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webrtc/modules/audio_coding/acm2/audio_coding_module_unittest.cc
Lionel Koenig a656b9d781 Use absolute capture timestamp from the beginning of payload 
This ensure the absolute capture timestamp from the first audio sample
encoded in the payload is used for the corresponding rtp header.

Bug: webrtc:42226041
Change-Id: Ib8f2e3a5df5c82c5806171bd5b36a26d92fbea72
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349265
Commit-Queue: Lionel Koenig <lionelk@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42281}
2024-05-13 08:10:56 +00:00

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/*
* 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 "modules/audio_coding/include/audio_coding_module.h"
#include <stdio.h>
#include <string.h>
#include <atomic>
#include <memory>
#include <vector>
#include "absl/strings/string_view.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/audio_codecs/opus/audio_decoder_multi_channel_opus.h"
#include "api/audio_codecs/opus/audio_decoder_opus.h"
#include "api/audio_codecs/opus/audio_encoder_multi_channel_opus.h"
#include "api/audio_codecs/opus/audio_encoder_opus.h"
#include "modules/audio_coding/acm2/acm_receive_test.h"
#include "modules/audio_coding/acm2/acm_send_test.h"
#include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
#include "modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
#include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
#include "modules/audio_coding/neteq/tools/audio_checksum.h"
#include "modules/audio_coding/neteq/tools/audio_loop.h"
#include "modules/audio_coding/neteq/tools/constant_pcm_packet_source.h"
#include "modules/audio_coding/neteq/tools/input_audio_file.h"
#include "modules/audio_coding/neteq/tools/output_audio_file.h"
#include "modules/audio_coding/neteq/tools/output_wav_file.h"
#include "modules/audio_coding/neteq/tools/packet.h"
#include "modules/audio_coding/neteq/tools/rtp_file_source.h"
#include "rtc_base/event.h"
#include "rtc_base/message_digest.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/platform_thread.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/system/arch.h"
#include "rtc_base/thread_annotations.h"
#include "system_wrappers/include/clock.h"
#include "system_wrappers/include/cpu_features_wrapper.h"
#include "system_wrappers/include/sleep.h"
#include "test/audio_decoder_proxy_factory.h"
#include "test/gtest.h"
#include "test/mock_audio_decoder.h"
#include "test/mock_audio_encoder.h"
#include "test/testsupport/file_utils.h"
#include "test/testsupport/rtc_expect_death.h"
using ::testing::_;
using ::testing::AtLeast;
using ::testing::Invoke;
namespace webrtc {
namespace {
const int kSampleRateHz = 16000;
const int kNumSamples10ms = kSampleRateHz / 100;
const int kFrameSizeMs = 10; // Multiple of 10.
const int kFrameSizeSamples = kFrameSizeMs / 10 * kNumSamples10ms;
const int kPayloadSizeBytes = kFrameSizeSamples * sizeof(int16_t);
const uint8_t kPayloadType = 111;
} // namespace
class RtpData {
public:
RtpData(int samples_per_packet, uint8_t payload_type)
: samples_per_packet_(samples_per_packet), payload_type_(payload_type) {}
virtual ~RtpData() {}
void Populate(RTPHeader* rtp_header) {
rtp_header->sequenceNumber = 0xABCD;
rtp_header->timestamp = 0xABCDEF01;
rtp_header->payloadType = payload_type_;
rtp_header->markerBit = false;
rtp_header->ssrc = 0x1234;
rtp_header->numCSRCs = 0;
}
void Forward(RTPHeader* rtp_header) {
++rtp_header->sequenceNumber;
rtp_header->timestamp += samples_per_packet_;
}
private:
int samples_per_packet_;
uint8_t payload_type_;
};
class PacketizationCallbackStubOldApi : public AudioPacketizationCallback {
public:
PacketizationCallbackStubOldApi()
: num_calls_(0),
last_frame_type_(AudioFrameType::kEmptyFrame),
last_payload_type_(-1),
last_timestamp_(0) {}
int32_t SendData(AudioFrameType frame_type,
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
size_t payload_len_bytes,
int64_t absolute_capture_timestamp_ms) override {
MutexLock lock(&mutex_);
++num_calls_;
last_frame_type_ = frame_type;
last_payload_type_ = payload_type;
last_timestamp_ = timestamp;
last_payload_vec_.assign(payload_data, payload_data + payload_len_bytes);
return 0;
}
int num_calls() const {
MutexLock lock(&mutex_);
return num_calls_;
}
int last_payload_len_bytes() const {
MutexLock lock(&mutex_);
return rtc::checked_cast<int>(last_payload_vec_.size());
}
AudioFrameType last_frame_type() const {
MutexLock lock(&mutex_);
return last_frame_type_;
}
int last_payload_type() const {
MutexLock lock(&mutex_);
return last_payload_type_;
}
uint32_t last_timestamp() const {
MutexLock lock(&mutex_);
return last_timestamp_;
}
void SwapBuffers(std::vector<uint8_t>* payload) {
MutexLock lock(&mutex_);
last_payload_vec_.swap(*payload);
}
private:
int num_calls_ RTC_GUARDED_BY(mutex_);
AudioFrameType last_frame_type_ RTC_GUARDED_BY(mutex_);
int last_payload_type_ RTC_GUARDED_BY(mutex_);
uint32_t last_timestamp_ RTC_GUARDED_BY(mutex_);
std::vector<uint8_t> last_payload_vec_ RTC_GUARDED_BY(mutex_);
mutable Mutex mutex_;
};
class AudioCodingModuleTestOldApi : public ::testing::Test {
protected:
AudioCodingModuleTestOldApi()
: rtp_utility_(new RtpData(kFrameSizeSamples, kPayloadType)),
clock_(Clock::GetRealTimeClock()) {}
~AudioCodingModuleTestOldApi() {}
void TearDown() {}
void SetUp() {
acm_ = AudioCodingModule::Create();
acm2::AcmReceiver::Config config;
config.clock = *clock_;
config.decoder_factory = CreateBuiltinAudioDecoderFactory();
acm_receiver_ = std::make_unique<acm2::AcmReceiver>(config);
rtp_utility_->Populate(&rtp_header_);
input_frame_.sample_rate_hz_ = kSampleRateHz;
input_frame_.num_channels_ = 1;
input_frame_.samples_per_channel_ = kSampleRateHz * 10 / 1000; // 10 ms.
static_assert(kSampleRateHz * 10 / 1000 <= AudioFrame::kMaxDataSizeSamples,
"audio frame too small");
input_frame_.Mute();
ASSERT_EQ(0, acm_->RegisterTransportCallback(&packet_cb_));
SetUpL16Codec();
}
// Set up L16 codec.
virtual void SetUpL16Codec() {
audio_format_ = SdpAudioFormat("L16", kSampleRateHz, 1);
pac_size_ = 160;
}
virtual void RegisterCodec() {
acm_receiver_->SetCodecs({{kPayloadType, *audio_format_}});
acm_->SetEncoder(CreateBuiltinAudioEncoderFactory()->MakeAudioEncoder(
kPayloadType, *audio_format_, absl::nullopt));
}
virtual void InsertPacketAndPullAudio() {
InsertPacket();
PullAudio();
}
virtual void InsertPacket() {
const uint8_t kPayload[kPayloadSizeBytes] = {0};
ASSERT_EQ(0, acm_receiver_->InsertPacket(rtp_header_,
rtc::ArrayView<const uint8_t>(
kPayload, kPayloadSizeBytes)));
rtp_utility_->Forward(&rtp_header_);
}
virtual void PullAudio() {
AudioFrame audio_frame;
bool muted;
ASSERT_EQ(0, acm_receiver_->GetAudio(-1, &audio_frame, &muted));
ASSERT_FALSE(muted);
}
virtual void InsertAudio() {
ASSERT_GE(acm_->Add10MsData(input_frame_), 0);
input_frame_.timestamp_ += kNumSamples10ms;
}
virtual void VerifyEncoding() {
int last_length = packet_cb_.last_payload_len_bytes();
EXPECT_TRUE(last_length == 2 * pac_size_ || last_length == 0)
<< "Last encoded packet was " << last_length << " bytes.";
}
virtual void InsertAudioAndVerifyEncoding() {
InsertAudio();
VerifyEncoding();
}
std::unique_ptr<RtpData> rtp_utility_;
std::unique_ptr<AudioCodingModule> acm_;
std::unique_ptr<acm2::AcmReceiver> acm_receiver_;
PacketizationCallbackStubOldApi packet_cb_;
RTPHeader rtp_header_;
AudioFrame input_frame_;
absl::optional<SdpAudioFormat> audio_format_;
int pac_size_ = -1;
Clock* clock_;
};
class AudioCodingModuleTestOldApiDeathTest
: public AudioCodingModuleTestOldApi {};
// The below test is temporarily disabled on Windows due to problems
// with clang debug builds.
// TODO(tommi): Re-enable when we've figured out what the problem is.
// http://crbug.com/615050
#if !defined(WEBRTC_WIN) && defined(__clang__) && RTC_DCHECK_IS_ON && \
GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
TEST_F(AudioCodingModuleTestOldApiDeathTest, FailOnZeroDesiredFrequency) {
AudioFrame audio_frame;
bool muted;
RTC_EXPECT_DEATH(acm_receiver_->GetAudio(0, &audio_frame, &muted),
"dst_sample_rate_hz");
}
#endif
// Checks that the transport callback is invoked once for each speech packet.
// Also checks that the frame type is kAudioFrameSpeech.
TEST_F(AudioCodingModuleTestOldApi, TransportCallbackIsInvokedForEachPacket) {
const int k10MsBlocksPerPacket = 3;
pac_size_ = k10MsBlocksPerPacket * kSampleRateHz / 100;
audio_format_->parameters["ptime"] = "30";
RegisterCodec();
const int kLoops = 10;
for (int i = 0; i < kLoops; ++i) {
EXPECT_EQ(i / k10MsBlocksPerPacket, packet_cb_.num_calls());
if (packet_cb_.num_calls() > 0)
EXPECT_EQ(AudioFrameType::kAudioFrameSpeech,
packet_cb_.last_frame_type());
InsertAudioAndVerifyEncoding();
}
EXPECT_EQ(kLoops / k10MsBlocksPerPacket, packet_cb_.num_calls());
EXPECT_EQ(AudioFrameType::kAudioFrameSpeech, packet_cb_.last_frame_type());
}
// Introduce this class to set different expectations on the number of encoded
// bytes. This class expects all encoded packets to be 9 bytes (matching one
// CNG SID frame) or 0 bytes. This test depends on `input_frame_` containing
// (near-)zero values. It also introduces a way to register comfort noise with
// a custom payload type.
class AudioCodingModuleTestWithComfortNoiseOldApi
: public AudioCodingModuleTestOldApi {
protected:
void RegisterCngCodec(int rtp_payload_type) {
acm_receiver_->SetCodecs({{kPayloadType, *audio_format_},
{rtp_payload_type, {"cn", kSampleRateHz, 1}}});
acm_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* enc) {
AudioEncoderCngConfig config;
config.speech_encoder = std::move(*enc);
config.num_channels = 1;
config.payload_type = rtp_payload_type;
config.vad_mode = Vad::kVadNormal;
*enc = CreateComfortNoiseEncoder(std::move(config));
});
}
void VerifyEncoding() override {
int last_length = packet_cb_.last_payload_len_bytes();
EXPECT_TRUE(last_length == 9 || last_length == 0)
<< "Last encoded packet was " << last_length << " bytes.";
}
void DoTest(int blocks_per_packet, int cng_pt) {
const int kLoops = 40;
// This array defines the expected frame types, and when they should arrive.
// We expect a frame to arrive each time the speech encoder would have
// produced a packet, and once every 100 ms the frame should be non-empty,
// that is contain comfort noise.
const struct {
int ix;
AudioFrameType type;
} expectation[] = {{2, AudioFrameType::kAudioFrameCN},
{5, AudioFrameType::kEmptyFrame},
{8, AudioFrameType::kEmptyFrame},
{11, AudioFrameType::kAudioFrameCN},
{14, AudioFrameType::kEmptyFrame},
{17, AudioFrameType::kEmptyFrame},
{20, AudioFrameType::kAudioFrameCN},
{23, AudioFrameType::kEmptyFrame},
{26, AudioFrameType::kEmptyFrame},
{29, AudioFrameType::kEmptyFrame},
{32, AudioFrameType::kAudioFrameCN},
{35, AudioFrameType::kEmptyFrame},
{38, AudioFrameType::kEmptyFrame}};
for (int i = 0; i < kLoops; ++i) {
int num_calls_before = packet_cb_.num_calls();
EXPECT_EQ(i / blocks_per_packet, num_calls_before);
InsertAudioAndVerifyEncoding();
int num_calls = packet_cb_.num_calls();
if (num_calls == num_calls_before + 1) {
EXPECT_EQ(expectation[num_calls - 1].ix, i);
EXPECT_EQ(expectation[num_calls - 1].type, packet_cb_.last_frame_type())
<< "Wrong frame type for lap " << i;
EXPECT_EQ(cng_pt, packet_cb_.last_payload_type());
} else {
EXPECT_EQ(num_calls, num_calls_before);
}
}
}
};
// Checks that the transport callback is invoked once per frame period of the
// underlying speech encoder, even when comfort noise is produced.
// Also checks that the frame type is kAudioFrameCN or kEmptyFrame.
TEST_F(AudioCodingModuleTestWithComfortNoiseOldApi,
TransportCallbackTestForComfortNoiseRegisterCngLast) {
const int k10MsBlocksPerPacket = 3;
pac_size_ = k10MsBlocksPerPacket * kSampleRateHz / 100;
audio_format_->parameters["ptime"] = "30";
RegisterCodec();
const int kCngPayloadType = 105;
RegisterCngCodec(kCngPayloadType);
DoTest(k10MsBlocksPerPacket, kCngPayloadType);
}
// A multi-threaded test for ACM that uses the PCM16b 16 kHz codec.
class AudioCodingModuleMtTestOldApi : public AudioCodingModuleTestOldApi {
protected:
static const int kNumPackets = 500;
static const int kNumPullCalls = 500;
AudioCodingModuleMtTestOldApi()
: AudioCodingModuleTestOldApi(),
send_count_(0),
insert_packet_count_(0),
pull_audio_count_(0),
next_insert_packet_time_ms_(0),
fake_clock_(new SimulatedClock(0)) {
clock_ = fake_clock_.get();
}
void SetUp() {
AudioCodingModuleTestOldApi::SetUp();
RegisterCodec(); // Must be called before the threads start below.
StartThreads();
}
void StartThreads() {
quit_.store(false);
const auto attributes =
rtc::ThreadAttributes().SetPriority(rtc::ThreadPriority::kRealtime);
send_thread_ = rtc::PlatformThread::SpawnJoinable(
[this] {
while (!quit_.load()) {
CbSendImpl();
}
},
"send", attributes);
insert_packet_thread_ = rtc::PlatformThread::SpawnJoinable(
[this] {
while (!quit_.load()) {
CbInsertPacketImpl();
}
},
"insert_packet", attributes);
pull_audio_thread_ = rtc::PlatformThread::SpawnJoinable(
[this] {
while (!quit_.load()) {
CbPullAudioImpl();
}
},
"pull_audio", attributes);
}
void TearDown() {
AudioCodingModuleTestOldApi::TearDown();
quit_.store(true);
pull_audio_thread_.Finalize();
send_thread_.Finalize();
insert_packet_thread_.Finalize();
}
bool RunTest() { return test_complete_.Wait(TimeDelta::Minutes(10)); }
virtual bool TestDone() {
if (packet_cb_.num_calls() > kNumPackets) {
MutexLock lock(&mutex_);
if (pull_audio_count_ > kNumPullCalls) {
// Both conditions for completion are met. End the test.
return true;
}
}
return false;
}
// The send thread doesn't have to care about the current simulated time,
// since only the AcmReceiver is using the clock.
void CbSendImpl() {
SleepMs(1);
if (HasFatalFailure()) {
// End the test early if a fatal failure (ASSERT_*) has occurred.
test_complete_.Set();
}
++send_count_;
InsertAudioAndVerifyEncoding();
if (TestDone()) {
test_complete_.Set();
}
}
void CbInsertPacketImpl() {
SleepMs(1);
{
MutexLock lock(&mutex_);
if (clock_->TimeInMilliseconds() < next_insert_packet_time_ms_) {
return;
}
next_insert_packet_time_ms_ += 10;
}
// Now we're not holding the crit sect when calling ACM.
++insert_packet_count_;
InsertPacket();
}
void CbPullAudioImpl() {
SleepMs(1);
{
MutexLock lock(&mutex_);
// Don't let the insert thread fall behind.
if (next_insert_packet_time_ms_ < clock_->TimeInMilliseconds()) {
return;
}
++pull_audio_count_;
}
// Now we're not holding the crit sect when calling ACM.
PullAudio();
fake_clock_->AdvanceTimeMilliseconds(10);
}
rtc::PlatformThread send_thread_;
rtc::PlatformThread insert_packet_thread_;
rtc::PlatformThread pull_audio_thread_;
// Used to force worker threads to stop looping.
std::atomic<bool> quit_;
rtc::Event test_complete_;
int send_count_;
int insert_packet_count_;
int pull_audio_count_ RTC_GUARDED_BY(mutex_);
Mutex mutex_;
int64_t next_insert_packet_time_ms_ RTC_GUARDED_BY(mutex_);
std::unique_ptr<SimulatedClock> fake_clock_;
};
#if defined(WEBRTC_IOS)
#define MAYBE_DoTest DISABLED_DoTest
#else
#define MAYBE_DoTest DoTest
#endif
TEST_F(AudioCodingModuleMtTestOldApi, MAYBE_DoTest) {
EXPECT_TRUE(RunTest());
}
class AudioPacketizationCallbackMock : public AudioPacketizationCallback {
public:
MOCK_METHOD(int32_t,
SendData,
(AudioFrameType frame_type,
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
size_t payload_len_bytes,
int64_t absolute_capture_timestamp_ms),
(override));
};
class AcmAbsoluteCaptureTimestamp : public ::testing::Test {
public:
AcmAbsoluteCaptureTimestamp() : audio_frame_(kSampleRateHz, kNumChannels) {}
protected:
static constexpr int kPTimeMs = 20;
static constexpr int kSampleRateHz = 48000;
static constexpr int kFrameSize = kSampleRateHz / 100;
static constexpr int kNumChannels = 2;
void SetUp() {
rtc::scoped_refptr<AudioEncoderFactory> codec_factory =
CreateBuiltinAudioEncoderFactory();
acm_ = AudioCodingModule::Create();
std::unique_ptr<AudioEncoder> encoder = codec_factory->MakeAudioEncoder(
111, SdpAudioFormat("OPUS", kSampleRateHz, kNumChannels),
absl::nullopt);
encoder->SetDtx(true);
encoder->SetReceiverFrameLengthRange(kPTimeMs, kPTimeMs);
acm_->SetEncoder(std::move(encoder));
acm_->RegisterTransportCallback(&transport_);
for (size_t k = 0; k < audio_.size(); ++k) {
audio_[k] = 10 * k;
}
}
const AudioFrame& GetAudioWithAbsoluteCaptureTimestamp(
int64_t absolute_capture_timestamp_ms) {
audio_frame_.ResetWithoutMuting();
audio_frame_.UpdateFrame(timestamp_, audio_.data(), kFrameSize,
kSampleRateHz,
AudioFrame::SpeechType::kNormalSpeech,
AudioFrame::VADActivity::kVadActive, kNumChannels);
audio_frame_.set_absolute_capture_timestamp_ms(
absolute_capture_timestamp_ms);
timestamp_ += kFrameSize;
return audio_frame_;
}
std::unique_ptr<AudioCodingModule> acm_;
AudioPacketizationCallbackMock transport_;
AudioFrame audio_frame_;
std::array<int16_t, kFrameSize * kNumChannels> audio_;
uint32_t timestamp_ = 9873546;
};
TEST_F(AcmAbsoluteCaptureTimestamp, HaveBeginningOfFrameCaptureTime) {
constexpr int64_t first_absolute_capture_timestamp_ms = 123456789;
int64_t absolute_capture_timestamp_ms = first_absolute_capture_timestamp_ms;
EXPECT_CALL(transport_,
SendData(_, _, _, _, _, first_absolute_capture_timestamp_ms))
.Times(1);
EXPECT_CALL(
transport_,
SendData(_, _, _, _, _, first_absolute_capture_timestamp_ms + kPTimeMs))
.Times(1);
for (int k = 0; k < 5; ++k) {
acm_->Add10MsData(
GetAudioWithAbsoluteCaptureTimestamp(absolute_capture_timestamp_ms));
absolute_capture_timestamp_ms += 10;
}
}
// Disabling all of these tests on iOS until file support has been added.
// See https://code.google.com/p/webrtc/issues/detail?id=4752 for details.
#if !defined(WEBRTC_IOS)
// This test verifies bit exactness for the send-side of ACM. The test setup is
// a chain of three different test classes:
//
// test::AcmSendTest -> AcmSenderBitExactness -> test::AcmReceiveTest
//
// The receiver side is driving the test by requesting new packets from
// AcmSenderBitExactness::NextPacket(). This method, in turn, asks for the
// packet from test::AcmSendTest::NextPacket, which inserts audio from the
// input file until one packet is produced. (The input file loops indefinitely.)
// Before passing the packet to the receiver, this test class verifies the
// packet header and updates a payload checksum with the new payload. The
// decoded output from the receiver is also verified with a (separate) checksum.
class AcmSenderBitExactnessOldApi : public ::testing::Test,
public test::PacketSource {
protected:
static const int kTestDurationMs = 1000;
AcmSenderBitExactnessOldApi()
: frame_size_rtp_timestamps_(0),
packet_count_(0),
payload_type_(0),
last_sequence_number_(0),
last_timestamp_(0),
payload_checksum_(rtc::MessageDigestFactory::Create(rtc::DIGEST_MD5)) {}
// Sets up the test::AcmSendTest object. Returns true on success, otherwise
// false.
bool SetUpSender(absl::string_view input_file_name, int source_rate) {
// Note that `audio_source_` will loop forever. The test duration is set
// explicitly by `kTestDurationMs`.
audio_source_.reset(new test::InputAudioFile(input_file_name));
send_test_.reset(new test::AcmSendTestOldApi(audio_source_.get(),
source_rate, kTestDurationMs));
return send_test_.get() != NULL;
}
// Registers a send codec in the test::AcmSendTest object. Returns true on
// success, false on failure.
bool RegisterSendCodec(absl::string_view payload_name,
int sampling_freq_hz,
int channels,
int payload_type,
int frame_size_samples,
int frame_size_rtp_timestamps) {
payload_type_ = payload_type;
frame_size_rtp_timestamps_ = frame_size_rtp_timestamps;
return send_test_->RegisterCodec(payload_name, sampling_freq_hz, channels,
payload_type, frame_size_samples);
}
void RegisterExternalSendCodec(
std::unique_ptr<AudioEncoder> external_speech_encoder,
int payload_type) {
payload_type_ = payload_type;
frame_size_rtp_timestamps_ = rtc::checked_cast<uint32_t>(
external_speech_encoder->Num10MsFramesInNextPacket() *
external_speech_encoder->RtpTimestampRateHz() / 100);
send_test_->RegisterExternalCodec(std::move(external_speech_encoder));
}
// Runs the test. SetUpSender() and RegisterSendCodec() must have been called
// before calling this method.
void Run(absl::string_view audio_checksum_ref,
absl::string_view payload_checksum_ref,
int expected_packets,
test::AcmReceiveTestOldApi::NumOutputChannels expected_channels,
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory = nullptr) {
if (!decoder_factory) {
decoder_factory = CreateBuiltinAudioDecoderFactory();
}
// Set up the receiver used to decode the packets and verify the decoded
// output.
test::AudioChecksum audio_checksum;
const std::string output_file_name =
webrtc::test::OutputPath() +
::testing::UnitTest::GetInstance()
->current_test_info()
->test_case_name() +
"_" + ::testing::UnitTest::GetInstance()->current_test_info()->name() +
"_output.wav";
const int kOutputFreqHz = 8000;
test::OutputWavFile output_file(output_file_name, kOutputFreqHz,
expected_channels);
// Have the output audio sent both to file and to the checksum calculator.
test::AudioSinkFork output(&audio_checksum, &output_file);
test::AcmReceiveTestOldApi receive_test(this, &output, kOutputFreqHz,
expected_channels, decoder_factory);
ASSERT_NO_FATAL_FAILURE(receive_test.RegisterDefaultCodecs());
// This is where the actual test is executed.
receive_test.Run();
// Extract and verify the audio checksum.
std::string checksum_string = audio_checksum.Finish();
ExpectChecksumEq(audio_checksum_ref, checksum_string);
// Extract and verify the payload checksum.
rtc::Buffer checksum_result(payload_checksum_->Size());
payload_checksum_->Finish(checksum_result.data(), checksum_result.size());
checksum_string = rtc::hex_encode(checksum_result);
ExpectChecksumEq(payload_checksum_ref, checksum_string);
// Verify number of packets produced.
EXPECT_EQ(expected_packets, packet_count_);
// Delete the output file.
remove(output_file_name.c_str());
}
// Helper: result must be one the "|"-separated checksums.
void ExpectChecksumEq(absl::string_view ref, absl::string_view result) {
if (ref.size() == result.size()) {
// Only one checksum: clearer message.
EXPECT_EQ(ref, result);
} else {
EXPECT_NE(ref.find(result), absl::string_view::npos)
<< result << " must be one of these:\n"
<< ref;
}
}
// Inherited from test::PacketSource.
std::unique_ptr<test::Packet> NextPacket() override {
auto packet = send_test_->NextPacket();
if (!packet)
return NULL;
VerifyPacket(packet.get());
// TODO(henrik.lundin) Save the packet to file as well.
// Pass it on to the caller. The caller becomes the owner of `packet`.
return packet;
}
// Verifies the packet.
void VerifyPacket(const test::Packet* packet) {
EXPECT_TRUE(packet->valid_header());
// (We can check the header fields even if valid_header() is false.)
EXPECT_EQ(payload_type_, packet->header().payloadType);
if (packet_count_ > 0) {
// This is not the first packet.
uint16_t sequence_number_diff =
packet->header().sequenceNumber - last_sequence_number_;
EXPECT_EQ(1, sequence_number_diff);
uint32_t timestamp_diff = packet->header().timestamp - last_timestamp_;
EXPECT_EQ(frame_size_rtp_timestamps_, timestamp_diff);
}
++packet_count_;
last_sequence_number_ = packet->header().sequenceNumber;
last_timestamp_ = packet->header().timestamp;
// Update the checksum.
payload_checksum_->Update(packet->payload(),
packet->payload_length_bytes());
}
void SetUpTest(absl::string_view codec_name,
int codec_sample_rate_hz,
int channels,
int payload_type,
int codec_frame_size_samples,
int codec_frame_size_rtp_timestamps) {
ASSERT_TRUE(SetUpSender(
channels == 1 ? kTestFileMono32kHz : kTestFileFakeStereo32kHz, 32000));
ASSERT_TRUE(RegisterSendCodec(codec_name, codec_sample_rate_hz, channels,
payload_type, codec_frame_size_samples,
codec_frame_size_rtp_timestamps));
}
void SetUpTestExternalEncoder(
std::unique_ptr<AudioEncoder> external_speech_encoder,
int payload_type) {
ASSERT_TRUE(send_test_);
RegisterExternalSendCodec(std::move(external_speech_encoder), payload_type);
}
std::unique_ptr<test::AcmSendTestOldApi> send_test_;
std::unique_ptr<test::InputAudioFile> audio_source_;
uint32_t frame_size_rtp_timestamps_;
int packet_count_;
uint8_t payload_type_;
uint16_t last_sequence_number_;
uint32_t last_timestamp_;
std::unique_ptr<rtc::MessageDigest> payload_checksum_;
const std::string kTestFileMono32kHz =
webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
const std::string kTestFileFakeStereo32kHz =
webrtc::test::ResourcePath("audio_coding/testfile_fake_stereo_32kHz",
"pcm");
const std::string kTestFileQuad48kHz = webrtc::test::ResourcePath(
"audio_coding/speech_4_channels_48k_one_second",
"wav");
};
class AcmSenderBitExactnessNewApi : public AcmSenderBitExactnessOldApi {};
TEST_F(AcmSenderBitExactnessOldApi, Pcm16_8000khz_10ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 8000, 1, 107, 80, 80));
Run(/*audio_checksum_ref=*/"3e43fd5d3c73a59e8118e68fbfafe2c7",
/*payload_checksum_ref=*/"c1edd36339ce0326cc4550041ad719a0",
/*expected_packets=*/100,
/*expected_channels=*/test::AcmReceiveTestOldApi::kMonoOutput);
}
TEST_F(AcmSenderBitExactnessOldApi, Pcm16_16000khz_10ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 16000, 1, 108, 160, 160));
Run(/*audio_checksum_ref=*/"608750138315cbab33d76d38e8367807",
/*payload_checksum_ref=*/"ad786526383178b08d80d6eee06e9bad",
/*expected_packets=*/100,
/*expected_channels=*/test::AcmReceiveTestOldApi::kMonoOutput);
}
TEST_F(AcmSenderBitExactnessOldApi, Pcm16_32000khz_10ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 32000, 1, 109, 320, 320));
Run(/*audio_checksum_ref=*/"02e9927ef5e4d2cd792a5df0bdee5e19",
/*payload_checksum_ref=*/"5ef82ea885e922263606c6fdbc49f651",
/*expected_packets=*/100,
/*expected_channels=*/test::AcmReceiveTestOldApi::kMonoOutput);
}
TEST_F(AcmSenderBitExactnessOldApi, Pcm16_stereo_8000khz_10ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 8000, 2, 111, 80, 80));
Run(/*audio_checksum_ref=*/"4ff38de045b19f64de9c7e229ba36317",
/*payload_checksum_ref=*/"62ce5adb0d4965d0a52ec98ae7f98974",
/*expected_packets=*/100,
/*expected_channels=*/test::AcmReceiveTestOldApi::kStereoOutput);
}
TEST_F(AcmSenderBitExactnessOldApi, Pcm16_stereo_16000khz_10ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 16000, 2, 112, 160, 160));
Run(/*audio_checksum_ref=*/"1ee35394cfca78ad6d55468441af36fa",
/*payload_checksum_ref=*/"41ca8edac4b8c71cd54fd9f25ec14870",
/*expected_packets=*/100,
/*expected_channels=*/test::AcmReceiveTestOldApi::kStereoOutput);
}
TEST_F(AcmSenderBitExactnessOldApi, Pcm16_stereo_32000khz_10ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 32000, 2, 113, 320, 320));
Run(/*audio_checksum_ref=*/"19cae34730a0f6a17cf4e76bf21b69d6",
/*payload_checksum_ref=*/"50e58502fb04421bf5b857dda4c96879",
/*expected_packets=*/100,
/*expected_channels=*/test::AcmReceiveTestOldApi::kStereoOutput);
}
TEST_F(AcmSenderBitExactnessOldApi, Pcmu_20ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("PCMU", 8000, 1, 0, 160, 160));
Run(/*audio_checksum_ref=*/"c8d1fc677f33c2022ec5f83c7f302280",
/*payload_checksum_ref=*/"8f9b8750bd80fe26b6cbf6659b89f0f9",
/*expected_packets=*/50,
/*expected_channels=*/test::AcmReceiveTestOldApi::kMonoOutput);
}
TEST_F(AcmSenderBitExactnessOldApi, Pcma_20ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("PCMA", 8000, 1, 8, 160, 160));
Run(/*audio_checksum_ref=*/"ae259cab624095270b7369e53a7b53a3",
/*payload_checksum_ref=*/"6ad745e55aa48981bfc790d0eeef2dd1",
/*expected_packets=*/50,
/*expected_channels=*/test::AcmReceiveTestOldApi::kMonoOutput);
}
TEST_F(AcmSenderBitExactnessOldApi, Pcmu_stereo_20ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("PCMU", 8000, 2, 110, 160, 160));
Run(/*audio_checksum_ref=*/"6ef2f57d4934714787fd0a834e3ea18e",
/*payload_checksum_ref=*/"60b6f25e8d1e74cb679cfe756dd9bca5",
/*expected_packets=*/50,
/*expected_channels=*/test::AcmReceiveTestOldApi::kStereoOutput);
}
TEST_F(AcmSenderBitExactnessOldApi, Pcma_stereo_20ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("PCMA", 8000, 2, 118, 160, 160));
Run(/*audio_checksum_ref=*/"f2e81d2531a805c40e61da5106b50006",
/*payload_checksum_ref=*/"92b282c83efd20e7eeef52ba40842cf7",
/*expected_packets=*/50,
/*expected_channels=*/test::AcmReceiveTestOldApi::kStereoOutput);
}
#if defined(WEBRTC_CODEC_ILBC) && defined(WEBRTC_LINUX) && \
defined(WEBRTC_ARCH_X86_64)
TEST_F(AcmSenderBitExactnessOldApi, Ilbc_30ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("ILBC", 8000, 1, 102, 240, 240));
Run(/*audio_checksum_ref=*/"a739434bec8a754e9356ce2115603ce5",
/*payload_checksum_ref=*/"cfae2e9f6aba96e145f2bcdd5050ce78",
/*expected_packets=*/33,
/*expected_channels=*/test::AcmReceiveTestOldApi::kMonoOutput);
}
#endif
#if defined(WEBRTC_LINUX) && defined(WEBRTC_ARCH_X86_64)
TEST_F(AcmSenderBitExactnessOldApi, G722_20ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("G722", 16000, 1, 9, 320, 160));
Run(/*audio_checksum_ref=*/"b875d9a3e41f5470857bdff02e3b368f",
/*payload_checksum_ref=*/"fc68a87e1380614e658087cb35d5ca10",
/*expected_packets=*/50,
/*expected_channels=*/test::AcmReceiveTestOldApi::kMonoOutput);
}
#endif
#if defined(WEBRTC_LINUX) && defined(WEBRTC_ARCH_X86_64)
TEST_F(AcmSenderBitExactnessOldApi, G722_stereo_20ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("G722", 16000, 2, 119, 320, 160));
Run(/*audio_checksum_ref=*/"02c427d73363b2f37853a0dd17fe1aba",
/*payload_checksum_ref=*/"66516152eeaa1e650ad94ff85f668dac",
/*expected_packets=*/50,
/*expected_channels=*/test::AcmReceiveTestOldApi::kStereoOutput);
}
#endif
namespace {
// Checksum depends on libopus being compiled with or without SSE.
const std::string audio_checksum =
"6a76fe2ffba057c06eb63239b3c47abe"
"|0c4f9d33b4a7379a34ee0c0d5718afe6";
const std::string payload_checksum =
"b43bdf7638b2bc2a5a6f30bdc640b9ed"
"|c30d463e7ed10bdd1da9045f80561f27";
} // namespace
#if defined(WEBRTC_LINUX) && defined(WEBRTC_ARCH_X86_64)
TEST_F(AcmSenderBitExactnessOldApi, Opus_stereo_20ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 2, 120, 960, 960));
Run(audio_checksum, payload_checksum, /*expected_packets=*/50,
/*expected_channels=*/test::AcmReceiveTestOldApi::kStereoOutput);
}
#endif
#if defined(WEBRTC_LINUX) && defined(WEBRTC_ARCH_X86_64)
TEST_F(AcmSenderBitExactnessNewApi, OpusFromFormat_stereo_20ms) {
const auto config = AudioEncoderOpus::SdpToConfig(
SdpAudioFormat("opus", 48000, 2, {{"stereo", "1"}}));
ASSERT_TRUE(SetUpSender(kTestFileFakeStereo32kHz, 32000));
ASSERT_NO_FATAL_FAILURE(SetUpTestExternalEncoder(
AudioEncoderOpus::MakeAudioEncoder(*config, 120), 120));
Run(audio_checksum, payload_checksum, /*expected_packets=*/50,
/*expected_channels=*/test::AcmReceiveTestOldApi::kStereoOutput);
}
#endif
// TODO(webrtc:8649): Disabled until the Encoder counterpart of
// https://webrtc-review.googlesource.com/c/src/+/129768 lands.
#if defined(WEBRTC_LINUX) && defined(WEBRTC_ARCH_X86_64)
TEST_F(AcmSenderBitExactnessNewApi, DISABLED_OpusManyChannels) {
constexpr int kNumChannels = 4;
constexpr int kOpusPayloadType = 120;
// Read a 4 channel file at 48kHz.
ASSERT_TRUE(SetUpSender(kTestFileQuad48kHz, 48000));
const auto sdp_format = SdpAudioFormat("multiopus", 48000, kNumChannels,
{{"channel_mapping", "0,1,2,3"},
{"coupled_streams", "2"},
{"num_streams", "2"}});
const auto encoder_config =
AudioEncoderMultiChannelOpus::SdpToConfig(sdp_format);
ASSERT_TRUE(encoder_config.has_value());
ASSERT_NO_FATAL_FAILURE(
SetUpTestExternalEncoder(AudioEncoderMultiChannelOpus::MakeAudioEncoder(
*encoder_config, kOpusPayloadType),
kOpusPayloadType));
const auto decoder_config =
AudioDecoderMultiChannelOpus::SdpToConfig(sdp_format);
const auto opus_decoder =
AudioDecoderMultiChannelOpus::MakeAudioDecoder(*decoder_config);
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory =
rtc::make_ref_counted<test::AudioDecoderProxyFactory>(opus_decoder.get());
// Set up an EXTERNAL DECODER to parse 4 channels.
Run("audio checksum check downstream|8051617907766bec5f4e4a4f7c6d5291",
"payload checksum check downstream|b09c52e44b2bdd9a0809e3a5b1623a76",
/*expected_packets=*/50,
/*expected_channels=*/test::AcmReceiveTestOldApi::kQuadOutput,
decoder_factory);
}
#endif
#if defined(WEBRTC_LINUX) && defined(WEBRTC_ARCH_X86_64)
TEST_F(AcmSenderBitExactnessNewApi, OpusFromFormat_stereo_20ms_voip) {
auto config = AudioEncoderOpus::SdpToConfig(
SdpAudioFormat("opus", 48000, 2, {{"stereo", "1"}}));
// If not set, default will be kAudio in case of stereo.
config->application = AudioEncoderOpusConfig::ApplicationMode::kVoip;
ASSERT_TRUE(SetUpSender(kTestFileFakeStereo32kHz, 32000));
ASSERT_NO_FATAL_FAILURE(SetUpTestExternalEncoder(
AudioEncoderOpus::MakeAudioEncoder(*config, 120), 120));
const std::string audio_maybe_sse =
"cb644fc17d9666a0f5986eef24818159"
"|4a74024473c7c729543c2790829b1e42";
const std::string payload_maybe_sse =
"ea48d94e43217793af9b7e15ece94e54"
"|bd93c492087093daf662cdd968f6cdda";
Run(audio_maybe_sse, payload_maybe_sse, /*expected_packets=*/50,
/*expected_channels=*/test::AcmReceiveTestOldApi::kStereoOutput);
}
#endif
// This test is for verifying the SetBitRate function. The bitrate is changed at
// the beginning, and the number of generated bytes are checked.
class AcmSetBitRateTest : public ::testing::Test {
protected:
static const int kTestDurationMs = 1000;
// Sets up the test::AcmSendTest object. Returns true on success, otherwise
// false.
bool SetUpSender() {
const std::string input_file_name =
webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
// Note that `audio_source_` will loop forever. The test duration is set
// explicitly by `kTestDurationMs`.
audio_source_.reset(new test::InputAudioFile(input_file_name));
static const int kSourceRateHz = 32000;
send_test_.reset(new test::AcmSendTestOldApi(
audio_source_.get(), kSourceRateHz, kTestDurationMs));
return send_test_.get();
}
// Registers a send codec in the test::AcmSendTest object. Returns true on
// success, false on failure.
virtual bool RegisterSendCodec(absl::string_view payload_name,
int sampling_freq_hz,
int channels,
int payload_type,
int frame_size_samples,
int frame_size_rtp_timestamps) {
return send_test_->RegisterCodec(payload_name, sampling_freq_hz, channels,
payload_type, frame_size_samples);
}
void RegisterExternalSendCodec(
std::unique_ptr<AudioEncoder> external_speech_encoder,
int payload_type) {
send_test_->RegisterExternalCodec(std::move(external_speech_encoder));
}
void RunInner(int min_expected_total_bits, int max_expected_total_bits) {
int nr_bytes = 0;
while (std::unique_ptr<test::Packet> next_packet =
send_test_->NextPacket()) {
nr_bytes += rtc::checked_cast<int>(next_packet->payload_length_bytes());
}
EXPECT_LE(min_expected_total_bits, nr_bytes * 8);
EXPECT_GE(max_expected_total_bits, nr_bytes * 8);
}
void SetUpTest(absl::string_view codec_name,
int codec_sample_rate_hz,
int channels,
int payload_type,
int codec_frame_size_samples,
int codec_frame_size_rtp_timestamps) {
ASSERT_TRUE(SetUpSender());
ASSERT_TRUE(RegisterSendCodec(codec_name, codec_sample_rate_hz, channels,
payload_type, codec_frame_size_samples,
codec_frame_size_rtp_timestamps));
}
std::unique_ptr<test::AcmSendTestOldApi> send_test_;
std::unique_ptr<test::InputAudioFile> audio_source_;
};
class AcmSetBitRateNewApi : public AcmSetBitRateTest {
protected:
// Runs the test. SetUpSender() must have been called and a codec must be set
// up before calling this method.
void Run(int min_expected_total_bits, int max_expected_total_bits) {
RunInner(min_expected_total_bits, max_expected_total_bits);
}
};
TEST_F(AcmSetBitRateNewApi, OpusFromFormat_48khz_20ms_10kbps) {
const auto config = AudioEncoderOpus::SdpToConfig(
SdpAudioFormat("opus", 48000, 2, {{"maxaveragebitrate", "10000"}}));
ASSERT_TRUE(SetUpSender());
RegisterExternalSendCodec(AudioEncoderOpus::MakeAudioEncoder(*config, 107),
107);
RunInner(7000, 12000);
}
TEST_F(AcmSetBitRateNewApi, OpusFromFormat_48khz_20ms_50kbps) {
const auto config = AudioEncoderOpus::SdpToConfig(
SdpAudioFormat("opus", 48000, 2, {{"maxaveragebitrate", "50000"}}));
ASSERT_TRUE(SetUpSender());
RegisterExternalSendCodec(AudioEncoderOpus::MakeAudioEncoder(*config, 107),
107);
RunInner(40000, 60000);
}
// Verify that it works when the data to send is mono and the encoder is set to
// send surround audio.
TEST_F(AudioCodingModuleTestOldApi, SendingMultiChannelForMonoInput) {
constexpr int kSampleRateHz = 48000;
constexpr int kSamplesPerChannel = kSampleRateHz * 10 / 1000;
audio_format_ = SdpAudioFormat({"multiopus",
kSampleRateHz,
6,
{{"minptime", "10"},
{"useinbandfec", "1"},
{"channel_mapping", "0,4,1,2,3,5"},
{"num_streams", "4"},
{"coupled_streams", "2"}}});
RegisterCodec();
input_frame_.sample_rate_hz_ = kSampleRateHz;
input_frame_.num_channels_ = 1;
input_frame_.samples_per_channel_ = kSamplesPerChannel;
for (size_t k = 0; k < 10; ++k) {
ASSERT_GE(acm_->Add10MsData(input_frame_), 0);
input_frame_.timestamp_ += kSamplesPerChannel;
}
}
// Verify that it works when the data to send is stereo and the encoder is set
// to send surround audio.
TEST_F(AudioCodingModuleTestOldApi, SendingMultiChannelForStereoInput) {
constexpr int kSampleRateHz = 48000;
constexpr int kSamplesPerChannel = (kSampleRateHz * 10) / 1000;
audio_format_ = SdpAudioFormat({"multiopus",
kSampleRateHz,
6,
{{"minptime", "10"},
{"useinbandfec", "1"},
{"channel_mapping", "0,4,1,2,3,5"},
{"num_streams", "4"},
{"coupled_streams", "2"}}});
RegisterCodec();
input_frame_.sample_rate_hz_ = kSampleRateHz;
input_frame_.num_channels_ = 2;
input_frame_.samples_per_channel_ = kSamplesPerChannel;
for (size_t k = 0; k < 10; ++k) {
ASSERT_GE(acm_->Add10MsData(input_frame_), 0);
input_frame_.timestamp_ += kSamplesPerChannel;
}
}
// Verify that it works when the data to send is mono and the encoder is set to
// send stereo audio.
TEST_F(AudioCodingModuleTestOldApi, SendingStereoForMonoInput) {
constexpr int kSampleRateHz = 48000;
constexpr int kSamplesPerChannel = (kSampleRateHz * 10) / 1000;
audio_format_ = SdpAudioFormat("L16", kSampleRateHz, 2);
RegisterCodec();
input_frame_.sample_rate_hz_ = kSampleRateHz;
input_frame_.num_channels_ = 1;
input_frame_.samples_per_channel_ = kSamplesPerChannel;
for (size_t k = 0; k < 10; ++k) {
ASSERT_GE(acm_->Add10MsData(input_frame_), 0);
input_frame_.timestamp_ += kSamplesPerChannel;
}
}
// Verify that it works when the data to send is stereo and the encoder is set
// to send mono audio.
TEST_F(AudioCodingModuleTestOldApi, SendingMonoForStereoInput) {
constexpr int kSampleRateHz = 48000;
constexpr int kSamplesPerChannel = (kSampleRateHz * 10) / 1000;
audio_format_ = SdpAudioFormat("L16", kSampleRateHz, 1);
RegisterCodec();
input_frame_.sample_rate_hz_ = kSampleRateHz;
input_frame_.num_channels_ = 1;
input_frame_.samples_per_channel_ = kSamplesPerChannel;
for (size_t k = 0; k < 10; ++k) {
ASSERT_GE(acm_->Add10MsData(input_frame_), 0);
input_frame_.timestamp_ += kSamplesPerChannel;
}
}
// The result on the Android platforms is inconsistent for this test case.
// On android_rel the result is different from android and android arm64 rel.
#if defined(WEBRTC_ANDROID)
#define MAYBE_OpusFromFormat_48khz_20ms_100kbps \
DISABLED_OpusFromFormat_48khz_20ms_100kbps
#else
#define MAYBE_OpusFromFormat_48khz_20ms_100kbps \
OpusFromFormat_48khz_20ms_100kbps
#endif
TEST_F(AcmSetBitRateNewApi, MAYBE_OpusFromFormat_48khz_20ms_100kbps) {
const auto config = AudioEncoderOpus::SdpToConfig(
SdpAudioFormat("opus", 48000, 2, {{"maxaveragebitrate", "100000"}}));
ASSERT_TRUE(SetUpSender());
RegisterExternalSendCodec(AudioEncoderOpus::MakeAudioEncoder(*config, 107),
107);
RunInner(80000, 120000);
}
TEST_F(AcmSenderBitExactnessOldApi, External_Pcmu_20ms) {
AudioEncoderPcmU::Config config;
config.frame_size_ms = 20;
config.num_channels = 1;
config.payload_type = 0;
AudioEncoderPcmU encoder(config);
auto mock_encoder = std::make_unique<MockAudioEncoder>();
// Set expectations on the mock encoder and also delegate the calls to the
// real encoder.
EXPECT_CALL(*mock_encoder, SampleRateHz())
.Times(AtLeast(1))
.WillRepeatedly(Invoke(&encoder, &AudioEncoderPcmU::SampleRateHz));
EXPECT_CALL(*mock_encoder, NumChannels())
.Times(AtLeast(1))
.WillRepeatedly(Invoke(&encoder, &AudioEncoderPcmU::NumChannels));
EXPECT_CALL(*mock_encoder, RtpTimestampRateHz())
.Times(AtLeast(1))
.WillRepeatedly(Invoke(&encoder, &AudioEncoderPcmU::RtpTimestampRateHz));
EXPECT_CALL(*mock_encoder, Num10MsFramesInNextPacket())
.Times(AtLeast(1))
.WillRepeatedly(
Invoke(&encoder, &AudioEncoderPcmU::Num10MsFramesInNextPacket));
EXPECT_CALL(*mock_encoder, GetTargetBitrate())
.Times(AtLeast(1))
.WillRepeatedly(Invoke(&encoder, &AudioEncoderPcmU::GetTargetBitrate));
EXPECT_CALL(*mock_encoder, EncodeImpl(_, _, _))
.Times(AtLeast(1))
.WillRepeatedly(Invoke(
&encoder, static_cast<AudioEncoder::EncodedInfo (AudioEncoder::*)(
uint32_t, rtc::ArrayView<const int16_t>, rtc::Buffer*)>(
&AudioEncoderPcmU::Encode)));
ASSERT_TRUE(SetUpSender(kTestFileMono32kHz, 32000));
ASSERT_NO_FATAL_FAILURE(
SetUpTestExternalEncoder(std::move(mock_encoder), config.payload_type));
Run("c8d1fc677f33c2022ec5f83c7f302280", "8f9b8750bd80fe26b6cbf6659b89f0f9",
50, test::AcmReceiveTestOldApi::kMonoOutput);
}
// This test fixture is implemented to run ACM and change the desired output
// frequency during the call. The input packets are simply PCM16b-wb encoded
// payloads with a constant value of `kSampleValue`. The test fixture itself
// acts as PacketSource in between the receive test class and the constant-
// payload packet source class. The output is both written to file, and analyzed
// in this test fixture.
class AcmSwitchingOutputFrequencyOldApi : public ::testing::Test,
public test::PacketSource,
public test::AudioSink {
protected:
static const size_t kTestNumPackets = 50;
static const int kEncodedSampleRateHz = 16000;
static const size_t kPayloadLenSamples = 30 * kEncodedSampleRateHz / 1000;
static const int kPayloadType = 108; // Default payload type for PCM16b-wb.
AcmSwitchingOutputFrequencyOldApi()
: first_output_(true),
num_packets_(0),
packet_source_(kPayloadLenSamples,
kSampleValue,
kEncodedSampleRateHz,
kPayloadType),
output_freq_2_(0),
has_toggled_(false) {}
void Run(int output_freq_1, int output_freq_2, int toggle_period_ms) {
// Set up the receiver used to decode the packets and verify the decoded
// output.
const std::string output_file_name =
webrtc::test::OutputPath() +
::testing::UnitTest::GetInstance()
->current_test_info()
->test_case_name() +
"_" + ::testing::UnitTest::GetInstance()->current_test_info()->name() +
"_output.pcm";
test::OutputAudioFile output_file(output_file_name);
// Have the output audio sent both to file and to the WriteArray method in
// this class.
test::AudioSinkFork output(this, &output_file);
test::AcmReceiveTestToggleOutputFreqOldApi receive_test(
this, &output, output_freq_1, output_freq_2, toggle_period_ms,
test::AcmReceiveTestOldApi::kMonoOutput);
ASSERT_NO_FATAL_FAILURE(receive_test.RegisterDefaultCodecs());
output_freq_2_ = output_freq_2;
// This is where the actual test is executed.
receive_test.Run();
// Delete output file.
remove(output_file_name.c_str());
}
// Inherited from test::PacketSource.
std::unique_ptr<test::Packet> NextPacket() override {
// Check if it is time to terminate the test. The packet source is of type
// ConstantPcmPacketSource, which is infinite, so we must end the test
// "manually".
if (num_packets_++ > kTestNumPackets) {
EXPECT_TRUE(has_toggled_);
return NULL; // Test ended.
}
// Get the next packet from the source.
return packet_source_.NextPacket();
}
// Inherited from test::AudioSink.
bool WriteArray(const int16_t* audio, size_t num_samples) override {
// Skip checking the first output frame, since it has a number of zeros
// due to how NetEq is initialized.
if (first_output_) {
first_output_ = false;
return true;
}
for (size_t i = 0; i < num_samples; ++i) {
EXPECT_EQ(kSampleValue, audio[i]);
}
if (num_samples ==
static_cast<size_t>(output_freq_2_ / 100)) // Size of 10 ms frame.
has_toggled_ = true;
// The return value does not say if the values match the expectation, just
// that the method could process the samples.
return true;
}
const int16_t kSampleValue = 1000;
bool first_output_;
size_t num_packets_;
test::ConstantPcmPacketSource packet_source_;
int output_freq_2_;
bool has_toggled_;
};
TEST_F(AcmSwitchingOutputFrequencyOldApi, TestWithoutToggling) {
Run(16000, 16000, 1000);
}
TEST_F(AcmSwitchingOutputFrequencyOldApi, Toggle16KhzTo32Khz) {
Run(16000, 32000, 1000);
}
TEST_F(AcmSwitchingOutputFrequencyOldApi, Toggle32KhzTo16Khz) {
Run(32000, 16000, 1000);
}
TEST_F(AcmSwitchingOutputFrequencyOldApi, Toggle16KhzTo8Khz) {
Run(16000, 8000, 1000);
}
TEST_F(AcmSwitchingOutputFrequencyOldApi, Toggle8KhzTo16Khz) {
Run(8000, 16000, 1000);
}
#endif
} // namespace webrtc
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