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/*
* Copyright (c) 2012 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/EncodeDecodeTest.h"
#include <stdio.h>
#include <stdlib.h>
#include <memory>
#include "absl/strings/string_view.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/environment/environment.h"
#include "api/environment/environment_factory.h"
#include "modules/audio_coding/include/audio_coding_module.h"
#include "rtc_base/strings/string_builder.h"
#include "test/gtest.h"
#include "test/testsupport/file_utils.h"
namespace webrtc {
namespace {
// Buffer size for stereo 48 kHz audio.
constexpr size_t kWebRtc10MsPcmAudio = 960;
} // namespace
TestPacketization::TestPacketization(RTPStream* rtpStream, uint16_t frequency)
: _rtpStream(rtpStream), _frequency(frequency), _seqNo(0) {}
TestPacketization::~TestPacketization() {}
int32_t TestPacketization::SendData(const AudioFrameType /* frameType */,
const uint8_t payloadType,
const uint32_t timeStamp,
const uint8_t* payloadData,
const size_t payloadSize,
int64_t absolute_capture_timestamp_ms) {
_rtpStream->Write(payloadType, timeStamp, _seqNo++, payloadData, payloadSize,
_frequency);
return 1;
}
Sender::Sender()
: _acm(NULL), _pcmFile(), _audioFrame(), _packetization(NULL) {}
void Sender::Setup(const Environment& env,
AudioCodingModule* acm,
RTPStream* rtpStream,
absl::string_view in_file_name,
int in_sample_rate,
int payload_type,
SdpAudioFormat format) {
// Open input file
const std::string file_name = webrtc::test::ResourcePath(in_file_name, "pcm");
_pcmFile.Open(file_name, in_sample_rate, "rb");
if (format.num_channels == 2) {
_pcmFile.ReadStereo(true);
}
// Set test length to 500 ms (50 blocks of 10 ms each).
_pcmFile.SetNum10MsBlocksToRead(50);
// Fast-forward 1 second (100 blocks) since the file starts with silence.
_pcmFile.FastForward(100);
acm->SetEncoder(CreateBuiltinAudioEncoderFactory()->Create(
env, format, {.payload_type = payload_type}));
_packetization = new TestPacketization(rtpStream, format.clockrate_hz);
EXPECT_EQ(0, acm->RegisterTransportCallback(_packetization));
_acm = acm;
}
void Sender::Teardown() {
_pcmFile.Close();
delete _packetization;
}
bool Sender::Add10MsData() {
if (!_pcmFile.EndOfFile()) {
EXPECT_GT(_pcmFile.Read10MsData(_audioFrame), 0);
int32_t ok = _acm->Add10MsData(_audioFrame);
EXPECT_GE(ok, 0);
return ok >= 0 ? true : false;
}
return false;
}
void Sender::Run() {
while (true) {
if (!Add10MsData()) {
break;
}
}
}
Receiver::Receiver()
: _playoutLengthSmpls(kWebRtc10MsPcmAudio),
_payloadSizeBytes(MAX_INCOMING_PAYLOAD) {}
void Receiver::Setup(acm2::AcmReceiver* acm_receiver,
RTPStream* rtpStream,
absl::string_view out_file_name,
size_t channels,
int file_num) {
if (channels == 1) {
acm_receiver->SetCodecs({{107, {"L16", 8000, 1}},
{108, {"L16", 16000, 1}},
{109, {"L16", 32000, 1}},
{0, {"PCMU", 8000, 1}},
{8, {"PCMA", 8000, 1}},
{102, {"ILBC", 8000, 1}},
{9, {"G722", 8000, 1}},
{120, {"OPUS", 48000, 2}},
{13, {"CN", 8000, 1}},
{98, {"CN", 16000, 1}},
{99, {"CN", 32000, 1}}});
} else {
ASSERT_EQ(channels, 2u);
acm_receiver->SetCodecs({{111, {"L16", 8000, 2}},
{112, {"L16", 16000, 2}},
{113, {"L16", 32000, 2}},
{110, {"PCMU", 8000, 2}},
{118, {"PCMA", 8000, 2}},
{119, {"G722", 8000, 2}},
{120, {"OPUS", 48000, 2, {{"stereo", "1"}}}}});
}
int playSampFreq;
std::string file_name;
rtc::StringBuilder file_stream;
file_stream << webrtc::test::OutputPath() << out_file_name << file_num
<< ".pcm";
file_name = file_stream.str();
_rtpStream = rtpStream;
playSampFreq = 32000;
_pcmFile.Open(file_name, 32000, "wb+");
_realPayloadSizeBytes = 0;
_playoutBuffer = new int16_t[kWebRtc10MsPcmAudio];
_frequency = playSampFreq;
_acm_receiver = acm_receiver;
_firstTime = true;
}
void Receiver::Teardown() {
delete[] _playoutBuffer;
_pcmFile.Close();
}
bool Receiver::IncomingPacket() {
if (!_rtpStream->EndOfFile()) {
if (_firstTime) {
_firstTime = false;
_realPayloadSizeBytes = _rtpStream->Read(&_rtpHeader, _incomingPayload,
_payloadSizeBytes, &_nextTime);
if (_realPayloadSizeBytes == 0) {
if (_rtpStream->EndOfFile()) {
_firstTime = true;
return true;
} else {
return false;
}
}
}
EXPECT_EQ(0, _acm_receiver->InsertPacket(
_rtpHeader,
rtc::ArrayView<const uint8_t>(_incomingPayload,
_realPayloadSizeBytes),
/*receive_time=*/Timestamp::Millis(_nextTime)));
_realPayloadSizeBytes = _rtpStream->Read(&_rtpHeader, _incomingPayload,
_payloadSizeBytes, &_nextTime);
if (_realPayloadSizeBytes == 0 && _rtpStream->EndOfFile()) {
_firstTime = true;
}
}
return true;
}
bool Receiver::PlayoutData() {
AudioFrame audioFrame;
bool muted;
int32_t ok = _acm_receiver->GetAudio(_frequency, &audioFrame, &muted);
if (muted) {
ADD_FAILURE();
return false;
}
EXPECT_EQ(0, ok);
if (ok < 0) {
return false;
}
if (_playoutLengthSmpls == 0) {
return false;
}
_pcmFile.Write10MsData(audioFrame.data(), audioFrame.samples_per_channel_ *
audioFrame.num_channels_);
return true;
}
void Receiver::Run() {
uint8_t counter500Ms = 50;
uint32_t clock = 0;
while (counter500Ms > 0) {
if (clock == 0 || clock >= _nextTime) {
EXPECT_TRUE(IncomingPacket());
if (clock == 0) {
clock = _nextTime;
}
}
if ((clock % 10) == 0) {
if (!PlayoutData()) {
clock++;
continue;
}
}
if (_rtpStream->EndOfFile()) {
counter500Ms--;
}
clock++;
}
}
EncodeDecodeTest::EncodeDecodeTest() = default;
void EncodeDecodeTest::Perform() {
const std::map<int, SdpAudioFormat> send_codecs = {
{107, {"L16", 8000, 1}}, {108, {"L16", 16000, 1}},
{109, {"L16", 32000, 1}}, {0, {"PCMU", 8000, 1}},
{8, {"PCMA", 8000, 1}},
// TODO(bugs.webrtc.org/345525069): Either fix/enable or remove G722 and iLBC.
#if defined(__has_feature) && !__has_feature(undefined_behavior_sanitizer)
#ifdef WEBRTC_CODEC_ILBC
{102, {"ILBC", 8000, 1}},
#endif
{9, {"G722", 8000, 1}},
#endif
};
const Environment env = CreateEnvironment();
int file_num = 0;
for (const auto& send_codec : send_codecs) {
RTPFile rtpFile;
std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create());
std::string fileName = webrtc::test::TempFilename(
webrtc::test::OutputPath(), "encode_decode_rtp");
rtpFile.Open(fileName.c_str(), "wb+");
rtpFile.WriteHeader();
Sender sender;
sender.Setup(env, acm.get(), &rtpFile, "audio_coding/testfile32kHz", 32000,
send_codec.first, send_codec.second);
sender.Run();
sender.Teardown();
rtpFile.Close();
rtpFile.Open(fileName.c_str(), "rb");
rtpFile.ReadHeader();
std::unique_ptr<acm2::AcmReceiver> acm_receiver =
std::make_unique<acm2::AcmReceiver>(
env, acm2::AcmReceiver::Config(CreateBuiltinAudioDecoderFactory()));
Receiver receiver;
receiver.Setup(acm_receiver.get(), &rtpFile, "encodeDecode_out", 1,
file_num);
receiver.Run();
receiver.Teardown();
rtpFile.Close();
file_num++;
}
}
} // namespace webrtc