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Migrate WebRTC test infra to ABSL_FLAG.

Browse Source 
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
Download Patch File Download Diff File Expand all files Collapse all files

15
modules/audio_coding/BUILD.gn
View File

@ -1593,10 +1593,10 @@ if (rtc_include_tests) {
":neteq_test_tools",
"../../api/audio_codecs:builtin_audio_decoder_factory",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
"../../test:fileutils",
"../../test:test_support",
"//testing/gtest",
"//third_party/abseil-cpp/absl/flags:flag",
]
}
@ -1617,7 +1617,7 @@ if (rtc_include_tests) {
"../../api/audio_codecs/ilbc:audio_encoder_ilbc",
"../../api/audio_codecs/isac:audio_encoder_isac",
"../../api/audio_codecs/opus:audio_encoder_opus",
"../../rtc_base:rtc_base_approved",
"../../rtc_base:safe_conversions",
]
sources = [
@ -1669,7 +1669,6 @@ if (rtc_include_tests) {
":neteq",
":neteq_test_tools",
":pcm16b",
"../../rtc_base:rtc_base_approved",
"//testing/gtest",
"//third_party/abseil-cpp/absl/flags:flag",
"//third_party/abseil-cpp/absl/flags:parse",
@ -1691,6 +1690,7 @@ if (rtc_include_tests) {
"../../rtc_base:rtc_base_approved",
"../../test:test_main",
"//testing/gtest",
"//third_party/abseil-cpp/absl/flags:flag",
]
}
@ -1704,8 +1704,10 @@ if (rtc_include_tests) {
deps = [
":neteq",
":neteq_test_support",
"../../rtc_base:rtc_base_approved",
"../../rtc_base:checks",
"../../test:test_support",
"//third_party/abseil-cpp/absl/flags:flag",
"//third_party/abseil-cpp/absl/flags:parse",
]
}
@ -1726,6 +1728,7 @@ if (rtc_include_tests) {
"../../test:fileutils",
"../../test:test_main",
"//testing/gtest",
"//third_party/abseil-cpp/absl/flags:flag",
]
}
@ -1743,6 +1746,7 @@ if (rtc_include_tests) {
"../../rtc_base:rtc_base_approved",
"../../test:test_main",
"//testing/gtest",
"//third_party/abseil-cpp/absl/flags:flag",
]
}
@ -1762,6 +1766,7 @@ if (rtc_include_tests) {
"../../test:fileutils",
"../../test:test_main",
"//testing/gtest",
"//third_party/abseil-cpp/absl/flags:flag",
]
}
@ -1781,6 +1786,7 @@ if (rtc_include_tests) {
"../../test:fileutils",
"../../test:test_main",
"//testing/gtest",
"//third_party/abseil-cpp/absl/flags:flag",
]
}
@ -2047,6 +2053,7 @@ if (rtc_include_tests) {
"codecs/opus/test",
"codecs/opus/test:test_unittest",
"//testing/gtest",
"//third_party/abseil-cpp/absl/flags:flag",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/types:optional",
]

12
modules/audio_coding/neteq/neteq_unittest.cc
View File

@ -20,6 +20,7 @@
#include <string>
#include <vector>
#include "absl/flags/flag.h"
#include "api/audio/audio_frame.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
@ -40,9 +41,6 @@
#include "test/gtest.h"
#include "test/testsupport/file_utils.h"
// This must come after test/gtest.h
#include "rtc_base/flags.h" // NOLINT(build/include)
#ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
RTC_PUSH_IGNORING_WUNDEF()
#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
@ -53,7 +51,7 @@ RTC_PUSH_IGNORING_WUNDEF()
RTC_POP_IGNORING_WUNDEF()
#endif
WEBRTC_DEFINE_bool(gen_ref, false, "Generate reference files.");
ABSL_FLAG(bool, gen_ref, false, "Generate reference files.");
namespace webrtc {
@ -470,7 +468,7 @@ TEST_F(NetEqDecodingTest, MAYBE_TestBitExactness) {
"3689c9f0ab9e50cefab3e44c37c3d7aa0de82ca4");
DecodeAndCompare(input_rtp_file, output_checksum, network_stats_checksum,
FLAG_gen_ref);
absl::GetFlag(FLAGS_gen_ref));
}
#if !defined(WEBRTC_IOS) && defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) && \
@ -499,7 +497,7 @@ TEST_F(NetEqDecodingTest, MAYBE_TestOpusBitExactness) {
"0b3d34baffaf651812ffaf06ea1b5ce45ea1c47a");
DecodeAndCompare(input_rtp_file, output_checksum, network_stats_checksum,
FLAG_gen_ref);
absl::GetFlag(FLAGS_gen_ref));
}
#if !defined(WEBRTC_IOS) && defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) && \
@ -523,7 +521,7 @@ TEST_F(NetEqDecodingTest, MAYBE_TestOpusDtxBitExactness) {
"bab58dc587d956f326056d7340c96eb9d2d3cc21";
DecodeAndCompare(input_rtp_file, output_checksum, network_stats_checksum,
FLAG_gen_ref);
absl::GetFlag(FLAGS_gen_ref));
}
// Use fax mode to avoid time-scaling. This is to simplify the testing of

17
modules/audio_coding/neteq/test/neteq_ilbc_quality_test.cc
View File

@ -10,13 +10,15 @@
#include <memory>
#include "absl/flags/flag.h"
#include "modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
#include "modules/audio_coding/neteq/tools/neteq_quality_test.h"
#include "rtc_base/checks.h"
#include "rtc_base/flags.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "test/testsupport/file_utils.h"
ABSL_FLAG(int, frame_size_ms, 20, "Codec frame size (milliseconds).");
using ::testing::InitGoogleTest;
namespace webrtc {
@ -24,28 +26,27 @@ namespace test {
namespace {
static const int kInputSampleRateKhz = 8;
static const int kOutputSampleRateKhz = 8;
WEBRTC_DEFINE_int(frame_size_ms, 20, "Codec frame size (milliseconds).");
} // namespace
class NetEqIlbcQualityTest : public NetEqQualityTest {
protected:
NetEqIlbcQualityTest()
: NetEqQualityTest(FLAG_frame_size_ms,
: NetEqQualityTest(absl::GetFlag(FLAGS_frame_size_ms),
kInputSampleRateKhz,
kOutputSampleRateKhz,
SdpAudioFormat("ilbc", 8000, 1)) {
// Flag validation
RTC_CHECK(FLAG_frame_size_ms == 20 || FLAG_frame_size_ms == 30 ||
FLAG_frame_size_ms == 40 || FLAG_frame_size_ms == 60)
RTC_CHECK(absl::GetFlag(FLAGS_frame_size_ms) == 20 ||
absl::GetFlag(FLAGS_frame_size_ms) == 30 ||
absl::GetFlag(FLAGS_frame_size_ms) == 40 ||
absl::GetFlag(FLAGS_frame_size_ms) == 60)
<< "Invalid frame size, should be 20, 30, 40, or 60 ms.";
}
void SetUp() override {
ASSERT_EQ(1u, channels_) << "iLBC supports only mono audio.";
AudioEncoderIlbcConfig config;
config.frame_size_ms = FLAG_frame_size_ms;
config.frame_size_ms = absl::GetFlag(FLAGS_frame_size_ms);
encoder_.reset(new AudioEncoderIlbcImpl(config, 102));
NetEqQualityTest::SetUp();
}

12
modules/audio_coding/neteq/test/neteq_isac_quality_test.cc
View File

@ -8,9 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "absl/flags/flag.h"
#include "modules/audio_coding/codecs/isac/fix/include/isacfix.h"
#include "modules/audio_coding/neteq/tools/neteq_quality_test.h"
#include "rtc_base/flags.h"
ABSL_FLAG(int, bit_rate_kbps, 32, "Target bit rate (kbps).");
using ::testing::InitGoogleTest;
@ -20,9 +22,6 @@ namespace {
static const int kIsacBlockDurationMs = 30;
static const int kIsacInputSamplingKhz = 16;
static const int kIsacOutputSamplingKhz = 16;
WEBRTC_DEFINE_int(bit_rate_kbps, 32, "Target bit rate (kbps).");
} // namespace
class NetEqIsacQualityTest : public NetEqQualityTest {
@ -46,9 +45,10 @@ NetEqIsacQualityTest::NetEqIsacQualityTest()
kIsacOutputSamplingKhz,
SdpAudioFormat("isac", 16000, 1)),
isac_encoder_(NULL),
bit_rate_kbps_(FLAG_bit_rate_kbps) {
bit_rate_kbps_(absl::GetFlag(FLAGS_bit_rate_kbps)) {
// Flag validation
RTC_CHECK(FLAG_bit_rate_kbps >= 10 && FLAG_bit_rate_kbps <= 32)
RTC_CHECK(absl::GetFlag(FLAGS_bit_rate_kbps) >= 10 &&
absl::GetFlag(FLAGS_bit_rate_kbps) <= 32)
<< "Invalid bit rate, should be between 10 and 32 kbps.";
}

77
modules/audio_coding/neteq/test/neteq_opus_quality_test.cc
View File

@ -8,10 +8,30 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "absl/flags/flag.h"
#include "modules/audio_coding/codecs/opus/opus_inst.h"
#include "modules/audio_coding/codecs/opus/opus_interface.h"
#include "modules/audio_coding/neteq/tools/neteq_quality_test.h"
#include "rtc_base/flags.h"
ABSL_FLAG(int, bit_rate_kbps, 32, "Target bit rate (kbps).");
ABSL_FLAG(int,
complexity,
10,
"Complexity: 0 ~ 10 -- defined as in Opus"
"specification.");
ABSL_FLAG(int, maxplaybackrate, 48000, "Maximum playback rate (Hz).");
ABSL_FLAG(int, application, 0, "Application mode: 0 -- VOIP, 1 -- Audio.");
ABSL_FLAG(int, reported_loss_rate, 10, "Reported percentile of packet loss.");
ABSL_FLAG(bool, fec, false, "Enable FEC for encoding (-nofec to disable).");
ABSL_FLAG(bool, dtx, false, "Enable DTX for encoding (-nodtx to disable).");
ABSL_FLAG(int, sub_packets, 1, "Number of sub packets to repacketize.");
using ::testing::InitGoogleTest;
@ -21,28 +41,6 @@ namespace {
static const int kOpusBlockDurationMs = 20;
static const int kOpusSamplingKhz = 48;
WEBRTC_DEFINE_int(bit_rate_kbps, 32, "Target bit rate (kbps).");
WEBRTC_DEFINE_int(complexity,
10,
"Complexity: 0 ~ 10 -- defined as in Opus"
"specification.");
WEBRTC_DEFINE_int(maxplaybackrate, 48000, "Maximum playback rate (Hz).");
WEBRTC_DEFINE_int(application, 0, "Application mode: 0 -- VOIP, 1 -- Audio.");
WEBRTC_DEFINE_int(reported_loss_rate,
10,
"Reported percentile of packet loss.");
WEBRTC_DEFINE_bool(fec, false, "Enable FEC for encoding (-nofec to disable).");
WEBRTC_DEFINE_bool(dtx, false, "Enable DTX for encoding (-nodtx to disable).");
WEBRTC_DEFINE_int(sub_packets, 1, "Number of sub packets to repacketize.");
} // namespace
class NetEqOpusQualityTest : public NetEqQualityTest {
@ -70,7 +68,7 @@ class NetEqOpusQualityTest : public NetEqQualityTest {
};
NetEqOpusQualityTest::NetEqOpusQualityTest()
: NetEqQualityTest(kOpusBlockDurationMs * FLAG_sub_packets,
: NetEqQualityTest(kOpusBlockDurationMs * absl::GetFlag(FLAGS_sub_packets),
kOpusSamplingKhz,
kOpusSamplingKhz,
SdpAudioFormat("opus", 48000, 2)),
@ -78,27 +76,32 @@ NetEqOpusQualityTest::NetEqOpusQualityTest()
repacketizer_(NULL),
sub_block_size_samples_(
static_cast<size_t>(kOpusBlockDurationMs * kOpusSamplingKhz)),
bit_rate_kbps_(FLAG_bit_rate_kbps),
fec_(FLAG_fec),
dtx_(FLAG_dtx),
complexity_(FLAG_complexity),
maxplaybackrate_(FLAG_maxplaybackrate),
target_loss_rate_(FLAG_reported_loss_rate),
sub_packets_(FLAG_sub_packets) {
bit_rate_kbps_(absl::GetFlag(FLAGS_bit_rate_kbps)),
fec_(absl::GetFlag(FLAGS_fec)),
dtx_(absl::GetFlag(FLAGS_dtx)),
complexity_(absl::GetFlag(FLAGS_complexity)),
maxplaybackrate_(absl::GetFlag(FLAGS_maxplaybackrate)),
target_loss_rate_(absl::GetFlag(FLAGS_reported_loss_rate)),
sub_packets_(absl::GetFlag(FLAGS_sub_packets)) {
// Flag validation
RTC_CHECK(FLAG_bit_rate_kbps >= 6 && FLAG_bit_rate_kbps <= 510)
RTC_CHECK(absl::GetFlag(FLAGS_bit_rate_kbps) >= 6 &&
absl::GetFlag(FLAGS_bit_rate_kbps) <= 510)
<< "Invalid bit rate, should be between 6 and 510 kbps.";
RTC_CHECK(FLAG_complexity >= -1 && FLAG_complexity <= 10)
RTC_CHECK(absl::GetFlag(FLAGS_complexity) >= -1 &&
absl::GetFlag(FLAGS_complexity) <= 10)
<< "Invalid complexity setting, should be between 0 and 10.";
RTC_CHECK(FLAG_application == 0 || FLAG_application == 1)
RTC_CHECK(absl::GetFlag(FLAGS_application) == 0 ||
absl::GetFlag(FLAGS_application) == 1)
<< "Invalid application mode, should be 0 or 1.";
RTC_CHECK(FLAG_reported_loss_rate >= 0 && FLAG_reported_loss_rate <= 100)
RTC_CHECK(absl::GetFlag(FLAGS_reported_loss_rate) >= 0 &&
absl::GetFlag(FLAGS_reported_loss_rate) <= 100)
<< "Invalid packet loss percentile, should be between 0 and 100.";
RTC_CHECK(FLAG_sub_packets >= 1 && FLAG_sub_packets <= 3)
RTC_CHECK(absl::GetFlag(FLAGS_sub_packets) >= 1 &&
absl::GetFlag(FLAGS_sub_packets) <= 3)
<< "Invalid number of sub packets, should be between 1 and 3.";
// Redefine decoder type if input is stereo.
@ -106,7 +109,7 @@ NetEqOpusQualityTest::NetEqOpusQualityTest()
audio_format_ = SdpAudioFormat(
"opus", 48000, 2, std::map<std::string, std::string>{{"stereo", "1"}});
}
application_ = FLAG_application;
application_ = absl::GetFlag(FLAGS_application);
}
void NetEqOpusQualityTest::SetUp() {

16
modules/audio_coding/neteq/test/neteq_pcm16b_quality_test.cc
View File

@ -10,13 +10,15 @@
#include <memory>
#include "absl/flags/flag.h"
#include "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
#include "modules/audio_coding/neteq/tools/neteq_quality_test.h"
#include "rtc_base/checks.h"
#include "rtc_base/flags.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "test/testsupport/file_utils.h"
ABSL_FLAG(int, frame_size_ms, 20, "Codec frame size (milliseconds).");
using ::testing::InitGoogleTest;
namespace webrtc {
@ -24,27 +26,25 @@ namespace test {
namespace {
static const int kInputSampleRateKhz = 48;
static const int kOutputSampleRateKhz = 48;
WEBRTC_DEFINE_int(frame_size_ms, 20, "Codec frame size (milliseconds).");
} // namespace
class NetEqPcm16bQualityTest : public NetEqQualityTest {
protected:
NetEqPcm16bQualityTest()
: NetEqQualityTest(FLAG_frame_size_ms,
: NetEqQualityTest(absl::GetFlag(FLAGS_frame_size_ms),
kInputSampleRateKhz,
kOutputSampleRateKhz,
SdpAudioFormat("l16", 48000, 1)) {
// Flag validation
RTC_CHECK(FLAG_frame_size_ms >= 10 && FLAG_frame_size_ms <= 60 &&
(FLAG_frame_size_ms % 10) == 0)
RTC_CHECK(absl::GetFlag(FLAGS_frame_size_ms) >= 10 &&
absl::GetFlag(FLAGS_frame_size_ms) <= 60 &&
(absl::GetFlag(FLAGS_frame_size_ms) % 10) == 0)
<< "Invalid frame size, should be 10, 20, ..., 60 ms.";
}
void SetUp() override {
AudioEncoderPcm16B::Config config;
config.frame_size_ms = FLAG_frame_size_ms;
config.frame_size_ms = absl::GetFlag(FLAGS_frame_size_ms);
config.sample_rate_hz = 48000;
config.num_channels = channels_;
encoder_.reset(new AudioEncoderPcm16B(config));

16
modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc
View File

@ -10,13 +10,15 @@
#include <memory>
#include "absl/flags/flag.h"
#include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
#include "modules/audio_coding/neteq/tools/neteq_quality_test.h"
#include "rtc_base/checks.h"
#include "rtc_base/flags.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "test/testsupport/file_utils.h"
ABSL_FLAG(int, frame_size_ms, 20, "Codec frame size (milliseconds).");
using ::testing::InitGoogleTest;
namespace webrtc {
@ -24,28 +26,26 @@ namespace test {
namespace {
static const int kInputSampleRateKhz = 8;
static const int kOutputSampleRateKhz = 8;
WEBRTC_DEFINE_int(frame_size_ms, 20, "Codec frame size (milliseconds).");
} // namespace
class NetEqPcmuQualityTest : public NetEqQualityTest {
protected:
NetEqPcmuQualityTest()
: NetEqQualityTest(FLAG_frame_size_ms,
: NetEqQualityTest(absl::GetFlag(FLAGS_frame_size_ms),
kInputSampleRateKhz,
kOutputSampleRateKhz,
SdpAudioFormat("pcmu", 8000, 1)) {
// Flag validation
RTC_CHECK(FLAG_frame_size_ms >= 10 && FLAG_frame_size_ms <= 60 &&
(FLAG_frame_size_ms % 10) == 0)
RTC_CHECK(absl::GetFlag(FLAGS_frame_size_ms) >= 10 &&
absl::GetFlag(FLAGS_frame_size_ms) <= 60 &&
(absl::GetFlag(FLAGS_frame_size_ms) % 10) == 0)
<< "Invalid frame size, should be 10, 20, ..., 60 ms.";
}
void SetUp() override {
ASSERT_EQ(1u, channels_) << "PCMu supports only mono audio.";
AudioEncoderPcmU::Config config;
config.frame_size_ms = FLAG_frame_size_ms;
config.frame_size_ms = absl::GetFlag(FLAGS_frame_size_ms);
encoder_.reset(new AudioEncoderPcmU(config));
NetEqQualityTest::SetUp();
}

32
modules/audio_coding/neteq/test/neteq_speed_test.cc
View File

@ -11,18 +11,21 @@
#include <stdio.h>
#include <iostream>
#include <vector>
#include "absl/flags/flag.h"
#include "absl/flags/parse.h"
#include "modules/audio_coding/neteq/tools/neteq_performance_test.h"
#include "rtc_base/flags.h"
#include "rtc_base/checks.h"
// Define command line flags.
WEBRTC_DEFINE_int(runtime_ms, 10000, "Simulated runtime in ms.");
WEBRTC_DEFINE_int(lossrate, 10, "Packet lossrate; drop every N packets.");
WEBRTC_DEFINE_float(drift, 0.1f, "Clockdrift factor.");
WEBRTC_DEFINE_bool(help, false, "Print this message.");
ABSL_FLAG(int, runtime_ms, 10000, "Simulated runtime in ms.");
ABSL_FLAG(int, lossrate, 10, "Packet lossrate; drop every N packets.");
ABSL_FLAG(float, drift, 0.1f, "Clockdrift factor.");
int main(int argc, char* argv[]) {
std::string program_name = argv[0];
std::vector<char*> args = absl::ParseCommandLine(argc, argv);
std::string program_name = args[0];
std::string usage =
"Tool for measuring the speed of NetEq.\n"
"Usage: " +
@ -32,21 +35,18 @@ int main(int argc, char* argv[]) {
" --lossrate=N drop every N packets; default is 10\n"
" --drift=F clockdrift factor between 0.0 and 1.0; "
"default is 0.1\n";
if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) || FLAG_help ||
argc != 1) {
if (args.size() != 1) {
printf("%s", usage.c_str());
if (FLAG_help) {
rtc::FlagList::Print(nullptr, false);
return 0;
}
return 1;
}
RTC_CHECK_GT(FLAG_runtime_ms, 0);
RTC_CHECK_GE(FLAG_lossrate, 0);
RTC_CHECK(FLAG_drift >= 0.0 && FLAG_drift < 1.0);
RTC_CHECK_GT(absl::GetFlag(FLAGS_runtime_ms), 0);
RTC_CHECK_GE(absl::GetFlag(FLAGS_lossrate), 0);
RTC_CHECK(absl::GetFlag(FLAGS_drift) >= 0.0 &&
absl::GetFlag(FLAGS_drift) < 1.0);
int64_t result = webrtc::test::NetEqPerformanceTest::Run(
FLAG_runtime_ms, FLAG_lossrate, FLAG_drift);
absl::GetFlag(FLAGS_runtime_ms), absl::GetFlag(FLAGS_lossrate),
absl::GetFlag(FLAGS_drift));
if (result <= 0) {
std::cout << "There was an error" << std::endl;
return -1;

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");