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Replace remaining gflags usages with rtc_base/flags

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Continued from https://codereview.webrtc.org/2995363002

BUG=webrtc:7644

Review-Url: https://codereview.webrtc.org/3005483002
Cr-Commit-Position: refs/heads/master@{#19624}
This commit is contained in:
oprypin
2017-08-31 03:21:39 -07:00
committed by Commit Bot
parent f2972ba166
commit 6e09d875fb
29 changed files with 849 additions and 803 deletions
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3
webrtc/logging/BUILD.gn
View File

@ -142,7 +142,6 @@ if (rtc_enable_protobuf) {
"../system_wrappers:field_trial_default",
"../system_wrappers:metrics_default",
"../test:rtp_test_utils",
"//third_party/gflags",
]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
@ -165,7 +164,6 @@ if (rtc_enable_protobuf) {
# TODO(kwiberg): Remove this dependency.
"../api/audio_codecs:audio_codecs_api",
"../modules/rtp_rtcp:rtp_rtcp",
"//third_party/gflags",
]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
@ -184,7 +182,6 @@ if (rtc_enable_protobuf) {
":rtc_event_log_impl",
":rtc_event_log_proto",
"../rtc_base:rtc_base_approved",
"//third_party/gflags",
]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).

46
webrtc/logging/rtc_event_log/rtc_event_log2rtp_dump.cc
View File

@ -8,17 +8,19 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include <string.h>
#include <iostream>
#include <memory>
#include <sstream>
#include <string>
#include "gflags/gflags.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log_parser.h"
#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "webrtc/rtc_base/checks.h"
#include "webrtc/rtc_base/flags.h"
#include "webrtc/test/rtp_file_writer.h"
namespace {
@ -44,6 +46,7 @@ DEFINE_string(ssrc,
"",
"Store only packets with this SSRC (decimal or hex, the latter "
"starting with 0x).");
DEFINE_bool(help, false, "Prints this message.");
// Parses the input string for a valid SSRC. If a valid SSRC is found, it is
// written to the output variable |ssrc|, and true is returned. Otherwise,
@ -73,22 +76,25 @@ int main(int argc, char* argv[]) {
"Tool for converting an RtcEventLog file to an RTP dump file.\n"
"Run " +
program_name +
" --helpshort for usage.\n"
" --help for usage.\n"
"Example usage:\n" +
program_name + " input.rel output.rtp\n";
google::SetUsageMessage(usage);
google::ParseCommandLineFlags(&argc, &argv, true);
if (argc != 3) {
std::cout << google::ProgramUsage();
return 0;
if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) ||
FLAG_help || argc != 3) {
std::cout << usage;
if (FLAG_help) {
rtc::FlagList::Print(nullptr, false);
return 0;
}
return 1;
}
std::string input_file = argv[1];
std::string output_file = argv[2];
uint32_t ssrc_filter = 0;
if (!FLAGS_ssrc.empty())
RTC_CHECK(ParseSsrc(FLAGS_ssrc, &ssrc_filter))
if (strlen(FLAG_ssrc) > 0)
RTC_CHECK(ParseSsrc(FLAG_ssrc, &ssrc_filter))
<< "Flag verification has failed.";
webrtc::ParsedRtcEventLog parsed_stream;
@ -116,7 +122,7 @@ int main(int argc, char* argv[]) {
// some required fields and we attempt to access them. We could consider
// a softer failure option, but it does not seem useful to generate
// RTP dumps based on broken event logs.
if (!FLAGS_nortp &&
if (!FLAG_nortp &&
parsed_stream.GetEventType(i) == webrtc::ParsedRtcEventLog::RTP_EVENT) {
webrtc::test::RtpPacket packet;
webrtc::PacketDirection direction;
@ -137,13 +143,13 @@ int main(int argc, char* argv[]) {
rtp_parser.Parse(&parsed_header);
MediaType media_type =
parsed_stream.GetMediaType(parsed_header.ssrc, direction);
if (FLAGS_noaudio && media_type == MediaType::AUDIO)
if (FLAG_noaudio && media_type == MediaType::AUDIO)
continue;
if (FLAGS_novideo && media_type == MediaType::VIDEO)
if (FLAG_novideo && media_type == MediaType::VIDEO)
continue;
if (FLAGS_nodata && media_type == MediaType::DATA)
if (FLAG_nodata && media_type == MediaType::DATA)
continue;
if (!FLAGS_ssrc.empty()) {
if (strlen(FLAG_ssrc) > 0) {
const uint32_t packet_ssrc =
webrtc::ByteReader<uint32_t>::ReadBigEndian(
reinterpret_cast<const uint8_t*>(packet.data + 8));
@ -154,7 +160,7 @@ int main(int argc, char* argv[]) {
rtp_writer->WritePacket(&packet);
rtp_counter++;
}
if (!FLAGS_nortcp &&
if (!FLAG_nortcp &&
parsed_stream.GetEventType(i) ==
webrtc::ParsedRtcEventLog::RTCP_EVENT) {
webrtc::test::RtpPacket packet;
@ -175,13 +181,13 @@ int main(int argc, char* argv[]) {
const uint32_t packet_ssrc = webrtc::ByteReader<uint32_t>::ReadBigEndian(
reinterpret_cast<const uint8_t*>(packet.data + 4));
MediaType media_type = parsed_stream.GetMediaType(packet_ssrc, direction);
if (FLAGS_noaudio && media_type == MediaType::AUDIO)
if (FLAG_noaudio && media_type == MediaType::AUDIO)
continue;
if (FLAGS_novideo && media_type == MediaType::VIDEO)
if (FLAG_novideo && media_type == MediaType::VIDEO)
continue;
if (FLAGS_nodata && media_type == MediaType::DATA)
if (FLAG_nodata && media_type == MediaType::DATA)
continue;
if (!FLAGS_ssrc.empty()) {
if (strlen(FLAG_ssrc) > 0) {
if (packet_ssrc != ssrc_filter)
continue;
}

20
webrtc/logging/rtc_event_log/rtc_event_log2stats.cc
View File

@ -19,9 +19,9 @@
#include <utility>
#include <vector>
#include "gflags/gflags.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
#include "webrtc/rtc_base/checks.h"
#include "webrtc/rtc_base/flags.h"
#include "webrtc/rtc_base/ignore_wundef.h"
#include "webrtc/rtc_base/logging.h"
@ -36,6 +36,8 @@ RTC_POP_IGNORING_WUNDEF()
namespace {
DEFINE_bool(help, false, "Prints this message.");
struct Stats {
int count = 0;
size_t total_size = 0;
@ -176,15 +178,17 @@ int main(int argc, char* argv[]) {
"Tool for file usage statistics from an RtcEventLog.\n"
"Run " +
program_name +
" --helpshort for usage.\n"
" --help for usage.\n"
"Example usage:\n" +
program_name + " input.rel\n";
google::SetUsageMessage(usage);
google::ParseCommandLineFlags(&argc, &argv, true);
if (argc != 2) {
std::cout << google::ProgramUsage();
return 0;
if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) ||
FLAG_help || argc != 2) {
std::cout << usage;
if (FLAG_help) {
rtc::FlagList::Print(nullptr, false);
return 0;
}
return 1;
}
std::string file_name = argv[1];

49
webrtc/logging/rtc_event_log/rtc_event_log2text.cc
View File

@ -8,13 +8,14 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include <string.h>
#include <iostream>
#include <map>
#include <sstream>
#include <string>
#include <utility> // pair
#include "gflags/gflags.h"
#include "webrtc/common_types.h"
#include "webrtc/config.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log_parser.h"
@ -35,6 +36,7 @@
#include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "webrtc/rtc_base/checks.h"
#include "webrtc/rtc_base/flags.h"
namespace {
@ -54,6 +56,7 @@ DEFINE_string(ssrc,
"",
"Print only packets with this SSRC (decimal or hex, the latter "
"starting with 0x).");
DEFINE_bool(help, false, "Prints this message.");
using MediaType = webrtc::ParsedRtcEventLog::MediaType;
@ -81,17 +84,17 @@ bool ParseSsrc(std::string str) {
bool ExcludePacket(webrtc::PacketDirection direction,
MediaType media_type,
uint32_t packet_ssrc) {
if (FLAGS_nooutgoing && direction == webrtc::kOutgoingPacket)
if (FLAG_nooutgoing && direction == webrtc::kOutgoingPacket)
return true;
if (FLAGS_noincoming && direction == webrtc::kIncomingPacket)
if (FLAG_noincoming && direction == webrtc::kIncomingPacket)
return true;
if (FLAGS_noaudio && media_type == MediaType::AUDIO)
if (FLAG_noaudio && media_type == MediaType::AUDIO)
return true;
if (FLAGS_novideo && media_type == MediaType::VIDEO)
if (FLAG_novideo && media_type == MediaType::VIDEO)
return true;
if (FLAGS_nodata && media_type == MediaType::DATA)
if (FLAG_nodata && media_type == MediaType::DATA)
return true;
if (!FLAGS_ssrc.empty() && packet_ssrc != filtered_ssrc)
if (strlen(FLAG_ssrc) > 0 && packet_ssrc != filtered_ssrc)
return true;
return false;
}
@ -357,20 +360,22 @@ int main(int argc, char* argv[]) {
"Tool for printing packet information from an RtcEventLog as text.\n"
"Run " +
program_name +
" --helpshort for usage.\n"
" --help for usage.\n"
"Example usage:\n" +
program_name + " input.rel\n";
google::SetUsageMessage(usage);
google::ParseCommandLineFlags(&argc, &argv, true);
if (argc != 2) {
std::cout << google::ProgramUsage();
return 0;
if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) ||
FLAG_help || argc != 2) {
std::cout << usage;
if (FLAG_help) {
rtc::FlagList::Print(nullptr, false);
return 0;
}
return 1;
}
std::string input_file = argv[1];
if (!FLAGS_ssrc.empty())
RTC_CHECK(ParseSsrc(FLAGS_ssrc)) << "Flag verification has failed.";
if (strlen(FLAG_ssrc) > 0)
RTC_CHECK(ParseSsrc(FLAG_ssrc)) << "Flag verification has failed.";
webrtc::RtpHeaderExtensionMap default_map = GetDefaultHeaderExtensionMap();
@ -381,7 +386,7 @@ int main(int argc, char* argv[]) {
}
for (size_t i = 0; i < parsed_stream.GetNumberOfEvents(); i++) {
if (!FLAGS_noconfig && !FLAGS_novideo && !FLAGS_noincoming &&
if (!FLAG_noconfig && !FLAG_novideo && !FLAG_noincoming &&
parsed_stream.GetEventType(i) ==
webrtc::ParsedRtcEventLog::VIDEO_RECEIVER_CONFIG_EVENT) {
webrtc::rtclog::StreamConfig config =
@ -402,7 +407,7 @@ int main(int argc, char* argv[]) {
}
std::cout << "}" << std::endl;
}
if (!FLAGS_noconfig && !FLAGS_novideo && !FLAGS_nooutgoing &&
if (!FLAG_noconfig && !FLAG_novideo && !FLAG_nooutgoing &&
parsed_stream.GetEventType(i) ==
webrtc::ParsedRtcEventLog::VIDEO_SENDER_CONFIG_EVENT) {
std::vector<webrtc::rtclog::StreamConfig> configs =
@ -425,7 +430,7 @@ int main(int argc, char* argv[]) {
std::cout << "}" << std::endl;
}
}
if (!FLAGS_noconfig && !FLAGS_noaudio && !FLAGS_noincoming &&
if (!FLAG_noconfig && !FLAG_noaudio && !FLAG_noincoming &&
parsed_stream.GetEventType(i) ==
webrtc::ParsedRtcEventLog::AUDIO_RECEIVER_CONFIG_EVENT) {
webrtc::rtclog::StreamConfig config =
@ -446,7 +451,7 @@ int main(int argc, char* argv[]) {
}
std::cout << "}" << std::endl;
}
if (!FLAGS_noconfig && !FLAGS_noaudio && !FLAGS_nooutgoing &&
if (!FLAG_noconfig && !FLAG_noaudio && !FLAG_nooutgoing &&
parsed_stream.GetEventType(i) ==
webrtc::ParsedRtcEventLog::AUDIO_SENDER_CONFIG_EVENT) {
webrtc::rtclog::StreamConfig config = parsed_stream.GetAudioSendConfig(i);
@ -465,7 +470,7 @@ int main(int argc, char* argv[]) {
}
std::cout << "}" << std::endl;
}
if (!FLAGS_nortp &&
if (!FLAG_nortp &&
parsed_stream.GetEventType(i) == webrtc::ParsedRtcEventLog::RTP_EVENT) {
size_t header_length;
size_t total_length;
@ -516,7 +521,7 @@ int main(int argc, char* argv[]) {
}
std::cout << std::endl;
}
if (!FLAGS_nortcp &&
if (!FLAG_nortcp &&
parsed_stream.GetEventType(i) ==
webrtc::ParsedRtcEventLog::RTCP_EVENT) {
size_t length;

7
webrtc/modules/audio_coding/BUILD.gn
View File

@ -1395,7 +1395,6 @@ if (rtc_include_tests) {
"../../test:test_support",
"../rtp_rtcp",
"//testing/gtest",
"//third_party/gflags:gflags",
]
} # delay_test
@ -1425,7 +1424,6 @@ if (rtc_include_tests) {
"../../test:test_support",
"../rtp_rtcp",
"//testing/gtest",
"//third_party/gflags:gflags",
]
} # insert_packet_with_timing
@ -1520,7 +1518,6 @@ if (rtc_include_tests) {
"../../rtc_base:rtc_base_approved",
"../../system_wrappers:system_wrappers_default",
"../../test:test_support",
"//third_party/gflags",
]
}
}
@ -1792,9 +1789,9 @@ if (rtc_include_tests) {
":neteq",
":neteq_test_tools",
":pcm16b",
"../../rtc_base:rtc_base_approved",
"../../system_wrappers:system_wrappers_default",
"//testing/gtest",
"//third_party/gflags:gflags",
]
if (!build_with_chromium && is_clang) {
@ -1832,9 +1829,9 @@ if (rtc_include_tests) {
":neteq",
":neteq_test_support",
"../..:webrtc_common",
"../../rtc_base:rtc_base_approved",
"../../system_wrappers:system_wrappers_default",
"../../test:test_support",
"//third_party/gflags",
]
}

59
webrtc/modules/audio_coding/neteq/test/neteq_speed_test.cc
View File

@ -12,43 +12,18 @@
#include <iostream>
#include "gflags/gflags.h"
#include "webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.h"
#include "webrtc/rtc_base/flags.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/typedefs.h"
// Flag validators.
static bool ValidateRuntime(const char* flagname, int value) {
if (value > 0) // Value is ok.
return true;
printf("Invalid value for --%s: %d\n", flagname, static_cast<int>(value));
return false;
}
static bool ValidateLossrate(const char* flagname, int value) {
if (value >= 0) // Value is ok.
return true;
printf("Invalid value for --%s: %d\n", flagname, static_cast<int>(value));
return false;
}
static bool ValidateDriftfactor(const char* flagname, double value) {
if (value >= 0.0 && value < 1.0) // Value is ok.
return true;
printf("Invalid value for --%s: %f\n", flagname, value);
return false;
}
// Define command line flags.
DEFINE_int32(runtime_ms, 10000, "Simulated runtime in ms.");
static const bool runtime_ms_dummy =
google::RegisterFlagValidator(&FLAGS_runtime_ms, &ValidateRuntime);
DEFINE_int32(lossrate, 10,
"Packet lossrate; drop every N packets.");
static const bool lossrate_dummy =
google::RegisterFlagValidator(&FLAGS_lossrate, &ValidateLossrate);
DEFINE_double(drift, 0.1,
DEFINE_int(runtime_ms, 10000, "Simulated runtime in ms.");
DEFINE_int(lossrate, 10,
"Packet lossrate; drop every N packets.");
DEFINE_float(drift, 0.1f,
"Clockdrift factor.");
static const bool drift_dummy =
google::RegisterFlagValidator(&FLAGS_drift, &ValidateDriftfactor);
DEFINE_bool(help, false, "Print this message.");
int main(int argc, char* argv[]) {
std::string program_name = argv[0];
@ -58,19 +33,23 @@ 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";
google::SetUsageMessage(usage);
google::ParseCommandLineFlags(&argc, &argv, true);
webrtc::test::SetExecutablePath(argv[0]);
if (argc != 1) {
// Print usage information.
std::cout << google::ProgramUsage();
return 0;
if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) ||
FLAG_help || argc != 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);
int64_t result =
webrtc::test::NetEqPerformanceTest::Run(FLAGS_runtime_ms, FLAGS_lossrate,
FLAGS_drift);
webrtc::test::NetEqPerformanceTest::Run(FLAG_runtime_ms, FLAG_lossrate,
FLAG_drift);
if (result <= 0) {
std::cout << "There was an error" << std::endl;
return -1;

286
webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
View File

@ -13,6 +13,7 @@
#include <limits.h> // For ULONG_MAX returned by strtoul.
#include <stdio.h>
#include <stdlib.h> // For strtoul.
#include <string.h>
#include <algorithm>
#include <ios>
@ -21,7 +22,6 @@
#include <numeric>
#include <string>
#include "gflags/gflags.h"
#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
#include "webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.h"
#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
@ -34,6 +34,7 @@
#include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/rtc_base/checks.h"
#include "webrtc/rtc_base/flags.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/typedefs.h"
@ -65,86 +66,51 @@ bool ParseSsrc(const std::string& str, uint32_t* ssrc) {
}
// Flag validators.
bool ValidatePayloadType(const char* flagname, int32_t value) {
bool ValidatePayloadType(int value) {
if (value >= 0 && value <= 127) // Value is ok.
return true;
printf("Invalid value for --%s: %d\n", flagname, static_cast<int>(value));
printf("Payload type must be between 0 and 127, not %d\n",
static_cast<int>(value));
return false;
}
bool ValidateSsrcValue(const char* flagname, const std::string& str) {
bool ValidateSsrcValue(const std::string& str) {
uint32_t dummy_ssrc;
return ParseSsrc(str, &dummy_ssrc);
if (ParseSsrc(str, &dummy_ssrc)) // Value is ok.
return true;
printf("Invalid SSRC: %s\n", str.c_str());
return false;
}
static bool ValidateExtensionId(const char* flagname, int32_t value) {
static bool ValidateExtensionId(int value) {
if (value > 0 && value <= 255) // Value is ok.
return true;
printf("Invalid value for --%s: %d\n", flagname, static_cast<int>(value));
printf("Extension ID must be between 1 and 255, not %d\n",
static_cast<int>(value));
return false;
}
// Define command line flags.
DEFINE_int32(pcmu, 0, "RTP payload type for PCM-u");
const bool pcmu_dummy =
google::RegisterFlagValidator(&FLAGS_pcmu, &ValidatePayloadType);
DEFINE_int32(pcma, 8, "RTP payload type for PCM-a");
const bool pcma_dummy =
google::RegisterFlagValidator(&FLAGS_pcma, &ValidatePayloadType);
DEFINE_int32(ilbc, 102, "RTP payload type for iLBC");
const bool ilbc_dummy =
google::RegisterFlagValidator(&FLAGS_ilbc, &ValidatePayloadType);
DEFINE_int32(isac, 103, "RTP payload type for iSAC");
const bool isac_dummy =
google::RegisterFlagValidator(&FLAGS_isac, &ValidatePayloadType);
DEFINE_int32(isac_swb, 104, "RTP payload type for iSAC-swb (32 kHz)");
const bool isac_swb_dummy =
google::RegisterFlagValidator(&FLAGS_isac_swb, &ValidatePayloadType);
DEFINE_int32(opus, 111, "RTP payload type for Opus");
const bool opus_dummy =
google::RegisterFlagValidator(&FLAGS_opus, &ValidatePayloadType);
DEFINE_int32(pcm16b, 93, "RTP payload type for PCM16b-nb (8 kHz)");
const bool pcm16b_dummy =
google::RegisterFlagValidator(&FLAGS_pcm16b, &ValidatePayloadType);
DEFINE_int32(pcm16b_wb, 94, "RTP payload type for PCM16b-wb (16 kHz)");
const bool pcm16b_wb_dummy =
google::RegisterFlagValidator(&FLAGS_pcm16b_wb, &ValidatePayloadType);
DEFINE_int32(pcm16b_swb32, 95, "RTP payload type for PCM16b-swb32 (32 kHz)");
const bool pcm16b_swb32_dummy =
google::RegisterFlagValidator(&FLAGS_pcm16b_swb32, &ValidatePayloadType);
DEFINE_int32(pcm16b_swb48, 96, "RTP payload type for PCM16b-swb48 (48 kHz)");
const bool pcm16b_swb48_dummy =
google::RegisterFlagValidator(&FLAGS_pcm16b_swb48, &ValidatePayloadType);
DEFINE_int32(g722, 9, "RTP payload type for G.722");
const bool g722_dummy =
google::RegisterFlagValidator(&FLAGS_g722, &ValidatePayloadType);
DEFINE_int32(avt, 106, "RTP payload type for AVT/DTMF (8 kHz)");
const bool avt_dummy =
google::RegisterFlagValidator(&FLAGS_avt, &ValidatePayloadType);
DEFINE_int32(avt_16, 114, "RTP payload type for AVT/DTMF (16 kHz)");
const bool avt_16_dummy =
google::RegisterFlagValidator(&FLAGS_avt_16, &ValidatePayloadType);
DEFINE_int32(avt_32, 115, "RTP payload type for AVT/DTMF (32 kHz)");
const bool avt_32_dummy =
google::RegisterFlagValidator(&FLAGS_avt_32, &ValidatePayloadType);
DEFINE_int32(avt_48, 116, "RTP payload type for AVT/DTMF (48 kHz)");
const bool avt_48_dummy =
google::RegisterFlagValidator(&FLAGS_avt_48, &ValidatePayloadType);
DEFINE_int32(red, 117, "RTP payload type for redundant audio (RED)");
const bool red_dummy =
google::RegisterFlagValidator(&FLAGS_red, &ValidatePayloadType);
DEFINE_int32(cn_nb, 13, "RTP payload type for comfort noise (8 kHz)");
const bool cn_nb_dummy =
google::RegisterFlagValidator(&FLAGS_cn_nb, &ValidatePayloadType);
DEFINE_int32(cn_wb, 98, "RTP payload type for comfort noise (16 kHz)");
const bool cn_wb_dummy =
google::RegisterFlagValidator(&FLAGS_cn_wb, &ValidatePayloadType);
DEFINE_int32(cn_swb32, 99, "RTP payload type for comfort noise (32 kHz)");
const bool cn_swb32_dummy =
google::RegisterFlagValidator(&FLAGS_cn_swb32, &ValidatePayloadType);
DEFINE_int32(cn_swb48, 100, "RTP payload type for comfort noise (48 kHz)");
const bool cn_swb48_dummy =
google::RegisterFlagValidator(&FLAGS_cn_swb48, &ValidatePayloadType);
DEFINE_int(pcmu, 0, "RTP payload type for PCM-u");
DEFINE_int(pcma, 8, "RTP payload type for PCM-a");
DEFINE_int(ilbc, 102, "RTP payload type for iLBC");
DEFINE_int(isac, 103, "RTP payload type for iSAC");
DEFINE_int(isac_swb, 104, "RTP payload type for iSAC-swb (32 kHz)");
DEFINE_int(opus, 111, "RTP payload type for Opus");
DEFINE_int(pcm16b, 93, "RTP payload type for PCM16b-nb (8 kHz)");
DEFINE_int(pcm16b_wb, 94, "RTP payload type for PCM16b-wb (16 kHz)");
DEFINE_int(pcm16b_swb32, 95, "RTP payload type for PCM16b-swb32 (32 kHz)");
DEFINE_int(pcm16b_swb48, 96, "RTP payload type for PCM16b-swb48 (48 kHz)");
DEFINE_int(g722, 9, "RTP payload type for G.722");
DEFINE_int(avt, 106, "RTP payload type for AVT/DTMF (8 kHz)");
DEFINE_int(avt_16, 114, "RTP payload type for AVT/DTMF (16 kHz)");
DEFINE_int(avt_32, 115, "RTP payload type for AVT/DTMF (32 kHz)");
DEFINE_int(avt_48, 116, "RTP payload type for AVT/DTMF (48 kHz)");
DEFINE_int(red, 117, "RTP payload type for redundant audio (RED)");
DEFINE_int(cn_nb, 13, "RTP payload type for comfort noise (8 kHz)");
DEFINE_int(cn_wb, 98, "RTP payload type for comfort noise (16 kHz)");
DEFINE_int(cn_swb32, 99, "RTP payload type for comfort noise (32 kHz)");
DEFINE_int(cn_swb48, 100, "RTP payload type for comfort noise (48 kHz)");
DEFINE_bool(codec_map, false, "Prints the mapping between RTP payload type and "
"codec");
DEFINE_string(replacement_audio_file, "",
@ -153,21 +119,13 @@ DEFINE_string(ssrc,
"",
"Only use packets with this SSRC (decimal or hex, the latter "
"starting with 0x)");
const bool hex_ssrc_dummy =
google::RegisterFlagValidator(&FLAGS_ssrc, &ValidateSsrcValue);
DEFINE_int32(audio_level, 1, "Extension ID for audio level (RFC 6464)");
const bool audio_level_dummy =
google::RegisterFlagValidator(&FLAGS_audio_level, &ValidateExtensionId);
DEFINE_int32(abs_send_time, 3, "Extension ID for absolute sender time");
const bool abs_send_time_dummy =
google::RegisterFlagValidator(&FLAGS_abs_send_time, &ValidateExtensionId);
DEFINE_int32(transport_seq_no, 5, "Extension ID for transport sequence number");
const bool transport_seq_no_dummy =
google::RegisterFlagValidator(&FLAGS_transport_seq_no,
&ValidateExtensionId);
DEFINE_int(audio_level, 1, "Extension ID for audio level (RFC 6464)");
DEFINE_int(abs_send_time, 3, "Extension ID for absolute sender time");
DEFINE_int(transport_seq_no, 5, "Extension ID for transport sequence number");
DEFINE_bool(matlabplot,
false,
"Generates a matlab script for plotting the delay profile");
DEFINE_bool(help, false, "Prints this message");
// Maps a codec type to a printable name string.
std::string CodecName(NetEqDecoder codec) {
@ -218,51 +176,51 @@ std::string CodecName(NetEqDecoder codec) {
}
}
void PrintCodecMappingEntry(NetEqDecoder codec, google::int32 flag) {
void PrintCodecMappingEntry(NetEqDecoder codec, int flag) {
std::cout << CodecName(codec) << ": " << flag << std::endl;
}
void PrintCodecMapping() {
PrintCodecMappingEntry(NetEqDecoder::kDecoderPCMu, FLAGS_pcmu);
PrintCodecMappingEntry(NetEqDecoder::kDecoderPCMa, FLAGS_pcma);
PrintCodecMappingEntry(NetEqDecoder::kDecoderILBC, FLAGS_ilbc);
PrintCodecMappingEntry(NetEqDecoder::kDecoderISAC, FLAGS_isac);
PrintCodecMappingEntry(NetEqDecoder::kDecoderISACswb, FLAGS_isac_swb);
PrintCodecMappingEntry(NetEqDecoder::kDecoderOpus, FLAGS_opus);
PrintCodecMappingEntry(NetEqDecoder::kDecoderPCM16B, FLAGS_pcm16b);
PrintCodecMappingEntry(NetEqDecoder::kDecoderPCM16Bwb, FLAGS_pcm16b_wb);
PrintCodecMappingEntry(NetEqDecoder::kDecoderPCMu, FLAG_pcmu);
PrintCodecMappingEntry(NetEqDecoder::kDecoderPCMa, FLAG_pcma);
PrintCodecMappingEntry(NetEqDecoder::kDecoderILBC, FLAG_ilbc);
PrintCodecMappingEntry(NetEqDecoder::kDecoderISAC, FLAG_isac);
PrintCodecMappingEntry(NetEqDecoder::kDecoderISACswb, FLAG_isac_swb);
PrintCodecMappingEntry(NetEqDecoder::kDecoderOpus, FLAG_opus);
PrintCodecMappingEntry(NetEqDecoder::kDecoderPCM16B, FLAG_pcm16b);
PrintCodecMappingEntry(NetEqDecoder::kDecoderPCM16Bwb, FLAG_pcm16b_wb);
PrintCodecMappingEntry(NetEqDecoder::kDecoderPCM16Bswb32kHz,
FLAGS_pcm16b_swb32);
FLAG_pcm16b_swb32);
PrintCodecMappingEntry(NetEqDecoder::kDecoderPCM16Bswb48kHz,
FLAGS_pcm16b_swb48);
PrintCodecMappingEntry(NetEqDecoder::kDecoderG722, FLAGS_g722);
PrintCodecMappingEntry(NetEqDecoder::kDecoderAVT, FLAGS_avt);
PrintCodecMappingEntry(NetEqDecoder::kDecoderAVT16kHz, FLAGS_avt_16);
PrintCodecMappingEntry(NetEqDecoder::kDecoderAVT32kHz, FLAGS_avt_32);
PrintCodecMappingEntry(NetEqDecoder::kDecoderAVT48kHz, FLAGS_avt_48);
PrintCodecMappingEntry(NetEqDecoder::kDecoderRED, FLAGS_red);
PrintCodecMappingEntry(NetEqDecoder::kDecoderCNGnb, FLAGS_cn_nb);
PrintCodecMappingEntry(NetEqDecoder::kDecoderCNGwb, FLAGS_cn_wb);
PrintCodecMappingEntry(NetEqDecoder::kDecoderCNGswb32kHz, FLAGS_cn_swb32);
PrintCodecMappingEntry(NetEqDecoder::kDecoderCNGswb48kHz, FLAGS_cn_swb48);
FLAG_pcm16b_swb48);
PrintCodecMappingEntry(NetEqDecoder::kDecoderG722, FLAG_g722);
PrintCodecMappingEntry(NetEqDecoder::kDecoderAVT, FLAG_avt);
PrintCodecMappingEntry(NetEqDecoder::kDecoderAVT16kHz, FLAG_avt_16);
PrintCodecMappingEntry(NetEqDecoder::kDecoderAVT32kHz, FLAG_avt_32);
PrintCodecMappingEntry(NetEqDecoder::kDecoderAVT48kHz, FLAG_avt_48);
PrintCodecMappingEntry(NetEqDecoder::kDecoderRED, FLAG_red);
PrintCodecMappingEntry(NetEqDecoder::kDecoderCNGnb, FLAG_cn_nb);
PrintCodecMappingEntry(NetEqDecoder::kDecoderCNGwb, FLAG_cn_wb);
PrintCodecMappingEntry(NetEqDecoder::kDecoderCNGswb32kHz, FLAG_cn_swb32);
PrintCodecMappingEntry(NetEqDecoder::kDecoderCNGswb48kHz, FLAG_cn_swb48);
}
rtc::Optional<int> CodecSampleRate(uint8_t payload_type) {
if (payload_type == FLAGS_pcmu || payload_type == FLAGS_pcma ||
payload_type == FLAGS_ilbc || payload_type == FLAGS_pcm16b ||
payload_type == FLAGS_cn_nb || payload_type == FLAGS_avt)
if (payload_type == FLAG_pcmu || payload_type == FLAG_pcma ||
payload_type == FLAG_ilbc || payload_type == FLAG_pcm16b ||
payload_type == FLAG_cn_nb || payload_type == FLAG_avt)
return rtc::Optional<int>(8000);
if (payload_type == FLAGS_isac || payload_type == FLAGS_pcm16b_wb ||
payload_type == FLAGS_g722 || payload_type == FLAGS_cn_wb ||
payload_type == FLAGS_avt_16)
if (payload_type == FLAG_isac || payload_type == FLAG_pcm16b_wb ||
payload_type == FLAG_g722 || payload_type == FLAG_cn_wb ||
payload_type == FLAG_avt_16)
return rtc::Optional<int>(16000);
if (payload_type == FLAGS_isac_swb || payload_type == FLAGS_pcm16b_swb32 ||
payload_type == FLAGS_cn_swb32 || payload_type == FLAGS_avt_32)
if (payload_type == FLAG_isac_swb || payload_type == FLAG_pcm16b_swb32 ||
payload_type == FLAG_cn_swb32 || payload_type == FLAG_avt_32)
return rtc::Optional<int>(32000);
if (payload_type == FLAGS_opus || payload_type == FLAGS_pcm16b_swb48 ||
payload_type == FLAGS_cn_swb48 || payload_type == FLAGS_avt_48)
if (payload_type == FLAG_opus || payload_type == FLAG_pcm16b_swb48 ||
payload_type == FLAG_cn_swb48 || payload_type == FLAG_avt_48)
return rtc::Optional<int>(48000);
if (payload_type == FLAGS_red)
if (payload_type == FLAG_red)
return rtc::Optional<int>(0);
return rtc::Optional<int>();
}
@ -460,31 +418,61 @@ class StatsGetter : public NetEqGetAudioCallback {
int RunTest(int argc, char* argv[]) {
std::string program_name = argv[0];
std::string usage = "Tool for decoding an RTP dump file using NetEq.\n"
"Run " + program_name + " --helpshort for usage.\n"
"Run " + program_name + " --help for usage.\n"
"Example usage:\n" + program_name +
" input.rtp output.{pcm, wav}\n";
google::SetUsageMessage(usage);
google::ParseCommandLineFlags(&argc, &argv, true);
if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true)) {
return 1;
}
if (FLAG_help) {
std::cout << usage;
rtc::FlagList::Print(nullptr, false);
return 0;
}
if (FLAGS_codec_map) {
if (FLAG_codec_map) {
PrintCodecMapping();
}
if (argc != 3) {
if (FLAGS_codec_map) {
if (FLAG_codec_map) {
// We have already printed the codec map. Just end the program.
return 0;
}
// Print usage information.
std::cout << google::ProgramUsage();
std::cout << usage;
return 0;
}
RTC_CHECK(ValidatePayloadType(FLAG_pcmu));
RTC_CHECK(ValidatePayloadType(FLAG_pcma));
RTC_CHECK(ValidatePayloadType(FLAG_ilbc));
RTC_CHECK(ValidatePayloadType(FLAG_isac));
RTC_CHECK(ValidatePayloadType(FLAG_isac_swb));
RTC_CHECK(ValidatePayloadType(FLAG_opus));
RTC_CHECK(ValidatePayloadType(FLAG_pcm16b));
RTC_CHECK(ValidatePayloadType(FLAG_pcm16b_wb));
RTC_CHECK(ValidatePayloadType(FLAG_pcm16b_swb32));
RTC_CHECK(ValidatePayloadType(FLAG_pcm16b_swb48));
RTC_CHECK(ValidatePayloadType(FLAG_g722));
RTC_CHECK(ValidatePayloadType(FLAG_avt));
RTC_CHECK(ValidatePayloadType(FLAG_avt_16));
RTC_CHECK(ValidatePayloadType(FLAG_avt_32));
RTC_CHECK(ValidatePayloadType(FLAG_avt_48));
RTC_CHECK(ValidatePayloadType(FLAG_red));
RTC_CHECK(ValidatePayloadType(FLAG_cn_nb));
RTC_CHECK(ValidatePayloadType(FLAG_cn_wb));
RTC_CHECK(ValidatePayloadType(FLAG_cn_swb32));
RTC_CHECK(ValidatePayloadType(FLAG_cn_swb48));
RTC_CHECK(ValidateSsrcValue(FLAG_ssrc));
RTC_CHECK(ValidateExtensionId(FLAG_audio_level));
RTC_CHECK(ValidateExtensionId(FLAG_abs_send_time));
RTC_CHECK(ValidateExtensionId(FLAG_transport_seq_no));
// Gather RTP header extensions in a map.
NetEqPacketSourceInput::RtpHeaderExtensionMap rtp_ext_map = {
{FLAGS_audio_level, kRtpExtensionAudioLevel},
{FLAGS_abs_send_time, kRtpExtensionAbsoluteSendTime},
{FLAGS_transport_seq_no, kRtpExtensionTransportSequenceNumber}};
{FLAG_audio_level, kRtpExtensionAudioLevel},
{FLAG_abs_send_time, kRtpExtensionAbsoluteSendTime},
{FLAG_transport_seq_no, kRtpExtensionTransportSequenceNumber}};
const std::string input_file_name = argv[1];
std::unique_ptr<NetEqInput> input;
@ -500,9 +488,9 @@ int RunTest(int argc, char* argv[]) {
RTC_CHECK(!input->ended()) << "Input file is empty";
// Check if an SSRC value was provided.
if (!FLAGS_ssrc.empty()) {
if (strlen(FLAG_ssrc) > 0) {
uint32_t ssrc;
RTC_CHECK(ParseSsrc(FLAGS_ssrc, &ssrc)) << "Flag verification has failed.";
RTC_CHECK(ParseSsrc(FLAG_ssrc, &ssrc)) << "Flag verification has failed.";
input.reset(new FilterSsrcInput(std::move(input), ssrc));
}
@ -557,39 +545,39 @@ int RunTest(int argc, char* argv[]) {
std::cout << "Output file: " << output_file_name << std::endl;
NetEqTest::DecoderMap codecs = {
{FLAGS_pcmu, std::make_pair(NetEqDecoder::kDecoderPCMu, "pcmu")},
{FLAGS_pcma, std::make_pair(NetEqDecoder::kDecoderPCMa, "pcma")},
{FLAGS_ilbc, std::make_pair(NetEqDecoder::kDecoderILBC, "ilbc")},
{FLAGS_isac, std::make_pair(NetEqDecoder::kDecoderISAC, "isac")},
{FLAGS_isac_swb,
{FLAG_pcmu, std::make_pair(NetEqDecoder::kDecoderPCMu, "pcmu")},
{FLAG_pcma, std::make_pair(NetEqDecoder::kDecoderPCMa, "pcma")},
{FLAG_ilbc, std::make_pair(NetEqDecoder::kDecoderILBC, "ilbc")},
{FLAG_isac, std::make_pair(NetEqDecoder::kDecoderISAC, "isac")},
{FLAG_isac_swb,
std::make_pair(NetEqDecoder::kDecoderISACswb, "isac-swb")},
{FLAGS_opus, std::make_pair(NetEqDecoder::kDecoderOpus, "opus")},
{FLAGS_pcm16b, std::make_pair(NetEqDecoder::kDecoderPCM16B, "pcm16-nb")},
{FLAGS_pcm16b_wb,
{FLAG_opus, std::make_pair(NetEqDecoder::kDecoderOpus, "opus")},
{FLAG_pcm16b, std::make_pair(NetEqDecoder::kDecoderPCM16B, "pcm16-nb")},
{FLAG_pcm16b_wb,
std::make_pair(NetEqDecoder::kDecoderPCM16Bwb, "pcm16-wb")},
{FLAGS_pcm16b_swb32,
{FLAG_pcm16b_swb32,
std::make_pair(NetEqDecoder::kDecoderPCM16Bswb32kHz, "pcm16-swb32")},
{FLAGS_pcm16b_swb48,
{FLAG_pcm16b_swb48,
std::make_pair(NetEqDecoder::kDecoderPCM16Bswb48kHz, "pcm16-swb48")},
{FLAGS_g722, std::make_pair(NetEqDecoder::kDecoderG722, "g722")},
{FLAGS_avt, std::make_pair(NetEqDecoder::kDecoderAVT, "avt")},
{FLAGS_avt_16, std::make_pair(NetEqDecoder::kDecoderAVT16kHz, "avt-16")},
{FLAGS_avt_32,
{FLAG_g722, std::make_pair(NetEqDecoder::kDecoderG722, "g722")},
{FLAG_avt, std::make_pair(NetEqDecoder::kDecoderAVT, "avt")},
{FLAG_avt_16, std::make_pair(NetEqDecoder::kDecoderAVT16kHz, "avt-16")},
{FLAG_avt_32,
std::make_pair(NetEqDecoder::kDecoderAVT32kHz, "avt-32")},
{FLAGS_avt_48,
{FLAG_avt_48,
std::make_pair(NetEqDecoder::kDecoderAVT48kHz, "avt-48")},
{FLAGS_red, std::make_pair(NetEqDecoder::kDecoderRED, "red")},
{FLAGS_cn_nb, std::make_pair(NetEqDecoder::kDecoderCNGnb, "cng-nb")},
{FLAGS_cn_wb, std::make_pair(NetEqDecoder::kDecoderCNGwb, "cng-wb")},
{FLAGS_cn_swb32,
{FLAG_red, std::make_pair(NetEqDecoder::kDecoderRED, "red")},
{FLAG_cn_nb, std::make_pair(NetEqDecoder::kDecoderCNGnb, "cng-nb")},
{FLAG_cn_wb, std::make_pair(NetEqDecoder::kDecoderCNGwb, "cng-wb")},
{FLAG_cn_swb32,
std::make_pair(NetEqDecoder::kDecoderCNGswb32kHz, "cng-swb32")},
{FLAGS_cn_swb48,
{FLAG_cn_swb48,
std::make_pair(NetEqDecoder::kDecoderCNGswb48kHz, "cng-swb48")}};
// Check if a replacement audio file was provided.
std::unique_ptr<AudioDecoder> replacement_decoder;
NetEqTest::ExtDecoderMap ext_codecs;
if (!FLAGS_replacement_audio_file.empty()) {
if (strlen(FLAG_replacement_audio_file) > 0) {
// Find largest unused payload type.
int replacement_pt = 127;
while (!(codecs.find(replacement_pt) == codecs.end() &&
@ -607,16 +595,16 @@ int RunTest(int argc, char* argv[]) {
};
std::set<uint8_t> cn_types = std_set_int32_to_uint8(
{FLAGS_cn_nb, FLAGS_cn_wb, FLAGS_cn_swb32, FLAGS_cn_swb48});
{FLAG_cn_nb, FLAG_cn_wb, FLAG_cn_swb32, FLAG_cn_swb48});
std::set<uint8_t> forbidden_types =
std_set_int32_to_uint8({FLAGS_g722, FLAGS_red, FLAGS_avt,
FLAGS_avt_16, FLAGS_avt_32, FLAGS_avt_48});
std_set_int32_to_uint8({FLAG_g722, FLAG_red, FLAG_avt,
FLAG_avt_16, FLAG_avt_32, FLAG_avt_48});
input.reset(new NetEqReplacementInput(std::move(input), replacement_pt,
cn_types, forbidden_types));
replacement_decoder.reset(new FakeDecodeFromFile(
std::unique_ptr<InputAudioFile>(
new InputAudioFile(FLAGS_replacement_audio_file)),
new InputAudioFile(FLAG_replacement_audio_file)),
48000, false));
NetEqTest::ExternalDecoderInfo ext_dec_info = {
replacement_decoder.get(), NetEqDecoder::kDecoderArbitrary,
@ -626,7 +614,7 @@ int RunTest(int argc, char* argv[]) {
NetEqTest::Callbacks callbacks;
std::unique_ptr<NetEqDelayAnalyzer> delay_analyzer;
if (FLAGS_matlabplot) {
if (FLAG_matlabplot) {
delay_analyzer.reset(new NetEqDelayAnalyzer);
}
@ -641,7 +629,7 @@ int RunTest(int argc, char* argv[]) {
int64_t test_duration_ms = test.Run();
if (FLAGS_matlabplot) {
if (FLAG_matlabplot) {
auto matlab_script_name = output_file_name;
std::replace(matlab_script_name.begin(), matlab_script_name.end(), '.',
'_');

68
webrtc/modules/audio_coding/neteq/tools/rtp_analyze.cc
View File

@ -14,34 +14,17 @@
#include <memory>
#include <vector>
#include "gflags/gflags.h"
#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
#include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"
// Flag validator.
static bool ValidatePayloadType(const char* flagname, int32_t value) {
if (value >= 0 && value <= 127) // Value is ok.
return true;
printf("Invalid value for --%s: %d\n", flagname, static_cast<int>(value));
return false;
}
static bool ValidateExtensionId(const char* flagname, int32_t value) {
if (value > 0 && value <= 255) // Value is ok.
return true;
printf("Invalid value for --%s: %d\n", flagname, static_cast<int>(value));
return false;
}
#include "webrtc/rtc_base/flags.h"
// Define command line flags.
DEFINE_int32(red, 117, "RTP payload type for RED");
static const bool red_dummy =
google::RegisterFlagValidator(&FLAGS_red, &ValidatePayloadType);
DEFINE_int32(audio_level, 1, "Extension ID for audio level (RFC 6464)");
static const bool audio_level_dummy =
google::RegisterFlagValidator(&FLAGS_audio_level, &ValidateExtensionId);
DEFINE_int32(abs_send_time, 3, "Extension ID for absolute sender time");
static const bool abs_send_time_dummy =
google::RegisterFlagValidator(&FLAGS_abs_send_time, &ValidateExtensionId);
DEFINE_int(red, 117, "RTP payload type for RED");
DEFINE_int(audio_level, -1, "Extension ID for audio level (RFC 6464); "
"-1 not to print audio level");
DEFINE_int(abs_send_time, -1, "Extension ID for absolute sender time; "
"-1 not to print absolute send time");
DEFINE_bool(help, false, "Print this message");
int main(int argc, char* argv[]) {
std::string program_name = argv[0];
@ -49,36 +32,43 @@ int main(int argc, char* argv[]) {
"Tool for parsing an RTP dump file to text output.\n"
"Run " +
program_name +
" --helpshort for usage.\n"
" --help for usage.\n"
"Example usage:\n" +
program_name + " input.rtp output.txt\n\n" +
"Output is sent to stdout if no output file is given." +
"Note that this tool can read files with our without payloads.";
google::SetUsageMessage(usage);
google::ParseCommandLineFlags(&argc, &argv, true);
if (argc != 2 && argc != 3) {
// Print usage information.
printf("%s", google::ProgramUsage());
return 0;
"Output is sent to stdout if no output file is given. " +
"Note that this tool can read files with or without payloads.\n";
if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) ||
FLAG_help || (argc != 2 && argc != 3)) {
printf("%s", usage.c_str());
if (FLAG_help) {
rtc::FlagList::Print(nullptr, false);
return 0;
}
return 1;
}
RTC_CHECK(FLAG_red >= 0 && FLAG_red <= 127); // Payload type
RTC_CHECK(FLAG_audio_level == -1 || // Default
(FLAG_audio_level > 0 && FLAG_audio_level <= 255)); // Extension ID
RTC_CHECK(FLAG_abs_send_time == -1 || // Default
(FLAG_abs_send_time > 0 && FLAG_abs_send_time <= 255)); // Extension ID
printf("Input file: %s\n", argv[1]);
std::unique_ptr<webrtc::test::RtpFileSource> file_source(
webrtc::test::RtpFileSource::Create(argv[1]));
assert(file_source.get());
// Set RTP extension IDs.
bool print_audio_level = false;
if (!google::GetCommandLineFlagInfoOrDie("audio_level").is_default) {
if (FLAG_audio_level != -1) {
print_audio_level = true;
file_source->RegisterRtpHeaderExtension(webrtc::kRtpExtensionAudioLevel,
FLAGS_audio_level);
FLAG_audio_level);
}
bool print_abs_send_time = false;
if (!google::GetCommandLineFlagInfoOrDie("abs_send_time").is_default) {
if (FLAG_abs_send_time != -1) {
print_abs_send_time = true;
file_source->RegisterRtpHeaderExtension(
webrtc::kRtpExtensionAbsoluteSendTime, FLAGS_abs_send_time);
webrtc::kRtpExtensionAbsoluteSendTime, FLAG_abs_send_time);
}
FILE* out_file;
@ -160,7 +150,7 @@ int main(int argc, char* argv[]) {
}
fprintf(out_file, "\n");
if (packet->header().payloadType == FLAGS_red) {
if (packet->header().payloadType == FLAG_red) {
std::list<webrtc::RTPHeader*> red_headers;
packet->ExtractRedHeaders(&red_headers);
while (!red_headers.empty()) {

55
webrtc/modules/audio_coding/test/delay_test.cc
View File

@ -10,11 +10,11 @@
#include <assert.h>
#include <math.h>
#include <string.h>
#include <iostream>
#include <memory>
#include "gflags/gflags.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
@ -22,19 +22,21 @@
#include "webrtc/modules/audio_coding/test/Channel.h"
#include "webrtc/modules/audio_coding/test/PCMFile.h"
#include "webrtc/modules/audio_coding/test/utility.h"
#include "webrtc/rtc_base/flags.h"
#include "webrtc/system_wrappers/include/event_wrapper.h"
#include "webrtc/test/gtest.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/typedefs.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_int(sample_rate_hz, 16000, "Sampling rate in Hertz.");
DEFINE_int(num_channels, 1, "Number of Channels.");
DEFINE_string(input_file, "", "Input file, PCM16 32 kHz, optional.");
DEFINE_int32(delay, 0, "Delay in millisecond.");
DEFINE_int(delay, 0, "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).");
DEFINE_bool(help, false, "Print this message.");
namespace webrtc {
@ -80,16 +82,16 @@ class DelayTest {
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;
if (strlen(FLAG_input_file) > 0)
file_name = FLAG_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_delay > 0) {
ASSERT_EQ(0, acm_b_->SetMinimumPlayoutDelay(FLAGS_delay)) <<
if (FLAG_delay > 0) {
ASSERT_EQ(0, acm_b_->SetMinimumPlayoutDelay(FLAG_delay)) <<
"Failed to set minimum delay.\n";
}
@ -166,8 +168,8 @@ class DelayTest {
void OpenOutFile(const char* output_id) {
std::stringstream file_stream;
file_stream << "delay_test_" << FLAGS_codec << "_" << FLAGS_sample_rate_hz
<< "Hz" << "_" << FLAGS_delay << "ms.pcm";
file_stream << "delay_test_" << FLAG_codec << "_" << FLAG_sample_rate_hz
<< "Hz" << "_" << FLAG_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");
@ -240,26 +242,33 @@ class DelayTest {
} // 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 (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true)) {
return 1;
}
if (FLAG_help) {
rtc::FlagList::Print(nullptr, false);
return 0;
}
if (FLAGS_sample_rate_hz != 8000 &&
FLAGS_sample_rate_hz != 16000 &&
FLAGS_sample_rate_hz != 32000 &&
FLAGS_sample_rate_hz != 48000) {
webrtc::TestSettings test_setting;
strcpy(test_setting.codec.name, FLAG_codec);
if (FLAG_sample_rate_hz != 8000 &&
FLAG_sample_rate_hz != 16000 &&
FLAG_sample_rate_hz != 32000 &&
FLAG_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) {
test_setting.codec.sample_rate_hz = FLAG_sample_rate_hz;
if (FLAG_num_channels < 1 || FLAG_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;
test_setting.codec.num_channels = FLAG_num_channels;
test_setting.acm.dtx = FLAG_dtx;
test_setting.acm.fec = FLAG_fec;
test_setting.packet_loss = FLAG_packet_loss;
webrtc::DelayTest delay_test;
delay_test.Initialize();

62
webrtc/modules/audio_coding/test/insert_packet_with_timing.cc
View File

@ -9,31 +9,32 @@
*/
#include <stdio.h>
#include <string.h>
#include <memory>
#include "gflags/gflags.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
#include "webrtc/modules/audio_coding/test/Channel.h"
#include "webrtc/modules/audio_coding/test/PCMFile.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/rtc_base/flags.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/test/gtest.h"
#include "webrtc/test/testsupport/fileutils.h"
// Codec.
DEFINE_string(codec, "opus", "Codec Name");
DEFINE_int32(codec_sample_rate_hz, 48000, "Sampling rate in Hertz.");
DEFINE_int32(codec_channels, 1, "Number of channels of the codec.");
DEFINE_int(codec_sample_rate_hz, 48000, "Sampling rate in Hertz.");
DEFINE_int(codec_channels, 1, "Number of channels of the codec.");
// PCM input/output.
DEFINE_string(input, "", "Input PCM file at 16 kHz.");
DEFINE_bool(input_stereo, false, "Input is stereo.");
DEFINE_int32(input_fs_hz, 32000, "Input sample rate Hz.");
DEFINE_int(input_fs_hz, 32000, "Input sample rate Hz.");
DEFINE_string(output, "insert_rtp_with_timing_out.pcm", "OutputFile");
DEFINE_int32(output_fs_hz, 32000, "Output sample rate Hz");
DEFINE_int(output_fs_hz, 32000, "Output sample rate Hz");
// Timing files
DEFINE_string(seq_num, "seq_num", "Sequence number file.");
@ -45,7 +46,9 @@ DEFINE_string(delay, "", "Log for delay.");
// Other setups
DEFINE_bool(verbose, false, "Verbosity.");
DEFINE_double(loss_rate, 0, "Rate of packet loss < 1");
DEFINE_float(loss_rate, 0, "Rate of packet loss < 1");
DEFINE_bool(help, false, "Prints this message.");
const int32_t kAudioPlayedOut = 0x00000001;
const int32_t kPacketPushedIn = 0x00000001 << 1;
@ -61,10 +64,10 @@ class InsertPacketWithTiming {
send_acm_(AudioCodingModule::Create(0, sender_clock_)),
receive_acm_(AudioCodingModule::Create(0, receiver_clock_)),
channel_(new Channel),
seq_num_fid_(fopen(FLAGS_seq_num.c_str(), "rt")),
send_ts_fid_(fopen(FLAGS_send_ts.c_str(), "rt")),
receive_ts_fid_(fopen(FLAGS_receive_ts.c_str(), "rt")),
pcm_out_fid_(fopen(FLAGS_output.c_str(), "wb")),
seq_num_fid_(fopen(FLAG_seq_num, "rt")),
send_ts_fid_(fopen(FLAG_send_ts, "rt")),
receive_ts_fid_(fopen(FLAG_receive_ts, "rt")),
pcm_out_fid_(fopen(FLAG_output, "wb")),
samples_in_1ms_(48),
num_10ms_in_codec_frame_(2), // Typical 20 ms frames.
time_to_insert_packet_ms_(3), // An arbitrary offset on pushing packet.
@ -90,9 +93,9 @@ class InsertPacketWithTiming {
next_receive_ts_ = ReceiveTimestamp();
CodecInst codec;
ASSERT_EQ(0, AudioCodingModule::Codec(FLAGS_codec.c_str(), &codec,
FLAGS_codec_sample_rate_hz,
FLAGS_codec_channels));
ASSERT_EQ(0, AudioCodingModule::Codec(FLAG_codec, &codec,
FLAG_codec_sample_rate_hz,
FLAG_codec_channels));
ASSERT_EQ(0, receive_acm_->InitializeReceiver());
ASSERT_EQ(0, send_acm_->RegisterSendCodec(codec));
ASSERT_EQ(true, receive_acm_->RegisterReceiveCodec(codec.pltype,
@ -105,27 +108,27 @@ class InsertPacketWithTiming {
channel_->RegisterReceiverACM(receive_acm_.get());
send_acm_->RegisterTransportCallback(channel_);
if (FLAGS_input.size() == 0) {
if (strlen(FLAG_input) == 0) {
std::string file_name = test::ResourcePath("audio_coding/testfile32kHz",
"pcm");
pcm_in_fid_.Open(file_name, 32000, "r", true); // auto-rewind
std::cout << "Input file " << file_name << " 32 kHz mono." << std::endl;
} else {
pcm_in_fid_.Open(FLAGS_input, static_cast<uint16_t>(FLAGS_input_fs_hz),
pcm_in_fid_.Open(FLAG_input, static_cast<uint16_t>(FLAG_input_fs_hz),
"r", true); // auto-rewind
std::cout << "Input file " << FLAGS_input << "at " << FLAGS_input_fs_hz
<< " Hz in " << ((FLAGS_input_stereo) ? "stereo." : "mono.")
std::cout << "Input file " << FLAG_input << "at " << FLAG_input_fs_hz
<< " Hz in " << ((FLAG_input_stereo) ? "stereo." : "mono.")
<< std::endl;
pcm_in_fid_.ReadStereo(FLAGS_input_stereo);
pcm_in_fid_.ReadStereo(FLAG_input_stereo);
}
ASSERT_TRUE(pcm_out_fid_ != NULL);
std::cout << "Output file " << FLAGS_output << " at " << FLAGS_output_fs_hz
std::cout << "Output file " << FLAG_output << " at " << FLAG_output_fs_hz
<< " Hz." << std::endl;
// Other setups
if (FLAGS_loss_rate > 0)
loss_threshold_ = RAND_MAX * FLAGS_loss_rate;
if (FLAG_loss_rate > 0)
loss_threshold_ = RAND_MAX * FLAG_loss_rate;
else
loss_threshold_ = 0;
}
@ -144,7 +147,7 @@ class InsertPacketWithTiming {
if (time_to_playout_audio_ms_ == 0) {
time_to_playout_audio_ms_ = kPlayoutPeriodMs;
bool muted;
receive_acm_->PlayoutData10Ms(static_cast<int>(FLAGS_output_fs_hz),
receive_acm_->PlayoutData10Ms(static_cast<int>(FLAG_output_fs_hz),
&frame_, &muted);
ASSERT_FALSE(muted);
fwrite(frame_.data(), sizeof(*frame_.data()),
@ -180,7 +183,7 @@ class InsertPacketWithTiming {
lost = true;
}
if (FLAGS_verbose) {
if (FLAG_verbose) {
if (!lost) {
std::cout << "\nInserting packet number " << seq_num
<< " timestamp " << ts << std::endl;
@ -279,13 +282,20 @@ class InsertPacketWithTiming {
} // webrtc
int main(int argc, char* argv[]) {
google::ParseCommandLineFlags(&argc, &argv, true);
if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true)) {
return 1;
}
if (FLAG_help) {
rtc::FlagList::Print(nullptr, false);
return 0;
}
webrtc::InsertPacketWithTiming test;
test.SetUp();
FILE* delay_log = NULL;
if (FLAGS_delay.size() > 0) {
delay_log = fopen(FLAGS_delay.c_str(), "wt");
if (strlen(FLAG_delay) > 0) {
delay_log = fopen(FLAG_delay, "wt");
if (delay_log == NULL) {
std::cout << "Cannot open the file to log delay values." << std::endl;
exit(1);

7
webrtc/modules/audio_processing/BUILD.gn
View File

@ -724,7 +724,6 @@ if (rtc_include_tests) {
"../../rtc_base:protobuf_utils",
"../../rtc_base:rtc_base_approved",
"../../system_wrappers:system_wrappers_default",
"//third_party/gflags:gflags",
]
} # unpack_aecdump
@ -755,7 +754,6 @@ if (rtc_include_tests) {
"aec_dump",
"aec_dump:aec_dump_impl",
"//testing/gtest",
"//third_party/gflags:gflags",
]
} # audioproc_f
}
@ -794,11 +792,11 @@ if (rtc_include_tests) {
"..:module_api",
"../..:webrtc_common",
"../../common_audio:common_audio",
"../../rtc_base:rtc_base_approved",
"../../system_wrappers:metrics_default",
"../../system_wrappers:system_wrappers",
"../../test:test_support",
"//testing/gtest",
"//third_party/gflags",
]
}
@ -828,7 +826,6 @@ if (rtc_include_tests) {
"../../common_audio:common_audio",
"../../rtc_base:rtc_base_approved",
"../../system_wrappers:metrics_default",
"//third_party/gflags",
]
}
@ -841,10 +838,10 @@ if (rtc_include_tests) {
deps = [
":audio_processing",
":audioproc_test_utils",
"../../rtc_base:rtc_base_approved",
"../../system_wrappers:metrics_default",
"../../test:test_support",
"//testing/gtest",
"//third_party/gflags",
]
}
}

26
webrtc/modules/audio_processing/beamformer/nonlinear_beamformer_test.cc
View File

@ -10,12 +10,12 @@
#include <vector>
#include "gflags/gflags.h"
#include "webrtc/common_audio/channel_buffer.h"
#include "webrtc/common_audio/wav_file.h"
#include "webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.h"
#include "webrtc/modules/audio_processing/test/test_utils.h"
#include "webrtc/rtc_base/checks.h"
#include "webrtc/rtc_base/flags.h"
#include "webrtc/rtc_base/format_macros.h"
DEFINE_string(i, "", "The name of the input file to read from.");
@ -24,6 +24,7 @@ DEFINE_string(mic_positions, "",
"Space delimited cartesian coordinates of microphones in meters. "
"The coordinates of each point are contiguous. "
"For a two element array: \"x1 y1 z1 x2 y2 z2\"");
DEFINE_bool(help, false, "Prints this message.");
namespace webrtc {
namespace {
@ -34,29 +35,36 @@ const int kChunkSizeMs = 1000 / kChunksPerSecond;
const char kUsage[] =
"Command-line tool to run beamforming on WAV files. The signal is passed\n"
"in as a single band, unlike the audio processing interface which splits\n"
"signals into multiple bands.";
"signals into multiple bands.\n";
} // namespace
int main(int argc, char* argv[]) {
google::SetUsageMessage(kUsage);
google::ParseCommandLineFlags(&argc, &argv, true);
if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) ||
FLAG_help || argc != 1) {
printf("%s", kUsage);
if (FLAG_help) {
rtc::FlagList::Print(nullptr, false);
return 0;
}
return 1;
}
WavReader in_file(FLAGS_i);
WavWriter out_file(FLAGS_o, in_file.sample_rate(), in_file.num_channels());
WavReader in_file(FLAG_i);
WavWriter out_file(FLAG_o, in_file.sample_rate(), in_file.num_channels());
const size_t num_mics = in_file.num_channels();
const std::vector<Point> array_geometry =
ParseArrayGeometry(FLAGS_mic_positions, num_mics);
ParseArrayGeometry(FLAG_mic_positions, num_mics);
RTC_CHECK_EQ(array_geometry.size(), num_mics);
NonlinearBeamformer bf(array_geometry, array_geometry.size());
bf.Initialize(kChunkSizeMs, in_file.sample_rate());
printf("Input file: %s\nChannels: %" PRIuS ", Sample rate: %d Hz\n\n",
FLAGS_i.c_str(), in_file.num_channels(), in_file.sample_rate());
FLAG_i, in_file.num_channels(), in_file.sample_rate());
printf("Output file: %s\nChannels: %" PRIuS ", Sample rate: %d Hz\n\n",
FLAGS_o.c_str(), out_file.num_channels(), out_file.sample_rate());
FLAG_o, out_file.num_channels(), out_file.sample_rate());
ChannelBuffer<float> buf(
rtc::CheckedDivExact(in_file.sample_rate(), kChunksPerSecond),

32
webrtc/modules/audio_processing/intelligibility/test/intelligibility_proc.cc
View File

@ -8,7 +8,6 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "gflags/gflags.h"
#include "webrtc/common_audio/channel_buffer.h"
#include "webrtc/common_audio/include/audio_util.h"
#include "webrtc/common_audio/wav_file.h"
@ -16,7 +15,7 @@
#include "webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h"
#include "webrtc/modules/audio_processing/noise_suppression_impl.h"
#include "webrtc/rtc_base/criticalsection.h"
#include "webrtc/test/gtest.h"
#include "webrtc/rtc_base/flags.h"
using std::complex;
@ -26,16 +25,24 @@ namespace {
DEFINE_string(clear_file, "speech.wav", "Input file with clear speech.");
DEFINE_string(noise_file, "noise.wav", "Input file with noise data.");
DEFINE_string(out_file, "proc_enhanced.wav", "Enhanced output file.");
DEFINE_bool(help, false, "Print this message.");
// void function for gtest
void void_main(int argc, char* argv[]) {
google::SetUsageMessage(
"\n\nInput files must be little-endian 16-bit signed raw PCM.\n");
google::ParseCommandLineFlags(&argc, &argv, true);
int int_main(int argc, char* argv[]) {
if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true)) {
return 1;
}
if (FLAG_help) {
rtc::FlagList::Print(nullptr, false);
return 0;
}
if (argc != 1) {
printf("\n\nInput files must be little-endian 16-bit signed raw PCM.\n");
return 0;
}
WavReader in_file(FLAGS_clear_file);
WavReader noise_file(FLAGS_noise_file);
WavWriter out_file(FLAGS_out_file, in_file.sample_rate(),
WavReader in_file(FLAG_clear_file);
WavReader noise_file(FLAG_noise_file);
WavWriter out_file(FLAG_out_file, in_file.sample_rate(),
in_file.num_channels());
rtc::CriticalSection crit;
NoiseSuppressionImpl ns(&crit);
@ -77,12 +84,13 @@ void void_main(int argc, char* argv[]) {
FloatToFloatS16(in.data(), in.size(), in.data());
out_file.WriteSamples(in.data(), in.size());
}
return 0;
}
} // namespace
} // namespace webrtc
int main(int argc, char* argv[]) {
webrtc::void_main(argc, argv);
return 0;
return webrtc::int_main(argc, argv);
}

328
webrtc/modules/audio_processing/test/audioproc_float.cc
View File

@ -13,11 +13,11 @@
#include <string.h>
#include "gflags/gflags.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/modules/audio_processing/test/aec_dump_based_simulator.h"
#include "webrtc/modules/audio_processing/test/audio_processing_simulator.h"
#include "webrtc/modules/audio_processing/test/wav_based_simulator.h"
#include "webrtc/rtc_base/flags.h"
namespace webrtc {
namespace test {
@ -32,7 +32,7 @@ const char kUsageDescription[] =
"\n\n"
"Command-line tool to simulate a call using the audio "
"processing module, either based on wav files or "
"protobuf debug dump recordings.";
"protobuf debug dump recordings.\n";
DEFINE_string(dump_input, "", "Aec dump input filename");
DEFINE_string(dump_output, "", "Aec dump output filename");
@ -41,126 +41,126 @@ DEFINE_string(o, "", "Forward stream output wav filename");
DEFINE_string(ri, "", "Reverse stream input wav filename");
DEFINE_string(ro, "", "Reverse stream output wav filename");
DEFINE_string(artificial_nearend, "", "Artificial nearend wav filename");
DEFINE_int32(output_num_channels,
kParameterNotSpecifiedValue,
"Number of forward stream output channels");
DEFINE_int32(reverse_output_num_channels,
kParameterNotSpecifiedValue,
"Number of Reverse stream output channels");
DEFINE_int32(output_sample_rate_hz,
kParameterNotSpecifiedValue,
"Forward stream output sample rate in Hz");
DEFINE_int32(reverse_output_sample_rate_hz,
kParameterNotSpecifiedValue,
"Reverse stream output sample rate in Hz");
DEFINE_int(output_num_channels,
kParameterNotSpecifiedValue,
"Number of forward stream output channels");
DEFINE_int(reverse_output_num_channels,
kParameterNotSpecifiedValue,
"Number of Reverse stream output channels");
DEFINE_int(output_sample_rate_hz,
kParameterNotSpecifiedValue,
"Forward stream output sample rate in Hz");
DEFINE_int(reverse_output_sample_rate_hz,
kParameterNotSpecifiedValue,
"Reverse stream output sample rate in Hz");
DEFINE_string(mic_positions,
"",
"Space delimited cartesian coordinates of microphones in "
"meters. The coordinates of each point are contiguous. For a "
"two element array: \"x1 y1 z1 x2 y2 z2\"");
DEFINE_int32(target_angle_degrees,
90,
"The azimuth of the target in degrees (0-359). Only applies to "
"beamforming.");
DEFINE_int(target_angle_degrees,
90,
"The azimuth of the target in degrees (0-359). Only applies to "
"beamforming.");
DEFINE_bool(fixed_interface,
false,
"Use the fixed interface when operating on wav files");
DEFINE_int32(aec,
kParameterNotSpecifiedValue,
"Activate (1) or deactivate(0) the echo canceller");
DEFINE_int32(aecm,
kParameterNotSpecifiedValue,
"Activate (1) or deactivate(0) the mobile echo controller");
DEFINE_int32(ed,
kParameterNotSpecifiedValue,
"Activate (1) or deactivate (0) the residual echo detector");
DEFINE_int(aec,
kParameterNotSpecifiedValue,
"Activate (1) or deactivate(0) the echo canceller");
DEFINE_int(aecm,
kParameterNotSpecifiedValue,
"Activate (1) or deactivate(0) the mobile echo controller");
DEFINE_int(ed,
kParameterNotSpecifiedValue,
"Activate (1) or deactivate (0) the residual echo detector");
DEFINE_string(ed_graph, "", "Output filename for graph of echo likelihood");
DEFINE_int32(agc,
kParameterNotSpecifiedValue,
"Activate (1) or deactivate(0) the AGC");
DEFINE_int32(agc2,
kParameterNotSpecifiedValue,
"Activate (1) or deactivate(0) the AGC2");
DEFINE_int32(hpf,
kParameterNotSpecifiedValue,
"Activate (1) or deactivate(0) the high-pass filter");
DEFINE_int32(ns,
kParameterNotSpecifiedValue,
"Activate (1) or deactivate(0) the noise suppressor");
DEFINE_int32(ts,
kParameterNotSpecifiedValue,
"Activate (1) or deactivate(0) the transient suppressor");
DEFINE_int32(bf,
kParameterNotSpecifiedValue,
"Activate (1) or deactivate(0) the beamformer");
DEFINE_int32(ie,
kParameterNotSpecifiedValue,
"Activate (1) or deactivate(0) the intelligibility enhancer");
DEFINE_int32(vad,
kParameterNotSpecifiedValue,
"Activate (1) or deactivate(0) the voice activity detector");
DEFINE_int32(le,
kParameterNotSpecifiedValue,
"Activate (1) or deactivate(0) the level estimator");
DEFINE_int(agc,
kParameterNotSpecifiedValue,
"Activate (1) or deactivate(0) the AGC");
DEFINE_int(agc2,
kParameterNotSpecifiedValue,
"Activate (1) or deactivate(0) the AGC2");
DEFINE_int(hpf,
kParameterNotSpecifiedValue,
"Activate (1) or deactivate(0) the high-pass filter");
DEFINE_int(ns,
kParameterNotSpecifiedValue,
"Activate (1) or deactivate(0) the noise suppressor");
DEFINE_int(ts,
kParameterNotSpecifiedValue,
"Activate (1) or deactivate(0) the transient suppressor");
DEFINE_int(bf,
kParameterNotSpecifiedValue,
"Activate (1) or deactivate(0) the beamformer");
DEFINE_int(ie,
kParameterNotSpecifiedValue,
"Activate (1) or deactivate(0) the intelligibility enhancer");
DEFINE_int(vad,
kParameterNotSpecifiedValue,
"Activate (1) or deactivate(0) the voice activity detector");
DEFINE_int(le,
kParameterNotSpecifiedValue,
"Activate (1) or deactivate(0) the level estimator");
DEFINE_bool(all_default,
false,
"Activate all of the default components (will be overridden by any "
"other settings)");
DEFINE_int32(aec_suppression_level,
kParameterNotSpecifiedValue,
"Set the aec suppression level (0-2)");
DEFINE_int32(delay_agnostic,
kParameterNotSpecifiedValue,
"Activate (1) or deactivate(0) the AEC delay agnostic mode");
DEFINE_int32(extended_filter,
kParameterNotSpecifiedValue,
"Activate (1) or deactivate(0) the AEC extended filter mode");
DEFINE_int32(drift_compensation,
kParameterNotSpecifiedValue,
"Activate (1) or deactivate(0) the drift compensation");
DEFINE_int32(aec3,
kParameterNotSpecifiedValue,
"Activate (1) or deactivate(0) the experimental AEC mode AEC3");
DEFINE_int32(lc,
kParameterNotSpecifiedValue,
"Activate (1) or deactivate(0) the level control");
DEFINE_int32(experimental_agc,
kParameterNotSpecifiedValue,
"Activate (1) or deactivate(0) the experimental AGC");
DEFINE_int32(
DEFINE_int(aec_suppression_level,
kParameterNotSpecifiedValue,
"Set the aec suppression level (0-2)");
DEFINE_int(delay_agnostic,
kParameterNotSpecifiedValue,
"Activate (1) or deactivate(0) the AEC delay agnostic mode");
DEFINE_int(extended_filter,
kParameterNotSpecifiedValue,
"Activate (1) or deactivate(0) the AEC extended filter mode");
DEFINE_int(drift_compensation,
kParameterNotSpecifiedValue,
"Activate (1) or deactivate(0) the drift compensation");
DEFINE_int(aec3,
kParameterNotSpecifiedValue,
"Activate (1) or deactivate(0) the experimental AEC mode AEC3");
DEFINE_int(lc,
kParameterNotSpecifiedValue,
"Activate (1) or deactivate(0) the level control");
DEFINE_int(experimental_agc,
kParameterNotSpecifiedValue,
"Activate (1) or deactivate(0) the experimental AGC");
DEFINE_int(
refined_adaptive_filter,
kParameterNotSpecifiedValue,
"Activate (1) or deactivate(0) the refined adaptive filter functionality");
DEFINE_int32(aecm_routing_mode,
kParameterNotSpecifiedValue,
"Specify the AECM routing mode (0-4)");
DEFINE_int32(aecm_comfort_noise,
kParameterNotSpecifiedValue,
"Activate (1) or deactivate(0) the AECM comfort noise");
DEFINE_int32(agc_mode,
kParameterNotSpecifiedValue,
"Specify the AGC mode (0-2)");
DEFINE_int32(agc_target_level,
kParameterNotSpecifiedValue,
"Specify the AGC target level (0-31)");
DEFINE_int32(agc_limiter,
kParameterNotSpecifiedValue,
"Activate (1) or deactivate(0) the level estimator");
DEFINE_int32(agc_compression_gain,
kParameterNotSpecifiedValue,
"Specify the AGC compression gain (0-90)");
DEFINE_int32(vad_likelihood,
kParameterNotSpecifiedValue,
"Specify the VAD likelihood (0-3)");
DEFINE_int32(ns_level,
kParameterNotSpecifiedValue,
"Specify the NS level (0-3)");
DEFINE_int32(stream_delay,
kParameterNotSpecifiedValue,
"Specify the stream delay in ms to use");
DEFINE_int32(stream_drift_samples,
kParameterNotSpecifiedValue,
"Specify the number of stream drift samples to use");
DEFINE_int(aecm_routing_mode,
kParameterNotSpecifiedValue,
"Specify the AECM routing mode (0-4)");
DEFINE_int(aecm_comfort_noise,
kParameterNotSpecifiedValue,
"Activate (1) or deactivate(0) the AECM comfort noise");
DEFINE_int(agc_mode,
kParameterNotSpecifiedValue,
"Specify the AGC mode (0-2)");
DEFINE_int(agc_target_level,
kParameterNotSpecifiedValue,
"Specify the AGC target level (0-31)");
DEFINE_int(agc_limiter,
kParameterNotSpecifiedValue,
"Activate (1) or deactivate(0) the level estimator");
DEFINE_int(agc_compression_gain,
kParameterNotSpecifiedValue,
"Specify the AGC compression gain (0-90)");
DEFINE_int(vad_likelihood,
kParameterNotSpecifiedValue,
"Specify the VAD likelihood (0-3)");
DEFINE_int(ns_level,
kParameterNotSpecifiedValue,
"Specify the NS level (0-3)");
DEFINE_int(stream_delay,
kParameterNotSpecifiedValue,
"Specify the stream delay in ms to use");
DEFINE_int(stream_drift_samples,
kParameterNotSpecifiedValue,
"Specify the number of stream drift samples to use");
DEFINE_bool(performance_report, false, "Report the APM performance ");
DEFINE_bool(verbose, false, "Produce verbose output");
DEFINE_bool(bitexactness_report,
@ -173,6 +173,7 @@ DEFINE_bool(store_intermediate_output,
false,
"Creates new output files after each init");
DEFINE_string(custom_call_order_file, "", "Custom process API call order file");
DEFINE_bool(help, false, "Print this message");
void SetSettingIfSpecified(const std::string value,
rtc::Optional<std::string>* parameter) {
@ -197,7 +198,7 @@ void SetSettingIfFlagSet(int32_t flag, rtc::Optional<bool>* parameter) {
SimulationSettings CreateSettings() {
SimulationSettings settings;
if (FLAGS_all_default) {
if (FLAG_all_default) {
settings.use_le = rtc::Optional<bool>(true);
settings.use_vad = rtc::Optional<bool>(true);
settings.use_ie = rtc::Optional<bool>(false);
@ -210,70 +211,70 @@ SimulationSettings CreateSettings() {
settings.use_aecm = rtc::Optional<bool>(false);
settings.use_ed = rtc::Optional<bool>(false);
}
SetSettingIfSpecified(FLAGS_dump_input, &settings.aec_dump_input_filename);
SetSettingIfSpecified(FLAGS_dump_output, &settings.aec_dump_output_filename);
SetSettingIfSpecified(FLAGS_i, &settings.input_filename);
SetSettingIfSpecified(FLAGS_o, &settings.output_filename);
SetSettingIfSpecified(FLAGS_ri, &settings.reverse_input_filename);
SetSettingIfSpecified(FLAGS_ro, &settings.reverse_output_filename);
SetSettingIfSpecified(FLAGS_artificial_nearend,
SetSettingIfSpecified(FLAG_dump_input, &settings.aec_dump_input_filename);
SetSettingIfSpecified(FLAG_dump_output, &settings.aec_dump_output_filename);
SetSettingIfSpecified(FLAG_i, &settings.input_filename);
SetSettingIfSpecified(FLAG_o, &settings.output_filename);
SetSettingIfSpecified(FLAG_ri, &settings.reverse_input_filename);
SetSettingIfSpecified(FLAG_ro, &settings.reverse_output_filename);
SetSettingIfSpecified(FLAG_artificial_nearend,
&settings.artificial_nearend_filename);
SetSettingIfSpecified(FLAGS_output_num_channels,
SetSettingIfSpecified(FLAG_output_num_channels,
&settings.output_num_channels);
SetSettingIfSpecified(FLAGS_reverse_output_num_channels,
SetSettingIfSpecified(FLAG_reverse_output_num_channels,
&settings.reverse_output_num_channels);
SetSettingIfSpecified(FLAGS_output_sample_rate_hz,
SetSettingIfSpecified(FLAG_output_sample_rate_hz,
&settings.output_sample_rate_hz);
SetSettingIfSpecified(FLAGS_reverse_output_sample_rate_hz,
SetSettingIfSpecified(FLAG_reverse_output_sample_rate_hz,
&settings.reverse_output_sample_rate_hz);
SetSettingIfSpecified(FLAGS_mic_positions, &settings.microphone_positions);
settings.target_angle_degrees = FLAGS_target_angle_degrees;
SetSettingIfFlagSet(FLAGS_aec, &settings.use_aec);
SetSettingIfFlagSet(FLAGS_aecm, &settings.use_aecm);
SetSettingIfFlagSet(FLAGS_ed, &settings.use_ed);
SetSettingIfSpecified(FLAGS_ed_graph, &settings.ed_graph_output_filename);
SetSettingIfFlagSet(FLAGS_agc, &settings.use_agc);
SetSettingIfFlagSet(FLAGS_agc2, &settings.use_agc2);
SetSettingIfFlagSet(FLAGS_hpf, &settings.use_hpf);
SetSettingIfFlagSet(FLAGS_ns, &settings.use_ns);
SetSettingIfFlagSet(FLAGS_ts, &settings.use_ts);
SetSettingIfFlagSet(FLAGS_bf, &settings.use_bf);
SetSettingIfFlagSet(FLAGS_ie, &settings.use_ie);
SetSettingIfFlagSet(FLAGS_vad, &settings.use_vad);
SetSettingIfFlagSet(FLAGS_le, &settings.use_le);
SetSettingIfSpecified(FLAGS_aec_suppression_level,
SetSettingIfSpecified(FLAG_mic_positions, &settings.microphone_positions);
settings.target_angle_degrees = FLAG_target_angle_degrees;
SetSettingIfFlagSet(FLAG_aec, &settings.use_aec);
SetSettingIfFlagSet(FLAG_aecm, &settings.use_aecm);
SetSettingIfFlagSet(FLAG_ed, &settings.use_ed);
SetSettingIfSpecified(FLAG_ed_graph, &settings.ed_graph_output_filename);
SetSettingIfFlagSet(FLAG_agc, &settings.use_agc);
SetSettingIfFlagSet(FLAG_agc2, &settings.use_agc2);
SetSettingIfFlagSet(FLAG_hpf, &settings.use_hpf);
SetSettingIfFlagSet(FLAG_ns, &settings.use_ns);
SetSettingIfFlagSet(FLAG_ts, &settings.use_ts);
SetSettingIfFlagSet(FLAG_bf, &settings.use_bf);
SetSettingIfFlagSet(FLAG_ie, &settings.use_ie);
SetSettingIfFlagSet(FLAG_vad, &settings.use_vad);
SetSettingIfFlagSet(FLAG_le, &settings.use_le);
SetSettingIfSpecified(FLAG_aec_suppression_level,
&settings.aec_suppression_level);
SetSettingIfFlagSet(FLAGS_delay_agnostic, &settings.use_delay_agnostic);
SetSettingIfFlagSet(FLAGS_extended_filter, &settings.use_extended_filter);
SetSettingIfFlagSet(FLAGS_drift_compensation,
SetSettingIfFlagSet(FLAG_delay_agnostic, &settings.use_delay_agnostic);
SetSettingIfFlagSet(FLAG_extended_filter, &settings.use_extended_filter);
SetSettingIfFlagSet(FLAG_drift_compensation,
&settings.use_drift_compensation);
SetSettingIfFlagSet(FLAGS_refined_adaptive_filter,
SetSettingIfFlagSet(FLAG_refined_adaptive_filter,
&settings.use_refined_adaptive_filter);
SetSettingIfFlagSet(FLAGS_aec3, &settings.use_aec3);
SetSettingIfFlagSet(FLAGS_lc, &settings.use_lc);
SetSettingIfFlagSet(FLAGS_experimental_agc, &settings.use_experimental_agc);
SetSettingIfSpecified(FLAGS_aecm_routing_mode, &settings.aecm_routing_mode);
SetSettingIfFlagSet(FLAGS_aecm_comfort_noise,
SetSettingIfFlagSet(FLAG_aec3, &settings.use_aec3);
SetSettingIfFlagSet(FLAG_lc, &settings.use_lc);
SetSettingIfFlagSet(FLAG_experimental_agc, &settings.use_experimental_agc);
SetSettingIfSpecified(FLAG_aecm_routing_mode, &settings.aecm_routing_mode);
SetSettingIfFlagSet(FLAG_aecm_comfort_noise,
&settings.use_aecm_comfort_noise);
SetSettingIfSpecified(FLAGS_agc_mode, &settings.agc_mode);
SetSettingIfSpecified(FLAGS_agc_target_level, &settings.agc_target_level);
SetSettingIfFlagSet(FLAGS_agc_limiter, &settings.use_agc_limiter);
SetSettingIfSpecified(FLAGS_agc_compression_gain,
SetSettingIfSpecified(FLAG_agc_mode, &settings.agc_mode);
SetSettingIfSpecified(FLAG_agc_target_level, &settings.agc_target_level);
SetSettingIfFlagSet(FLAG_agc_limiter, &settings.use_agc_limiter);
SetSettingIfSpecified(FLAG_agc_compression_gain,
&settings.agc_compression_gain);
SetSettingIfSpecified(FLAGS_vad_likelihood, &settings.vad_likelihood);
SetSettingIfSpecified(FLAGS_ns_level, &settings.ns_level);
SetSettingIfSpecified(FLAGS_stream_delay, &settings.stream_delay);
SetSettingIfSpecified(FLAGS_stream_drift_samples,
SetSettingIfSpecified(FLAG_vad_likelihood, &settings.vad_likelihood);
SetSettingIfSpecified(FLAG_ns_level, &settings.ns_level);
SetSettingIfSpecified(FLAG_stream_delay, &settings.stream_delay);
SetSettingIfSpecified(FLAG_stream_drift_samples,
&settings.stream_drift_samples);
SetSettingIfSpecified(FLAGS_custom_call_order_file,
SetSettingIfSpecified(FLAG_custom_call_order_file,
&settings.custom_call_order_filename);
settings.report_performance = FLAGS_performance_report;
settings.use_verbose_logging = FLAGS_verbose;
settings.report_bitexactness = FLAGS_bitexactness_report;
settings.discard_all_settings_in_aecdump = FLAGS_discard_settings_in_aecdump;
settings.fixed_interface = FLAGS_fixed_interface;
settings.store_intermediate_output = FLAGS_store_intermediate_output;
settings.report_performance = FLAG_performance_report;
settings.use_verbose_logging = FLAG_verbose;
settings.report_bitexactness = FLAG_bitexactness_report;
settings.discard_all_settings_in_aecdump = FLAG_discard_settings_in_aecdump;
settings.fixed_interface = FLAG_fixed_interface;
settings.store_intermediate_output = FLAG_store_intermediate_output;
return settings;
}
@ -422,8 +423,15 @@ void PerformBasicParameterSanityChecks(const SimulationSettings& settings) {
} // namespace
int main(int argc, char* argv[]) {
google::SetUsageMessage(kUsageDescription);
google::ParseCommandLineFlags(&argc, &argv, true);
if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) ||
FLAG_help || argc != 1) {
printf("%s", kUsageDescription);
if (FLAG_help) {
rtc::FlagList::Print(nullptr, false);
return 0;
}
return 1;
}
SimulationSettings settings = CreateSettings();
PerformBasicParameterSanityChecks(settings);

1
webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn
View File

@ -24,7 +24,6 @@ rtc_executable("conversational_speech_generator") {
":lib",
"../../../../rtc_base:rtc_base_approved",
"../../../../test:test_support",
"//third_party/gflags",
]
}

33
webrtc/modules/audio_processing/test/conversational_speech/generator.cc
View File

@ -10,12 +10,12 @@
#include <iostream>
#include "gflags/gflags.h"
#include "webrtc/modules/audio_processing/test/conversational_speech/config.h"
#include "webrtc/modules/audio_processing/test/conversational_speech/multiend_call.h"
#include "webrtc/modules/audio_processing/test/conversational_speech/simulator.h"
#include "webrtc/modules/audio_processing/test/conversational_speech/timing.h"
#include "webrtc/modules/audio_processing/test/conversational_speech/wavreader_factory.h"
#include "webrtc/rtc_base/flags.h"
#include "webrtc/rtc_base/ptr_util.h"
#include "webrtc/test/testsupport/fileutils.h"
@ -23,14 +23,6 @@ namespace webrtc {
namespace test {
namespace {
// Adapting DirExists/FileExists interfaces to DEFINE_validator.
auto dir_exists = [](const char* c, const std::string& dirpath) {
return DirExists(dirpath);
};
auto file_exists = [](const char* c, const std::string& filepath) {
return FileExists(filepath);
};
const char kUsageDescription[] =
"Usage: conversational_speech_generator\n"
" -i <path/to/source/audiotracks>\n"
@ -38,21 +30,30 @@ const char kUsageDescription[] =
" -o <output/path>\n"
"\n\n"
"Command-line tool to generate multiple-end audio tracks to simulate "
"conversational speech with two or more participants.";
"conversational speech with two or more participants.\n";
DEFINE_string(i, "", "Directory containing the speech turn wav files");
DEFINE_validator(i, dir_exists);
DEFINE_string(t, "", "Path to the timing text file");
DEFINE_validator(t, file_exists);
DEFINE_string(o, "", "Output wav files destination path");
DEFINE_validator(o, dir_exists);
DEFINE_bool(help, false, "Prints this message");
} // namespace
int main(int argc, char* argv[]) {
google::SetUsageMessage(kUsageDescription);
google::ParseCommandLineFlags(&argc, &argv, true);
conversational_speech::Config config(FLAGS_i, FLAGS_t, FLAGS_o);
if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) ||
FLAG_help || argc != 1) {
printf("%s", kUsageDescription);
if (FLAG_help) {
rtc::FlagList::Print(nullptr, false);
return 0;
}
return 1;
}
RTC_CHECK(DirExists(FLAG_i));
RTC_CHECK(FileExists(FLAG_t));
RTC_CHECK(DirExists(FLAG_o));
conversational_speech::Config config(FLAG_i, FLAG_t, FLAG_o);
// Load timing.
std::vector<conversational_speech::Turn> timing =

2
webrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_gencfgs.py
View File

@ -27,7 +27,7 @@ def _GenerateDefaultOverridden(config_override):
The default settings are loaded via "-all_default".
Check "src/webrtc/modules/audio_processing/test/audioproc_float.cc" and search
for "if (FLAGS_all_default) {".
for "if (FLAG_all_default) {".
For instance, in 55eb6d621489730084927868fed195d3645a9ec9 the default is this:
settings.use_aec = rtc::Optional<bool>(true);

80
webrtc/modules/audio_processing/test/unpack.cc
View File

@ -17,9 +17,9 @@
#include <memory>
#include "gflags/gflags.h"
#include "webrtc/modules/audio_processing/test/protobuf_utils.h"
#include "webrtc/modules/audio_processing/test/test_utils.h"
#include "webrtc/rtc_base/flags.h"
#include "webrtc/rtc_base/format_macros.h"
#include "webrtc/rtc_base/ignore_wundef.h"
#include "webrtc/typedefs.h"
@ -45,6 +45,7 @@ DEFINE_bool(raw, false, "Write raw data instead of a WAV file.");
DEFINE_bool(text,
false,
"Write non-audio files as text files instead of binary files.");
DEFINE_bool(help, false, "Print this message.");
#define PRINT_CONFIG(field_name) \
if (msg.has_##field_name()) { \
@ -70,11 +71,14 @@ int do_main(int argc, char* argv[]) {
std::string program_name = argv[0];
std::string usage = "Commandline tool to unpack audioproc debug files.\n"
"Example usage:\n" + program_name + " debug_dump.pb\n";
google::SetUsageMessage(usage);
google::ParseCommandLineFlags(&argc, &argv, true);
if (argc < 2) {
printf("%s", google::ProgramUsage());
if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) ||
FLAG_help || argc < 2) {
printf("%s", usage.c_str());
if (FLAG_help) {
rtc::FlagList::Print(nullptr, false);
return 0;
}
return 1;
}
@ -95,7 +99,7 @@ int do_main(int argc, char* argv[]) {
std::unique_ptr<RawFile> input_raw_file;
std::unique_ptr<RawFile> output_raw_file;
FILE* settings_file = OpenFile(FLAGS_settings_file, "wb");
FILE* settings_file = OpenFile(FLAG_settings_file, "wb");
while (ReadMessageFromFile(debug_file, &event_msg)) {
if (event_msg.type() == Event::REVERSE_STREAM) {
@ -106,8 +110,9 @@ int do_main(int argc, char* argv[]) {
const ReverseStream msg = event_msg.reverse_stream();
if (msg.has_data()) {
if (FLAGS_raw && !reverse_raw_file) {
reverse_raw_file.reset(new RawFile(FLAGS_reverse_file + ".pcm"));
if (FLAG_raw && !reverse_raw_file) {
reverse_raw_file.reset(new RawFile(std::string(FLAG_reverse_file) +
".pcm"));
}
// TODO(aluebs): Replace "num_reverse_channels *
// reverse_samples_per_channel" with "msg.data().size() /
@ -118,8 +123,9 @@ int do_main(int argc, char* argv[]) {
reverse_wav_file.get(),
reverse_raw_file.get());
} else if (msg.channel_size() > 0) {
if (FLAGS_raw && !reverse_raw_file) {
reverse_raw_file.reset(new RawFile(FLAGS_reverse_file + ".float"));
if (FLAG_raw && !reverse_raw_file) {
reverse_raw_file.reset(new RawFile(std::string(FLAG_reverse_file) +
".float"));
}
std::unique_ptr<const float* []> data(
new const float* [num_reverse_channels]);
@ -141,16 +147,18 @@ int do_main(int argc, char* argv[]) {
const Stream msg = event_msg.stream();
if (msg.has_input_data()) {
if (FLAGS_raw && !input_raw_file) {
input_raw_file.reset(new RawFile(FLAGS_input_file + ".pcm"));
if (FLAG_raw && !input_raw_file) {
input_raw_file.reset(new RawFile(std::string(FLAG_input_file) +
".pcm"));
}
WriteIntData(reinterpret_cast<const int16_t*>(msg.input_data().data()),
num_input_channels * input_samples_per_channel,
input_wav_file.get(),
input_raw_file.get());
} else if (msg.input_channel_size() > 0) {
if (FLAGS_raw && !input_raw_file) {
input_raw_file.reset(new RawFile(FLAGS_input_file + ".float"));
if (FLAG_raw && !input_raw_file) {
input_raw_file.reset(new RawFile(std::string(FLAG_input_file) +
".float"));
}
std::unique_ptr<const float* []> data(
new const float* [num_input_channels]);
@ -165,16 +173,18 @@ int do_main(int argc, char* argv[]) {
}
if (msg.has_output_data()) {
if (FLAGS_raw && !output_raw_file) {
output_raw_file.reset(new RawFile(FLAGS_output_file + ".pcm"));
if (FLAG_raw && !output_raw_file) {
output_raw_file.reset(new RawFile(std::string(FLAG_output_file) +
".pcm"));
}
WriteIntData(reinterpret_cast<const int16_t*>(msg.output_data().data()),
num_output_channels * output_samples_per_channel,
output_wav_file.get(),
output_raw_file.get());
} else if (msg.output_channel_size() > 0) {
if (FLAGS_raw && !output_raw_file) {
output_raw_file.reset(new RawFile(FLAGS_output_file + ".float"));
if (FLAG_raw && !output_raw_file) {
output_raw_file.reset(new RawFile(std::string(FLAG_output_file) +
".float"));
}
std::unique_ptr<const float* []> data(
new const float* [num_output_channels]);
@ -189,45 +199,45 @@ int do_main(int argc, char* argv[]) {
output_raw_file.get());
}
if (FLAGS_full) {
if (FLAG_full) {
if (msg.has_delay()) {
static FILE* delay_file = OpenFile(FLAGS_delay_file, "wb");
static FILE* delay_file = OpenFile(FLAG_delay_file, "wb");
int32_t delay = msg.delay();
if (FLAGS_text) {
if (FLAG_text) {
fprintf(delay_file, "%d\n", delay);
} else {
WriteData(&delay, sizeof(delay), delay_file, FLAGS_delay_file);
WriteData(&delay, sizeof(delay), delay_file, FLAG_delay_file);
}
}
if (msg.has_drift()) {
static FILE* drift_file = OpenFile(FLAGS_drift_file, "wb");
static FILE* drift_file = OpenFile(FLAG_drift_file, "wb");
int32_t drift = msg.drift();
if (FLAGS_text) {
if (FLAG_text) {
fprintf(drift_file, "%d\n", drift);
} else {
WriteData(&drift, sizeof(drift), drift_file, FLAGS_drift_file);
WriteData(&drift, sizeof(drift), drift_file, FLAG_drift_file);
}
}
if (msg.has_level()) {
static FILE* level_file = OpenFile(FLAGS_level_file, "wb");
static FILE* level_file = OpenFile(FLAG_level_file, "wb");
int32_t level = msg.level();
if (FLAGS_text) {
if (FLAG_text) {
fprintf(level_file, "%d\n", level);
} else {
WriteData(&level, sizeof(level), level_file, FLAGS_level_file);
WriteData(&level, sizeof(level), level_file, FLAG_level_file);
}
}
if (msg.has_keypress()) {
static FILE* keypress_file = OpenFile(FLAGS_keypress_file, "wb");
static FILE* keypress_file = OpenFile(FLAG_keypress_file, "wb");
bool keypress = msg.keypress();
if (FLAGS_text) {
if (FLAG_text) {
fprintf(keypress_file, "%d\n", keypress);
} else {
WriteData(&keypress, sizeof(keypress), keypress_file,
FLAGS_keypress_file);
FLAG_keypress_file);
}
}
}
@ -304,21 +314,21 @@ int do_main(int argc, char* argv[]) {
output_samples_per_channel =
static_cast<size_t>(output_sample_rate / 100);
if (!FLAGS_raw) {
if (!FLAG_raw) {
// The WAV files need to be reset every time, because they cant change
// their sample rate or number of channels.
std::stringstream reverse_name;
reverse_name << FLAGS_reverse_file << frame_count << ".wav";
reverse_name << FLAG_reverse_file << frame_count << ".wav";
reverse_wav_file.reset(new WavWriter(reverse_name.str(),
reverse_sample_rate,
num_reverse_channels));
std::stringstream input_name;
input_name << FLAGS_input_file << frame_count << ".wav";
input_name << FLAG_input_file << frame_count << ".wav";
input_wav_file.reset(new WavWriter(input_name.str(),
input_sample_rate,
num_input_channels));
std::stringstream output_name;
output_name << FLAGS_output_file << frame_count << ".wav";
output_name << FLAG_output_file << frame_count << ".wav";
output_wav_file.reset(new WavWriter(output_name.str(),
output_sample_rate,
num_output_channels));

87
webrtc/modules/audio_processing/transient/transient_suppression_test.cc
View File

@ -12,14 +12,15 @@
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <memory>
#include <string>
#include "gflags/gflags.h"
#include "webrtc/common_audio/include/audio_util.h"
#include "webrtc/modules/audio_processing/agc/agc.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/rtc_base/flags.h"
#include "webrtc/test/gtest.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/typedefs.h"
@ -32,31 +33,20 @@ DEFINE_string(reference_file_name,
"",
"PCM file that contains the reference signal.");
static bool ValidatePositiveInt(const char* flagname, int32_t value) {
if (value <= 0) {
printf("%s must be a positive integer.\n", flagname);
return false;
}
return true;
}
DEFINE_int32(chunk_size_ms,
10,
"Time between each chunk of samples in milliseconds.");
static const bool chunk_size_ms_dummy =
google::RegisterFlagValidator(&FLAGS_chunk_size_ms, &ValidatePositiveInt);
DEFINE_int(chunk_size_ms,
10,
"Time between each chunk of samples in milliseconds.");
DEFINE_int32(sample_rate_hz,
16000,
"Sampling frequency of the signal in Hertz.");
static const bool sample_rate_hz_dummy =
google::RegisterFlagValidator(&FLAGS_sample_rate_hz, &ValidatePositiveInt);
DEFINE_int32(detection_rate_hz,
0,
"Sampling frequency of the detection signal in Hertz.");
DEFINE_int(sample_rate_hz,
16000,
"Sampling frequency of the signal in Hertz.");
DEFINE_int(detection_rate_hz,
0,
"Sampling frequency of the detection signal in Hertz.");
DEFINE_int32(num_channels, 1, "Number of channels.");
static const bool num_channels_dummy =
google::RegisterFlagValidator(&FLAGS_num_channels, &ValidatePositiveInt);
DEFINE_int(num_channels, 1, "Number of channels.");
DEFINE_bool(help, false, "Print this message.");
namespace webrtc {
@ -146,19 +136,19 @@ static void WritePCM(FILE* f,
void void_main() {
// TODO(aluebs): Remove all FileWrappers.
// Prepare the input file.
FILE* in_file = fopen(FLAGS_in_file_name.c_str(), "rb");
FILE* in_file = fopen(FLAG_in_file_name, "rb");
ASSERT_TRUE(in_file != NULL);
// Prepare the detection file.
FILE* detection_file = NULL;
if (!FLAGS_detection_file_name.empty()) {
detection_file = fopen(FLAGS_detection_file_name.c_str(), "rb");
if (strlen(FLAG_detection_file_name) > 0) {
detection_file = fopen(FLAG_detection_file_name, "rb");
}
// Prepare the reference file.
FILE* reference_file = NULL;
if (!FLAGS_reference_file_name.empty()) {
reference_file = fopen(FLAGS_reference_file_name.c_str(), "rb");
if (strlen(FLAG_reference_file_name) > 0) {
reference_file = fopen(FLAG_reference_file_name, "rb");
}
// Prepare the output file.
@ -166,27 +156,27 @@ void void_main() {
FILE* out_file = fopen(out_file_name.c_str(), "wb");
ASSERT_TRUE(out_file != NULL);
int detection_rate_hz = FLAGS_detection_rate_hz;
int detection_rate_hz = FLAG_detection_rate_hz;
if (detection_rate_hz == 0) {
detection_rate_hz = FLAGS_sample_rate_hz;
detection_rate_hz = FLAG_sample_rate_hz;
}
Agc agc;
TransientSuppressor suppressor;
suppressor.Initialize(
FLAGS_sample_rate_hz, detection_rate_hz, FLAGS_num_channels);
FLAG_sample_rate_hz, detection_rate_hz, FLAG_num_channels);
const size_t audio_buffer_size =
FLAGS_chunk_size_ms * FLAGS_sample_rate_hz / 1000;
FLAG_chunk_size_ms * FLAG_sample_rate_hz / 1000;
const size_t detection_buffer_size =
FLAGS_chunk_size_ms * detection_rate_hz / 1000;
FLAG_chunk_size_ms * detection_rate_hz / 1000;
// int16 and float variants of the same data.
std::unique_ptr<int16_t[]> audio_buffer_i(
new int16_t[FLAGS_num_channels * audio_buffer_size]);
new int16_t[FLAG_num_channels * audio_buffer_size]);
std::unique_ptr<float[]> audio_buffer_f(
new float[FLAGS_num_channels * audio_buffer_size]);
new float[FLAG_num_channels * audio_buffer_size]);
std::unique_ptr<float[]> detection_buffer, reference_buffer;
@ -197,7 +187,7 @@ void void_main() {
while (ReadBuffers(in_file,
audio_buffer_size,
FLAGS_num_channels,
FLAG_num_channels,
audio_buffer_i.get(),
detection_file,
detection_buffer_size,
@ -207,17 +197,17 @@ void void_main() {
ASSERT_EQ(0,
agc.Process(audio_buffer_i.get(),
static_cast<int>(audio_buffer_size),
FLAGS_sample_rate_hz))
FLAG_sample_rate_hz))
<< "The AGC could not process the frame";
for (size_t i = 0; i < FLAGS_num_channels * audio_buffer_size; ++i) {
for (size_t i = 0; i < FLAG_num_channels * audio_buffer_size; ++i) {
audio_buffer_f[i] = audio_buffer_i[i];
}
ASSERT_EQ(0,
suppressor.Suppress(audio_buffer_f.get(),
audio_buffer_size,
FLAGS_num_channels,
FLAG_num_channels,
detection_buffer.get(),
detection_buffer_size,
reference_buffer.get(),
@ -228,7 +218,7 @@ void void_main() {
// Write result to out file.
WritePCM(
out_file, audio_buffer_size, FLAGS_num_channels, audio_buffer_f.get());
out_file, audio_buffer_size, FLAG_num_channels, audio_buffer_f.get());
}
fclose(in_file);
@ -244,8 +234,19 @@ void void_main() {
} // namespace webrtc
int main(int argc, char* argv[]) {
google::SetUsageMessage(webrtc::kUsage);
google::ParseCommandLineFlags(&argc, &argv, true);
if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) ||
FLAG_help || argc != 1) {
printf("%s", webrtc::kUsage);
if (FLAG_help) {
rtc::FlagList::Print(nullptr, false);
return 0;
}
return 1;
}
RTC_CHECK_GT(FLAG_chunk_size_ms, 0);
RTC_CHECK_GT(FLAG_sample_rate_hz, 0);
RTC_CHECK_GT(FLAG_num_channels, 0);
webrtc::void_main();
return 0;
}

1
webrtc/modules/remote_bitrate_estimator/BUILD.gn
View File

@ -225,7 +225,6 @@ if (rtc_include_tests) {
"../../test:test_main",
"//testing/gmock",
"//testing/gtest",
"//third_party/gflags",
]
if (!build_with_chromium && is_clang) {

22
webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.cc
View File

@ -16,11 +16,11 @@
#include <sstream>
#include <string>
#include "gflags/gflags.h"
#include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h"
#include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
#include "webrtc/rtc_base/flags.h"
#include "webrtc/test/rtp_file_reader.h"
namespace flags {
@ -30,17 +30,17 @@ DEFINE_string(extension_type,
"Extension type, either abs for absolute send time or tsoffset "
"for timestamp offset.");
std::string ExtensionType() {
return static_cast<std::string>(FLAGS_extension_type);
return static_cast<std::string>(FLAG_extension_type);
}
DEFINE_int32(extension_id, 3, "Extension id.");
DEFINE_int(extension_id, 3, "Extension id.");
int ExtensionId() {
return static_cast<int>(FLAGS_extension_id);
return static_cast<int>(FLAG_extension_id);
}
DEFINE_string(input_file, "", "Input file.");
std::string InputFile() {
return static_cast<std::string>(FLAGS_input_file);
return static_cast<std::string>(FLAG_input_file);
}
DEFINE_string(ssrc_filter,
@ -48,7 +48,7 @@ DEFINE_string(ssrc_filter,
"Comma-separated list of SSRCs in hexadecimal which are to be "
"used as input to the BWE (only applicable to pcap files).");
std::set<uint32_t> SsrcFilter() {
std::string ssrc_filter_string = static_cast<std::string>(FLAGS_ssrc_filter);
std::string ssrc_filter_string = static_cast<std::string>(FLAG_ssrc_filter);
if (ssrc_filter_string.empty())
return std::set<uint32_t>();
std::stringstream ss;
@ -64,6 +64,8 @@ std::set<uint32_t> SsrcFilter() {
}
return ssrcs;
}
DEFINE_bool(help, false, "Print this message.");
} // namespace flags
bool ParseArgsAndSetupEstimator(int argc,
@ -74,7 +76,13 @@ bool ParseArgsAndSetupEstimator(int argc,
webrtc::RtpHeaderParser** parser,
webrtc::RemoteBitrateEstimator** estimator,
std::string* estimator_used) {
google::ParseCommandLineFlags(&argc, &argv, true);
if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true)) {
return 1;
}
if (FLAG_help) {
rtc::FlagList::Print(nullptr, false);
return 0;
}
std::string filename = flags::InputFile();
std::set<uint32_t> ssrc_filter = flags::SsrcFilter();

1
webrtc/modules/video_coding/BUILD.gn
View File

@ -350,7 +350,6 @@ if (rtc_include_tests) {
"../../test:video_test_common",
"../../test:video_test_support",
"../video_capture",
"//third_party/gflags",
]
} # video_quality_measurement

205
webrtc/modules/video_coding/codecs/tools/video_quality_measurement.cc
View File

@ -11,6 +11,7 @@
#include <assert.h>
#include <stdio.h>
#include <time.h>
#include <string.h>
#include <stdarg.h>
#include <sys/stat.h> // To check for directory existence.
@ -19,13 +20,13 @@
#define S_ISDIR(mode) (((mode)&S_IFMT) == S_IFDIR)
#endif
#include "gflags/gflags.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/video_coding/codecs/test/packet_manipulator.h"
#include "webrtc/modules/video_coding/codecs/test/stats.h"
#include "webrtc/modules/video_coding/codecs/test/videoprocessor.h"
#include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
#include "webrtc/modules/video_coding/include/video_coding.h"
#include "webrtc/rtc_base/flags.h"
#include "webrtc/rtc_base/format_macros.h"
#include "webrtc/test/testsupport/frame_reader.h"
#include "webrtc/test/testsupport/frame_writer.h"
@ -43,12 +44,12 @@ DEFINE_string(input_filename,
"Input file. "
"The source video file to be encoded and decoded. Must be in "
".yuv format");
DEFINE_int32(width, -1, "Width in pixels of the frames in the input file.");
DEFINE_int32(height, -1, "Height in pixels of the frames in the input file.");
DEFINE_int32(framerate,
30,
"Frame rate of the input file, in FPS "
"(frames-per-second). ");
DEFINE_int(width, -1, "Width in pixels of the frames in the input file.");
DEFINE_int(height, -1, "Height in pixels of the frames in the input file.");
DEFINE_int(framerate,
30,
"Frame rate of the input file, in FPS "
"(frames-per-second). ");
DEFINE_string(output_dir,
".",
"Output directory. "
@ -76,40 +77,40 @@ DEFINE_string(output_filename,
"The name of the output video file resulting of the processing "
"of the source file. By default this is the same name as the "
"input file with '_out' appended before the extension.");
DEFINE_int32(bitrate, 500, "Bit rate in kilobits/second.");
DEFINE_int32(keyframe_interval,
0,
"Forces a keyframe every Nth frame. "
"0 means the encoder decides when to insert keyframes. Note that "
"the encoder may create a keyframe in other locations in addition "
"to the interval that is set using this parameter.");
DEFINE_int32(temporal_layers,
0,
"The number of temporal layers to use "
"(VP8 specific codec setting). Must be 0-4.");
DEFINE_int32(packet_size,
1500,
"Simulated network packet size in bytes (MTU). "
"Used for packet loss simulation.");
DEFINE_int32(max_payload_size,
1440,
"Max payload size in bytes for the "
"encoder.");
DEFINE_int(bitrate, 500, "Bit rate in kilobits/second.");
DEFINE_int(keyframe_interval,
0,
"Forces a keyframe every Nth frame. "
"0 means the encoder decides when to insert keyframes. Note that "
"the encoder may create a keyframe in other locations in addition "
"to the interval that is set using this parameter.");
DEFINE_int(temporal_layers,
0,
"The number of temporal layers to use "
"(VP8 specific codec setting). Must be 0-4.");
DEFINE_int(packet_size,
1500,
"Simulated network packet size in bytes (MTU). "
"Used for packet loss simulation.");
DEFINE_int(max_payload_size,
1440,
"Max payload size in bytes for the "
"encoder.");
DEFINE_string(packet_loss_mode,
"uniform",
"Packet loss mode. Two different "
"packet loss models are supported: uniform or burst. This "
"setting has no effect unless packet_loss_rate is >0. ");
DEFINE_double(packet_loss_probability,
0.0,
"Packet loss probability. A value "
"between 0.0 and 1.0 that defines the probability of a packet "
"being lost. 0.1 means 10% and so on.");
DEFINE_int32(packet_loss_burst_length,
1,
"Packet loss burst length. Defines "
"how many packets will be lost in a burst when a packet has been "
"decided to be lost. Must be >=1.");
DEFINE_float(packet_loss_probability,
0.0f,
"Packet loss probability. A value "
"between 0.0 and 1.0 that defines the probability of a packet "
"being lost. 0.1 means 10% and so on.");
DEFINE_int(packet_loss_burst_length,
1,
"Packet loss burst length. Defines "
"how many packets will be lost in a burst when a packet has been "
"decided to be lost. Must be >=1.");
DEFINE_bool(csv,
false,
"CSV output. Enabling this will output all frame "
@ -126,12 +127,13 @@ DEFINE_bool(verbose,
"Verbose mode. Prints a lot of debugging info. "
"Suitable for tracking progress but not for capturing output. "
"Disable with --noverbose flag.");
DEFINE_bool(help, false, "Prints this message.");
// Custom log method that only prints if the verbose flag is given.
// Supports all the standard printf parameters and formatting (just forwarded).
int Log(const char* format, ...) {
int result = 0;
if (FLAGS_verbose) {
if (FLAG_verbose) {
va_list args;
va_start(args, format);
result = vprintf(format, args);
@ -143,55 +145,58 @@ int Log(const char* format, ...) {
// Validates the arguments given as command line flags and fills in the
// TestConfig struct with all configurations needed for video processing.
// Returns 0 if everything is OK, otherwise an exit code.
int HandleCommandLineFlags(webrtc::test::TestConfig* config) {
int HandleCommandLineFlags(webrtc::test::TestConfig* config,
const std::string& usage) {
// Validate the mandatory flags:
if (FLAGS_input_filename.empty() || FLAGS_width == -1 || FLAGS_height == -1) {
printf("%s\n", google::ProgramUsage());
if (strlen(FLAG_input_filename) == 0 ||
FLAG_width == -1 || FLAG_height == -1) {
printf("%s\n", usage.c_str());
return 1;
}
config->name = FLAGS_test_name;
config->description = FLAGS_test_description;
config->name = FLAG_test_name;
config->description = FLAG_test_description;
// Verify the input file exists and is readable.
FILE* test_file;
test_file = fopen(FLAGS_input_filename.c_str(), "rb");
test_file = fopen(FLAG_input_filename, "rb");
if (test_file == NULL) {
fprintf(stderr, "Cannot read the specified input file: %s\n",
FLAGS_input_filename.c_str());
FLAG_input_filename);
return 2;
}
fclose(test_file);
config->input_filename = FLAGS_input_filename;
config->input_filename = FLAG_input_filename;
// Verify the output dir exists.
struct stat dir_info;
if (!(stat(FLAGS_output_dir.c_str(), &dir_info) == 0 &&
if (!(stat(FLAG_output_dir, &dir_info) == 0 &&
S_ISDIR(dir_info.st_mode))) {
fprintf(stderr, "Cannot find output directory: %s\n",
FLAGS_output_dir.c_str());
FLAG_output_dir);
return 3;
}
config->output_dir = FLAGS_output_dir;
config->output_dir = FLAG_output_dir;
// Manufacture an output filename if none was given.
if (FLAGS_output_filename.empty()) {
if (strlen(FLAG_output_filename) == 0) {
// Cut out the filename without extension from the given input file
// (which may include a path)
size_t startIndex = FLAGS_input_filename.find_last_of("/") + 1;
size_t startIndex = config->input_filename.find_last_of("/") + 1;
if (startIndex == 0) {
startIndex = 0;
}
FLAGS_output_filename =
FLAGS_input_filename.substr(
startIndex, FLAGS_input_filename.find_last_of(".") - startIndex) +
config->output_filename =
config->input_filename.substr(
startIndex, config->input_filename.find_last_of(".") - startIndex) +
"_out.yuv";
} else {
config->output_filename = FLAG_output_filename;
}
// Verify output file can be written.
if (FLAGS_output_dir == ".") {
config->output_filename = FLAGS_output_filename;
} else {
config->output_filename = FLAGS_output_dir + "/" + FLAGS_output_filename;
if (config->output_dir != ".") {
config->output_filename =
config->output_dir + "/" + config->output_filename;
}
test_file = fopen(config->output_filename.c_str(), "wb");
if (test_file == NULL) {
@ -202,99 +207,98 @@ int HandleCommandLineFlags(webrtc::test::TestConfig* config) {
fclose(test_file);
// Check single core flag.
config->use_single_core = FLAGS_use_single_core;
config->use_single_core = FLAG_use_single_core;
// Get codec specific configuration.
webrtc::test::CodecSettings(webrtc::kVideoCodecVP8, &config->codec_settings);
// Check the temporal layers.
if (FLAGS_temporal_layers < 0 ||
FLAGS_temporal_layers > webrtc::kMaxTemporalStreams) {
if (FLAG_temporal_layers < 0 ||
FLAG_temporal_layers > webrtc::kMaxTemporalStreams) {
fprintf(stderr, "Temporal layers number must be 0-4, was: %d\n",
FLAGS_temporal_layers);
FLAG_temporal_layers);
return 13;
}
config->codec_settings.VP8()->numberOfTemporalLayers = FLAGS_temporal_layers;
config->codec_settings.VP8()->numberOfTemporalLayers = FLAG_temporal_layers;
// Check the bit rate.
if (FLAGS_bitrate <= 0) {
fprintf(stderr, "Bit rate must be >0 kbps, was: %d\n", FLAGS_bitrate);
if (FLAG_bitrate <= 0) {
fprintf(stderr, "Bit rate must be >0 kbps, was: %d\n", FLAG_bitrate);
return 5;
}
config->codec_settings.startBitrate = FLAGS_bitrate;
config->codec_settings.startBitrate = FLAG_bitrate;
// Check the keyframe interval.
if (FLAGS_keyframe_interval < 0) {
if (FLAG_keyframe_interval < 0) {
fprintf(stderr, "Keyframe interval must be >=0, was: %d\n",
FLAGS_keyframe_interval);
FLAG_keyframe_interval);
return 6;
}
config->keyframe_interval = FLAGS_keyframe_interval;
config->keyframe_interval = FLAG_keyframe_interval;
// Check packet size and max payload size.
if (FLAGS_packet_size <= 0) {
if (FLAG_packet_size <= 0) {
fprintf(stderr, "Packet size must be >0 bytes, was: %d\n",
FLAGS_packet_size);
FLAG_packet_size);
return 7;
}
config->networking_config.packet_size_in_bytes =
static_cast<size_t>(FLAGS_packet_size);
static_cast<size_t>(FLAG_packet_size);
if (FLAGS_max_payload_size <= 0) {
if (FLAG_max_payload_size <= 0) {
fprintf(stderr, "Max payload size must be >0 bytes, was: %d\n",
FLAGS_max_payload_size);
FLAG_max_payload_size);
return 8;
}
config->networking_config.max_payload_size_in_bytes =
static_cast<size_t>(FLAGS_max_payload_size);
static_cast<size_t>(FLAG_max_payload_size);
// Check the width and height
if (FLAGS_width <= 0 || FLAGS_height <= 0) {
if (FLAG_width <= 0 || FLAG_height <= 0) {
fprintf(stderr, "Width and height must be >0.");
return 9;
}
config->codec_settings.width = FLAGS_width;
config->codec_settings.height = FLAGS_height;
config->codec_settings.maxFramerate = FLAGS_framerate;
config->codec_settings.width = FLAG_width;
config->codec_settings.height = FLAG_height;
config->codec_settings.maxFramerate = FLAG_framerate;
// Calculate the size of each frame to read (according to YUV spec).
config->frame_length_in_bytes =
3 * config->codec_settings.width * config->codec_settings.height / 2;
// Check packet loss settings
if (FLAGS_packet_loss_mode != "uniform" &&
FLAGS_packet_loss_mode != "burst") {
if (strcmp(FLAG_packet_loss_mode, "uniform") == 0) {
config->networking_config.packet_loss_mode = webrtc::test::kUniform;
} else if (strcmp(FLAG_packet_loss_mode, "burst") == 0) {
config->networking_config.packet_loss_mode = webrtc::test::kBurst;
} else {
fprintf(stderr,
"Unsupported packet loss mode, must be 'uniform' or "
"'burst'\n.");
return 10;
}
config->networking_config.packet_loss_mode = webrtc::test::kUniform;
if (FLAGS_packet_loss_mode == "burst") {
config->networking_config.packet_loss_mode = webrtc::test::kBurst;
}
if (FLAGS_packet_loss_probability < 0.0 ||
FLAGS_packet_loss_probability > 1.0) {
if (FLAG_packet_loss_probability < 0.0 ||
FLAG_packet_loss_probability > 1.0) {
fprintf(stderr,
"Invalid packet loss probability. Must be 0.0 - 1.0, "
"was: %f\n",
FLAGS_packet_loss_probability);
FLAG_packet_loss_probability);
return 11;
}
config->networking_config.packet_loss_probability =
FLAGS_packet_loss_probability;
FLAG_packet_loss_probability;
if (FLAGS_packet_loss_burst_length < 1) {
if (FLAG_packet_loss_burst_length < 1) {
fprintf(stderr,
"Invalid packet loss burst length, must be >=1, "
"was: %d\n",
FLAGS_packet_loss_burst_length);
FLAG_packet_loss_burst_length);
return 12;
}
config->networking_config.packet_loss_burst_length =
FLAGS_packet_loss_burst_length;
config->verbose = FLAGS_verbose;
FLAG_packet_loss_burst_length;
config->verbose = FLAG_verbose;
return 0;
}
@ -464,18 +468,21 @@ int main(int argc, char* argv[]) {
"Quality test application for video comparisons.\n"
"Run " +
program_name +
" --helpshort for usage.\n"
" --help for usage.\n"
"Example usage:\n" +
program_name +
" --input_filename=filename.yuv --width=352 --height=288\n";
google::SetUsageMessage(usage);
google::ParseCommandLineFlags(&argc, &argv, true);
rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true);
if (FLAG_help) {
rtc::FlagList::Print(nullptr, false);
return 0;
}
// Create TestConfig.
webrtc::test::TestConfig config;
int return_code = HandleCommandLineFlags(&config);
int return_code = HandleCommandLineFlags(&config, usage);
// Exit if an invalid argument is supplied.
if (return_code != 0) {
return return_code;
@ -500,7 +507,7 @@ int main(int argc, char* argv[]) {
&packet_reader, config.networking_config, config.verbose);
// By default the packet manipulator is seeded with a fixed random.
// If disabled we must generate a new seed.
if (FLAGS_disable_fixed_random_seed) {
if (FLAG_disable_fixed_random_seed) {
packet_manipulator.InitializeRandomSeed(time(NULL));
}
webrtc::test::VideoProcessor* processor = new webrtc::test::VideoProcessor(
@ -539,10 +546,10 @@ int main(int argc, char* argv[]) {
webrtc::test::QualityMetricsResult psnr_result;
CalculatePsnrVideoMetrics(&config, &psnr_result);
if (FLAGS_csv) {
if (FLAG_csv) {
PrintCsvOutput(stats, ssim_result, psnr_result);
}
if (FLAGS_python) {
if (FLAG_python) {
PrintPythonOutput(config, stats, ssim_result, psnr_result);
}
delete processor;

1
webrtc/video/BUILD.gn
View File

@ -223,7 +223,6 @@ if (rtc_include_tests) {
"../test:test_support",
"../test:video_test_common",
"../test:video_test_support",
"//third_party/gflags",
]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).

150
webrtc/video/replay.cc
View File

@ -14,13 +14,14 @@
#include <memory>
#include <sstream>
#include "gflags/gflags.h"
#include "webrtc/api/video_codecs/video_decoder.h"
#include "webrtc/call/call.h"
#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
#include "webrtc/rtc_base/checks.h"
#include "webrtc/rtc_base/flags.h"
#include "webrtc/rtc_base/string_to_number.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/system_wrappers/include/sleep.h"
#include "webrtc/test/call_test.h"
@ -36,121 +37,113 @@
#include "webrtc/test/video_renderer.h"
#include "webrtc/typedefs.h"
namespace {
static bool ValidatePayloadType(int32_t payload_type) {
return payload_type > 0 && payload_type <= 127;
}
static bool ValidateSsrc(const char* ssrc_string) {
return rtc::StringToNumber<uint32_t>(ssrc_string).has_value();
}
static bool ValidateOptionalPayloadType(int32_t payload_type) {
return payload_type == -1 || ValidatePayloadType(payload_type);
}
static bool ValidateRtpHeaderExtensionId(int32_t extension_id) {
return extension_id >= -1 && extension_id < 15;
}
bool ValidateInputFilenameNotEmpty(const std::string& string) {
return !string.empty();
}
} // namespace
namespace webrtc {
namespace flags {
// TODO(pbos): Multiple receivers.
// Flag for payload type.
static bool ValidatePayloadType(const char* flagname, int32_t payload_type) {
return payload_type > 0 && payload_type <= 127;
}
DEFINE_int32(payload_type, test::CallTest::kPayloadTypeVP8, "Payload type");
static int PayloadType() { return static_cast<int>(FLAGS_payload_type); }
static const bool payload_dummy =
google::RegisterFlagValidator(&FLAGS_payload_type, &ValidatePayloadType);
DEFINE_int(payload_type, test::CallTest::kPayloadTypeVP8, "Payload type");
static int PayloadType() { return static_cast<int>(FLAG_payload_type); }
DEFINE_int32(payload_type_rtx,
test::CallTest::kSendRtxPayloadType,
"RTX payload type");
DEFINE_int(payload_type_rtx,
test::CallTest::kSendRtxPayloadType,
"RTX payload type");
static int PayloadTypeRtx() {
return static_cast<int>(FLAGS_payload_type_rtx);
return static_cast<int>(FLAG_payload_type_rtx);
}
static const bool payload_rtx_dummy =
google::RegisterFlagValidator(&FLAGS_payload_type_rtx,
&ValidatePayloadType);
// Flag for SSRC.
static bool ValidateSsrc(const char* flagname, uint64_t ssrc) {
return ssrc > 0 && ssrc <= 0xFFFFFFFFu;
const std::string& DefaultSsrc() {
static const std::string ssrc = std::to_string(
test::CallTest::kVideoSendSsrcs[0]);
return ssrc;
}
DEFINE_string(ssrc, DefaultSsrc().c_str(), "Incoming SSRC");
static uint32_t Ssrc() {
return rtc::StringToNumber<uint32_t>(FLAG_ssrc).value();
}
DEFINE_uint64(ssrc, test::CallTest::kVideoSendSsrcs[0], "Incoming SSRC");
static uint32_t Ssrc() { return static_cast<uint32_t>(FLAGS_ssrc); }
static const bool ssrc_dummy =
google::RegisterFlagValidator(&FLAGS_ssrc, &ValidateSsrc);
DEFINE_uint64(ssrc_rtx, test::CallTest::kSendRtxSsrcs[0], "Incoming RTX SSRC");
const std::string& DefaultSsrcRtx() {
static const std::string ssrc_rtx = std::to_string(
test::CallTest::kSendRtxSsrcs[0]);
return ssrc_rtx;
}
DEFINE_string(ssrc_rtx, DefaultSsrcRtx().c_str(), "Incoming RTX SSRC");
static uint32_t SsrcRtx() {
return static_cast<uint32_t>(FLAGS_ssrc_rtx);
}
static const bool ssrc_rtx_dummy =
google::RegisterFlagValidator(&FLAGS_ssrc_rtx, &ValidateSsrc);
static bool ValidateOptionalPayloadType(const char* flagname,
int32_t payload_type) {
return payload_type == -1 || ValidatePayloadType(flagname, payload_type);
return rtc::StringToNumber<uint32_t>(FLAG_ssrc_rtx).value();
}
// Flag for RED payload type.
DEFINE_int32(red_payload_type, -1, "RED payload type");
DEFINE_int(red_payload_type, -1, "RED payload type");
static int RedPayloadType() {
return static_cast<int>(FLAGS_red_payload_type);
return static_cast<int>(FLAG_red_payload_type);
}
static const bool red_dummy =
google::RegisterFlagValidator(&FLAGS_red_payload_type,
&ValidateOptionalPayloadType);
// Flag for ULPFEC payload type.
DEFINE_int32(fec_payload_type, -1, "ULPFEC payload type");
DEFINE_int(fec_payload_type, -1, "ULPFEC payload type");
static int FecPayloadType() {
return static_cast<int>(FLAGS_fec_payload_type);
return static_cast<int>(FLAG_fec_payload_type);
}
static const bool fec_dummy =
google::RegisterFlagValidator(&FLAGS_fec_payload_type,
&ValidateOptionalPayloadType);
// Flag for abs-send-time id.
static bool ValidateRtpHeaderExtensionId(const char* flagname,
int32_t extension_id) {
return extension_id >= -1 || extension_id < 15;
}
DEFINE_int32(abs_send_time_id, -1, "RTP extension ID for abs-send-time");
static int AbsSendTimeId() { return static_cast<int>(FLAGS_abs_send_time_id); }
static const bool abs_send_time_dummy =
google::RegisterFlagValidator(&FLAGS_abs_send_time_id,
&ValidateRtpHeaderExtensionId);
DEFINE_int(abs_send_time_id, -1, "RTP extension ID for abs-send-time");
static int AbsSendTimeId() { return static_cast<int>(FLAG_abs_send_time_id); }
// Flag for transmission-offset id.
DEFINE_int32(transmission_offset_id,
-1,
"RTP extension ID for transmission-offset");
DEFINE_int(transmission_offset_id,
-1,
"RTP extension ID for transmission-offset");
static int TransmissionOffsetId() {
return static_cast<int>(FLAGS_transmission_offset_id);
return static_cast<int>(FLAG_transmission_offset_id);
}
static const bool timestamp_offset_dummy =
google::RegisterFlagValidator(&FLAGS_transmission_offset_id,
&ValidateRtpHeaderExtensionId);
// Flag for rtpdump input file.
bool ValidateInputFilenameNotEmpty(const char* flagname,
const std::string& string) {
return !string.empty();
}
DEFINE_string(input_file, "", "input file");
static std::string InputFile() {
return static_cast<std::string>(FLAGS_input_file);
return static_cast<std::string>(FLAG_input_file);
}
static const bool input_file_dummy =
google::RegisterFlagValidator(&FLAGS_input_file,
&ValidateInputFilenameNotEmpty);
// Flag for raw output files.
DEFINE_string(out_base, "", "Basename (excluding .jpg) for raw output");
static std::string OutBase() {
return static_cast<std::string>(FLAGS_out_base);
return static_cast<std::string>(FLAG_out_base);
}
DEFINE_string(decoder_bitstream_filename, "", "Decoder bitstream output file");
static std::string DecoderBitstreamFilename() {
return static_cast<std::string>(FLAGS_decoder_bitstream_filename);
return static_cast<std::string>(FLAG_decoder_bitstream_filename);
}
// Flag for video codec.
DEFINE_string(codec, "VP8", "Video codec");
static std::string Codec() { return static_cast<std::string>(FLAGS_codec); }
static std::string Codec() { return static_cast<std::string>(FLAG_codec); }
DEFINE_bool(help, false, "Print this message.");
} // namespace flags
static const uint32_t kReceiverLocalSsrc = 0x123456;
@ -330,7 +323,24 @@ void RtpReplay() {
int main(int argc, char* argv[]) {
::testing::InitGoogleTest(&argc, argv);
google::ParseCommandLineFlags(&argc, &argv, true);
if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true)) {
return 1;
}
if (webrtc::flags::FLAG_help) {
rtc::FlagList::Print(nullptr, false);
return 0;
}
RTC_CHECK(ValidatePayloadType(webrtc::flags::FLAG_payload_type));
RTC_CHECK(ValidatePayloadType(webrtc::flags::FLAG_payload_type_rtx));
RTC_CHECK(ValidateSsrc(webrtc::flags::FLAG_ssrc));
RTC_CHECK(ValidateSsrc(webrtc::flags::FLAG_ssrc_rtx));
RTC_CHECK(ValidateOptionalPayloadType(webrtc::flags::FLAG_red_payload_type));
RTC_CHECK(ValidateOptionalPayloadType(webrtc::flags::FLAG_fec_payload_type));
RTC_CHECK(ValidateRtpHeaderExtensionId(webrtc::flags::FLAG_abs_send_time_id));
RTC_CHECK(ValidateRtpHeaderExtensionId(
webrtc::flags::FLAG_transmission_offset_id));
RTC_CHECK(ValidateInputFilenameNotEmpty(webrtc::flags::FLAG_input_file));
webrtc::test::RunTest(webrtc::RtpReplay);
return 0;

1
webrtc/voice_engine/BUILD.gn
View File

@ -260,7 +260,6 @@ if (rtc_include_tests) {
"../test/:video_test_common",
"//testing/gmock",
"//testing/gtest",
"//third_party/gflags",
]
sources = [

5
webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc
View File

@ -11,11 +11,14 @@
#include <memory>
#include "webrtc/rtc_base/criticalsection.h"
#include "webrtc/rtc_base/flags.h"
#include "webrtc/system_wrappers/include/event_wrapper.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/voice_engine/test/auto_test/fixtures/after_streaming_fixture.h"
#include "webrtc/voice_engine/test/auto_test/voe_standard_test.h"
DECLARE_bool(include_timing_dependent_tests);
class TestRtpObserver : public webrtc::VoERTPObserver {
public:
TestRtpObserver() : changed_ssrc_event_(webrtc::EventWrapper::Create()) {}
@ -85,7 +88,7 @@ class RtpRtcpTest : public AfterStreamingFixture {
};
TEST_F(RtpRtcpTest, RemoteRtcpCnameHasPropagatedToRemoteSide) {
if (!FLAGS_include_timing_dependent_tests) {
if (!FLAG_include_timing_dependent_tests) {
TEST_LOG("Skipping test - running in slow execution environment...\n");
return;
}

12
webrtc/voice_engine/test/auto_test/voe_standard_test.cc
View File

@ -14,6 +14,7 @@
#include <stdio.h>
#include <string.h>
#include "webrtc/rtc_base/flags.h"
#include "webrtc/system_wrappers/include/event_wrapper.h"
#include "webrtc/typedefs.h"
#include "webrtc/voice_engine/test/auto_test/automated_mode.h"
@ -26,6 +27,7 @@ DEFINE_bool(include_timing_dependent_tests, true,
DEFINE_bool(automated, false,
"If true, we'll run the automated tests we have in noninteractive "
"mode.");
DEFINE_bool(help, false, "Print this message.");
namespace webrtc {
namespace voetest {
@ -101,9 +103,15 @@ int main(int argc, char** argv) {
// This function and RunInAutomatedMode is defined in automated_mode.cc
// to avoid macro clashes with googletest (for instance ASSERT_TRUE).
webrtc::voetest::InitializeGoogleTest(&argc, argv);
google::ParseCommandLineFlags(&argc, &argv, true);
if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true)) {
return 1;
}
if (FLAG_help) {
rtc::FlagList::Print(nullptr, false);
return 0;
}
if (FLAGS_automated) {
if (FLAG_automated) {
return webrtc::voetest::RunInAutomatedMode();
}

3
webrtc/voice_engine/test/auto_test/voe_standard_test.h
View File

@ -14,12 +14,9 @@
#include <stdio.h>
#include <string>
#include "gflags/gflags.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/voice_engine/test/auto_test/voe_test_common.h"
DECLARE_bool(include_timing_dependent_tests);
namespace webrtc {
namespace voetest {