modules/audio_coding/codecs/opus/opus_fec_test.cc - src - Git at Google

 /*
  *  Copyright (c) 2014 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 <memory>

 #include "modules/audio_coding/codecs/opus/opus_interface.h"
 #include "rtc_base/format_macros.h"
 #include "test/gtest.h"
 #include "test/testsupport/file_utils.h"

 using std::get;
 using std::string;
 using std::tuple;
 using ::testing::TestWithParam;

 namespace webrtc {

 // Define coding parameter as <channels, bit_rate, filename, extension>.
 typedef tuple<size_t, int, string, string> coding_param;
 typedef struct mode mode;

 struct mode {
   bool fec;
   uint8_t target_packet_loss_rate;
 };

 const int kOpusBlockDurationMs = 20;
 const int kOpusSamplingKhz = 48;

 class OpusFecTest : public TestWithParam<coding_param> {
  protected:
   OpusFecTest();

   void SetUp() override;
   void TearDown() override;

   virtual void EncodeABlock();

   virtual void DecodeABlock(bool lost_previous, bool lost_current);

   int block_duration_ms_;
   int sampling_khz_;
   size_t block_length_sample_;

   size_t channels_;
   int bit_rate_;

   size_t data_pointer_;
   size_t loop_length_samples_;
   size_t max_bytes_;
   size_t encoded_bytes_;

   WebRtcOpusEncInst* opus_encoder_;
   WebRtcOpusDecInst* opus_decoder_;

   string in_filename_;

   std::unique_ptr<int16_t[]> in_data_;
   std::unique_ptr<int16_t[]> out_data_;
   std::unique_ptr<uint8_t[]> bit_stream_;
 };

 void OpusFecTest::SetUp() {
   channels_ = get<0>(GetParam());
   bit_rate_ = get<1>(GetParam());
   printf("Coding %" RTC_PRIuS " channel signal at %d bps.\n", channels_,
          bit_rate_);

   in_filename_ = test::ResourcePath(get<2>(GetParam()), get<3>(GetParam()));

   FILE* fp = fopen(in_filename_.c_str(), "rb");
   ASSERT_FALSE(fp == NULL);

   // Obtain file size.
   fseek(fp, 0, SEEK_END);
   loop_length_samples_ = ftell(fp) / sizeof(int16_t);
   rewind(fp);

   // Allocate memory to contain the whole file.
   in_data_.reset(
       new int16_t[loop_length_samples_ + block_length_sample_ * channels_]);

   // Copy the file into the buffer.
   ASSERT_EQ(fread(&in_data_[0], sizeof(int16_t), loop_length_samples_, fp),
             loop_length_samples_);
   fclose(fp);

   // The audio will be used in a looped manner. To ease the acquisition of an
   // audio frame that crosses the end of the excerpt, we add an extra block
   // length of samples to the end of the array, starting over again from the
   // beginning of the array. Audio frames cross the end of the excerpt always
   // appear as a continuum of memory.
   memcpy(&in_data_[loop_length_samples_], &in_data_[0],
          block_length_sample_ * channels_ * sizeof(int16_t));

   // Maximum number of bytes in output bitstream.
   max_bytes_ = block_length_sample_ * channels_ * sizeof(int16_t);

   out_data_.reset(new int16_t[2 * block_length_sample_ * channels_]);
   bit_stream_.reset(new uint8_t[max_bytes_]);

   // If channels_ == 1, use Opus VOIP mode, otherwise, audio mode.
   int app = channels_ == 1 ? 0 : 1;

   // Create encoder memory.
   EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_, channels_, app, 48000));
   EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_, channels_, 48000));
   // Set bitrate.
   EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_, bit_rate_));
 }

 void OpusFecTest::TearDown() {
   // Free memory.
   EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
   EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
 }

 OpusFecTest::OpusFecTest()
     : block_duration_ms_(kOpusBlockDurationMs),
       sampling_khz_(kOpusSamplingKhz),
       block_length_sample_(
           static_cast<size_t>(block_duration_ms_ * sampling_khz_)),
       data_pointer_(0),
       max_bytes_(0),
       encoded_bytes_(0),
       opus_encoder_(NULL),
       opus_decoder_(NULL) {}

 void OpusFecTest::EncodeABlock() {
   int value =
       WebRtcOpus_Encode(opus_encoder_, &in_data_[data_pointer_],
                         block_length_sample_, max_bytes_, &bit_stream_[0]);
   EXPECT_GT(value, 0);

   encoded_bytes_ = static_cast<size_t>(value);
 }

 void OpusFecTest::DecodeABlock(bool lost_previous, bool lost_current) {
   int16_t audio_type;
   int value_1 = 0, value_2 = 0;

   if (lost_previous) {
     // Decode previous frame.
     if (!lost_current &&
         WebRtcOpus_PacketHasFec(&bit_stream_[0], encoded_bytes_) == 1) {
       value_1 =
           WebRtcOpus_DecodeFec(opus_decoder_, &bit_stream_[0], encoded_bytes_,
                                &out_data_[0], &audio_type);
     } else {
       // Call decoder PLC.
       while (value_1 < static_cast<int>(block_length_sample_)) {
         int ret = WebRtcOpus_Decode(opus_decoder_, NULL, 0, &out_data_[value_1],
                                     &audio_type);
         EXPECT_EQ(ret, sampling_khz_ * 10);  // Should return 10 ms of samples.
         value_1 += ret;
       }
     }
     EXPECT_EQ(static_cast<int>(block_length_sample_), value_1);
   }

   if (!lost_current) {
     // Decode current frame.
     value_2 = WebRtcOpus_Decode(opus_decoder_, &bit_stream_[0], encoded_bytes_,
                                 &out_data_[value_1 * channels_], &audio_type);
     EXPECT_EQ(static_cast<int>(block_length_sample_), value_2);
   }
 }

 TEST_P(OpusFecTest, RandomPacketLossTest) {
   const int kDurationMs = 200000;
   int time_now_ms, fec_frames;
   int actual_packet_loss_rate;
   bool lost_current, lost_previous;
   mode mode_set[3] = {{true, 0}, {false, 0}, {true, 50}};

   lost_current = false;
   for (int i = 0; i < 3; i++) {
     if (mode_set[i].fec) {
       EXPECT_EQ(0, WebRtcOpus_EnableFec(opus_encoder_));
       EXPECT_EQ(0, WebRtcOpus_SetPacketLossRate(
                        opus_encoder_, mode_set[i].target_packet_loss_rate));
       printf("FEC is ON, target at packet loss rate %d percent.\n",
              mode_set[i].target_packet_loss_rate);
     } else {
       EXPECT_EQ(0, WebRtcOpus_DisableFec(opus_encoder_));
       printf("FEC is OFF.\n");
     }
     // In this test, we let the target packet loss rate match the actual rate.
     actual_packet_loss_rate = mode_set[i].target_packet_loss_rate;
     // Run every mode a certain time.
     time_now_ms = 0;
     fec_frames = 0;
     while (time_now_ms < kDurationMs) {
       // Encode & decode.
       EncodeABlock();

       // Check if payload has FEC.
       int fec = WebRtcOpus_PacketHasFec(&bit_stream_[0], encoded_bytes_);

       // If FEC is disabled or the target packet loss rate is set to 0, there
       // should be no FEC in the bit stream.
       if (!mode_set[i].fec || mode_set[i].target_packet_loss_rate == 0) {
         EXPECT_EQ(fec, 0);
       } else if (fec == 1) {
         fec_frames++;
       }

       lost_previous = lost_current;
       lost_current = rand() < actual_packet_loss_rate * (RAND_MAX / 100);
       DecodeABlock(lost_previous, lost_current);

       time_now_ms += block_duration_ms_;

       // |data_pointer_| is incremented and wrapped across
       // |loop_length_samples_|.
       data_pointer_ = (data_pointer_ + block_length_sample_ * channels_) %
                       loop_length_samples_;
     }
     if (mode_set[i].fec) {
       printf("%.2f percent frames has FEC.\n",
              static_cast<float>(fec_frames) * block_duration_ms_ / 2000);
     }
   }
 }

 const coding_param param_set[] = {
     std::make_tuple(1,
                     64000,
                     string("audio_coding/testfile32kHz"),
                     string("pcm")),
     std::make_tuple(1,
                     32000,
                     string("audio_coding/testfile32kHz"),
                     string("pcm")),
     std::make_tuple(2,
                     64000,
                     string("audio_coding/teststereo32kHz"),
                     string("pcm"))};

 // 64 kbps, stereo
 INSTANTIATE_TEST_SUITE_P(AllTest, OpusFecTest, ::testing::ValuesIn(param_set));

 }  // namespace webrtc
	/*
	* Copyright (c) 2014 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 <memory>

	#include "modules/audio_coding/codecs/opus/opus_interface.h"
	#include "rtc_base/format_macros.h"
	#include "test/gtest.h"
	#include "test/testsupport/file_utils.h"

	using std::get;
	using std::string;
	using std::tuple;
	using ::testing::TestWithParam;

	namespace webrtc {

	// Define coding parameter as <channels, bit_rate, filename, extension>.
	typedef tuple<size_t, int, string, string> coding_param;
	typedef struct mode mode;

	struct mode {
	bool fec;
	uint8_t target_packet_loss_rate;
	};

	const int kOpusBlockDurationMs = 20;
	const int kOpusSamplingKhz = 48;

	class OpusFecTest : public TestWithParam<coding_param> {
	protected:
	OpusFecTest();

	void SetUp() override;
	void TearDown() override;

	virtual void EncodeABlock();

	virtual void DecodeABlock(bool lost_previous, bool lost_current);

	int block_duration_ms_;
	int sampling_khz_;
	size_t block_length_sample_;

	size_t channels_;
	int bit_rate_;

	size_t data_pointer_;
	size_t loop_length_samples_;
	size_t max_bytes_;
	size_t encoded_bytes_;

	WebRtcOpusEncInst* opus_encoder_;
	WebRtcOpusDecInst* opus_decoder_;

	string in_filename_;

	std::unique_ptr<int16_t[]> in_data_;
	std::unique_ptr<int16_t[]> out_data_;
	std::unique_ptr<uint8_t[]> bit_stream_;
	};

	void OpusFecTest::SetUp() {
	channels_ = get<0>(GetParam());
	bit_rate_ = get<1>(GetParam());
	printf("Coding %" RTC_PRIuS " channel signal at %d bps.\n", channels_,
	bit_rate_);

	in_filename_ = test::ResourcePath(get<2>(GetParam()), get<3>(GetParam()));

	FILE* fp = fopen(in_filename_.c_str(), "rb");
	ASSERT_FALSE(fp == NULL);

	// Obtain file size.
	fseek(fp, 0, SEEK_END);
	loop_length_samples_ = ftell(fp) / sizeof(int16_t);
	rewind(fp);

	// Allocate memory to contain the whole file.
	in_data_.reset(
	new int16_t[loop_length_samples_ + block_length_sample_ * channels_]);

	// Copy the file into the buffer.
	ASSERT_EQ(fread(&in_data_[0], sizeof(int16_t), loop_length_samples_, fp),
	loop_length_samples_);
	fclose(fp);

	// The audio will be used in a looped manner. To ease the acquisition of an
	// audio frame that crosses the end of the excerpt, we add an extra block
	// length of samples to the end of the array, starting over again from the
	// beginning of the array. Audio frames cross the end of the excerpt always
	// appear as a continuum of memory.
	memcpy(&in_data_[loop_length_samples_], &in_data_[0],
	block_length_sample_ * channels_ * sizeof(int16_t));

	// Maximum number of bytes in output bitstream.
	max_bytes_ = block_length_sample_ * channels_ * sizeof(int16_t);

	out_data_.reset(new int16_t[2 * block_length_sample_ * channels_]);
	bit_stream_.reset(new uint8_t[max_bytes_]);

	// If channels_ == 1, use Opus VOIP mode, otherwise, audio mode.
	int app = channels_ == 1 ? 0 : 1;

	// Create encoder memory.
	EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_, channels_, app, 48000));
	EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_, channels_, 48000));
	// Set bitrate.
	EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_, bit_rate_));
	}

	void OpusFecTest::TearDown() {
	// Free memory.
	EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
	EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
	}

	OpusFecTest::OpusFecTest()
	: block_duration_ms_(kOpusBlockDurationMs),
	sampling_khz_(kOpusSamplingKhz),
	block_length_sample_(
	static_cast<size_t>(block_duration_ms_ * sampling_khz_)),
	data_pointer_(0),
	max_bytes_(0),
	encoded_bytes_(0),
	opus_encoder_(NULL),
	opus_decoder_(NULL) {}

	void OpusFecTest::EncodeABlock() {
	int value =
	WebRtcOpus_Encode(opus_encoder_, &in_data_[data_pointer_],
	block_length_sample_, max_bytes_, &bit_stream_[0]);
	EXPECT_GT(value, 0);

	encoded_bytes_ = static_cast<size_t>(value);
	}

	void OpusFecTest::DecodeABlock(bool lost_previous, bool lost_current) {
	int16_t audio_type;
	int value_1 = 0, value_2 = 0;

	if (lost_previous) {
	// Decode previous frame.
	if (!lost_current &&
	WebRtcOpus_PacketHasFec(&bit_stream_[0], encoded_bytes_) == 1) {
	value_1 =
	WebRtcOpus_DecodeFec(opus_decoder_, &bit_stream_[0], encoded_bytes_,
	&out_data_[0], &audio_type);
	} else {
	// Call decoder PLC.
	while (value_1 < static_cast<int>(block_length_sample_)) {
	int ret = WebRtcOpus_Decode(opus_decoder_, NULL, 0, &out_data_[value_1],
	&audio_type);
	EXPECT_EQ(ret, sampling_khz_ * 10); // Should return 10 ms of samples.
	value_1 += ret;
	}
	}
	EXPECT_EQ(static_cast<int>(block_length_sample_), value_1);
	}

	if (!lost_current) {
	// Decode current frame.
	value_2 = WebRtcOpus_Decode(opus_decoder_, &bit_stream_[0], encoded_bytes_,
	&out_data_[value_1 * channels_], &audio_type);
	EXPECT_EQ(static_cast<int>(block_length_sample_), value_2);
	}
	}

	TEST_P(OpusFecTest, RandomPacketLossTest) {
	const int kDurationMs = 200000;
	int time_now_ms, fec_frames;
	int actual_packet_loss_rate;
	bool lost_current, lost_previous;
	mode mode_set[3] = {{true, 0}, {false, 0}, {true, 50}};

	lost_current = false;
	for (int i = 0; i < 3; i++) {
	if (mode_set[i].fec) {
	EXPECT_EQ(0, WebRtcOpus_EnableFec(opus_encoder_));
	EXPECT_EQ(0, WebRtcOpus_SetPacketLossRate(
	opus_encoder_, mode_set[i].target_packet_loss_rate));
	printf("FEC is ON, target at packet loss rate %d percent.\n",
	mode_set[i].target_packet_loss_rate);
	} else {
	EXPECT_EQ(0, WebRtcOpus_DisableFec(opus_encoder_));
	printf("FEC is OFF.\n");
	}
	// In this test, we let the target packet loss rate match the actual rate.
	actual_packet_loss_rate = mode_set[i].target_packet_loss_rate;
	// Run every mode a certain time.
	time_now_ms = 0;
	fec_frames = 0;
	while (time_now_ms < kDurationMs) {
	// Encode & decode.
	EncodeABlock();

	// Check if payload has FEC.
	int fec = WebRtcOpus_PacketHasFec(&bit_stream_[0], encoded_bytes_);

	// If FEC is disabled or the target packet loss rate is set to 0, there
	// should be no FEC in the bit stream.
	if (!mode_set[i].fec \|\| mode_set[i].target_packet_loss_rate == 0) {
	EXPECT_EQ(fec, 0);
	} else if (fec == 1) {
	fec_frames++;
	}

	lost_previous = lost_current;
	lost_current = rand() < actual_packet_loss_rate * (RAND_MAX / 100);
	DecodeABlock(lost_previous, lost_current);

	time_now_ms += block_duration_ms_;

	// \|data_pointer_\| is incremented and wrapped across
	// \|loop_length_samples_\|.
	data_pointer_ = (data_pointer_ + block_length_sample_ * channels_) %
	loop_length_samples_;
	}
	if (mode_set[i].fec) {
	printf("%.2f percent frames has FEC.\n",
	static_cast<float>(fec_frames) * block_duration_ms_ / 2000);
	}
	}
	}

	const coding_param param_set[] = {
	std::make_tuple(1,
	64000,
	string("audio_coding/testfile32kHz"),
	string("pcm")),
	std::make_tuple(1,
	32000,
	string("audio_coding/testfile32kHz"),
	string("pcm")),
	std::make_tuple(2,
	64000,
	string("audio_coding/teststereo32kHz"),
	string("pcm"))};

	// 64 kbps, stereo
	INSTANTIATE_TEST_SUITE_P(AllTest, OpusFecTest, ::testing::ValuesIn(param_set));

	} // namespace webrtc