446 lines
15 KiB
C++
446 lines
15 KiB
C++
/*
|
|
* 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 "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(AudioCodingModule* acm) {
|
|
receiver_acm_ = acm;
|
|
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_->IncomingPacket(payload_data_, payload_size, rtp_header);
|
|
|
|
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()
|
|
: acm_a_(AudioCodingModule::Create(
|
|
AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))),
|
|
acm_b_(AudioCodingModule::Create(
|
|
AudioCodingModule::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_a_->InitializeReceiver();
|
|
acm_b_->InitializeReceiver();
|
|
|
|
acm_b_->SetReceiveCodecs({{103, {"ISAC", 16000, 1}},
|
|
{104, {"ISAC", 32000, 1}},
|
|
{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.
|
|
test_count_++;
|
|
OpenOutFile(test_count_);
|
|
char codec_g722[] = "G722";
|
|
RegisterSendCodec('A', codec_g722, 16000, 64000, 160, 0);
|
|
Run(channel_a_to_b_);
|
|
RegisterSendCodec('A', codec_g722, 16000, 64000, 320, 0);
|
|
Run(channel_a_to_b_);
|
|
RegisterSendCodec('A', codec_g722, 16000, 64000, 480, 0);
|
|
Run(channel_a_to_b_);
|
|
RegisterSendCodec('A', codec_g722, 16000, 64000, 640, 0);
|
|
Run(channel_a_to_b_);
|
|
RegisterSendCodec('A', codec_g722, 16000, 64000, 800, 0);
|
|
Run(channel_a_to_b_);
|
|
RegisterSendCodec('A', codec_g722, 16000, 64000, 960, 0);
|
|
Run(channel_a_to_b_);
|
|
outfile_b_.Close();
|
|
#ifdef WEBRTC_CODEC_ILBC
|
|
test_count_++;
|
|
OpenOutFile(test_count_);
|
|
char codec_ilbc[] = "ILBC";
|
|
RegisterSendCodec('A', codec_ilbc, 8000, 13300, 240, 0);
|
|
Run(channel_a_to_b_);
|
|
RegisterSendCodec('A', codec_ilbc, 8000, 13300, 480, 0);
|
|
Run(channel_a_to_b_);
|
|
RegisterSendCodec('A', codec_ilbc, 8000, 15200, 160, 0);
|
|
Run(channel_a_to_b_);
|
|
RegisterSendCodec('A', codec_ilbc, 8000, 15200, 320, 0);
|
|
Run(channel_a_to_b_);
|
|
outfile_b_.Close();
|
|
#endif
|
|
#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
|
|
test_count_++;
|
|
OpenOutFile(test_count_);
|
|
char codec_isac[] = "ISAC";
|
|
RegisterSendCodec('A', codec_isac, 16000, -1, 480, kVariableSize);
|
|
Run(channel_a_to_b_);
|
|
RegisterSendCodec('A', codec_isac, 16000, -1, 960, kVariableSize);
|
|
Run(channel_a_to_b_);
|
|
RegisterSendCodec('A', codec_isac, 16000, 15000, 480, kVariableSize);
|
|
Run(channel_a_to_b_);
|
|
RegisterSendCodec('A', codec_isac, 16000, 32000, 960, kVariableSize);
|
|
Run(channel_a_to_b_);
|
|
outfile_b_.Close();
|
|
#endif
|
|
#ifdef WEBRTC_CODEC_ISAC
|
|
test_count_++;
|
|
OpenOutFile(test_count_);
|
|
RegisterSendCodec('A', codec_isac, 32000, -1, 960, kVariableSize);
|
|
Run(channel_a_to_b_);
|
|
RegisterSendCodec('A', codec_isac, 32000, 56000, 960, kVariableSize);
|
|
Run(channel_a_to_b_);
|
|
RegisterSendCodec('A', codec_isac, 32000, 37000, 960, kVariableSize);
|
|
Run(channel_a_to_b_);
|
|
RegisterSendCodec('A', codec_isac, 32000, 32000, 960, kVariableSize);
|
|
Run(channel_a_to_b_);
|
|
outfile_b_.Close();
|
|
#endif
|
|
test_count_++;
|
|
OpenOutFile(test_count_);
|
|
char codec_l16[] = "L16";
|
|
RegisterSendCodec('A', codec_l16, 8000, 128000, 80, 0);
|
|
Run(channel_a_to_b_);
|
|
RegisterSendCodec('A', codec_l16, 8000, 128000, 160, 0);
|
|
Run(channel_a_to_b_);
|
|
RegisterSendCodec('A', codec_l16, 8000, 128000, 240, 0);
|
|
Run(channel_a_to_b_);
|
|
RegisterSendCodec('A', codec_l16, 8000, 128000, 320, 0);
|
|
Run(channel_a_to_b_);
|
|
outfile_b_.Close();
|
|
|
|
test_count_++;
|
|
OpenOutFile(test_count_);
|
|
RegisterSendCodec('A', codec_l16, 16000, 256000, 160, 0);
|
|
Run(channel_a_to_b_);
|
|
RegisterSendCodec('A', codec_l16, 16000, 256000, 320, 0);
|
|
Run(channel_a_to_b_);
|
|
RegisterSendCodec('A', codec_l16, 16000, 256000, 480, 0);
|
|
Run(channel_a_to_b_);
|
|
RegisterSendCodec('A', codec_l16, 16000, 256000, 640, 0);
|
|
Run(channel_a_to_b_);
|
|
outfile_b_.Close();
|
|
|
|
test_count_++;
|
|
OpenOutFile(test_count_);
|
|
RegisterSendCodec('A', codec_l16, 32000, 512000, 320, 0);
|
|
Run(channel_a_to_b_);
|
|
RegisterSendCodec('A', codec_l16, 32000, 512000, 640, 0);
|
|
Run(channel_a_to_b_);
|
|
outfile_b_.Close();
|
|
|
|
test_count_++;
|
|
OpenOutFile(test_count_);
|
|
char codec_pcma[] = "PCMA";
|
|
RegisterSendCodec('A', codec_pcma, 8000, 64000, 80, 0);
|
|
Run(channel_a_to_b_);
|
|
RegisterSendCodec('A', codec_pcma, 8000, 64000, 160, 0);
|
|
Run(channel_a_to_b_);
|
|
RegisterSendCodec('A', codec_pcma, 8000, 64000, 240, 0);
|
|
Run(channel_a_to_b_);
|
|
RegisterSendCodec('A', codec_pcma, 8000, 64000, 320, 0);
|
|
Run(channel_a_to_b_);
|
|
RegisterSendCodec('A', codec_pcma, 8000, 64000, 400, 0);
|
|
Run(channel_a_to_b_);
|
|
RegisterSendCodec('A', codec_pcma, 8000, 64000, 480, 0);
|
|
Run(channel_a_to_b_);
|
|
|
|
char codec_pcmu[] = "PCMU";
|
|
RegisterSendCodec('A', codec_pcmu, 8000, 64000, 80, 0);
|
|
Run(channel_a_to_b_);
|
|
RegisterSendCodec('A', codec_pcmu, 8000, 64000, 160, 0);
|
|
Run(channel_a_to_b_);
|
|
RegisterSendCodec('A', codec_pcmu, 8000, 64000, 240, 0);
|
|
Run(channel_a_to_b_);
|
|
RegisterSendCodec('A', codec_pcmu, 8000, 64000, 320, 0);
|
|
Run(channel_a_to_b_);
|
|
RegisterSendCodec('A', codec_pcmu, 8000, 64000, 400, 0);
|
|
Run(channel_a_to_b_);
|
|
RegisterSendCodec('A', 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('A', codec_opus, 48000, 6000, 480, kVariableSize);
|
|
Run(channel_a_to_b_);
|
|
RegisterSendCodec('A', codec_opus, 48000, 20000, 480 * 2, kVariableSize);
|
|
Run(channel_a_to_b_);
|
|
RegisterSendCodec('A', codec_opus, 48000, 32000, 480 * 4, kVariableSize);
|
|
Run(channel_a_to_b_);
|
|
RegisterSendCodec('A', codec_opus, 48000, 48000, 480, kVariableSize);
|
|
Run(channel_a_to_b_);
|
|
RegisterSendCodec('A', codec_opus, 48000, 64000, 480 * 4, kVariableSize);
|
|
Run(channel_a_to_b_);
|
|
RegisterSendCodec('A', codec_opus, 48000, 96000, 480 * 6, kVariableSize);
|
|
Run(channel_a_to_b_);
|
|
RegisterSendCodec('A', codec_opus, 48000, 500000, 480 * 2, kVariableSize);
|
|
Run(channel_a_to_b_);
|
|
outfile_b_.Close();
|
|
#endif
|
|
}
|
|
|
|
// Register Codec to use in the test
|
|
//
|
|
// Input: side - which ACM to use, 'A' or 'B'
|
|
// 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 side,
|
|
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.
|
|
// If iSAC runs in adaptive mode, packet size in samples can change on the
|
|
// fly, so we exclude this test by setting |packet_size_samples_| to -1.
|
|
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, "ISAC") && (rate == -1)) {
|
|
packet_size_samples_ = -1;
|
|
} 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;
|
|
}
|
|
|
|
// Set pointer to the ACM where to register the codec.
|
|
AudioCodingModule* my_acm = NULL;
|
|
switch (side) {
|
|
case 'A': {
|
|
my_acm = acm_a_.get();
|
|
break;
|
|
}
|
|
case 'B': {
|
|
my_acm = acm_b_.get();
|
|
break;
|
|
}
|
|
default: {
|
|
break;
|
|
}
|
|
}
|
|
ASSERT_TRUE(my_acm != NULL);
|
|
|
|
auto factory = CreateBuiltinAudioEncoderFactory();
|
|
constexpr int payload_type = 17;
|
|
SdpAudioFormat format = {codec_name, clockrate_hz, num_channels};
|
|
format.parameters["ptime"] = rtc::ToString(rtc::CheckedDivExact(
|
|
packet_size, rtc::CheckedDivExact(sampling_freq_hz, 1000)));
|
|
my_acm->SetEncoder(
|
|
factory->MakeAudioEncoder(payload_type, format, absl::nullopt));
|
|
}
|
|
|
|
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_->PlayoutData10Ms(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
|