modules/audio_coding/codecs/isac/unittest.cc - src - Git at Google

 /*
  *  Copyright (c) 2015 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 <algorithm>
 #include <numeric>
 #include <vector>

 #include "modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h"
 #include "modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
 #include "modules/audio_coding/neteq/tools/input_audio_file.h"
 #include "rtc_base/buffer.h"
 #include "rtc_base/numerics/safe_conversions.h"
 #include "rtc_base/strings/string_builder.h"
 #include "test/gtest.h"
 #include "test/testsupport/file_utils.h"

 namespace webrtc {

 namespace {

 const int kIsacNumberOfSamples = 32 * 60;  // 60 ms at 32 kHz

 std::vector<int16_t> LoadSpeechData() {
   webrtc::test::InputAudioFile input_file(
       webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"));
   std::vector<int16_t> speech_data(kIsacNumberOfSamples);
   input_file.Read(kIsacNumberOfSamples, speech_data.data());
   return speech_data;
 }

 template <typename T>
 IsacBandwidthInfo GetBwInfo(typename T::instance_type* inst) {
   IsacBandwidthInfo bi;
   T::GetBandwidthInfo(inst, &bi);
   EXPECT_TRUE(bi.in_use);
   return bi;
 }

 // Encodes one packet. Returns the packet duration in milliseconds.
 template <typename T>
 int EncodePacket(typename T::instance_type* inst,
                  const IsacBandwidthInfo* bi,
                  const int16_t* speech_data,
                  rtc::Buffer* output) {
   output->SetSize(1000);
   for (int duration_ms = 10;; duration_ms += 10) {
     if (bi)
       T::SetBandwidthInfo(inst, bi);
     int encoded_bytes = T::Encode(inst, speech_data, output->data());
     if (encoded_bytes > 0 || duration_ms >= 60) {
       EXPECT_GT(encoded_bytes, 0);
       EXPECT_LE(static_cast<size_t>(encoded_bytes), output->size());
       output->SetSize(encoded_bytes);
       return duration_ms;
     }
   }
 }

 template <typename T>
 std::vector<int16_t> DecodePacket(typename T::instance_type* inst,
                                   const rtc::Buffer& encoded) {
   std::vector<int16_t> decoded(kIsacNumberOfSamples);
   int16_t speech_type;
   int nsamples = T::DecodeInternal(inst, encoded.data(), encoded.size(),
                                    &decoded.front(), &speech_type);
   EXPECT_GT(nsamples, 0);
   EXPECT_LE(static_cast<size_t>(nsamples), decoded.size());
   decoded.resize(nsamples);
   return decoded;
 }

 class BoundedCapacityChannel final {
  public:
   BoundedCapacityChannel(int sample_rate_hz, int rate_bits_per_second)
       : current_time_rtp_(0),
         channel_rate_bytes_per_sample_(rate_bits_per_second /
                                        (8.0 * sample_rate_hz)) {}

   // Simulate sending the given number of bytes at the given RTP time. Returns
   // the new current RTP time after the sending is done.
   int Send(int send_time_rtp, int nbytes) {
     current_time_rtp_ = std::max(current_time_rtp_, send_time_rtp) +
                         nbytes / channel_rate_bytes_per_sample_;
     return current_time_rtp_;
   }

  private:
   int current_time_rtp_;
   // The somewhat strange unit for channel rate, bytes per sample, is because
   // RTP time is measured in samples:
   const double channel_rate_bytes_per_sample_;
 };

 // Test that the iSAC encoder produces identical output whether or not we use a
 // conjoined encoder+decoder pair or a separate encoder and decoder that
 // communicate BW estimation info explicitly.
 template <typename T, bool adaptive>
 void TestGetSetBandwidthInfo(const int16_t* speech_data,
                              int rate_bits_per_second,
                              int sample_rate_hz,
                              int frame_size_ms) {
   const int bit_rate = 32000;

   // Conjoined encoder/decoder pair:
   typename T::instance_type* encdec;
   ASSERT_EQ(0, T::Create(&encdec));
   ASSERT_EQ(0, T::EncoderInit(encdec, adaptive ? 0 : 1));
   T::DecoderInit(encdec);
   ASSERT_EQ(0, T::SetEncSampRate(encdec, sample_rate_hz));
   if (adaptive)
     ASSERT_EQ(0, T::ControlBwe(encdec, bit_rate, frame_size_ms, false));
   else
     ASSERT_EQ(0, T::Control(encdec, bit_rate, frame_size_ms));

   // Disjoint encoder/decoder pair:
   typename T::instance_type* enc;
   ASSERT_EQ(0, T::Create(&enc));
   ASSERT_EQ(0, T::EncoderInit(enc, adaptive ? 0 : 1));
   ASSERT_EQ(0, T::SetEncSampRate(enc, sample_rate_hz));
   if (adaptive)
     ASSERT_EQ(0, T::ControlBwe(enc, bit_rate, frame_size_ms, false));
   else
     ASSERT_EQ(0, T::Control(enc, bit_rate, frame_size_ms));
   typename T::instance_type* dec;
   ASSERT_EQ(0, T::Create(&dec));
   T::DecoderInit(dec);
   T::SetInitialBweBottleneck(dec, bit_rate);
   T::SetEncSampRateInDecoder(dec, sample_rate_hz);

   // 0. Get initial BW info from decoder.
   auto bi = GetBwInfo<T>(dec);

   BoundedCapacityChannel channel1(sample_rate_hz, rate_bits_per_second),
       channel2(sample_rate_hz, rate_bits_per_second);

   int elapsed_time_ms = 0;
   for (int i = 0; elapsed_time_ms < 10000; ++i) {
     rtc::StringBuilder ss;
     ss << " i = " << i;
     SCOPED_TRACE(ss.str());

     // 1. Encode 3 * 10 ms or 6 * 10 ms. The separate encoder is given the BW
     // info before each encode call.
     rtc::Buffer bitstream1, bitstream2;
     int duration1_ms =
         EncodePacket<T>(encdec, nullptr, speech_data, &bitstream1);
     int duration2_ms = EncodePacket<T>(enc, &bi, speech_data, &bitstream2);
     EXPECT_EQ(duration1_ms, duration2_ms);
     if (adaptive)
       EXPECT_TRUE(duration1_ms == 30 || duration1_ms == 60);
     else
       EXPECT_EQ(frame_size_ms, duration1_ms);
     ASSERT_EQ(bitstream1.size(), bitstream2.size());
     EXPECT_EQ(bitstream1, bitstream2);

     // 2. Deliver the encoded data to the decoders.
     const int send_time = elapsed_time_ms * (sample_rate_hz / 1000);
     EXPECT_EQ(0, T::UpdateBwEstimate(
                      encdec, bitstream1.data(), bitstream1.size(), i, send_time,
                      channel1.Send(send_time,
                                    rtc::checked_cast<int>(bitstream1.size()))));
     EXPECT_EQ(0, T::UpdateBwEstimate(
                      dec, bitstream2.data(), bitstream2.size(), i, send_time,
                      channel2.Send(send_time,
                                    rtc::checked_cast<int>(bitstream2.size()))));

     // 3. Decode, and get new BW info from the separate decoder.
     ASSERT_EQ(0, T::SetDecSampRate(encdec, sample_rate_hz));
     ASSERT_EQ(0, T::SetDecSampRate(dec, sample_rate_hz));
     auto decoded1 = DecodePacket<T>(encdec, bitstream1);
     auto decoded2 = DecodePacket<T>(dec, bitstream2);
     EXPECT_EQ(decoded1, decoded2);
     bi = GetBwInfo<T>(dec);

     elapsed_time_ms += duration1_ms;
   }

   EXPECT_EQ(0, T::Free(encdec));
   EXPECT_EQ(0, T::Free(enc));
   EXPECT_EQ(0, T::Free(dec));
 }

 enum class IsacType { Fix, Float };

 std::ostream& operator<<(std::ostream& os, IsacType t) {
   os << (t == IsacType::Fix ? "fix" : "float");
   return os;
 }

 struct IsacTestParam {
   IsacType isac_type;
   bool adaptive;
   int channel_rate_bits_per_second;
   int sample_rate_hz;
   int frame_size_ms;

   friend std::ostream& operator<<(std::ostream& os, const IsacTestParam& itp) {
     os << '{' << itp.isac_type << ','
        << (itp.adaptive ? "adaptive" : "nonadaptive") << ','
        << itp.channel_rate_bits_per_second << ',' << itp.sample_rate_hz << ','
        << itp.frame_size_ms << '}';
     return os;
   }
 };

 class IsacCommonTest : public testing::TestWithParam<IsacTestParam> {};

 }  // namespace

 TEST_P(IsacCommonTest, GetSetBandwidthInfo) {
   auto p = GetParam();
   auto test_fun = [p] {
     if (p.isac_type == IsacType::Fix) {
       if (p.adaptive)
         return TestGetSetBandwidthInfo<IsacFix, true>;
       else
         return TestGetSetBandwidthInfo<IsacFix, false>;
     } else {
       if (p.adaptive)
         return TestGetSetBandwidthInfo<IsacFloat, true>;
       else
         return TestGetSetBandwidthInfo<IsacFloat, false>;
     }
   }();
   test_fun(LoadSpeechData().data(), p.channel_rate_bits_per_second,
            p.sample_rate_hz, p.frame_size_ms);
 }

 std::vector<IsacTestParam> TestCases() {
   static const IsacType types[] = {IsacType::Fix, IsacType::Float};
   static const bool adaptives[] = {true, false};
   static const int channel_rates[] = {12000, 15000, 19000, 22000};
   static const int sample_rates[] = {16000, 32000};
   static const int frame_sizes[] = {30, 60};
   std::vector<IsacTestParam> cases;
   for (IsacType type : types)
     for (bool adaptive : adaptives)
       for (int channel_rate : channel_rates)
         for (int sample_rate : sample_rates)
           if (!(type == IsacType::Fix && sample_rate == 32000))
             for (int frame_size : frame_sizes)
               if (!(sample_rate == 32000 && frame_size == 60))
                 cases.push_back(
                     {type, adaptive, channel_rate, sample_rate, frame_size});
   return cases;
 }

 INSTANTIATE_TEST_SUITE_P(, IsacCommonTest, testing::ValuesIn(TestCases()));

 }  // namespace webrtc
	/*
	* Copyright (c) 2015 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 <algorithm>
	#include <numeric>
	#include <vector>

	#include "modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h"
	#include "modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
	#include "modules/audio_coding/neteq/tools/input_audio_file.h"
	#include "rtc_base/buffer.h"
	#include "rtc_base/numerics/safe_conversions.h"
	#include "rtc_base/strings/string_builder.h"
	#include "test/gtest.h"
	#include "test/testsupport/file_utils.h"

	namespace webrtc {

	namespace {

	const int kIsacNumberOfSamples = 32 * 60; // 60 ms at 32 kHz

	std::vector<int16_t> LoadSpeechData() {
	webrtc::test::InputAudioFile input_file(
	webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"));
	std::vector<int16_t> speech_data(kIsacNumberOfSamples);
	input_file.Read(kIsacNumberOfSamples, speech_data.data());
	return speech_data;
	}

	template <typename T>
	IsacBandwidthInfo GetBwInfo(typename T::instance_type* inst) {
	IsacBandwidthInfo bi;
	T::GetBandwidthInfo(inst, &bi);
	EXPECT_TRUE(bi.in_use);
	return bi;
	}

	// Encodes one packet. Returns the packet duration in milliseconds.
	template <typename T>
	int EncodePacket(typename T::instance_type* inst,
	const IsacBandwidthInfo* bi,
	const int16_t* speech_data,
	rtc::Buffer* output) {
	output->SetSize(1000);
	for (int duration_ms = 10;; duration_ms += 10) {
	if (bi)
	T::SetBandwidthInfo(inst, bi);
	int encoded_bytes = T::Encode(inst, speech_data, output->data());
	if (encoded_bytes > 0 \|\| duration_ms >= 60) {
	EXPECT_GT(encoded_bytes, 0);
	EXPECT_LE(static_cast<size_t>(encoded_bytes), output->size());
	output->SetSize(encoded_bytes);
	return duration_ms;
	}
	}
	}

	template <typename T>
	std::vector<int16_t> DecodePacket(typename T::instance_type* inst,
	const rtc::Buffer& encoded) {
	std::vector<int16_t> decoded(kIsacNumberOfSamples);
	int16_t speech_type;
	int nsamples = T::DecodeInternal(inst, encoded.data(), encoded.size(),
	&decoded.front(), &speech_type);
	EXPECT_GT(nsamples, 0);
	EXPECT_LE(static_cast<size_t>(nsamples), decoded.size());
	decoded.resize(nsamples);
	return decoded;
	}

	class BoundedCapacityChannel final {
	public:
	BoundedCapacityChannel(int sample_rate_hz, int rate_bits_per_second)
	: current_time_rtp_(0),
	channel_rate_bytes_per_sample_(rate_bits_per_second /
	(8.0 * sample_rate_hz)) {}

	// Simulate sending the given number of bytes at the given RTP time. Returns
	// the new current RTP time after the sending is done.
	int Send(int send_time_rtp, int nbytes) {
	current_time_rtp_ = std::max(current_time_rtp_, send_time_rtp) +
	nbytes / channel_rate_bytes_per_sample_;
	return current_time_rtp_;
	}

	private:
	int current_time_rtp_;
	// The somewhat strange unit for channel rate, bytes per sample, is because
	// RTP time is measured in samples:
	const double channel_rate_bytes_per_sample_;
	};

	// Test that the iSAC encoder produces identical output whether or not we use a
	// conjoined encoder+decoder pair or a separate encoder and decoder that
	// communicate BW estimation info explicitly.
	template <typename T, bool adaptive>
	void TestGetSetBandwidthInfo(const int16_t* speech_data,
	int rate_bits_per_second,
	int sample_rate_hz,
	int frame_size_ms) {
	const int bit_rate = 32000;

	// Conjoined encoder/decoder pair:
	typename T::instance_type* encdec;
	ASSERT_EQ(0, T::Create(&encdec));
	ASSERT_EQ(0, T::EncoderInit(encdec, adaptive ? 0 : 1));
	T::DecoderInit(encdec);
	ASSERT_EQ(0, T::SetEncSampRate(encdec, sample_rate_hz));
	if (adaptive)
	ASSERT_EQ(0, T::ControlBwe(encdec, bit_rate, frame_size_ms, false));
	else
	ASSERT_EQ(0, T::Control(encdec, bit_rate, frame_size_ms));

	// Disjoint encoder/decoder pair:
	typename T::instance_type* enc;
	ASSERT_EQ(0, T::Create(&enc));
	ASSERT_EQ(0, T::EncoderInit(enc, adaptive ? 0 : 1));
	ASSERT_EQ(0, T::SetEncSampRate(enc, sample_rate_hz));
	if (adaptive)
	ASSERT_EQ(0, T::ControlBwe(enc, bit_rate, frame_size_ms, false));
	else
	ASSERT_EQ(0, T::Control(enc, bit_rate, frame_size_ms));
	typename T::instance_type* dec;
	ASSERT_EQ(0, T::Create(&dec));
	T::DecoderInit(dec);
	T::SetInitialBweBottleneck(dec, bit_rate);
	T::SetEncSampRateInDecoder(dec, sample_rate_hz);

	// 0. Get initial BW info from decoder.
	auto bi = GetBwInfo<T>(dec);

	BoundedCapacityChannel channel1(sample_rate_hz, rate_bits_per_second),
	channel2(sample_rate_hz, rate_bits_per_second);

	int elapsed_time_ms = 0;
	for (int i = 0; elapsed_time_ms < 10000; ++i) {
	rtc::StringBuilder ss;
	ss << " i = " << i;
	SCOPED_TRACE(ss.str());

	// 1. Encode 3 * 10 ms or 6 * 10 ms. The separate encoder is given the BW
	// info before each encode call.
	rtc::Buffer bitstream1, bitstream2;
	int duration1_ms =
	EncodePacket<T>(encdec, nullptr, speech_data, &bitstream1);
	int duration2_ms = EncodePacket<T>(enc, &bi, speech_data, &bitstream2);
	EXPECT_EQ(duration1_ms, duration2_ms);
	if (adaptive)
	EXPECT_TRUE(duration1_ms == 30 \|\| duration1_ms == 60);
	else
	EXPECT_EQ(frame_size_ms, duration1_ms);
	ASSERT_EQ(bitstream1.size(), bitstream2.size());
	EXPECT_EQ(bitstream1, bitstream2);

	// 2. Deliver the encoded data to the decoders.
	const int send_time = elapsed_time_ms * (sample_rate_hz / 1000);
	EXPECT_EQ(0, T::UpdateBwEstimate(
	encdec, bitstream1.data(), bitstream1.size(), i, send_time,
	channel1.Send(send_time,
	rtc::checked_cast<int>(bitstream1.size()))));
	EXPECT_EQ(0, T::UpdateBwEstimate(
	dec, bitstream2.data(), bitstream2.size(), i, send_time,
	channel2.Send(send_time,
	rtc::checked_cast<int>(bitstream2.size()))));

	// 3. Decode, and get new BW info from the separate decoder.
	ASSERT_EQ(0, T::SetDecSampRate(encdec, sample_rate_hz));
	ASSERT_EQ(0, T::SetDecSampRate(dec, sample_rate_hz));
	auto decoded1 = DecodePacket<T>(encdec, bitstream1);
	auto decoded2 = DecodePacket<T>(dec, bitstream2);
	EXPECT_EQ(decoded1, decoded2);
	bi = GetBwInfo<T>(dec);

	elapsed_time_ms += duration1_ms;
	}

	EXPECT_EQ(0, T::Free(encdec));
	EXPECT_EQ(0, T::Free(enc));
	EXPECT_EQ(0, T::Free(dec));
	}

	enum class IsacType { Fix, Float };

	std::ostream& operator<<(std::ostream& os, IsacType t) {
	os << (t == IsacType::Fix ? "fix" : "float");
	return os;
	}

	struct IsacTestParam {
	IsacType isac_type;
	bool adaptive;
	int channel_rate_bits_per_second;
	int sample_rate_hz;
	int frame_size_ms;

	friend std::ostream& operator<<(std::ostream& os, const IsacTestParam& itp) {
	os << '{' << itp.isac_type << ','
	<< (itp.adaptive ? "adaptive" : "nonadaptive") << ','
	<< itp.channel_rate_bits_per_second << ',' << itp.sample_rate_hz << ','
	<< itp.frame_size_ms << '}';
	return os;
	}
	};

	class IsacCommonTest : public testing::TestWithParam<IsacTestParam> {};

	} // namespace

	TEST_P(IsacCommonTest, GetSetBandwidthInfo) {
	auto p = GetParam();
	auto test_fun = [p] {
	if (p.isac_type == IsacType::Fix) {
	if (p.adaptive)
	return TestGetSetBandwidthInfo<IsacFix, true>;
	else
	return TestGetSetBandwidthInfo<IsacFix, false>;
	} else {
	if (p.adaptive)
	return TestGetSetBandwidthInfo<IsacFloat, true>;
	else
	return TestGetSetBandwidthInfo<IsacFloat, false>;
	}
	}();
	test_fun(LoadSpeechData().data(), p.channel_rate_bits_per_second,
	p.sample_rate_hz, p.frame_size_ms);
	}

	std::vector<IsacTestParam> TestCases() {
	static const IsacType types[] = {IsacType::Fix, IsacType::Float};
	static const bool adaptives[] = {true, false};
	static const int channel_rates[] = {12000, 15000, 19000, 22000};
	static const int sample_rates[] = {16000, 32000};
	static const int frame_sizes[] = {30, 60};
	std::vector<IsacTestParam> cases;
	for (IsacType type : types)
	for (bool adaptive : adaptives)
	for (int channel_rate : channel_rates)
	for (int sample_rate : sample_rates)
	if (!(type == IsacType::Fix && sample_rate == 32000))
	for (int frame_size : frame_sizes)
	if (!(sample_rate == 32000 && frame_size == 60))
	cases.push_back(
	{type, adaptive, channel_rate, sample_rate, frame_size});
	return cases;
	}

	INSTANTIATE_TEST_SUITE_P(, IsacCommonTest, testing::ValuesIn(TestCases()));

	} // namespace webrtc