/*
* 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 <assert.h>
#include <math.h>
#include <iostream>
#include "gflags/gflags.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/common.h"
#include "webrtc/common_types.h"
#include "webrtc/engine_configurations.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
#include "webrtc/modules/audio_coding/main/test/Channel.h"
#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
#include "webrtc/modules/audio_coding/main/test/utility.h"
#include "webrtc/system_wrappers/interface/event_wrapper.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/test/testsupport/fileutils.h"
DEFINE_string(codec, "isac", "Codec Name");
DEFINE_int32(sample_rate_hz, 16000, "Sampling rate in Hertz.");
DEFINE_int32(num_channels, 1, "Number of Channels.");
DEFINE_string(input_file, "", "Input file, PCM16 32 kHz, optional.");
DEFINE_int32(delay, 0, "Delay in millisecond.");
DEFINE_int32(init_delay, 0, "Initial delay in millisecond.");
DEFINE_bool(dtx, false, "Enable DTX at the sender side.");
DEFINE_bool(packet_loss, false, "Apply packet loss, c.f. Channel{.cc, .h}.");
DEFINE_bool(fec, false, "Use Forward Error Correction (FEC).");
namespace webrtc {
namespace {
struct CodecSettings {
char name[50];
int sample_rate_hz;
int num_channels;
};
struct AcmSettings {
bool dtx;
bool fec;
};
struct TestSettings {
CodecSettings codec;
AcmSettings acm;
bool packet_loss;
};
} // namespace
class DelayTest {
public:
DelayTest()
: acm_a_(AudioCodingModule::Create(0)),
acm_b_(AudioCodingModule::Create(1)),
channel_a2b_(new Channel),
test_cntr_(0),
encoding_sample_rate_hz_(8000) {}
~DelayTest() {
if (channel_a2b_ != NULL) {
delete channel_a2b_;
channel_a2b_ = NULL;
}
in_file_a_.Close();
}
void Initialize() {
test_cntr_ = 0;
std::string file_name = webrtc::test::ResourcePath(
"audio_coding/testfile32kHz", "pcm");
if (FLAGS_input_file.size() > 0)
file_name = FLAGS_input_file;
in_file_a_.Open(file_name, 32000, "rb");
ASSERT_EQ(0, acm_a_->InitializeReceiver()) <<
"Couldn't initialize receiver.\n";
ASSERT_EQ(0, acm_b_->InitializeReceiver()) <<
"Couldn't initialize receiver.\n";
if (FLAGS_init_delay > 0) {
ASSERT_EQ(0, acm_b_->SetInitialPlayoutDelay(FLAGS_init_delay)) <<
"Failed to set initial delay.\n";
}
if (FLAGS_delay > 0) {
ASSERT_EQ(0, acm_b_->SetMinimumPlayoutDelay(FLAGS_delay)) <<
"Failed to set minimum delay.\n";
}
int num_encoders = acm_a_->NumberOfCodecs();
CodecInst my_codec_param;
for (int n = 0; n < num_encoders; n++) {
EXPECT_EQ(0, acm_b_->Codec(n, &my_codec_param)) <<
"Failed to get codec.";
if (STR_CASE_CMP(my_codec_param.plname, "opus") == 0)
my_codec_param.channels = 1;
else if (my_codec_param.channels > 1)
continue;
if (STR_CASE_CMP(my_codec_param.plname, "CN") == 0 &&
my_codec_param.plfreq == 48000)
continue;
if (STR_CASE_CMP(my_codec_param.plname, "telephone-event") == 0)
continue;
ASSERT_EQ(0, acm_b_->RegisterReceiveCodec(my_codec_param)) <<
"Couldn't register receive codec.\n";
}
// Create and connect the channel
ASSERT_EQ(0, acm_a_->RegisterTransportCallback(channel_a2b_)) <<
"Couldn't register Transport callback.\n";
channel_a2b_->RegisterReceiverACM(acm_b_.get());
}
void Perform(const TestSettings* config, size_t num_tests, int duration_sec,
const char* output_prefix) {
for (size_t n = 0; n < num_tests; ++n) {
ApplyConfig(config[n]);
Run(duration_sec, output_prefix);
}
}
private:
void ApplyConfig(const TestSettings& config) {
printf("====================================\n");
printf("Test %d \n"
"Codec: %s, %d kHz, %d channel(s)\n"
"ACM: DTX %s, FEC %s\n"
"Channel: %s\n",
++test_cntr_, config.codec.name, config.codec.sample_rate_hz,
config.codec.num_channels, config.acm.dtx ? "on" : "off",
config.acm.fec ? "on" : "off",
config.packet_loss ? "with packet-loss" : "no packet-loss");
SendCodec(config.codec);
ConfigAcm(config.acm);
ConfigChannel(config.packet_loss);
}
void SendCodec(const CodecSettings& config) {
CodecInst my_codec_param;
ASSERT_EQ(0, AudioCodingModule::Codec(
config.name, &my_codec_param, config.sample_rate_hz,
config.num_channels)) << "Specified codec is not supported.\n";
encoding_sample_rate_hz_ = my_codec_param.plfreq;
ASSERT_EQ(0, acm_a_->RegisterSendCodec(my_codec_param)) <<
"Failed to register send-codec.\n";
}
void ConfigAcm(const AcmSettings& config) {
ASSERT_EQ(0, acm_a_->SetVAD(config.dtx, config.dtx, VADAggr)) <<
"Failed to set VAD.\n";
ASSERT_EQ(0, acm_a_->SetREDStatus(config.fec)) <<
"Failed to set RED.\n";
}
void ConfigChannel(bool packet_loss) {
channel_a2b_->SetFECTestWithPacketLoss(packet_loss);
}
void OpenOutFile(const char* output_id) {
std::stringstream file_stream;
file_stream << "delay_test_" << FLAGS_codec << "_" << FLAGS_sample_rate_hz
<< "Hz" << "_" << FLAGS_init_delay << "ms_" << FLAGS_delay << "ms.pcm";
std::cout << "Output file: " << file_stream.str() << std::endl << std::endl;
std::string file_name = webrtc::test::OutputPath() + file_stream.str();
out_file_b_.Open(file_name.c_str(), 32000, "wb");
}
void Run(int duration_sec, const char* output_prefix) {
OpenOutFile(output_prefix);
AudioFrame audio_frame;
uint32_t out_freq_hz_b = out_file_b_.SamplingFrequency();
int num_frames = 0;
int in_file_frames = 0;
uint32_t playout_ts;
uint32_t received_ts;
double average_delay = 0;
double inst_delay_sec = 0;
while (num_frames < (duration_sec * 100)) {
if (in_file_a_.EndOfFile()) {
in_file_a_.Rewind();
}
// Print delay information every 16 frame
if ((num_frames & 0x3F) == 0x3F) {
ACMNetworkStatistics statistics;
acm_b_->NetworkStatistics(&statistics);
fprintf(stdout, "delay: min=%3d max=%3d mean=%3d median=%3d"
" ts-based average = %6.3f, "
"curr buff-lev = %4u opt buff-lev = %4u \n",
statistics.minWaitingTimeMs, statistics.maxWaitingTimeMs,
statistics.meanWaitingTimeMs, statistics.medianWaitingTimeMs,
average_delay, statistics.currentBufferSize,
statistics.preferredBufferSize);
fflush (stdout);
}
in_file_a_.Read10MsData(audio_frame);
ASSERT_EQ(0, acm_a_->Add10MsData(audio_frame));
ASSERT_LE(0, acm_a_->Process());
ASSERT_EQ(0, acm_b_->PlayoutData10Ms(out_freq_hz_b, &audio_frame));
out_file_b_.Write10MsData(
audio_frame.data_,
audio_frame.samples_per_channel_ * audio_frame.num_channels_);
acm_b_->PlayoutTimestamp(&playout_ts);
received_ts = channel_a2b_->LastInTimestamp();
inst_delay_sec = static_cast<uint32_t>(received_ts - playout_ts)
/ static_cast<double>(encoding_sample_rate_hz_);
if (num_frames > 10)
average_delay = 0.95 * average_delay + 0.05 * inst_delay_sec;
++num_frames;
++in_file_frames;
}
out_file_b_.Close();
}
scoped_ptr<AudioCodingModule> acm_a_;
scoped_ptr<AudioCodingModule> acm_b_;
Channel* channel_a2b_;
PCMFile in_file_a_;
PCMFile out_file_b_;
int test_cntr_;
int encoding_sample_rate_hz_;
};
} // namespace webrtc
int main(int argc, char* argv[]) {
google::ParseCommandLineFlags(&argc, &argv, true);
webrtc::TestSettings test_setting;
strcpy(test_setting.codec.name, FLAGS_codec.c_str());
if (FLAGS_sample_rate_hz != 8000 &&
FLAGS_sample_rate_hz != 16000 &&
FLAGS_sample_rate_hz != 32000 &&
FLAGS_sample_rate_hz != 48000) {
std::cout << "Invalid sampling rate.\n";
return 1;
}
test_setting.codec.sample_rate_hz = FLAGS_sample_rate_hz;
if (FLAGS_num_channels < 1 || FLAGS_num_channels > 2) {
std::cout << "Only mono and stereo are supported.\n";
return 1;
}
test_setting.codec.num_channels = FLAGS_num_channels;
test_setting.acm.dtx = FLAGS_dtx;
test_setting.acm.fec = FLAGS_fec;
test_setting.packet_loss = FLAGS_packet_loss;
webrtc::DelayTest delay_test;
delay_test.Initialize();
delay_test.Perform(&test_setting, 1, 240, "delay_test");
return 0;
}