modules/audio_coding/acm2/acm_receiver_unittest.cc - mirrors/cros/chromiumos/third_party/webrtc-apm - 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/acm2/acm_receiver.h"

 #include <algorithm>  // std::min
 #include <memory>

 #include "api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "api/audio_codecs/builtin_audio_encoder_factory.h"
 #include "modules/audio_coding/acm2/rent_a_codec.h"
 #include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
 #include "modules/audio_coding/include/audio_coding_module.h"
 #include "modules/audio_coding/neteq/tools/rtp_generator.h"
 #include "modules/include/module_common_types.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/numerics/safe_conversions.h"
 #include "system_wrappers/include/clock.h"
 #include "test/gtest.h"
 #include "test/testsupport/fileutils.h"

 namespace webrtc {

 namespace acm2 {

 class AcmReceiverTestOldApi : public AudioPacketizationCallback,
                               public ::testing::Test {
  protected:
   AcmReceiverTestOldApi()
       : timestamp_(0),
         packet_sent_(false),
         last_packet_send_timestamp_(timestamp_),
         last_frame_type_(kEmptyFrame) {
     config_.decoder_factory = decoder_factory_;
   }

   ~AcmReceiverTestOldApi() {}

   void SetUp() override {
     acm_.reset(AudioCodingModule::Create(config_));
     receiver_.reset(new AcmReceiver(config_));
     ASSERT_TRUE(receiver_.get() != NULL);
     ASSERT_TRUE(acm_.get() != NULL);
     acm_->InitializeReceiver();
     acm_->RegisterTransportCallback(this);

     rtp_header_.header.sequenceNumber = 0;
     rtp_header_.header.timestamp = 0;
     rtp_header_.header.markerBit = false;
     rtp_header_.header.ssrc = 0x12345678;  // Arbitrary.
     rtp_header_.header.numCSRCs = 0;
     rtp_header_.header.payloadType = 0;
     rtp_header_.frameType = kAudioFrameSpeech;
   }

   void TearDown() override {}

   AudioCodecInfo SetEncoder(int payload_type,
                             const SdpAudioFormat& format,
                             const std::map<int, int> cng_payload_types = {}) {
     // Create the speech encoder.
     AudioCodecInfo info = encoder_factory_->QueryAudioEncoder(format).value();
     std::unique_ptr<AudioEncoder> enc =
         encoder_factory_->MakeAudioEncoder(payload_type, format, absl::nullopt);

     // If we have a compatible CN specification, stack a CNG on top.
     auto it = cng_payload_types.find(info.sample_rate_hz);
     if (it != cng_payload_types.end()) {
       AudioEncoderCngConfig config;
       config.speech_encoder = std::move(enc);
       config.num_channels = 1;
       config.payload_type = it->second;
       config.vad_mode = Vad::kVadNormal;
       enc = CreateComfortNoiseEncoder(std::move(config));
     }

     // Actually start using the new encoder.
     acm_->SetEncoder(std::move(enc));
     return info;
   }

   int InsertOnePacketOfSilence(const AudioCodecInfo& info) {
     // Frame setup according to the codec.
     AudioFrame frame;
     frame.sample_rate_hz_ = info.sample_rate_hz;
     frame.samples_per_channel_ = info.sample_rate_hz / 100;  // 10 ms.
     frame.num_channels_ = info.num_channels;
     frame.Mute();
     packet_sent_ = false;
     last_packet_send_timestamp_ = timestamp_;
     int num_10ms_frames = 0;
     while (!packet_sent_) {
       frame.timestamp_ = timestamp_;
       timestamp_ += rtc::checked_cast<uint32_t>(frame.samples_per_channel_);
       EXPECT_GE(acm_->Add10MsData(frame), 0);
       ++num_10ms_frames;
     }
     return num_10ms_frames;
   }

   template <size_t N>
   void AddSetOfCodecs(rtc::ArrayView<SdpAudioFormat> formats) {
     static int payload_type = 0;
     for (const auto& format : formats) {
       EXPECT_TRUE(receiver_->AddCodec(payload_type++, format));
     }
   }

   int SendData(FrameType frame_type,
                uint8_t payload_type,
                uint32_t timestamp,
                const uint8_t* payload_data,
                size_t payload_len_bytes,
                const RTPFragmentationHeader* fragmentation) override {
     if (frame_type == kEmptyFrame)
       return 0;

     rtp_header_.header.payloadType = payload_type;
     rtp_header_.frameType = frame_type;
     rtp_header_.header.timestamp = timestamp;

     int ret_val = receiver_->InsertPacket(
         rtp_header_,
         rtc::ArrayView<const uint8_t>(payload_data, payload_len_bytes));
     if (ret_val < 0) {
       assert(false);
       return -1;
     }
     rtp_header_.header.sequenceNumber++;
     packet_sent_ = true;
     last_frame_type_ = frame_type;
     return 0;
   }

   const rtc::scoped_refptr<AudioEncoderFactory> encoder_factory_ =
       CreateBuiltinAudioEncoderFactory();
   const rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_ =
       CreateBuiltinAudioDecoderFactory();
   AudioCodingModule::Config config_;
   std::unique_ptr<AcmReceiver> receiver_;
   std::unique_ptr<AudioCodingModule> acm_;
   WebRtcRTPHeader rtp_header_;
   uint32_t timestamp_;
   bool packet_sent_;  // Set when SendData is called reset when inserting audio.
   uint32_t last_packet_send_timestamp_;
   FrameType last_frame_type_;
 };

 #if defined(WEBRTC_ANDROID)
 #define MAYBE_AddCodecGetCodec DISABLED_AddCodecGetCodec
 #else
 #define MAYBE_AddCodecGetCodec AddCodecGetCodec
 #endif
 TEST_F(AcmReceiverTestOldApi, MAYBE_AddCodecGetCodec) {
   const std::vector<AudioCodecSpec> codecs =
       decoder_factory_->GetSupportedDecoders();

   // Add codec.
   for (size_t n = 0; n < codecs.size(); ++n) {
     if (n & 0x1) {  // Just add codecs with odd index.
       const int payload_type = rtc::checked_cast<int>(n);
       EXPECT_TRUE(receiver_->AddCodec(payload_type, codecs[n].format));
     }
   }
   // Get codec and compare.
   for (size_t n = 0; n < codecs.size(); ++n) {
     const int payload_type = rtc::checked_cast<int>(n);
     if (n & 0x1) {
       // Codecs with odd index should match the reference.
       EXPECT_EQ(absl::make_optional(codecs[n].format),
                 receiver_->DecoderByPayloadType(payload_type));
     } else {
       // Codecs with even index are not registered.
       EXPECT_EQ(absl::nullopt, receiver_->DecoderByPayloadType(payload_type));
     }
   }
 }

 #if defined(WEBRTC_ANDROID)
 #define MAYBE_AddCodecChangePayloadType DISABLED_AddCodecChangePayloadType
 #else
 #define MAYBE_AddCodecChangePayloadType AddCodecChangePayloadType
 #endif
 TEST_F(AcmReceiverTestOldApi, MAYBE_AddCodecChangePayloadType) {
   const SdpAudioFormat format("giraffe", 8000, 1);

   // Register the same codec with different payloads.
   EXPECT_EQ(true, receiver_->AddCodec(17, format));
   EXPECT_EQ(true, receiver_->AddCodec(18, format));

   // Both payload types should exist.
   EXPECT_EQ(absl::make_optional(format), receiver_->DecoderByPayloadType(17));
   EXPECT_EQ(absl::make_optional(format), receiver_->DecoderByPayloadType(18));
 }

 #if defined(WEBRTC_ANDROID)
 #define MAYBE_AddCodecChangeCodecId DISABLED_AddCodecChangeCodecId
 #else
 #define MAYBE_AddCodecChangeCodecId AddCodecChangeCodecId
 #endif
 TEST_F(AcmReceiverTestOldApi, AddCodecChangeCodecId) {
   const SdpAudioFormat format1("giraffe", 8000, 1);
   const SdpAudioFormat format2("gnu", 16000, 1);

   // Register the same payload type with different codec ID.
   EXPECT_EQ(true, receiver_->AddCodec(17, format1));
   EXPECT_EQ(true, receiver_->AddCodec(17, format2));

   // Make sure that the last codec is used.
   EXPECT_EQ(absl::make_optional(format2), receiver_->DecoderByPayloadType(17));
 }

 #if defined(WEBRTC_ANDROID)
 #define MAYBE_AddCodecRemoveCodec DISABLED_AddCodecRemoveCodec
 #else
 #define MAYBE_AddCodecRemoveCodec AddCodecRemoveCodec
 #endif
 TEST_F(AcmReceiverTestOldApi, MAYBE_AddCodecRemoveCodec) {
   EXPECT_EQ(true, receiver_->AddCodec(17, SdpAudioFormat("giraffe", 8000, 1)));

   // Remove non-existing codec should not fail. ACM1 legacy.
   EXPECT_EQ(0, receiver_->RemoveCodec(18));

   // Remove an existing codec.
   EXPECT_EQ(0, receiver_->RemoveCodec(17));

   // Ask for the removed codec, must fail.
   EXPECT_EQ(absl::nullopt, receiver_->DecoderByPayloadType(17));
 }

 #if defined(WEBRTC_ANDROID)
 #define MAYBE_SampleRate DISABLED_SampleRate
 #else
 #define MAYBE_SampleRate SampleRate
 #endif
 TEST_F(AcmReceiverTestOldApi, MAYBE_SampleRate) {
   const std::vector<SdpAudioFormat> codecs = {{"ISAC", 16000, 1},
                                               {"ISAC", 32000, 1}};
   for (size_t i = 0; i < codecs.size(); ++i) {
     const int payload_type = rtc::checked_cast<int>(i);
     EXPECT_EQ(true, receiver_->AddCodec(payload_type, codecs[i]));
   }

   constexpr int kOutSampleRateHz = 8000;  // Different than codec sample rate.
   for (size_t i = 0; i < codecs.size(); ++i) {
     const int payload_type = rtc::checked_cast<int>(i);
     const int num_10ms_frames =
         InsertOnePacketOfSilence(SetEncoder(payload_type, codecs[i]));
     for (int k = 0; k < num_10ms_frames; ++k) {
       AudioFrame frame;
       bool muted;
       EXPECT_EQ(0, receiver_->GetAudio(kOutSampleRateHz, &frame, &muted));
     }
     EXPECT_EQ(encoder_factory_->QueryAudioEncoder(codecs[i])->sample_rate_hz,
               receiver_->last_output_sample_rate_hz());
   }
 }

 class AcmReceiverTestFaxModeOldApi : public AcmReceiverTestOldApi {
  protected:
   AcmReceiverTestFaxModeOldApi() {
     config_.neteq_config.for_test_no_time_stretching = true;
   }

   void RunVerifyAudioFrame(const SdpAudioFormat& codec) {
     // Make sure "fax mode" is enabled. This will avoid delay changes unless the
     // packet-loss concealment is made. We do this in order to make the
     // timestamp increments predictable; in normal mode, NetEq may decide to do
     // accelerate or pre-emptive expand operations after some time, offsetting
     // the timestamp.
     EXPECT_TRUE(config_.neteq_config.for_test_no_time_stretching);

     constexpr int payload_type = 17;
     EXPECT_TRUE(receiver_->AddCodec(payload_type, codec));

     const AudioCodecInfo info = SetEncoder(payload_type, codec);
     const int output_sample_rate_hz = info.sample_rate_hz;
     const size_t output_channels = info.num_channels;
     const size_t samples_per_ms = rtc::checked_cast<size_t>(
         rtc::CheckedDivExact(output_sample_rate_hz, 1000));
     const AudioFrame::VADActivity expected_vad_activity =
         output_sample_rate_hz > 16000 ? AudioFrame::kVadActive
                                       : AudioFrame::kVadPassive;

     // Expect the first output timestamp to be 5*fs/8000 samples before the
     // first inserted timestamp (because of NetEq's look-ahead). (This value is
     // defined in Expand::overlap_length_.)
     uint32_t expected_output_ts =
         last_packet_send_timestamp_ -
         rtc::CheckedDivExact(5 * output_sample_rate_hz, 8000);

     AudioFrame frame;
     bool muted;
     EXPECT_EQ(0, receiver_->GetAudio(output_sample_rate_hz, &frame, &muted));
     // Expect timestamp = 0 before first packet is inserted.
     EXPECT_EQ(0u, frame.timestamp_);
     for (int i = 0; i < 5; ++i) {
       const int num_10ms_frames = InsertOnePacketOfSilence(info);
       for (int k = 0; k < num_10ms_frames; ++k) {
         EXPECT_EQ(0,
                   receiver_->GetAudio(output_sample_rate_hz, &frame, &muted));
         EXPECT_EQ(expected_output_ts, frame.timestamp_);
         expected_output_ts += rtc::checked_cast<uint32_t>(10 * samples_per_ms);
         EXPECT_EQ(10 * samples_per_ms, frame.samples_per_channel_);
         EXPECT_EQ(output_sample_rate_hz, frame.sample_rate_hz_);
         EXPECT_EQ(output_channels, frame.num_channels_);
         EXPECT_EQ(AudioFrame::kNormalSpeech, frame.speech_type_);
         EXPECT_EQ(expected_vad_activity, frame.vad_activity_);
         EXPECT_FALSE(muted);
       }
     }
   }
 };

 #if defined(WEBRTC_ANDROID)
 #define MAYBE_VerifyAudioFramePCMU DISABLED_VerifyAudioFramePCMU
 #else
 #define MAYBE_VerifyAudioFramePCMU VerifyAudioFramePCMU
 #endif
 TEST_F(AcmReceiverTestFaxModeOldApi, MAYBE_VerifyAudioFramePCMU) {
   RunVerifyAudioFrame({"PCMU", 8000, 1});
 }

 #if defined(WEBRTC_ANDROID)
 #define MAYBE_VerifyAudioFrameISAC DISABLED_VerifyAudioFrameISAC
 #else
 #define MAYBE_VerifyAudioFrameISAC VerifyAudioFrameISAC
 #endif
 TEST_F(AcmReceiverTestFaxModeOldApi, MAYBE_VerifyAudioFrameISAC) {
   RunVerifyAudioFrame({"ISAC", 16000, 1});
 }

 #if defined(WEBRTC_ANDROID)
 #define MAYBE_VerifyAudioFrameOpus DISABLED_VerifyAudioFrameOpus
 #else
 #define MAYBE_VerifyAudioFrameOpus VerifyAudioFrameOpus
 #endif
 TEST_F(AcmReceiverTestFaxModeOldApi, MAYBE_VerifyAudioFrameOpus) {
   RunVerifyAudioFrame({"opus", 48000, 2});
 }

 #if defined(WEBRTC_ANDROID)
 #define MAYBE_PostdecodingVad DISABLED_PostdecodingVad
 #else
 #define MAYBE_PostdecodingVad PostdecodingVad
 #endif
 TEST_F(AcmReceiverTestOldApi, MAYBE_PostdecodingVad) {
   EXPECT_TRUE(config_.neteq_config.enable_post_decode_vad);
   constexpr int payload_type = 34;
   const SdpAudioFormat codec = {"L16", 16000, 1};
   const AudioCodecInfo info = SetEncoder(payload_type, codec);
   EXPECT_TRUE(receiver_->AddCodec(payload_type, codec));
   constexpr int kNumPackets = 5;
   AudioFrame frame;
   for (int n = 0; n < kNumPackets; ++n) {
     const int num_10ms_frames = InsertOnePacketOfSilence(info);
     for (int k = 0; k < num_10ms_frames; ++k) {
       bool muted;
       ASSERT_EQ(0, receiver_->GetAudio(info.sample_rate_hz, &frame, &muted));
     }
   }
   EXPECT_EQ(AudioFrame::kVadPassive, frame.vad_activity_);
 }

 class AcmReceiverTestPostDecodeVadPassiveOldApi : public AcmReceiverTestOldApi {
  protected:
   AcmReceiverTestPostDecodeVadPassiveOldApi() {
     config_.neteq_config.enable_post_decode_vad = false;
   }
 };

 #if defined(WEBRTC_ANDROID)
 #define MAYBE_PostdecodingVad DISABLED_PostdecodingVad
 #else
 #define MAYBE_PostdecodingVad PostdecodingVad
 #endif
 TEST_F(AcmReceiverTestPostDecodeVadPassiveOldApi, MAYBE_PostdecodingVad) {
   EXPECT_FALSE(config_.neteq_config.enable_post_decode_vad);
   constexpr int payload_type = 34;
   const SdpAudioFormat codec = {"L16", 16000, 1};
   const AudioCodecInfo info = SetEncoder(payload_type, codec);
   encoder_factory_->QueryAudioEncoder(codec).value();
   EXPECT_TRUE(receiver_->AddCodec(payload_type, codec));
   const int kNumPackets = 5;
   AudioFrame frame;
   for (int n = 0; n < kNumPackets; ++n) {
     const int num_10ms_frames = InsertOnePacketOfSilence(info);
     for (int k = 0; k < num_10ms_frames; ++k) {
       bool muted;
       ASSERT_EQ(0, receiver_->GetAudio(info.sample_rate_hz, &frame, &muted));
     }
   }
   EXPECT_EQ(AudioFrame::kVadUnknown, frame.vad_activity_);
 }

 #if defined(WEBRTC_ANDROID)
 #define MAYBE_LastAudioCodec DISABLED_LastAudioCodec
 #else
 #define MAYBE_LastAudioCodec LastAudioCodec
 #endif
 #if defined(WEBRTC_CODEC_ISAC)
 TEST_F(AcmReceiverTestOldApi, MAYBE_LastAudioCodec) {
   const std::vector<SdpAudioFormat> codecs = {{"ISAC", 16000, 1},
                                               {"PCMA", 8000, 1},
                                               {"ISAC", 32000, 1},
                                               {"L16", 32000, 1}};
   for (size_t i = 0; i < codecs.size(); ++i) {
     const int payload_type = rtc::checked_cast<int>(i);
     EXPECT_TRUE(receiver_->AddCodec(payload_type, codecs[i]));
   }

   const std::map<int, int> cng_payload_types = {
       {8000, 100}, {16000, 101}, {32000, 102}};
   for (const auto& x : cng_payload_types) {
     const int sample_rate_hz = x.first;
     const int payload_type = x.second;
     EXPECT_TRUE(receiver_->AddCodec(payload_type, {"CN", sample_rate_hz, 1}));
   }

   // No audio payload is received.
   EXPECT_EQ(absl::nullopt, receiver_->LastAudioFormat());

   // Start with sending DTX.
   packet_sent_ = false;
   InsertOnePacketOfSilence(
       SetEncoder(0, codecs[0], cng_payload_types));  // Enough to test
                                                      // with one codec.
   ASSERT_TRUE(packet_sent_);
   EXPECT_EQ(kAudioFrameCN, last_frame_type_);

   // Has received, only, DTX. Last Audio codec is undefined.
   EXPECT_EQ(absl::nullopt, receiver_->LastAudioFormat());
   EXPECT_FALSE(receiver_->last_packet_sample_rate_hz());

   for (size_t i = 0; i < codecs.size(); ++i) {
     // Set DTX off to send audio payload.
     packet_sent_ = false;
     const int payload_type = rtc::checked_cast<int>(i);
     const AudioCodecInfo info_without_cng = SetEncoder(payload_type, codecs[i]);
     InsertOnePacketOfSilence(info_without_cng);

     // Sanity check if Actually an audio payload received, and it should be
     // of type "speech."
     ASSERT_TRUE(packet_sent_);
     ASSERT_EQ(kAudioFrameSpeech, last_frame_type_);
     EXPECT_EQ(info_without_cng.sample_rate_hz,
               receiver_->last_packet_sample_rate_hz());

     // Set VAD on to send DTX. Then check if the "Last Audio codec" returns
     // the expected codec. Encode repeatedly until a DTX is sent.
     const AudioCodecInfo info_with_cng =
         SetEncoder(payload_type, codecs[i], cng_payload_types);
     while (last_frame_type_ != kAudioFrameCN) {
       packet_sent_ = false;
       InsertOnePacketOfSilence(info_with_cng);
       ASSERT_TRUE(packet_sent_);
     }
     EXPECT_EQ(info_with_cng.sample_rate_hz,
               receiver_->last_packet_sample_rate_hz());
     EXPECT_EQ(codecs[i], receiver_->LastAudioFormat());
   }
 }
 #endif

 }  // namespace acm2

 }  // 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/acm2/acm_receiver.h"

	#include <algorithm> // std::min
	#include <memory>

	#include "api/audio_codecs/builtin_audio_decoder_factory.h"
	#include "api/audio_codecs/builtin_audio_encoder_factory.h"
	#include "modules/audio_coding/acm2/rent_a_codec.h"
	#include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
	#include "modules/audio_coding/include/audio_coding_module.h"
	#include "modules/audio_coding/neteq/tools/rtp_generator.h"
	#include "modules/include/module_common_types.h"
	#include "rtc_base/checks.h"
	#include "rtc_base/numerics/safe_conversions.h"
	#include "system_wrappers/include/clock.h"
	#include "test/gtest.h"
	#include "test/testsupport/fileutils.h"

	namespace webrtc {

	namespace acm2 {

	class AcmReceiverTestOldApi : public AudioPacketizationCallback,
	public ::testing::Test {
	protected:
	AcmReceiverTestOldApi()
	: timestamp_(0),
	packet_sent_(false),
	last_packet_send_timestamp_(timestamp_),
	last_frame_type_(kEmptyFrame) {
	config_.decoder_factory = decoder_factory_;
	}

	~AcmReceiverTestOldApi() {}

	void SetUp() override {
	acm_.reset(AudioCodingModule::Create(config_));
	receiver_.reset(new AcmReceiver(config_));
	ASSERT_TRUE(receiver_.get() != NULL);
	ASSERT_TRUE(acm_.get() != NULL);
	acm_->InitializeReceiver();
	acm_->RegisterTransportCallback(this);

	rtp_header_.header.sequenceNumber = 0;
	rtp_header_.header.timestamp = 0;
	rtp_header_.header.markerBit = false;
	rtp_header_.header.ssrc = 0x12345678; // Arbitrary.
	rtp_header_.header.numCSRCs = 0;
	rtp_header_.header.payloadType = 0;
	rtp_header_.frameType = kAudioFrameSpeech;
	}

	void TearDown() override {}

	AudioCodecInfo SetEncoder(int payload_type,
	const SdpAudioFormat& format,
	const std::map<int, int> cng_payload_types = {}) {
	// Create the speech encoder.
	AudioCodecInfo info = encoder_factory_->QueryAudioEncoder(format).value();
	std::unique_ptr<AudioEncoder> enc =
	encoder_factory_->MakeAudioEncoder(payload_type, format, absl::nullopt);

	// If we have a compatible CN specification, stack a CNG on top.
	auto it = cng_payload_types.find(info.sample_rate_hz);
	if (it != cng_payload_types.end()) {
	AudioEncoderCngConfig config;
	config.speech_encoder = std::move(enc);
	config.num_channels = 1;
	config.payload_type = it->second;
	config.vad_mode = Vad::kVadNormal;
	enc = CreateComfortNoiseEncoder(std::move(config));
	}

	// Actually start using the new encoder.
	acm_->SetEncoder(std::move(enc));
	return info;
	}

	int InsertOnePacketOfSilence(const AudioCodecInfo& info) {
	// Frame setup according to the codec.
	AudioFrame frame;
	frame.sample_rate_hz_ = info.sample_rate_hz;
	frame.samples_per_channel_ = info.sample_rate_hz / 100; // 10 ms.
	frame.num_channels_ = info.num_channels;
	frame.Mute();
	packet_sent_ = false;
	last_packet_send_timestamp_ = timestamp_;
	int num_10ms_frames = 0;
	while (!packet_sent_) {
	frame.timestamp_ = timestamp_;
	timestamp_ += rtc::checked_cast<uint32_t>(frame.samples_per_channel_);
	EXPECT_GE(acm_->Add10MsData(frame), 0);
	++num_10ms_frames;
	}
	return num_10ms_frames;
	}

	template <size_t N>
	void AddSetOfCodecs(rtc::ArrayView<SdpAudioFormat> formats) {
	static int payload_type = 0;
	for (const auto& format : formats) {
	EXPECT_TRUE(receiver_->AddCodec(payload_type++, format));
	}
	}

	int SendData(FrameType frame_type,
	uint8_t payload_type,
	uint32_t timestamp,
	const uint8_t* payload_data,
	size_t payload_len_bytes,
	const RTPFragmentationHeader* fragmentation) override {
	if (frame_type == kEmptyFrame)
	return 0;

	rtp_header_.header.payloadType = payload_type;
	rtp_header_.frameType = frame_type;
	rtp_header_.header.timestamp = timestamp;

	int ret_val = receiver_->InsertPacket(
	rtp_header_,
	rtc::ArrayView<const uint8_t>(payload_data, payload_len_bytes));
	if (ret_val < 0) {
	assert(false);
	return -1;
	}
	rtp_header_.header.sequenceNumber++;
	packet_sent_ = true;
	last_frame_type_ = frame_type;
	return 0;
	}

	const rtc::scoped_refptr<AudioEncoderFactory> encoder_factory_ =
	CreateBuiltinAudioEncoderFactory();
	const rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_ =
	CreateBuiltinAudioDecoderFactory();
	AudioCodingModule::Config config_;
	std::unique_ptr<AcmReceiver> receiver_;
	std::unique_ptr<AudioCodingModule> acm_;
	WebRtcRTPHeader rtp_header_;
	uint32_t timestamp_;
	bool packet_sent_; // Set when SendData is called reset when inserting audio.
	uint32_t last_packet_send_timestamp_;
	FrameType last_frame_type_;
	};

	#if defined(WEBRTC_ANDROID)
	#define MAYBE_AddCodecGetCodec DISABLED_AddCodecGetCodec
	#else
	#define MAYBE_AddCodecGetCodec AddCodecGetCodec
	#endif
	TEST_F(AcmReceiverTestOldApi, MAYBE_AddCodecGetCodec) {
	const std::vector<AudioCodecSpec> codecs =
	decoder_factory_->GetSupportedDecoders();

	// Add codec.
	for (size_t n = 0; n < codecs.size(); ++n) {
	if (n & 0x1) { // Just add codecs with odd index.
	const int payload_type = rtc::checked_cast<int>(n);
	EXPECT_TRUE(receiver_->AddCodec(payload_type, codecs[n].format));
	}
	}
	// Get codec and compare.
	for (size_t n = 0; n < codecs.size(); ++n) {
	const int payload_type = rtc::checked_cast<int>(n);
	if (n & 0x1) {
	// Codecs with odd index should match the reference.
	EXPECT_EQ(absl::make_optional(codecs[n].format),
	receiver_->DecoderByPayloadType(payload_type));
	} else {
	// Codecs with even index are not registered.
	EXPECT_EQ(absl::nullopt, receiver_->DecoderByPayloadType(payload_type));
	}
	}
	}

	#if defined(WEBRTC_ANDROID)
	#define MAYBE_AddCodecChangePayloadType DISABLED_AddCodecChangePayloadType
	#else
	#define MAYBE_AddCodecChangePayloadType AddCodecChangePayloadType
	#endif
	TEST_F(AcmReceiverTestOldApi, MAYBE_AddCodecChangePayloadType) {
	const SdpAudioFormat format("giraffe", 8000, 1);

	// Register the same codec with different payloads.
	EXPECT_EQ(true, receiver_->AddCodec(17, format));
	EXPECT_EQ(true, receiver_->AddCodec(18, format));

	// Both payload types should exist.
	EXPECT_EQ(absl::make_optional(format), receiver_->DecoderByPayloadType(17));
	EXPECT_EQ(absl::make_optional(format), receiver_->DecoderByPayloadType(18));
	}

	#if defined(WEBRTC_ANDROID)
	#define MAYBE_AddCodecChangeCodecId DISABLED_AddCodecChangeCodecId
	#else
	#define MAYBE_AddCodecChangeCodecId AddCodecChangeCodecId
	#endif
	TEST_F(AcmReceiverTestOldApi, AddCodecChangeCodecId) {
	const SdpAudioFormat format1("giraffe", 8000, 1);
	const SdpAudioFormat format2("gnu", 16000, 1);

	// Register the same payload type with different codec ID.
	EXPECT_EQ(true, receiver_->AddCodec(17, format1));
	EXPECT_EQ(true, receiver_->AddCodec(17, format2));

	// Make sure that the last codec is used.
	EXPECT_EQ(absl::make_optional(format2), receiver_->DecoderByPayloadType(17));
	}

	#if defined(WEBRTC_ANDROID)
	#define MAYBE_AddCodecRemoveCodec DISABLED_AddCodecRemoveCodec
	#else
	#define MAYBE_AddCodecRemoveCodec AddCodecRemoveCodec
	#endif
	TEST_F(AcmReceiverTestOldApi, MAYBE_AddCodecRemoveCodec) {
	EXPECT_EQ(true, receiver_->AddCodec(17, SdpAudioFormat("giraffe", 8000, 1)));

	// Remove non-existing codec should not fail. ACM1 legacy.
	EXPECT_EQ(0, receiver_->RemoveCodec(18));

	// Remove an existing codec.
	EXPECT_EQ(0, receiver_->RemoveCodec(17));

	// Ask for the removed codec, must fail.
	EXPECT_EQ(absl::nullopt, receiver_->DecoderByPayloadType(17));
	}

	#if defined(WEBRTC_ANDROID)
	#define MAYBE_SampleRate DISABLED_SampleRate
	#else
	#define MAYBE_SampleRate SampleRate
	#endif
	TEST_F(AcmReceiverTestOldApi, MAYBE_SampleRate) {
	const std::vector<SdpAudioFormat> codecs = {{"ISAC", 16000, 1},
	{"ISAC", 32000, 1}};
	for (size_t i = 0; i < codecs.size(); ++i) {
	const int payload_type = rtc::checked_cast<int>(i);
	EXPECT_EQ(true, receiver_->AddCodec(payload_type, codecs[i]));
	}

	constexpr int kOutSampleRateHz = 8000; // Different than codec sample rate.
	for (size_t i = 0; i < codecs.size(); ++i) {
	const int payload_type = rtc::checked_cast<int>(i);
	const int num_10ms_frames =
	InsertOnePacketOfSilence(SetEncoder(payload_type, codecs[i]));
	for (int k = 0; k < num_10ms_frames; ++k) {
	AudioFrame frame;
	bool muted;
	EXPECT_EQ(0, receiver_->GetAudio(kOutSampleRateHz, &frame, &muted));
	}
	EXPECT_EQ(encoder_factory_->QueryAudioEncoder(codecs[i])->sample_rate_hz,
	receiver_->last_output_sample_rate_hz());
	}
	}

	class AcmReceiverTestFaxModeOldApi : public AcmReceiverTestOldApi {
	protected:
	AcmReceiverTestFaxModeOldApi() {
	config_.neteq_config.for_test_no_time_stretching = true;
	}

	void RunVerifyAudioFrame(const SdpAudioFormat& codec) {
	// Make sure "fax mode" is enabled. This will avoid delay changes unless the
	// packet-loss concealment is made. We do this in order to make the
	// timestamp increments predictable; in normal mode, NetEq may decide to do
	// accelerate or pre-emptive expand operations after some time, offsetting
	// the timestamp.
	EXPECT_TRUE(config_.neteq_config.for_test_no_time_stretching);

	constexpr int payload_type = 17;
	EXPECT_TRUE(receiver_->AddCodec(payload_type, codec));

	const AudioCodecInfo info = SetEncoder(payload_type, codec);
	const int output_sample_rate_hz = info.sample_rate_hz;
	const size_t output_channels = info.num_channels;
	const size_t samples_per_ms = rtc::checked_cast<size_t>(
	rtc::CheckedDivExact(output_sample_rate_hz, 1000));
	const AudioFrame::VADActivity expected_vad_activity =
	output_sample_rate_hz > 16000 ? AudioFrame::kVadActive
	: AudioFrame::kVadPassive;

	// Expect the first output timestamp to be 5*fs/8000 samples before the
	// first inserted timestamp (because of NetEq's look-ahead). (This value is
	// defined in Expand::overlap_length_.)
	uint32_t expected_output_ts =
	last_packet_send_timestamp_ -
	rtc::CheckedDivExact(5 * output_sample_rate_hz, 8000);

	AudioFrame frame;
	bool muted;
	EXPECT_EQ(0, receiver_->GetAudio(output_sample_rate_hz, &frame, &muted));
	// Expect timestamp = 0 before first packet is inserted.
	EXPECT_EQ(0u, frame.timestamp_);
	for (int i = 0; i < 5; ++i) {
	const int num_10ms_frames = InsertOnePacketOfSilence(info);
	for (int k = 0; k < num_10ms_frames; ++k) {
	EXPECT_EQ(0,
	receiver_->GetAudio(output_sample_rate_hz, &frame, &muted));
	EXPECT_EQ(expected_output_ts, frame.timestamp_);
	expected_output_ts += rtc::checked_cast<uint32_t>(10 * samples_per_ms);
	EXPECT_EQ(10 * samples_per_ms, frame.samples_per_channel_);
	EXPECT_EQ(output_sample_rate_hz, frame.sample_rate_hz_);
	EXPECT_EQ(output_channels, frame.num_channels_);
	EXPECT_EQ(AudioFrame::kNormalSpeech, frame.speech_type_);
	EXPECT_EQ(expected_vad_activity, frame.vad_activity_);
	EXPECT_FALSE(muted);
	}
	}
	}
	};

	#if defined(WEBRTC_ANDROID)
	#define MAYBE_VerifyAudioFramePCMU DISABLED_VerifyAudioFramePCMU
	#else
	#define MAYBE_VerifyAudioFramePCMU VerifyAudioFramePCMU
	#endif
	TEST_F(AcmReceiverTestFaxModeOldApi, MAYBE_VerifyAudioFramePCMU) {
	RunVerifyAudioFrame({"PCMU", 8000, 1});
	}

	#if defined(WEBRTC_ANDROID)
	#define MAYBE_VerifyAudioFrameISAC DISABLED_VerifyAudioFrameISAC
	#else
	#define MAYBE_VerifyAudioFrameISAC VerifyAudioFrameISAC
	#endif
	TEST_F(AcmReceiverTestFaxModeOldApi, MAYBE_VerifyAudioFrameISAC) {
	RunVerifyAudioFrame({"ISAC", 16000, 1});
	}

	#if defined(WEBRTC_ANDROID)
	#define MAYBE_VerifyAudioFrameOpus DISABLED_VerifyAudioFrameOpus
	#else
	#define MAYBE_VerifyAudioFrameOpus VerifyAudioFrameOpus
	#endif
	TEST_F(AcmReceiverTestFaxModeOldApi, MAYBE_VerifyAudioFrameOpus) {
	RunVerifyAudioFrame({"opus", 48000, 2});
	}

	#if defined(WEBRTC_ANDROID)
	#define MAYBE_PostdecodingVad DISABLED_PostdecodingVad
	#else
	#define MAYBE_PostdecodingVad PostdecodingVad
	#endif
	TEST_F(AcmReceiverTestOldApi, MAYBE_PostdecodingVad) {
	EXPECT_TRUE(config_.neteq_config.enable_post_decode_vad);
	constexpr int payload_type = 34;
	const SdpAudioFormat codec = {"L16", 16000, 1};
	const AudioCodecInfo info = SetEncoder(payload_type, codec);
	EXPECT_TRUE(receiver_->AddCodec(payload_type, codec));
	constexpr int kNumPackets = 5;
	AudioFrame frame;
	for (int n = 0; n < kNumPackets; ++n) {
	const int num_10ms_frames = InsertOnePacketOfSilence(info);
	for (int k = 0; k < num_10ms_frames; ++k) {
	bool muted;
	ASSERT_EQ(0, receiver_->GetAudio(info.sample_rate_hz, &frame, &muted));
	}
	}
	EXPECT_EQ(AudioFrame::kVadPassive, frame.vad_activity_);
	}

	class AcmReceiverTestPostDecodeVadPassiveOldApi : public AcmReceiverTestOldApi {
	protected:
	AcmReceiverTestPostDecodeVadPassiveOldApi() {
	config_.neteq_config.enable_post_decode_vad = false;
	}
	};

	#if defined(WEBRTC_ANDROID)
	#define MAYBE_PostdecodingVad DISABLED_PostdecodingVad
	#else
	#define MAYBE_PostdecodingVad PostdecodingVad
	#endif
	TEST_F(AcmReceiverTestPostDecodeVadPassiveOldApi, MAYBE_PostdecodingVad) {
	EXPECT_FALSE(config_.neteq_config.enable_post_decode_vad);
	constexpr int payload_type = 34;
	const SdpAudioFormat codec = {"L16", 16000, 1};
	const AudioCodecInfo info = SetEncoder(payload_type, codec);
	encoder_factory_->QueryAudioEncoder(codec).value();
	EXPECT_TRUE(receiver_->AddCodec(payload_type, codec));
	const int kNumPackets = 5;
	AudioFrame frame;
	for (int n = 0; n < kNumPackets; ++n) {
	const int num_10ms_frames = InsertOnePacketOfSilence(info);
	for (int k = 0; k < num_10ms_frames; ++k) {
	bool muted;
	ASSERT_EQ(0, receiver_->GetAudio(info.sample_rate_hz, &frame, &muted));
	}
	}
	EXPECT_EQ(AudioFrame::kVadUnknown, frame.vad_activity_);
	}

	#if defined(WEBRTC_ANDROID)
	#define MAYBE_LastAudioCodec DISABLED_LastAudioCodec
	#else
	#define MAYBE_LastAudioCodec LastAudioCodec
	#endif
	#if defined(WEBRTC_CODEC_ISAC)
	TEST_F(AcmReceiverTestOldApi, MAYBE_LastAudioCodec) {
	const std::vector<SdpAudioFormat> codecs = {{"ISAC", 16000, 1},
	{"PCMA", 8000, 1},
	{"ISAC", 32000, 1},
	{"L16", 32000, 1}};
	for (size_t i = 0; i < codecs.size(); ++i) {
	const int payload_type = rtc::checked_cast<int>(i);
	EXPECT_TRUE(receiver_->AddCodec(payload_type, codecs[i]));
	}

	const std::map<int, int> cng_payload_types = {
	{8000, 100}, {16000, 101}, {32000, 102}};
	for (const auto& x : cng_payload_types) {
	const int sample_rate_hz = x.first;
	const int payload_type = x.second;
	EXPECT_TRUE(receiver_->AddCodec(payload_type, {"CN", sample_rate_hz, 1}));
	}

	// No audio payload is received.
	EXPECT_EQ(absl::nullopt, receiver_->LastAudioFormat());

	// Start with sending DTX.
	packet_sent_ = false;
	InsertOnePacketOfSilence(
	SetEncoder(0, codecs[0], cng_payload_types)); // Enough to test
	// with one codec.
	ASSERT_TRUE(packet_sent_);
	EXPECT_EQ(kAudioFrameCN, last_frame_type_);

	// Has received, only, DTX. Last Audio codec is undefined.
	EXPECT_EQ(absl::nullopt, receiver_->LastAudioFormat());
	EXPECT_FALSE(receiver_->last_packet_sample_rate_hz());

	for (size_t i = 0; i < codecs.size(); ++i) {
	// Set DTX off to send audio payload.
	packet_sent_ = false;
	const int payload_type = rtc::checked_cast<int>(i);
	const AudioCodecInfo info_without_cng = SetEncoder(payload_type, codecs[i]);
	InsertOnePacketOfSilence(info_without_cng);

	// Sanity check if Actually an audio payload received, and it should be
	// of type "speech."
	ASSERT_TRUE(packet_sent_);
	ASSERT_EQ(kAudioFrameSpeech, last_frame_type_);
	EXPECT_EQ(info_without_cng.sample_rate_hz,
	receiver_->last_packet_sample_rate_hz());

	// Set VAD on to send DTX. Then check if the "Last Audio codec" returns
	// the expected codec. Encode repeatedly until a DTX is sent.
	const AudioCodecInfo info_with_cng =
	SetEncoder(payload_type, codecs[i], cng_payload_types);
	while (last_frame_type_ != kAudioFrameCN) {
	packet_sent_ = false;
	InsertOnePacketOfSilence(info_with_cng);
	ASSERT_TRUE(packet_sent_);
	}
	EXPECT_EQ(info_with_cng.sample_rate_hz,
	receiver_->last_packet_sample_rate_hz());
	EXPECT_EQ(codecs[i], receiver_->LastAudioFormat());
	}
	}
	#endif

	} // namespace acm2

	} // namespace webrtc