_ref;
	if (std::abs(encoding_rtf - encoding_ref) < encoding_abs_error) {
	SUCCEED();
	break; // No need for retry.
	} else if (k == RETRIES - 1) {
	// Nb: If returned value is consistently lower (better),
	// one just have to update the reference.
	EXPECT_NEAR(encoding_rtf, encoding_ref, encoding_abs_error)
	<< "Mismatch for bitrate " << bit_rates[i] << " and complexity "
	<< complexities[j];
	EXPECT_NEAR(decoding_rtf, decoding_ref, decoding_abs_error)
	<< "Mismatch for bitrate " << bit_rates[i] << " and complexity "
	<< complexities[j];
	}
	#endif
	}
	#if REGENERATE_REFERENCES
	fprintf(stderr, "%d, %d, %.2f, %.2f\n", bit_rates[i], complexities[j],
	encoding_best, decoding_best);
	#endif
	}
	}
	}