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Migrate WebRTC test infra to ABSL_FLAG.

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This is the last CL required to migrate WebRTC to ABSL_FLAG, rtc::Flag
will be removed soon after this one lands.

Bug: webrtc:10616
Change-Id: I2807cec39e28a2737d2c49e2dc23f2a6f98d08f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145727
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28606}
This commit is contained in:
Mirko Bonadei
2019-07-18 13:44:12 +02:00
committed by Commit Bot
parent 63741c7fa1
commit 2ab97f6f8e
48 changed files with 1959 additions and 1705 deletions
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184
modules/audio_coding/neteq/tools/neteq_quality_test.cc
View File

@ -14,12 +14,75 @@
#include <cmath>
#include "absl/flags/flag.h"
#include "modules/audio_coding/neteq/tools/neteq_quality_test.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/resample_input_audio_file.h"
#include "rtc_base/checks.h"
#include "test/testsupport/file_utils.h"
const std::string& DefaultInFilename() {
static const std::string path =
::webrtc::test::ResourcePath("audio_coding/speech_mono_16kHz", "pcm");
return path;
}
const std::string& DefaultOutFilename() {
static const std::string path =
::webrtc::test::OutputPath() + "neteq_quality_test_out.pcm";
return path;
}
ABSL_FLAG(
std::string,
in_filename,
DefaultInFilename(),
"Filename for input audio (specify sample rate with --input_sample_rate, "
"and channels with --channels).");
ABSL_FLAG(int, input_sample_rate, 16000, "Sample rate of input file in Hz.");
ABSL_FLAG(int, channels, 1, "Number of channels in input audio.");
ABSL_FLAG(std::string,
out_filename,
DefaultOutFilename(),
"Name of output audio file.");
ABSL_FLAG(
int,
runtime_ms,
10000,
"Simulated runtime (milliseconds). -1 will consume the complete file.");
ABSL_FLAG(int, packet_loss_rate, 10, "Percentile of packet loss.");
ABSL_FLAG(int,
random_loss_mode,
::webrtc::test::kUniformLoss,
"Random loss mode: 0--no loss, 1--uniform loss, 2--Gilbert Elliot "
"loss, 3--fixed loss.");
ABSL_FLAG(int,
burst_length,
30,
"Burst length in milliseconds, only valid for Gilbert Elliot loss.");
ABSL_FLAG(float, drift_factor, 0.0, "Time drift factor.");
ABSL_FLAG(int,
preload_packets,
1,
"Preload the buffer with this many packets.");
ABSL_FLAG(std::string,
loss_events,
"",
"List of loss events time and duration separated by comma: "
"<first_event_time> <first_event_duration>, <second_event_time> "
"<second_event_duration>, ...");
namespace webrtc {
namespace test {
@ -28,17 +91,6 @@ const int kOutputSizeMs = 10;
const int kInitSeed = 0x12345678;
const int kPacketLossTimeUnitMs = 10;
const std::string& DefaultInFilename() {
static const std::string path =
ResourcePath("audio_coding/speech_mono_16kHz", "pcm");
return path;
}
const std::string& DefaultOutFilename() {
static const std::string path = OutputPath() + "neteq_quality_test_out.pcm";
return path;
}
// Common validator for file names.
static bool ValidateFilename(const std::string& value, bool is_output) {
if (!is_output) {
@ -53,51 +105,6 @@ static bool ValidateFilename(const std::string& value, bool is_output) {
return true;
}
WEBRTC_DEFINE_string(
in_filename,
DefaultInFilename().c_str(),
"Filename for input audio (specify sample rate with --input_sample_rate, "
"and channels with --channels).");
WEBRTC_DEFINE_int(input_sample_rate, 16000, "Sample rate of input file in Hz.");
WEBRTC_DEFINE_int(channels, 1, "Number of channels in input audio.");
WEBRTC_DEFINE_string(out_filename,
DefaultOutFilename().c_str(),
"Name of output audio file.");
WEBRTC_DEFINE_int(
runtime_ms,
10000,
"Simulated runtime (milliseconds). -1 will consume the complete file.");
WEBRTC_DEFINE_int(packet_loss_rate, 10, "Percentile of packet loss.");
WEBRTC_DEFINE_int(
random_loss_mode,
kUniformLoss,
"Random loss mode: 0--no loss, 1--uniform loss, 2--Gilbert Elliot "
"loss, 3--fixed loss.");
WEBRTC_DEFINE_int(
burst_length,
30,
"Burst length in milliseconds, only valid for Gilbert Elliot loss.");
WEBRTC_DEFINE_float(drift_factor, 0.0, "Time drift factor.");
WEBRTC_DEFINE_int(preload_packets,
1,
"Preload the buffer with this many packets.");
WEBRTC_DEFINE_string(
loss_events,
"",
"List of loss events time and duration separated by comma: "
"<first_event_time> <first_event_duration>, <second_event_time> "
"<second_event_duration>, ...");
// ProbTrans00Solver() is to calculate the transition probability from no-loss
// state to itself in a modified Gilbert Elliot packet loss model. The result is
// to achieve the target packet loss rate |loss_rate|, when a packet is not
@ -148,11 +155,11 @@ NetEqQualityTest::NetEqQualityTest(
const SdpAudioFormat& format,
const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory)
: audio_format_(format),
channels_(static_cast<size_t>(FLAG_channels)),
channels_(absl::GetFlag(FLAGS_channels)),
decoded_time_ms_(0),
decodable_time_ms_(0),
drift_factor_(FLAG_drift_factor),
packet_loss_rate_(FLAG_packet_loss_rate),
drift_factor_(absl::GetFlag(FLAGS_drift_factor)),
packet_loss_rate_(absl::GetFlag(FLAGS_packet_loss_rate)),
block_duration_ms_(block_duration_ms),
in_sampling_khz_(in_sampling_khz),
out_sampling_khz_(out_sampling_khz),
@ -160,45 +167,50 @@ NetEqQualityTest::NetEqQualityTest(
static_cast<size_t>(in_sampling_khz_ * block_duration_ms_)),
payload_size_bytes_(0),
max_payload_bytes_(0),
in_file_(new ResampleInputAudioFile(FLAG_in_filename,
FLAG_input_sample_rate,
in_sampling_khz * 1000,
FLAG_runtime_ms > 0)),
in_file_(
new ResampleInputAudioFile(absl::GetFlag(FLAGS_in_filename),
absl::GetFlag(FLAGS_input_sample_rate),
in_sampling_khz * 1000,
absl::GetFlag(FLAGS_runtime_ms) > 0)),
rtp_generator_(
new RtpGenerator(in_sampling_khz_, 0, 0, decodable_time_ms_)),
total_payload_size_bytes_(0) {
// Flag validation
RTC_CHECK(ValidateFilename(FLAG_in_filename, false))
RTC_CHECK(ValidateFilename(absl::GetFlag(FLAGS_in_filename), false))
<< "Invalid input filename.";
RTC_CHECK(FLAG_input_sample_rate == 8000 || FLAG_input_sample_rate == 16000 ||
FLAG_input_sample_rate == 32000 || FLAG_input_sample_rate == 48000)
RTC_CHECK(absl::GetFlag(FLAGS_input_sample_rate) == 8000 ||
absl::GetFlag(FLAGS_input_sample_rate) == 16000 ||
absl::GetFlag(FLAGS_input_sample_rate) == 32000 ||
absl::GetFlag(FLAGS_input_sample_rate) == 48000)
<< "Invalid sample rate should be 8000, 16000, 32000 or 48000 Hz.";
RTC_CHECK_EQ(FLAG_channels, 1)
RTC_CHECK_EQ(absl::GetFlag(FLAGS_channels), 1)
<< "Invalid number of channels, current support only 1.";
RTC_CHECK(ValidateFilename(FLAG_out_filename, true))
RTC_CHECK(ValidateFilename(absl::GetFlag(FLAGS_out_filename), true))
<< "Invalid output filename.";
RTC_CHECK(FLAG_packet_loss_rate >= 0 && FLAG_packet_loss_rate <= 100)
RTC_CHECK(absl::GetFlag(FLAGS_packet_loss_rate) >= 0 &&
absl::GetFlag(FLAGS_packet_loss_rate) <= 100)
<< "Invalid packet loss percentile, should be between 0 and 100.";
RTC_CHECK(FLAG_random_loss_mode >= 0 && FLAG_random_loss_mode < kLastLossMode)
RTC_CHECK(absl::GetFlag(FLAGS_random_loss_mode) >= 0 &&
absl::GetFlag(FLAGS_random_loss_mode) < kLastLossMode)
<< "Invalid random packet loss mode, should be between 0 and "
<< kLastLossMode - 1 << ".";
RTC_CHECK_GE(FLAG_burst_length, kPacketLossTimeUnitMs)
RTC_CHECK_GE(absl::GetFlag(FLAGS_burst_length), kPacketLossTimeUnitMs)
<< "Invalid burst length, should be greater than or equal to "
<< kPacketLossTimeUnitMs << " ms.";
RTC_CHECK_GT(FLAG_drift_factor, -0.1)
RTC_CHECK_GT(absl::GetFlag(FLAGS_drift_factor), -0.1)
<< "Invalid drift factor, should be greater than -0.1.";
RTC_CHECK_GE(FLAG_preload_packets, 0)
RTC_CHECK_GE(absl::GetFlag(FLAGS_preload_packets), 0)
<< "Invalid number of packets to preload; must be non-negative.";
const std::string out_filename = FLAG_out_filename;
const std::string out_filename = absl::GetFlag(FLAGS_out_filename);
const std::string log_filename = out_filename + ".log";
log_file_.open(log_filename.c_str(), std::ofstream::out);
RTC_CHECK(log_file_.is_open());
@ -283,7 +295,7 @@ void NetEqQualityTest::SetUp() {
rtp_generator_->set_drift_factor(drift_factor_);
int units = block_duration_ms_ / kPacketLossTimeUnitMs;
switch (FLAG_random_loss_mode) {
switch (absl::GetFlag(FLAGS_random_loss_mode)) {
case kUniformLoss: {
// |unit_loss_rate| is the packet loss rate for each unit time interval
// (kPacketLossTimeUnitMs). Since a packet loss event is generated if any
@ -297,8 +309,8 @@ void NetEqQualityTest::SetUp() {
break;
}
case kGilbertElliotLoss: {
// |FLAG_burst_length| should be integer times of kPacketLossTimeUnitMs.
ASSERT_EQ(0, FLAG_burst_length % kPacketLossTimeUnitMs);
// |FLAGS_burst_length| should be integer times of kPacketLossTimeUnitMs.
ASSERT_EQ(0, absl::GetFlag(FLAGS_burst_length) % kPacketLossTimeUnitMs);
// We do not allow 100 percent packet loss in Gilbert Elliot model, which
// makes no sense.
@ -316,14 +328,15 @@ void NetEqQualityTest::SetUp() {
// prob_trans_00 ^ (units - 1) = (loss_rate - 1) / prob_trans_10 *
// prob_trans_00 + (1 - loss_rate) * (1 + 1 / prob_trans_10).
double loss_rate = 0.01f * packet_loss_rate_;
double prob_trans_10 = 1.0f * kPacketLossTimeUnitMs / FLAG_burst_length;
double prob_trans_10 =
1.0f * kPacketLossTimeUnitMs / absl::GetFlag(FLAGS_burst_length);
double prob_trans_00 = ProbTrans00Solver(units, loss_rate, prob_trans_10);
loss_model_.reset(
new GilbertElliotLoss(1.0f - prob_trans_10, 1.0f - prob_trans_00));
break;
}
case kFixedLoss: {
std::istringstream loss_events_stream(FLAG_loss_events);
std::istringstream loss_events_stream(absl::GetFlag(FLAGS_loss_events));
std::string loss_event_string;
std::set<FixedLossEvent, FixedLossEventCmp> loss_events;
while (std::getline(loss_events_stream, loss_event_string, ',')) {
@ -415,15 +428,18 @@ int NetEqQualityTest::DecodeBlock() {
void NetEqQualityTest::Simulate() {
int audio_size_samples;
bool end_of_input = false;
int runtime_ms = FLAG_runtime_ms >= 0 ? FLAG_runtime_ms : INT_MAX;
int runtime_ms = absl::GetFlag(FLAGS_runtime_ms) >= 0
? absl::GetFlag(FLAGS_runtime_ms)
: INT_MAX;
while (!end_of_input && decoded_time_ms_ < runtime_ms) {
// Preload the buffer if needed.
while (decodable_time_ms_ - FLAG_preload_packets * block_duration_ms_ <
while (decodable_time_ms_ -
absl::GetFlag(FLAGS_preload_packets) * block_duration_ms_ <
decoded_time_ms_) {
if (!in_file_->Read(in_size_samples_ * channels_, &in_data_[0])) {
end_of_input = true;
ASSERT_TRUE(end_of_input && FLAG_runtime_ms < 0);
ASSERT_TRUE(end_of_input && absl::GetFlag(FLAGS_runtime_ms) < 0);
break;
}
payload_.Clear();
@ -438,8 +454,8 @@ void NetEqQualityTest::Simulate() {
}
}
Log() << "Average bit rate was "
<< 8.0f * total_payload_size_bytes_ / FLAG_runtime_ms << " kbps"
<< std::endl;
<< 8.0f * total_payload_size_bytes_ / absl::GetFlag(FLAGS_runtime_ms)
<< " kbps" << std::endl;
}
} // namespace test

1
modules/audio_coding/neteq/tools/neteq_quality_test.h
View File

@ -19,7 +19,6 @@
#include "modules/audio_coding/neteq/tools/audio_sink.h"
#include "modules/audio_coding/neteq/tools/input_audio_file.h"
#include "modules/audio_coding/neteq/tools/rtp_generator.h"
#include "rtc_base/flags.h"
#include "test/gtest.h"
namespace webrtc {

1
modules/audio_coding/neteq/tools/neteq_test_factory.cc
View File

@ -39,7 +39,6 @@
#include "modules/audio_coding/neteq/tools/output_wav_file.h"
#include "modules/audio_coding/neteq/tools/rtp_file_source.h"
#include "rtc_base/checks.h"
#include "rtc_base/flags.h"
#include "rtc_base/ref_counted_object.h"
#include "test/function_audio_decoder_factory.h"
#include "test/testsupport/file_utils.h"

1
modules/audio_coding/neteq/tools/rtp_analyze.cc
View File

@ -18,7 +18,6 @@
#include "absl/flags/parse.h"
#include "modules/audio_coding/neteq/tools/packet.h"
#include "modules/audio_coding/neteq/tools/rtp_file_source.h"
#include "rtc_base/flags.h"
ABSL_FLAG(int, red, 117, "RTP payload type for RED");
ABSL_FLAG(int,

1
modules/audio_coding/neteq/tools/rtp_encode.cc
View File

@ -35,7 +35,6 @@
#include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/audio_coding/neteq/tools/input_audio_file.h"
#include "rtc_base/flags.h"
#include "rtc_base/numerics/safe_conversions.h"
ABSL_FLAG(bool, list_codecs, false, "Enumerate all codecs");