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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/TestAllCodecs.h"
#include <cstdio>
#include <limits>
#include <string>
#include "absl/strings/match.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/environment/environment_factory.h"
#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/logging.h"
#include "rtc_base/string_encode.h"
#include "rtc_base/strings/string_builder.h"
#include "test/gtest.h"
#include "test/testsupport/file_utils.h"
// Description of the test:
// In this test we set up a one-way communication channel from a participant
// called "a" to a participant called "b".
// a -> channel_a_to_b -> b
//
// The test loops through all available mono codecs, encode at "a" sends over
// the channel, and decodes at "b".
#define CHECK_ERROR(f) \
do { \
EXPECT_GE(f, 0) << "Error Calling API"; \
} while (0)
namespace {
const size_t kVariableSize = std::numeric_limits<size_t>::max();
}
namespace webrtc {
// Class for simulating packet handling.
TestPack::TestPack()
: receiver_acm_(NULL),
sequence_number_(0),
timestamp_diff_(0),
last_in_timestamp_(0),
total_bytes_(0),
payload_size_(0) {}
TestPack::~TestPack() {}
void TestPack::RegisterReceiverACM(acm2::AcmReceiver* acm_receiver) {
receiver_acm_ = acm_receiver;
return;
}
int32_t TestPack::SendData(AudioFrameType frame_type,
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
size_t payload_size,
int64_t absolute_capture_timestamp_ms) {
RTPHeader rtp_header;
int32_t status;
rtp_header.markerBit = false;
rtp_header.ssrc = 0;
rtp_header.sequenceNumber = sequence_number_++;
rtp_header.payloadType = payload_type;
rtp_header.timestamp = timestamp;
if (frame_type == AudioFrameType::kEmptyFrame) {
// Skip this frame.
return 0;
}
// Only run mono for all test cases.
memcpy(payload_data_, payload_data, payload_size);
status = receiver_acm_->InsertPacket(
rtp_header, rtc::ArrayView<const uint8_t>(payload_data_, payload_size),
/*receive_time=*/Timestamp::MinusInfinity());
payload_size_ = payload_size;
timestamp_diff_ = timestamp - last_in_timestamp_;
last_in_timestamp_ = timestamp;
total_bytes_ += payload_size;
return status;
}
size_t TestPack::payload_size() {
return payload_size_;
}
uint32_t TestPack::timestamp_diff() {
return timestamp_diff_;
}
void TestPack::reset_payload_size() {
payload_size_ = 0;
}
TestAllCodecs::TestAllCodecs()
: env_(CreateEnvironment()),
acm_a_(AudioCodingModule::Create()),
acm_b_(std::make_unique<acm2::AcmReceiver>(
env_,
acm2::AcmReceiver::Config(CreateBuiltinAudioDecoderFactory()))),
channel_a_to_b_(NULL),
test_count_(0),
packet_size_samples_(0),
packet_size_bytes_(0) {}
TestAllCodecs::~TestAllCodecs() {
if (channel_a_to_b_ != NULL) {
delete channel_a_to_b_;
channel_a_to_b_ = NULL;
}
}
void TestAllCodecs::Perform() {
const std::string file_name =
webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
infile_a_.Open(file_name, 32000, "rb");
acm_b_->SetCodecs({{107, {"L16", 8000, 1}},
{108, {"L16", 16000, 1}},
{109, {"L16", 32000, 1}},
{111, {"L16", 8000, 2}},
{112, {"L16", 16000, 2}},
{113, {"L16", 32000, 2}},
{0, {"PCMU", 8000, 1}},
{110, {"PCMU", 8000, 2}},
{8, {"PCMA", 8000, 1}},
{118, {"PCMA", 8000, 2}},
{102, {"ILBC", 8000, 1}},
{9, {"G722", 8000, 1}},
{119, {"G722", 8000, 2}},
{120, {"OPUS", 48000, 2, {{"stereo", "1"}}}},
{13, {"CN", 8000, 1}},
{98, {"CN", 16000, 1}},
{99, {"CN", 32000, 1}}});
// Create and connect the channel
channel_a_to_b_ = new TestPack;
acm_a_->RegisterTransportCallback(channel_a_to_b_);
channel_a_to_b_->RegisterReceiverACM(acm_b_.get());
// All codecs are tested for all allowed sampling frequencies, rates and
// packet sizes.
// TODO(bugs.webrtc.org/345525069): Either fix/enable or remove G722.
#if defined(__has_feature) && !__has_feature(undefined_behavior_sanitizer)
test_count_++;
OpenOutFile(test_count_);
char codec_g722[] = "G722";
RegisterSendCodec(codec_g722, 16000, 64000, 160, 0);
Run(channel_a_to_b_);
RegisterSendCodec(codec_g722, 16000, 64000, 320, 0);
Run(channel_a_to_b_);
RegisterSendCodec(codec_g722, 16000, 64000, 480, 0);
Run(channel_a_to_b_);
RegisterSendCodec(codec_g722, 16000, 64000, 640, 0);
Run(channel_a_to_b_);
RegisterSendCodec(codec_g722, 16000, 64000, 800, 0);
Run(channel_a_to_b_);
RegisterSendCodec(codec_g722, 16000, 64000, 960, 0);
Run(channel_a_to_b_);
outfile_b_.Close();
#endif
// TODO(bugs.webrtc.org/345525069): Either fix/enable or remove iLBC.
#if defined(__has_feature) && !__has_feature(undefined_behavior_sanitizer)
#ifdef WEBRTC_CODEC_ILBC
test_count_++;
OpenOutFile(test_count_);
char codec_ilbc[] = "ILBC";
RegisterSendCodec(codec_ilbc, 8000, 13300, 240, 0);
Run(channel_a_to_b_);
RegisterSendCodec(codec_ilbc, 8000, 13300, 480, 0);
Run(channel_a_to_b_);
RegisterSendCodec(codec_ilbc, 8000, 15200, 160, 0);
Run(channel_a_to_b_);
RegisterSendCodec(codec_ilbc, 8000, 15200, 320, 0);
Run(channel_a_to_b_);
outfile_b_.Close();
#endif
#endif
test_count_++;
OpenOutFile(test_count_);
char codec_l16[] = "L16";
RegisterSendCodec(codec_l16, 8000, 128000, 80, 0);
Run(channel_a_to_b_);
RegisterSendCodec(codec_l16, 8000, 128000, 160, 0);
Run(channel_a_to_b_);
RegisterSendCodec(codec_l16, 8000, 128000, 240, 0);
Run(channel_a_to_b_);
RegisterSendCodec(codec_l16, 8000, 128000, 320, 0);
Run(channel_a_to_b_);
outfile_b_.Close();
test_count_++;
OpenOutFile(test_count_);
RegisterSendCodec(codec_l16, 16000, 256000, 160, 0);
Run(channel_a_to_b_);
RegisterSendCodec(codec_l16, 16000, 256000, 320, 0);
Run(channel_a_to_b_);
RegisterSendCodec(codec_l16, 16000, 256000, 480, 0);
Run(channel_a_to_b_);
RegisterSendCodec(codec_l16, 16000, 256000, 640, 0);
Run(channel_a_to_b_);
outfile_b_.Close();
test_count_++;
OpenOutFile(test_count_);
RegisterSendCodec(codec_l16, 32000, 512000, 320, 0);
Run(channel_a_to_b_);
RegisterSendCodec(codec_l16, 32000, 512000, 640, 0);
Run(channel_a_to_b_);
outfile_b_.Close();
test_count_++;
OpenOutFile(test_count_);
char codec_pcma[] = "PCMA";
RegisterSendCodec(codec_pcma, 8000, 64000, 80, 0);
Run(channel_a_to_b_);
RegisterSendCodec(codec_pcma, 8000, 64000, 160, 0);
Run(channel_a_to_b_);
RegisterSendCodec(codec_pcma, 8000, 64000, 240, 0);
Run(channel_a_to_b_);
RegisterSendCodec(codec_pcma, 8000, 64000, 320, 0);
Run(channel_a_to_b_);
RegisterSendCodec(codec_pcma, 8000, 64000, 400, 0);
Run(channel_a_to_b_);
RegisterSendCodec(codec_pcma, 8000, 64000, 480, 0);
Run(channel_a_to_b_);
char codec_pcmu[] = "PCMU";
RegisterSendCodec(codec_pcmu, 8000, 64000, 80, 0);
Run(channel_a_to_b_);
RegisterSendCodec(codec_pcmu, 8000, 64000, 160, 0);
Run(channel_a_to_b_);
RegisterSendCodec(codec_pcmu, 8000, 64000, 240, 0);
Run(channel_a_to_b_);
RegisterSendCodec(codec_pcmu, 8000, 64000, 320, 0);
Run(channel_a_to_b_);
RegisterSendCodec(codec_pcmu, 8000, 64000, 400, 0);
Run(channel_a_to_b_);
RegisterSendCodec(codec_pcmu, 8000, 64000, 480, 0);
Run(channel_a_to_b_);
outfile_b_.Close();
#ifdef WEBRTC_CODEC_OPUS
test_count_++;
OpenOutFile(test_count_);
char codec_opus[] = "OPUS";
RegisterSendCodec(codec_opus, 48000, 6000, 480, kVariableSize);
Run(channel_a_to_b_);
RegisterSendCodec(codec_opus, 48000, 20000, 480 * 2, kVariableSize);
Run(channel_a_to_b_);
RegisterSendCodec(codec_opus, 48000, 32000, 480 * 4, kVariableSize);
Run(channel_a_to_b_);
RegisterSendCodec(codec_opus, 48000, 48000, 480, kVariableSize);
Run(channel_a_to_b_);
RegisterSendCodec(codec_opus, 48000, 64000, 480 * 4, kVariableSize);
Run(channel_a_to_b_);
RegisterSendCodec(codec_opus, 48000, 96000, 480 * 6, kVariableSize);
Run(channel_a_to_b_);
RegisterSendCodec(codec_opus, 48000, 500000, 480 * 2, kVariableSize);
Run(channel_a_to_b_);
outfile_b_.Close();
#endif
}
// Register Codec to use in the test
//
// Input: codec_name - name to use when register the codec
// sampling_freq_hz - sampling frequency in Herz
// rate - bitrate in bytes
// packet_size - packet size in samples
// extra_byte - if extra bytes needed compared to the bitrate
// used when registering, can be an internal header
// set to kVariableSize if the codec is a variable
// rate codec
void TestAllCodecs::RegisterSendCodec(char* codec_name,
int32_t sampling_freq_hz,
int rate,
int packet_size,
size_t extra_byte) {
// Store packet-size in samples, used to validate the received packet.
// If G.722, store half the size to compensate for the timestamp bug in the
// RFC for G.722.
int clockrate_hz = sampling_freq_hz;
size_t num_channels = 1;
if (absl::EqualsIgnoreCase(codec_name, "G722")) {
packet_size_samples_ = packet_size / 2;
clockrate_hz = sampling_freq_hz / 2;
} else if (absl::EqualsIgnoreCase(codec_name, "OPUS")) {
packet_size_samples_ = packet_size;
num_channels = 2;
} else {
packet_size_samples_ = packet_size;
}
// Store the expected packet size in bytes, used to validate the received
// packet. If variable rate codec (extra_byte == -1), set to -1.
if (extra_byte != kVariableSize) {
// Add 0.875 to always round up to a whole byte
packet_size_bytes_ =
static_cast<size_t>(static_cast<float>(packet_size * rate) /
static_cast<float>(sampling_freq_hz * 8) +
0.875) +
extra_byte;
} else {
// Packets will have a variable size.
packet_size_bytes_ = kVariableSize;
}
auto factory = CreateBuiltinAudioEncoderFactory();
SdpAudioFormat format = {codec_name, clockrate_hz, num_channels};
format.parameters["ptime"] = rtc::ToString(rtc::CheckedDivExact(
packet_size, rtc::CheckedDivExact(sampling_freq_hz, 1000)));
acm_a_->SetEncoder(factory->Create(env_, format, {.payload_type = 17}));
}
void TestAllCodecs::Run(TestPack* channel) {
AudioFrame audio_frame;
int32_t out_freq_hz = outfile_b_.SamplingFrequency();
size_t receive_size;
uint32_t timestamp_diff;
channel->reset_payload_size();
int error_count = 0;
int counter = 0;
// Set test length to 500 ms (50 blocks of 10 ms each).
infile_a_.SetNum10MsBlocksToRead(50);
// Fast-forward 1 second (100 blocks) since the file starts with silence.
infile_a_.FastForward(100);
while (!infile_a_.EndOfFile()) {
// Add 10 msec to ACM.
infile_a_.Read10MsData(audio_frame);
CHECK_ERROR(acm_a_->Add10MsData(audio_frame));
// Verify that the received packet size matches the settings.
receive_size = channel->payload_size();
if (receive_size) {
if ((receive_size != packet_size_bytes_) &&
(packet_size_bytes_ != kVariableSize)) {
error_count++;
}
// Verify that the timestamp is updated with expected length. The counter
// is used to avoid problems when switching codec or frame size in the
// test.
timestamp_diff = channel->timestamp_diff();
if ((counter > 10) &&
(static_cast<int>(timestamp_diff) != packet_size_samples_) &&
(packet_size_samples_ > -1))
error_count++;
}
// Run received side of ACM.
bool muted;
CHECK_ERROR(acm_b_->GetAudio(out_freq_hz, &audio_frame, &muted));
ASSERT_FALSE(muted);
// Write output speech to file.
outfile_b_.Write10MsData(audio_frame.data(),
audio_frame.samples_per_channel_);
// Update loop counter
counter++;
}
EXPECT_EQ(0, error_count);
if (infile_a_.EndOfFile()) {
infile_a_.Rewind();
}
}
void TestAllCodecs::OpenOutFile(int test_number) {
std::string filename = webrtc::test::OutputPath();
rtc::StringBuilder test_number_str;
test_number_str << test_number;
filename += "testallcodecs_out_";
filename += test_number_str.str();
filename += ".pcm";
outfile_b_.Open(filename, 32000, "wb");
}
} // namespace webrtc