modules/audio_coding/test/opus_test.cc - src - Git at Google

 /*
  *  Copyright (c) 2013 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 "modules/audio_coding/test/opus_test.h"

 #include <string>

 #include "api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "modules/audio_coding/codecs/opus/opus_interface.h"
 #include "modules/audio_coding/include/audio_coding_module_typedefs.h"
 #include "modules/audio_coding/test/TestStereo.h"
 #include "test/gtest.h"
 #include "test/testsupport/file_utils.h"

 namespace webrtc {

 OpusTest::OpusTest()
     : acm_receiver_(AudioCodingModule::Create(
           AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))),
       channel_a2b_(NULL),
       counter_(0),
       payload_type_(255),
       rtp_timestamp_(0) {}

 OpusTest::~OpusTest() {
   if (channel_a2b_ != NULL) {
     delete channel_a2b_;
     channel_a2b_ = NULL;
   }
   if (opus_mono_encoder_ != NULL) {
     WebRtcOpus_EncoderFree(opus_mono_encoder_);
     opus_mono_encoder_ = NULL;
   }
   if (opus_stereo_encoder_ != NULL) {
     WebRtcOpus_EncoderFree(opus_stereo_encoder_);
     opus_stereo_encoder_ = NULL;
   }
   if (opus_mono_decoder_ != NULL) {
     WebRtcOpus_DecoderFree(opus_mono_decoder_);
     opus_mono_decoder_ = NULL;
   }
   if (opus_stereo_decoder_ != NULL) {
     WebRtcOpus_DecoderFree(opus_stereo_decoder_);
     opus_stereo_decoder_ = NULL;
   }
 }

 void OpusTest::Perform() {
 #ifndef WEBRTC_CODEC_OPUS
   // Opus isn't defined, exit.
   return;
 #else
   uint16_t frequency_hz;
   size_t audio_channels;
   int16_t test_cntr = 0;

   // Open both mono and stereo test files in 32 kHz.
   const std::string file_name_stereo =
       webrtc::test::ResourcePath("audio_coding/teststereo32kHz", "pcm");
   const std::string file_name_mono =
       webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
   frequency_hz = 32000;
   in_file_stereo_.Open(file_name_stereo, frequency_hz, "rb");
   in_file_stereo_.ReadStereo(true);
   in_file_mono_.Open(file_name_mono, frequency_hz, "rb");
   in_file_mono_.ReadStereo(false);

   // Create Opus encoders for mono and stereo.
   ASSERT_GT(WebRtcOpus_EncoderCreate(&opus_mono_encoder_, 1, 0, 48000), -1);
   ASSERT_GT(WebRtcOpus_EncoderCreate(&opus_stereo_encoder_, 2, 1, 48000), -1);

   // Create Opus decoders for mono and stereo for stand-alone testing of Opus.
   ASSERT_GT(WebRtcOpus_DecoderCreate(&opus_mono_decoder_, 1, 48000), -1);
   ASSERT_GT(WebRtcOpus_DecoderCreate(&opus_stereo_decoder_, 2, 48000), -1);
   WebRtcOpus_DecoderInit(opus_mono_decoder_);
   WebRtcOpus_DecoderInit(opus_stereo_decoder_);

   ASSERT_TRUE(acm_receiver_.get() != NULL);
   EXPECT_EQ(0, acm_receiver_->InitializeReceiver());

   // Register Opus stereo as receiving codec.
   constexpr int kOpusPayloadType = 120;
   const SdpAudioFormat kOpusFormatStereo("opus", 48000, 2, {{"stereo", "1"}});
   payload_type_ = kOpusPayloadType;
   acm_receiver_->SetReceiveCodecs({{kOpusPayloadType, kOpusFormatStereo}});

   // Create and connect the channel.
   channel_a2b_ = new TestPackStereo;
   channel_a2b_->RegisterReceiverACM(acm_receiver_.get());

   //
   // Test Stereo.
   //

   channel_a2b_->set_codec_mode(kStereo);
   audio_channels = 2;
   test_cntr++;
   OpenOutFile(test_cntr);

   // Run Opus with 2.5 ms frame size.
   Run(channel_a2b_, audio_channels, 64000, 120);

   // Run Opus with 5 ms frame size.
   Run(channel_a2b_, audio_channels, 64000, 240);

   // Run Opus with 10 ms frame size.
   Run(channel_a2b_, audio_channels, 64000, 480);

   // Run Opus with 20 ms frame size.
   Run(channel_a2b_, audio_channels, 64000, 960);

   // Run Opus with 40 ms frame size.
   Run(channel_a2b_, audio_channels, 64000, 1920);

   // Run Opus with 60 ms frame size.
   Run(channel_a2b_, audio_channels, 64000, 2880);

   out_file_.Close();
   out_file_standalone_.Close();

   //
   // Test Opus stereo with packet-losses.
   //

   test_cntr++;
   OpenOutFile(test_cntr);

   // Run Opus with 20 ms frame size, 1% packet loss.
   Run(channel_a2b_, audio_channels, 64000, 960, 1);

   // Run Opus with 20 ms frame size, 5% packet loss.
   Run(channel_a2b_, audio_channels, 64000, 960, 5);

   // Run Opus with 20 ms frame size, 10% packet loss.
   Run(channel_a2b_, audio_channels, 64000, 960, 10);

   out_file_.Close();
   out_file_standalone_.Close();

   //
   // Test Mono.
   //
   channel_a2b_->set_codec_mode(kMono);
   audio_channels = 1;
   test_cntr++;
   OpenOutFile(test_cntr);

   // Register Opus mono as receiving codec.
   const SdpAudioFormat kOpusFormatMono("opus", 48000, 2);
   acm_receiver_->SetReceiveCodecs({{kOpusPayloadType, kOpusFormatMono}});

   // Run Opus with 2.5 ms frame size.
   Run(channel_a2b_, audio_channels, 32000, 120);

   // Run Opus with 5 ms frame size.
   Run(channel_a2b_, audio_channels, 32000, 240);

   // Run Opus with 10 ms frame size.
   Run(channel_a2b_, audio_channels, 32000, 480);

   // Run Opus with 20 ms frame size.
   Run(channel_a2b_, audio_channels, 32000, 960);

   // Run Opus with 40 ms frame size.
   Run(channel_a2b_, audio_channels, 32000, 1920);

   // Run Opus with 60 ms frame size.
   Run(channel_a2b_, audio_channels, 32000, 2880);

   out_file_.Close();
   out_file_standalone_.Close();

   //
   // Test Opus mono with packet-losses.
   //
   test_cntr++;
   OpenOutFile(test_cntr);

   // Run Opus with 20 ms frame size, 1% packet loss.
   Run(channel_a2b_, audio_channels, 64000, 960, 1);

   // Run Opus with 20 ms frame size, 5% packet loss.
   Run(channel_a2b_, audio_channels, 64000, 960, 5);

   // Run Opus with 20 ms frame size, 10% packet loss.
   Run(channel_a2b_, audio_channels, 64000, 960, 10);

   // Close the files.
   in_file_stereo_.Close();
   in_file_mono_.Close();
   out_file_.Close();
   out_file_standalone_.Close();
 #endif
 }

 void OpusTest::Run(TestPackStereo* channel,
                    size_t channels,
                    int bitrate,
                    size_t frame_length,
                    int percent_loss) {
   AudioFrame audio_frame;
   int32_t out_freq_hz_b = out_file_.SamplingFrequency();
   const size_t kBufferSizeSamples = 480 * 12 * 2;  // 120 ms stereo audio.
   int16_t audio[kBufferSizeSamples];
   int16_t out_audio[kBufferSizeSamples];
   int16_t audio_type;
   size_t written_samples = 0;
   size_t read_samples = 0;
   size_t decoded_samples = 0;
   bool first_packet = true;
   uint32_t start_time_stamp = 0;

   channel->reset_payload_size();
   counter_ = 0;

   // Set encoder rate.
   EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_mono_encoder_, bitrate));
   EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_stereo_encoder_, bitrate));

 #if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) || defined(WEBRTC_ARCH_ARM)
   // If we are on Android, iOS and/or ARM, use a lower complexity setting as
   // default.
   const int kOpusComplexity5 = 5;
   EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_mono_encoder_, kOpusComplexity5));
   EXPECT_EQ(0,
             WebRtcOpus_SetComplexity(opus_stereo_encoder_, kOpusComplexity5));
 #endif

   // Fast-forward 1 second (100 blocks) since the files start with silence.
   in_file_stereo_.FastForward(100);
   in_file_mono_.FastForward(100);

   // Limit the runtime to 1000 blocks of 10 ms each.
   for (size_t audio_length = 0; audio_length < 1000; audio_length += 10) {
     bool lost_packet = false;

     // Get 10 msec of audio.
     if (channels == 1) {
       if (in_file_mono_.EndOfFile()) {
         break;
       }
       in_file_mono_.Read10MsData(audio_frame);
     } else {
       if (in_file_stereo_.EndOfFile()) {
         break;
       }
       in_file_stereo_.Read10MsData(audio_frame);
     }

     // If input audio is sampled at 32 kHz, resampling to 48 kHz is required.
     EXPECT_EQ(480, resampler_.Resample10Msec(
                        audio_frame.data(), audio_frame.sample_rate_hz_, 48000,
                        channels, kBufferSizeSamples - written_samples,
                        &audio[written_samples]));
     written_samples += 480 * channels;

     // Sometimes we need to loop over the audio vector to produce the right
     // number of packets.
     size_t loop_encode =
         (written_samples - read_samples) / (channels * frame_length);

     if (loop_encode > 0) {
       const size_t kMaxBytes = 1000;  // Maximum number of bytes for one packet.
       size_t bitstream_len_byte;
       uint8_t bitstream[kMaxBytes];
       for (size_t i = 0; i < loop_encode; i++) {
         int bitstream_len_byte_int = WebRtcOpus_Encode(
             (channels == 1) ? opus_mono_encoder_ : opus_stereo_encoder_,
             &audio[read_samples], frame_length, kMaxBytes, bitstream);
         ASSERT_GE(bitstream_len_byte_int, 0);
         bitstream_len_byte = static_cast<size_t>(bitstream_len_byte_int);

         // Simulate packet loss by setting |packet_loss_| to "true" in
         // |percent_loss| percent of the loops.
         // TODO(tlegrand): Move handling of loss simulation to TestPackStereo.
         if (percent_loss > 0) {
           if (counter_ == floor((100 / percent_loss) + 0.5)) {
             counter_ = 0;
             lost_packet = true;
             channel->set_lost_packet(true);
           } else {
             lost_packet = false;
             channel->set_lost_packet(false);
           }
           counter_++;
         }

         // Run stand-alone Opus decoder, or decode PLC.
         if (channels == 1) {
           if (!lost_packet) {
             decoded_samples += WebRtcOpus_Decode(
                 opus_mono_decoder_, bitstream, bitstream_len_byte,
                 &out_audio[decoded_samples * channels], &audio_type);
           } else {
             // Call decoder PLC.
             constexpr int kPlcDurationMs = 10;
             constexpr int kPlcSamples = 48 * kPlcDurationMs;
             size_t total_plc_samples = 0;
             while (total_plc_samples < frame_length) {
               int ret = WebRtcOpus_Decode(
                   opus_mono_decoder_, NULL, 0,
                   &out_audio[decoded_samples * channels], &audio_type);
               EXPECT_EQ(ret, kPlcSamples);
               decoded_samples += ret;
               total_plc_samples += ret;
             }
             EXPECT_EQ(total_plc_samples, frame_length);
           }
         } else {
           if (!lost_packet) {
             decoded_samples += WebRtcOpus_Decode(
                 opus_stereo_decoder_, bitstream, bitstream_len_byte,
                 &out_audio[decoded_samples * channels], &audio_type);
           } else {
             // Call decoder PLC.
             constexpr int kPlcDurationMs = 10;
             constexpr int kPlcSamples = 48 * kPlcDurationMs;
             size_t total_plc_samples = 0;
             while (total_plc_samples < frame_length) {
               int ret = WebRtcOpus_Decode(
                   opus_stereo_decoder_, NULL, 0,
                   &out_audio[decoded_samples * channels], &audio_type);
               EXPECT_EQ(ret, kPlcSamples);
               decoded_samples += ret;
               total_plc_samples += ret;
             }
             EXPECT_EQ(total_plc_samples, frame_length);
           }
         }

         // Send data to the channel. "channel" will handle the loss simulation.
         channel->SendData(AudioFrameType::kAudioFrameSpeech, payload_type_,
                           rtp_timestamp_, bitstream, bitstream_len_byte, 0);
         if (first_packet) {
           first_packet = false;
           start_time_stamp = rtp_timestamp_;
         }
         rtp_timestamp_ += static_cast<uint32_t>(frame_length);
         read_samples += frame_length * channels;
       }
       if (read_samples == written_samples) {
         read_samples = 0;
         written_samples = 0;
       }
     }

     // Run received side of ACM.
     bool muted;
     ASSERT_EQ(
         0, acm_receiver_->PlayoutData10Ms(out_freq_hz_b, &audio_frame, &muted));
     ASSERT_FALSE(muted);

     // Write output speech to file.
     out_file_.Write10MsData(
         audio_frame.data(),
         audio_frame.samples_per_channel_ * audio_frame.num_channels_);

     // Write stand-alone speech to file.
     out_file_standalone_.Write10MsData(out_audio, decoded_samples * channels);

     if (audio_frame.timestamp_ > start_time_stamp) {
       // Number of channels should be the same for both stand-alone and
       // ACM-decoding.
       EXPECT_EQ(audio_frame.num_channels_, channels);
     }

     decoded_samples = 0;
   }

   if (in_file_mono_.EndOfFile()) {
     in_file_mono_.Rewind();
   }
   if (in_file_stereo_.EndOfFile()) {
     in_file_stereo_.Rewind();
   }
   // Reset in case we ended with a lost packet.
   channel->set_lost_packet(false);
 }

 void OpusTest::OpenOutFile(int test_number) {
   std::string file_name;
   std::stringstream file_stream;
   file_stream << webrtc::test::OutputPath() << "opustest_out_" << test_number
               << ".pcm";
   file_name = file_stream.str();
   out_file_.Open(file_name, 48000, "wb");
   file_stream.str("");
   file_name = file_stream.str();
   file_stream << webrtc::test::OutputPath() << "opusstandalone_out_"
               << test_number << ".pcm";
   file_name = file_stream.str();
   out_file_standalone_.Open(file_name, 48000, "wb");
 }

 }  // namespace webrtc
	/*
	* Copyright (c) 2013 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 "modules/audio_coding/test/opus_test.h"

	#include <string>

	#include "api/audio_codecs/builtin_audio_decoder_factory.h"
	#include "modules/audio_coding/codecs/opus/opus_interface.h"
	#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
	#include "modules/audio_coding/test/TestStereo.h"
	#include "test/gtest.h"
	#include "test/testsupport/file_utils.h"

	namespace webrtc {

	OpusTest::OpusTest()
	: acm_receiver_(AudioCodingModule::Create(
	AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))),
	channel_a2b_(NULL),
	counter_(0),
	payload_type_(255),
	rtp_timestamp_(0) {}

	OpusTest::~OpusTest() {
	if (channel_a2b_ != NULL) {
	delete channel_a2b_;
	channel_a2b_ = NULL;
	}
	if (opus_mono_encoder_ != NULL) {
	WebRtcOpus_EncoderFree(opus_mono_encoder_);
	opus_mono_encoder_ = NULL;
	}
	if (opus_stereo_encoder_ != NULL) {
	WebRtcOpus_EncoderFree(opus_stereo_encoder_);
	opus_stereo_encoder_ = NULL;
	}
	if (opus_mono_decoder_ != NULL) {
	WebRtcOpus_DecoderFree(opus_mono_decoder_);
	opus_mono_decoder_ = NULL;
	}
	if (opus_stereo_decoder_ != NULL) {
	WebRtcOpus_DecoderFree(opus_stereo_decoder_);
	opus_stereo_decoder_ = NULL;
	}
	}

	void OpusTest::Perform() {
	#ifndef WEBRTC_CODEC_OPUS
	// Opus isn't defined, exit.
	return;
	#else
	uint16_t frequency_hz;
	size_t audio_channels;
	int16_t test_cntr = 0;

	// Open both mono and stereo test files in 32 kHz.
	const std::string file_name_stereo =
	webrtc::test::ResourcePath("audio_coding/teststereo32kHz", "pcm");
	const std::string file_name_mono =
	webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
	frequency_hz = 32000;
	in_file_stereo_.Open(file_name_stereo, frequency_hz, "rb");
	in_file_stereo_.ReadStereo(true);
	in_file_mono_.Open(file_name_mono, frequency_hz, "rb");
	in_file_mono_.ReadStereo(false);

	// Create Opus encoders for mono and stereo.
	ASSERT_GT(WebRtcOpus_EncoderCreate(&opus_mono_encoder_, 1, 0, 48000), -1);
	ASSERT_GT(WebRtcOpus_EncoderCreate(&opus_stereo_encoder_, 2, 1, 48000), -1);

	// Create Opus decoders for mono and stereo for stand-alone testing of Opus.
	ASSERT_GT(WebRtcOpus_DecoderCreate(&opus_mono_decoder_, 1, 48000), -1);
	ASSERT_GT(WebRtcOpus_DecoderCreate(&opus_stereo_decoder_, 2, 48000), -1);
	WebRtcOpus_DecoderInit(opus_mono_decoder_);
	WebRtcOpus_DecoderInit(opus_stereo_decoder_);

	ASSERT_TRUE(acm_receiver_.get() != NULL);
	EXPECT_EQ(0, acm_receiver_->InitializeReceiver());

	// Register Opus stereo as receiving codec.
	constexpr int kOpusPayloadType = 120;
	const SdpAudioFormat kOpusFormatStereo("opus", 48000, 2, {{"stereo", "1"}});
	payload_type_ = kOpusPayloadType;
	acm_receiver_->SetReceiveCodecs({{kOpusPayloadType, kOpusFormatStereo}});

	// Create and connect the channel.
	channel_a2b_ = new TestPackStereo;
	channel_a2b_->RegisterReceiverACM(acm_receiver_.get());

	//
	// Test Stereo.
	//

	channel_a2b_->set_codec_mode(kStereo);
	audio_channels = 2;
	test_cntr++;
	OpenOutFile(test_cntr);

	// Run Opus with 2.5 ms frame size.
	Run(channel_a2b_, audio_channels, 64000, 120);

	// Run Opus with 5 ms frame size.
	Run(channel_a2b_, audio_channels, 64000, 240);

	// Run Opus with 10 ms frame size.
	Run(channel_a2b_, audio_channels, 64000, 480);

	// Run Opus with 20 ms frame size.
	Run(channel_a2b_, audio_channels, 64000, 960);

	// Run Opus with 40 ms frame size.
	Run(channel_a2b_, audio_channels, 64000, 1920);

	// Run Opus with 60 ms frame size.
	Run(channel_a2b_, audio_channels, 64000, 2880);

	out_file_.Close();
	out_file_standalone_.Close();

	//
	// Test Opus stereo with packet-losses.
	//

	test_cntr++;
	OpenOutFile(test_cntr);

	// Run Opus with 20 ms frame size, 1% packet loss.
	Run(channel_a2b_, audio_channels, 64000, 960, 1);

	// Run Opus with 20 ms frame size, 5% packet loss.
	Run(channel_a2b_, audio_channels, 64000, 960, 5);

	// Run Opus with 20 ms frame size, 10% packet loss.
	Run(channel_a2b_, audio_channels, 64000, 960, 10);

	out_file_.Close();
	out_file_standalone_.Close();

	//
	// Test Mono.
	//
	channel_a2b_->set_codec_mode(kMono);
	audio_channels = 1;
	test_cntr++;
	OpenOutFile(test_cntr);

	// Register Opus mono as receiving codec.
	const SdpAudioFormat kOpusFormatMono("opus", 48000, 2);
	acm_receiver_->SetReceiveCodecs({{kOpusPayloadType, kOpusFormatMono}});

	// Run Opus with 2.5 ms frame size.
	Run(channel_a2b_, audio_channels, 32000, 120);

	// Run Opus with 5 ms frame size.
	Run(channel_a2b_, audio_channels, 32000, 240);

	// Run Opus with 10 ms frame size.
	Run(channel_a2b_, audio_channels, 32000, 480);

	// Run Opus with 20 ms frame size.
	Run(channel_a2b_, audio_channels, 32000, 960);

	// Run Opus with 40 ms frame size.
	Run(channel_a2b_, audio_channels, 32000, 1920);

	// Run Opus with 60 ms frame size.
	Run(channel_a2b_, audio_channels, 32000, 2880);

	out_file_.Close();
	out_file_standalone_.Close();

	//
	// Test Opus mono with packet-losses.
	//
	test_cntr++;
	OpenOutFile(test_cntr);

	// Run Opus with 20 ms frame size, 1% packet loss.
	Run(channel_a2b_, audio_channels, 64000, 960, 1);

	// Run Opus with 20 ms frame size, 5% packet loss.
	Run(channel_a2b_, audio_channels, 64000, 960, 5);

	// Run Opus with 20 ms frame size, 10% packet loss.
	Run(channel_a2b_, audio_channels, 64000, 960, 10);

	// Close the files.
	in_file_stereo_.Close();
	in_file_mono_.Close();
	out_file_.Close();
	out_file_standalone_.Close();
	#endif
	}

	void OpusTest::Run(TestPackStereo* channel,
	size_t channels,
	int bitrate,
	size_t frame_length,
	int percent_loss) {
	AudioFrame audio_frame;
	int32_t out_freq_hz_b = out_file_.SamplingFrequency();
	const size_t kBufferSizeSamples = 480 * 12 * 2; // 120 ms stereo audio.
	int16_t audio[kBufferSizeSamples];
	int16_t out_audio[kBufferSizeSamples];
	int16_t audio_type;
	size_t written_samples = 0;
	size_t read_samples = 0;
	size_t decoded_samples = 0;
	bool first_packet = true;
	uint32_t start_time_stamp = 0;

	channel->reset_payload_size();
	counter_ = 0;

	// Set encoder rate.
	EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_mono_encoder_, bitrate));
	EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_stereo_encoder_, bitrate));

	#if defined(WEBRTC_ANDROID) \|\| defined(WEBRTC_IOS) \|\| defined(WEBRTC_ARCH_ARM)
	// If we are on Android, iOS and/or ARM, use a lower complexity setting as
	// default.
	const int kOpusComplexity5 = 5;
	EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_mono_encoder_, kOpusComplexity5));
	EXPECT_EQ(0,
	WebRtcOpus_SetComplexity(opus_stereo_encoder_, kOpusComplexity5));
	#endif

	// Fast-forward 1 second (100 blocks) since the files start with silence.
	in_file_stereo_.FastForward(100);
	in_file_mono_.FastForward(100);

	// Limit the runtime to 1000 blocks of 10 ms each.
	for (size_t audio_length = 0; audio_length < 1000; audio_length += 10) {
	bool lost_packet = false;

	// Get 10 msec of audio.
	if (channels == 1) {
	if (in_file_mono_.EndOfFile()) {
	break;
	}
	in_file_mono_.Read10MsData(audio_frame);
	} else {
	if (in_file_stereo_.EndOfFile()) {
	break;
	}
	in_file_stereo_.Read10MsData(audio_frame);
	}

	// If input audio is sampled at 32 kHz, resampling to 48 kHz is required.
	EXPECT_EQ(480, resampler_.Resample10Msec(
	audio_frame.data(), audio_frame.sample_rate_hz_, 48000,
	channels, kBufferSizeSamples - written_samples,
	&audio[written_samples]));
	written_samples += 480 * channels;

	// Sometimes we need to loop over the audio vector to produce the right
	// number of packets.
	size_t loop_encode =
	(written_samples - read_samples) / (channels * frame_length);

	if (loop_encode > 0) {
	const size_t kMaxBytes = 1000; // Maximum number of bytes for one packet.
	size_t bitstream_len_byte;
	uint8_t bitstream[kMaxBytes];
	for (size_t i = 0; i < loop_encode; i++) {
	int bitstream_len_byte_int = WebRtcOpus_Encode(
	(channels == 1) ? opus_mono_encoder_ : opus_stereo_encoder_,
	&audio[read_samples], frame_length, kMaxBytes, bitstream);
	ASSERT_GE(bitstream_len_byte_int, 0);
	bitstream_len_byte = static_cast<size_t>(bitstream_len_byte_int);

	// Simulate packet loss by setting \|packet_loss_\| to "true" in
	// \|percent_loss\| percent of the loops.
	// TODO(tlegrand): Move handling of loss simulation to TestPackStereo.
	if (percent_loss > 0) {
	if (counter_ == floor((100 / percent_loss) + 0.5)) {
	counter_ = 0;
	lost_packet = true;
	channel->set_lost_packet(true);
	} else {
	lost_packet = false;
	channel->set_lost_packet(false);
	}
	counter_++;
	}

	// Run stand-alone Opus decoder, or decode PLC.
	if (channels == 1) {
	if (!lost_packet) {
	decoded_samples += WebRtcOpus_Decode(
	opus_mono_decoder_, bitstream, bitstream_len_byte,
	&out_audio[decoded_samples * channels], &audio_type);
	} else {
	// Call decoder PLC.
	constexpr int kPlcDurationMs = 10;
	constexpr int kPlcSamples = 48 * kPlcDurationMs;
	size_t total_plc_samples = 0;
	while (total_plc_samples < frame_length) {
	int ret = WebRtcOpus_Decode(
	opus_mono_decoder_, NULL, 0,
	&out_audio[decoded_samples * channels], &audio_type);
	EXPECT_EQ(ret, kPlcSamples);
	decoded_samples += ret;
	total_plc_samples += ret;
	}
	EXPECT_EQ(total_plc_samples, frame_length);
	}
	} else {
	if (!lost_packet) {
	decoded_samples += WebRtcOpus_Decode(
	opus_stereo_decoder_, bitstream, bitstream_len_byte,
	&out_audio[decoded_samples * channels], &audio_type);
	} else {
	// Call decoder PLC.
	constexpr int kPlcDurationMs = 10;
	constexpr int kPlcSamples = 48 * kPlcDurationMs;
	size_t total_plc_samples = 0;
	while (total_plc_samples < frame_length) {
	int ret = WebRtcOpus_Decode(
	opus_stereo_decoder_, NULL, 0,
	&out_audio[decoded_samples * channels], &audio_type);
	EXPECT_EQ(ret, kPlcSamples);
	decoded_samples += ret;
	total_plc_samples += ret;
	}
	EXPECT_EQ(total_plc_samples, frame_length);
	}
	}

	// Send data to the channel. "channel" will handle the loss simulation.
	channel->SendData(AudioFrameType::kAudioFrameSpeech, payload_type_,
	rtp_timestamp_, bitstream, bitstream_len_byte, 0);
	if (first_packet) {
	first_packet = false;
	start_time_stamp = rtp_timestamp_;
	}
	rtp_timestamp_ += static_cast<uint32_t>(frame_length);
	read_samples += frame_length * channels;
	}
	if (read_samples == written_samples) {
	read_samples = 0;
	written_samples = 0;
	}
	}

	// Run received side of ACM.
	bool muted;
	ASSERT_EQ(
	0, acm_receiver_->PlayoutData10Ms(out_freq_hz_b, &audio_frame, &muted));
	ASSERT_FALSE(muted);

	// Write output speech to file.
	out_file_.Write10MsData(
	audio_frame.data(),
	audio_frame.samples_per_channel_ * audio_frame.num_channels_);

	// Write stand-alone speech to file.
	out_file_standalone_.Write10MsData(out_audio, decoded_samples * channels);

	if (audio_frame.timestamp_ > start_time_stamp) {
	// Number of channels should be the same for both stand-alone and
	// ACM-decoding.
	EXPECT_EQ(audio_frame.num_channels_, channels);
	}

	decoded_samples = 0;
	}

	if (in_file_mono_.EndOfFile()) {
	in_file_mono_.Rewind();
	}
	if (in_file_stereo_.EndOfFile()) {
	in_file_stereo_.Rewind();
	}
	// Reset in case we ended with a lost packet.
	channel->set_lost_packet(false);
	}

	void OpusTest::OpenOutFile(int test_number) {
	std::string file_name;
	std::stringstream file_stream;
	file_stream << webrtc::test::OutputPath() << "opustest_out_" << test_number
	<< ".pcm";
	file_name = file_stream.str();
	out_file_.Open(file_name, 48000, "wb");
	file_stream.str("");
	file_name = file_stream.str();
	file_stream << webrtc::test::OutputPath() << "opusstandalone_out_"
	<< test_number << ".pcm";
	file_name = file_stream.str();
	out_file_standalone_.Open(file_name, 48000, "wb");
	}

	} // namespace webrtc