Switch neteq tools to ABSL_FLAG.
Bug: webrtc:10616
Change-Id: I2aa688f0976d5618347e402f25d8701b0cf5a360
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144027
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28442}
diff --git a/DEPS b/DEPS
index dc92d38..1521e49 100644
--- a/DEPS
+++ b/DEPS
@@ -1638,4 +1638,7 @@
"+absl/strings/string_view.h",
"+absl/types/optional.h",
"+absl/types/variant.h",
+
+ # Abseil flags are allowed in tests and tools.
+ "+absl/flags",
]
diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index 9afc0e4..3976600 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -1498,6 +1498,8 @@
"../../rtc_base:stringutils",
"../../system_wrappers:field_trial",
"../../test:field_trial",
+ "//third_party/abseil-cpp/absl/flags:flag",
+ "//third_party/abseil-cpp/absl/flags:parse",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
@@ -1605,6 +1607,8 @@
testonly = true
deps = audio_coding_deps + [
+ "//third_party/abseil-cpp/absl/flags:flag",
+ "//third_party/abseil-cpp/absl/flags:parse",
"//third_party/abseil-cpp/absl/memory",
":audio_coding",
":audio_encoder_cng",
@@ -1670,6 +1674,8 @@
":pcm16b",
"../../rtc_base:rtc_base_approved",
"//testing/gtest",
+ "//third_party/abseil-cpp/absl/flags:flag",
+ "//third_party/abseil-cpp/absl/flags:parse",
]
}
diff --git a/modules/audio_coding/neteq/tools/neteq_rtpplay.cc b/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
index 4868a95..ceebfd5 100644
--- a/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
+++ b/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
@@ -11,136 +11,150 @@
#include <iostream>
#include <string>
+#include "absl/flags/flag.h"
+#include "absl/flags/parse.h"
#include "absl/strings/string_view.h"
#include "absl/types/optional.h"
#include "modules/audio_coding/neteq/tools/neteq_test.h"
#include "modules/audio_coding/neteq/tools/neteq_test_factory.h"
-#include "rtc_base/flags.h"
#include "rtc_base/strings/string_builder.h"
#include "system_wrappers/include/field_trial.h"
#include "test/field_trial.h"
-namespace {
-
using TestConfig = webrtc::test::NetEqTestFactory::Config;
-WEBRTC_DEFINE_bool(codec_map,
- false,
- "Prints the mapping between RTP payload type and "
- "codec");
-WEBRTC_DEFINE_string(
- force_fieldtrials,
- "",
- "Field trials control experimental feature code which can be forced. "
- "E.g. running with --force_fieldtrials=WebRTC-FooFeature/Enable/"
- " will assign the group Enable to field trial WebRTC-FooFeature.");
-WEBRTC_DEFINE_bool(help, false, "Prints this message");
-// Define command line flags.
-WEBRTC_DEFINE_int(pcmu,
- TestConfig::default_pcmu(),
- "RTP payload type for PCM-u");
-WEBRTC_DEFINE_int(pcma,
- TestConfig::default_pcma(),
- "RTP payload type for PCM-a");
-WEBRTC_DEFINE_int(ilbc,
- TestConfig::default_ilbc(),
- "RTP payload type for iLBC");
-WEBRTC_DEFINE_int(isac,
- TestConfig::default_isac(),
- "RTP payload type for iSAC");
-WEBRTC_DEFINE_int(isac_swb,
- TestConfig::default_isac_swb(),
- "RTP payload type for iSAC-swb (32 kHz)");
-WEBRTC_DEFINE_int(opus,
- TestConfig::default_opus(),
- "RTP payload type for Opus");
-WEBRTC_DEFINE_int(pcm16b,
- TestConfig::default_pcm16b(),
- "RTP payload type for PCM16b-nb (8 kHz)");
-WEBRTC_DEFINE_int(pcm16b_wb,
- TestConfig::default_pcm16b_wb(),
- "RTP payload type for PCM16b-wb (16 kHz)");
-WEBRTC_DEFINE_int(pcm16b_swb32,
- TestConfig::default_pcm16b_swb32(),
- "RTP payload type for PCM16b-swb32 (32 kHz)");
-WEBRTC_DEFINE_int(pcm16b_swb48,
- TestConfig::default_pcm16b_swb48(),
- "RTP payload type for PCM16b-swb48 (48 kHz)");
-WEBRTC_DEFINE_int(g722,
- TestConfig::default_g722(),
- "RTP payload type for G.722");
-WEBRTC_DEFINE_int(avt,
- TestConfig::default_avt(),
- "RTP payload type for AVT/DTMF (8 kHz)");
-WEBRTC_DEFINE_int(avt_16,
- TestConfig::default_avt_16(),
- "RTP payload type for AVT/DTMF (16 kHz)");
-WEBRTC_DEFINE_int(avt_32,
- TestConfig::default_avt_32(),
- "RTP payload type for AVT/DTMF (32 kHz)");
-WEBRTC_DEFINE_int(avt_48,
- TestConfig::default_avt_48(),
- "RTP payload type for AVT/DTMF (48 kHz)");
-WEBRTC_DEFINE_int(red,
- TestConfig::default_red(),
- "RTP payload type for redundant audio (RED)");
-WEBRTC_DEFINE_int(cn_nb,
- TestConfig::default_cn_nb(),
- "RTP payload type for comfort noise (8 kHz)");
-WEBRTC_DEFINE_int(cn_wb,
- TestConfig::default_cn_wb(),
- "RTP payload type for comfort noise (16 kHz)");
-WEBRTC_DEFINE_int(cn_swb32,
- TestConfig::default_cn_swb32(),
- "RTP payload type for comfort noise (32 kHz)");
-WEBRTC_DEFINE_int(cn_swb48,
- TestConfig::default_cn_swb48(),
- "RTP payload type for comfort noise (48 kHz)");
-WEBRTC_DEFINE_string(replacement_audio_file,
- "",
- "A PCM file that will be used to populate dummy"
- " RTP packets");
-WEBRTC_DEFINE_string(
- ssrc,
- "",
- "Only use packets with this SSRC (decimal or hex, the latter "
- "starting with 0x)");
-WEBRTC_DEFINE_int(audio_level,
- TestConfig::default_audio_level(),
- "Extension ID for audio level (RFC 6464)");
-WEBRTC_DEFINE_int(abs_send_time,
- TestConfig::default_abs_send_time(),
- "Extension ID for absolute sender time");
-WEBRTC_DEFINE_int(transport_seq_no,
- TestConfig::default_transport_seq_no(),
- "Extension ID for transport sequence number");
-WEBRTC_DEFINE_int(video_content_type,
- TestConfig::default_video_content_type(),
- "Extension ID for video content type");
-WEBRTC_DEFINE_int(video_timing,
- TestConfig::default_video_timing(),
- "Extension ID for video timing");
-WEBRTC_DEFINE_string(output_files_base_name,
- "",
- "Custom path used as prefix for the output files - i.e., "
- "matlab plot, python plot, text log.");
-WEBRTC_DEFINE_bool(matlabplot,
- false,
- "Generates a matlab script for plotting the delay profile");
-WEBRTC_DEFINE_bool(pythonplot,
- false,
- "Generates a python script for plotting the delay profile");
-WEBRTC_DEFINE_bool(textlog,
- false,
- "Generates a text log describing the simulation on a "
- "step-by-step basis.");
-WEBRTC_DEFINE_bool(concealment_events, false, "Prints concealment events");
-WEBRTC_DEFINE_int(max_nr_packets_in_buffer,
- TestConfig::default_max_nr_packets_in_buffer(),
- "Maximum allowed number of packets in the buffer");
-WEBRTC_DEFINE_bool(enable_fast_accelerate,
- false,
- "Enables jitter buffer fast accelerate");
+ABSL_FLAG(bool,
+ codec_map,
+ false,
+ "Prints the mapping between RTP payload type and "
+ "codec");
+ABSL_FLAG(std::string,
+ force_fieldtrials,
+ "",
+ "Field trials control experimental feature code which can be forced. "
+ "E.g. running with --force_fieldtrials=WebRTC-FooFeature/Enable/"
+ " will assign the group Enable to field trial WebRTC-FooFeature.");
+ABSL_FLAG(int, pcmu, TestConfig::default_pcmu(), "RTP payload type for PCM-u");
+ABSL_FLAG(int, pcma, TestConfig::default_pcma(), "RTP payload type for PCM-a");
+ABSL_FLAG(int, ilbc, TestConfig::default_ilbc(), "RTP payload type for iLBC");
+ABSL_FLAG(int, isac, TestConfig::default_isac(), "RTP payload type for iSAC");
+ABSL_FLAG(int,
+ isac_swb,
+ TestConfig::default_isac_swb(),
+ "RTP payload type for iSAC-swb (32 kHz)");
+ABSL_FLAG(int, opus, TestConfig::default_opus(), "RTP payload type for Opus");
+ABSL_FLAG(int,
+ pcm16b,
+ TestConfig::default_pcm16b(),
+ "RTP payload type for PCM16b-nb (8 kHz)");
+ABSL_FLAG(int,
+ pcm16b_wb,
+ TestConfig::default_pcm16b_wb(),
+ "RTP payload type for PCM16b-wb (16 kHz)");
+ABSL_FLAG(int,
+ pcm16b_swb32,
+ TestConfig::default_pcm16b_swb32(),
+ "RTP payload type for PCM16b-swb32 (32 kHz)");
+ABSL_FLAG(int,
+ pcm16b_swb48,
+ TestConfig::default_pcm16b_swb48(),
+ "RTP payload type for PCM16b-swb48 (48 kHz)");
+ABSL_FLAG(int, g722, TestConfig::default_g722(), "RTP payload type for G.722");
+ABSL_FLAG(int,
+ avt,
+ TestConfig::default_avt(),
+ "RTP payload type for AVT/DTMF (8 kHz)");
+ABSL_FLAG(int,
+ avt_16,
+ TestConfig::default_avt_16(),
+ "RTP payload type for AVT/DTMF (16 kHz)");
+ABSL_FLAG(int,
+ avt_32,
+ TestConfig::default_avt_32(),
+ "RTP payload type for AVT/DTMF (32 kHz)");
+ABSL_FLAG(int,
+ avt_48,
+ TestConfig::default_avt_48(),
+ "RTP payload type for AVT/DTMF (48 kHz)");
+ABSL_FLAG(int,
+ red,
+ TestConfig::default_red(),
+ "RTP payload type for redundant audio (RED)");
+ABSL_FLAG(int,
+ cn_nb,
+ TestConfig::default_cn_nb(),
+ "RTP payload type for comfort noise (8 kHz)");
+ABSL_FLAG(int,
+ cn_wb,
+ TestConfig::default_cn_wb(),
+ "RTP payload type for comfort noise (16 kHz)");
+ABSL_FLAG(int,
+ cn_swb32,
+ TestConfig::default_cn_swb32(),
+ "RTP payload type for comfort noise (32 kHz)");
+ABSL_FLAG(int,
+ cn_swb48,
+ TestConfig::default_cn_swb48(),
+ "RTP payload type for comfort noise (48 kHz)");
+ABSL_FLAG(std::string,
+ replacement_audio_file,
+ "",
+ "A PCM file that will be used to populate dummy"
+ " RTP packets");
+ABSL_FLAG(std::string,
+ ssrc,
+ "",
+ "Only use packets with this SSRC (decimal or hex, the latter "
+ "starting with 0x)");
+ABSL_FLAG(int,
+ audio_level,
+ TestConfig::default_audio_level(),
+ "Extension ID for audio level (RFC 6464)");
+ABSL_FLAG(int,
+ abs_send_time,
+ TestConfig::default_abs_send_time(),
+ "Extension ID for absolute sender time");
+ABSL_FLAG(int,
+ transport_seq_no,
+ TestConfig::default_transport_seq_no(),
+ "Extension ID for transport sequence number");
+ABSL_FLAG(int,
+ video_content_type,
+ TestConfig::default_video_content_type(),
+ "Extension ID for video content type");
+ABSL_FLAG(int,
+ video_timing,
+ TestConfig::default_video_timing(),
+ "Extension ID for video timing");
+ABSL_FLAG(std::string,
+ output_files_base_name,
+ "",
+ "Custom path used as prefix for the output files - i.e., "
+ "matlab plot, python plot, text log.");
+ABSL_FLAG(bool,
+ matlabplot,
+ false,
+ "Generates a matlab script for plotting the delay profile");
+ABSL_FLAG(bool,
+ pythonplot,
+ false,
+ "Generates a python script for plotting the delay profile");
+ABSL_FLAG(bool,
+ textlog,
+ false,
+ "Generates a text log describing the simulation on a "
+ "step-by-step basis.");
+ABSL_FLAG(bool, concealment_events, false, "Prints concealment events");
+ABSL_FLAG(int,
+ max_nr_packets_in_buffer,
+ TestConfig::default_max_nr_packets_in_buffer(),
+ "Maximum allowed number of packets in the buffer");
+ABSL_FLAG(bool,
+ enable_fast_accelerate,
+ false,
+ "Enables jitter buffer fast accelerate");
+
+namespace {
// Parses the input string for a valid SSRC (at the start of the string). If a
// valid SSRC is found, it is written to the output variable |ssrc|, and true is
@@ -195,26 +209,30 @@
}
void PrintCodecMapping() {
- PrintCodecMappingEntry("PCM-u", FLAG_pcmu);
- PrintCodecMappingEntry("PCM-a", FLAG_pcma);
- PrintCodecMappingEntry("iLBC", FLAG_ilbc);
- PrintCodecMappingEntry("iSAC", FLAG_isac);
- PrintCodecMappingEntry("iSAC-swb (32 kHz)", FLAG_isac_swb);
- PrintCodecMappingEntry("Opus", FLAG_opus);
- PrintCodecMappingEntry("PCM16b-nb (8 kHz)", FLAG_pcm16b);
- PrintCodecMappingEntry("PCM16b-wb (16 kHz)", FLAG_pcm16b_wb);
- PrintCodecMappingEntry("PCM16b-swb32 (32 kHz)", FLAG_pcm16b_swb32);
- PrintCodecMappingEntry("PCM16b-swb48 (48 kHz)", FLAG_pcm16b_swb48);
- PrintCodecMappingEntry("G.722", FLAG_g722);
- PrintCodecMappingEntry("AVT/DTMF (8 kHz)", FLAG_avt);
- PrintCodecMappingEntry("AVT/DTMF (16 kHz)", FLAG_avt_16);
- PrintCodecMappingEntry("AVT/DTMF (32 kHz)", FLAG_avt_32);
- PrintCodecMappingEntry("AVT/DTMF (48 kHz)", FLAG_avt_48);
- PrintCodecMappingEntry("redundant audio (RED)", FLAG_red);
- PrintCodecMappingEntry("comfort noise (8 kHz)", FLAG_cn_nb);
- PrintCodecMappingEntry("comfort noise (16 kHz)", FLAG_cn_wb);
- PrintCodecMappingEntry("comfort noise (32 kHz)", FLAG_cn_swb32);
- PrintCodecMappingEntry("comfort noise (48 kHz)", FLAG_cn_swb48);
+ PrintCodecMappingEntry("PCM-u", absl::GetFlag(FLAGS_pcmu));
+ PrintCodecMappingEntry("PCM-a", absl::GetFlag(FLAGS_pcma));
+ PrintCodecMappingEntry("iLBC", absl::GetFlag(FLAGS_ilbc));
+ PrintCodecMappingEntry("iSAC", absl::GetFlag(FLAGS_isac));
+ PrintCodecMappingEntry("iSAC-swb (32 kHz)", absl::GetFlag(FLAGS_isac_swb));
+ PrintCodecMappingEntry("Opus", absl::GetFlag(FLAGS_opus));
+ PrintCodecMappingEntry("PCM16b-nb (8 kHz)", absl::GetFlag(FLAGS_pcm16b));
+ PrintCodecMappingEntry("PCM16b-wb (16 kHz)", absl::GetFlag(FLAGS_pcm16b_wb));
+ PrintCodecMappingEntry("PCM16b-swb32 (32 kHz)",
+ absl::GetFlag(FLAGS_pcm16b_swb32));
+ PrintCodecMappingEntry("PCM16b-swb48 (48 kHz)",
+ absl::GetFlag(FLAGS_pcm16b_swb48));
+ PrintCodecMappingEntry("G.722", absl::GetFlag(FLAGS_g722));
+ PrintCodecMappingEntry("AVT/DTMF (8 kHz)", absl::GetFlag(FLAGS_avt));
+ PrintCodecMappingEntry("AVT/DTMF (16 kHz)", absl::GetFlag(FLAGS_avt_16));
+ PrintCodecMappingEntry("AVT/DTMF (32 kHz)", absl::GetFlag(FLAGS_avt_32));
+ PrintCodecMappingEntry("AVT/DTMF (48 kHz)", absl::GetFlag(FLAGS_avt_48));
+ PrintCodecMappingEntry("redundant audio (RED)", absl::GetFlag(FLAGS_red));
+ PrintCodecMappingEntry("comfort noise (8 kHz)", absl::GetFlag(FLAGS_cn_nb));
+ PrintCodecMappingEntry("comfort noise (16 kHz)", absl::GetFlag(FLAGS_cn_wb));
+ PrintCodecMappingEntry("comfort noise (32 kHz)",
+ absl::GetFlag(FLAGS_cn_swb32));
+ PrintCodecMappingEntry("comfort noise (48 kHz)",
+ absl::GetFlag(FLAGS_cn_swb48));
}
bool ValidateOutputFilesOptions(bool textlog,
@@ -268,116 +286,111 @@
} // namespace
int main(int argc, char* argv[]) {
+ std::vector<char*> args = absl::ParseCommandLine(argc, argv);
webrtc::test::NetEqTestFactory factory;
- std::string program_name = argv[0];
std::string usage =
"Tool for decoding an RTP dump file using NetEq.\n"
- "Run " +
- program_name +
- " --help for usage.\n"
- "Example usage:\n" +
- program_name + " input.rtp [output.{pcm, wav}]\n";
- if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true)) {
- exit(1);
- }
- if (FLAG_help) {
- std::cout << usage;
- rtc::FlagList::Print(nullptr, false);
- exit(0);
- }
- if (FLAG_codec_map) {
+ "Example usage:\n"
+ "./neteq_rtpplay input.rtp [output.{pcm, wav}]\n";
+ if (absl::GetFlag(FLAGS_codec_map)) {
PrintCodecMapping();
exit(0);
}
- if (argc < 2 || argc > 3) { // The output audio file is optional.
+ if (args.size() != 2 &&
+ args.size() != 3) { // The output audio file is optional.
// Print usage information.
std::cout << usage;
exit(0);
}
- const std::string output_audio_filename((argc == 3) ? argv[2] : "");
- const std::string output_files_base_name(FLAG_output_files_base_name);
+ const std::string output_audio_filename((args.size() == 3) ? args[2] : "");
+ const std::string output_files_base_name(
+ absl::GetFlag(FLAGS_output_files_base_name));
RTC_CHECK(ValidateOutputFilesOptions(
- FLAG_textlog, FLAG_matlabplot || FLAG_pythonplot, output_files_base_name,
- output_audio_filename));
- 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));
- RTC_CHECK(ValidateExtensionId(FLAG_video_content_type));
- RTC_CHECK(ValidateExtensionId(FLAG_video_timing));
+ absl::GetFlag(FLAGS_textlog),
+ absl::GetFlag(FLAGS_matlabplot) || absl::GetFlag(FLAGS_pythonplot),
+ output_files_base_name, output_audio_filename));
+ RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_pcmu)));
+ RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_pcma)));
+ RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_ilbc)));
+ RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_isac)));
+ RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_isac_swb)));
+ RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_opus)));
+ RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_pcm16b)));
+ RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_pcm16b_wb)));
+ RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_pcm16b_swb32)));
+ RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_pcm16b_swb48)));
+ RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_g722)));
+ RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_avt)));
+ RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_avt_16)));
+ RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_avt_32)));
+ RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_avt_48)));
+ RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_red)));
+ RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_cn_nb)));
+ RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_cn_wb)));
+ RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_cn_swb32)));
+ RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_cn_swb48)));
+ RTC_CHECK(ValidateSsrcValue(absl::GetFlag(FLAGS_ssrc)));
+ RTC_CHECK(ValidateExtensionId(absl::GetFlag(FLAGS_audio_level)));
+ RTC_CHECK(ValidateExtensionId(absl::GetFlag(FLAGS_abs_send_time)));
+ RTC_CHECK(ValidateExtensionId(absl::GetFlag(FLAGS_transport_seq_no)));
+ RTC_CHECK(ValidateExtensionId(absl::GetFlag(FLAGS_video_content_type)));
+ RTC_CHECK(ValidateExtensionId(absl::GetFlag(FLAGS_video_timing)));
- webrtc::field_trial::InitFieldTrialsFromString(FLAG_force_fieldtrials);
+ webrtc::field_trial::InitFieldTrialsFromString(
+ absl::GetFlag(FLAGS_force_fieldtrials).c_str());
webrtc::test::NetEqTestFactory::Config config;
- config.pcmu = FLAG_pcmu;
- config.pcma = FLAG_pcma;
- config.ilbc = FLAG_ilbc;
- config.isac = FLAG_isac;
- config.isac_swb = FLAG_isac_swb;
- config.opus = FLAG_opus;
- config.pcm16b = FLAG_pcm16b;
- config.pcm16b_wb = FLAG_pcm16b_wb;
- config.pcm16b_swb32 = FLAG_pcm16b_swb32;
- config.pcm16b_swb48 = FLAG_pcm16b_swb48;
- config.g722 = FLAG_g722;
- config.avt = FLAG_avt;
- config.avt_16 = FLAG_avt_16;
- config.avt_32 = FLAG_avt_32;
- config.avt_48 = FLAG_avt_48;
- config.red = FLAG_red;
- config.cn_nb = FLAG_cn_nb;
- config.cn_wb = FLAG_cn_wb;
- config.cn_swb32 = FLAG_cn_swb32;
- config.cn_swb48 = FLAG_cn_swb48;
- config.replacement_audio_file = FLAG_replacement_audio_file;
- config.audio_level = FLAG_audio_level;
- config.abs_send_time = FLAG_abs_send_time;
- config.transport_seq_no = FLAG_transport_seq_no;
- config.video_content_type = FLAG_video_content_type;
- config.video_timing = FLAG_video_timing;
- config.matlabplot = FLAG_matlabplot;
- config.pythonplot = FLAG_pythonplot;
- config.concealment_events = FLAG_concealment_events;
- config.max_nr_packets_in_buffer = FLAG_max_nr_packets_in_buffer;
- config.enable_fast_accelerate = FLAG_enable_fast_accelerate;
+ config.pcmu = absl::GetFlag(FLAGS_pcmu);
+ config.pcma = absl::GetFlag(FLAGS_pcma);
+ config.ilbc = absl::GetFlag(FLAGS_ilbc);
+ config.isac = absl::GetFlag(FLAGS_isac);
+ config.isac_swb = absl::GetFlag(FLAGS_isac_swb);
+ config.opus = absl::GetFlag(FLAGS_opus);
+ config.pcm16b = absl::GetFlag(FLAGS_pcm16b);
+ config.pcm16b_wb = absl::GetFlag(FLAGS_pcm16b_wb);
+ config.pcm16b_swb32 = absl::GetFlag(FLAGS_pcm16b_swb32);
+ config.pcm16b_swb48 = absl::GetFlag(FLAGS_pcm16b_swb48);
+ config.g722 = absl::GetFlag(FLAGS_g722);
+ config.avt = absl::GetFlag(FLAGS_avt);
+ config.avt_16 = absl::GetFlag(FLAGS_avt_16);
+ config.avt_32 = absl::GetFlag(FLAGS_avt_32);
+ config.avt_48 = absl::GetFlag(FLAGS_avt_48);
+ config.red = absl::GetFlag(FLAGS_red);
+ config.cn_nb = absl::GetFlag(FLAGS_cn_nb);
+ config.cn_wb = absl::GetFlag(FLAGS_cn_wb);
+ config.cn_swb32 = absl::GetFlag(FLAGS_cn_swb32);
+ config.cn_swb48 = absl::GetFlag(FLAGS_cn_swb48);
+ config.replacement_audio_file = absl::GetFlag(FLAGS_replacement_audio_file);
+ config.audio_level = absl::GetFlag(FLAGS_audio_level);
+ config.abs_send_time = absl::GetFlag(FLAGS_abs_send_time);
+ config.transport_seq_no = absl::GetFlag(FLAGS_transport_seq_no);
+ config.video_content_type = absl::GetFlag(FLAGS_video_content_type);
+ config.video_timing = absl::GetFlag(FLAGS_video_timing);
+ config.matlabplot = absl::GetFlag(FLAGS_matlabplot);
+ config.pythonplot = absl::GetFlag(FLAGS_pythonplot);
+ config.concealment_events = absl::GetFlag(FLAGS_concealment_events);
+ config.max_nr_packets_in_buffer =
+ absl::GetFlag(FLAGS_max_nr_packets_in_buffer);
+ config.enable_fast_accelerate = absl::GetFlag(FLAGS_enable_fast_accelerate);
if (!output_audio_filename.empty()) {
config.output_audio_filename = output_audio_filename;
}
- config.textlog_filename =
- CreateOptionalOutputFileName(FLAG_textlog, output_files_base_name,
- output_audio_filename, ".text_log.txt");
+ config.textlog_filename = CreateOptionalOutputFileName(
+ absl::GetFlag(FLAGS_textlog), output_files_base_name,
+ output_audio_filename, ".text_log.txt");
config.plot_scripts_basename = CreateOptionalOutputFileName(
- FLAG_matlabplot || FLAG_pythonplot, output_files_base_name,
- output_audio_filename, "");
+ absl::GetFlag(FLAGS_matlabplot) || absl::GetFlag(FLAGS_pythonplot),
+ output_files_base_name, output_audio_filename, "");
// Check if an SSRC value was provided.
- if (strlen(FLAG_ssrc) > 0) {
+ if (absl::GetFlag(FLAGS_ssrc).size() > 0) {
uint32_t ssrc;
- RTC_CHECK(ParseSsrc(FLAG_ssrc, &ssrc)) << "Flag verification has failed.";
+ RTC_CHECK(ParseSsrc(absl::GetFlag(FLAGS_ssrc), &ssrc))
+ << "Flag verification has failed.";
config.ssrc_filter = absl::make_optional(ssrc);
}
std::unique_ptr<webrtc::test::NetEqTest> test =
- factory.InitializeTestFromFile(/*input_filename=*/argv[1], config);
+ factory.InitializeTestFromFile(/*input_filename=*/args[1], config);
RTC_CHECK(test) << "ERROR: Unable to run test";
test->Run();
return 0;
diff --git a/modules/audio_coding/neteq/tools/rtp_analyze.cc b/modules/audio_coding/neteq/tools/rtp_analyze.cc
index 9d3041e..e71aee0 100644
--- a/modules/audio_coding/neteq/tools/rtp_analyze.cc
+++ b/modules/audio_coding/neteq/tools/rtp_analyze.cc
@@ -14,76 +14,73 @@
#include <memory>
#include <vector>
+#include "absl/flags/flag.h"
+#include "absl/flags/parse.h"
#include "modules/audio_coding/neteq/tools/packet.h"
#include "modules/audio_coding/neteq/tools/rtp_file_source.h"
#include "rtc_base/flags.h"
-// Define command line flags.
-WEBRTC_DEFINE_int(red, 117, "RTP payload type for RED");
-WEBRTC_DEFINE_int(audio_level,
- -1,
- "Extension ID for audio level (RFC 6464); "
- "-1 not to print audio level");
-WEBRTC_DEFINE_int(abs_send_time,
- -1,
- "Extension ID for absolute sender time; "
- "-1 not to print absolute send time");
-WEBRTC_DEFINE_bool(help, false, "Print this message");
+ABSL_FLAG(int, red, 117, "RTP payload type for RED");
+ABSL_FLAG(int,
+ audio_level,
+ -1,
+ "Extension ID for audio level (RFC 6464); "
+ "-1 not to print audio level");
+ABSL_FLAG(int,
+ abs_send_time,
+ -1,
+ "Extension ID for absolute sender time; "
+ "-1 not to print absolute send time");
int main(int argc, char* argv[]) {
- std::string program_name = argv[0];
+ std::vector<char*> args = absl::ParseCommandLine(argc, argv);
std::string usage =
"Tool for parsing an RTP dump file to text output.\n"
- "Run " +
- program_name +
- " --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. " +
+ "Example usage:\n"
+ "./rtp_analyze 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 or without payloads.\n";
- if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) || FLAG_help ||
- (argc != 2 && argc != 3)) {
+ if (args.size() != 2 && args.size() != 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
+ RTC_CHECK(absl::GetFlag(FLAGS_red) >= 0 &&
+ absl::GetFlag(FLAGS_red) <= 127); // Payload type
+ RTC_CHECK(absl::GetFlag(FLAGS_audio_level) == -1 || // Default
+ (absl::GetFlag(FLAGS_audio_level) > 0 &&
+ absl::GetFlag(FLAGS_audio_level) <= 255)); // Extension ID
+ RTC_CHECK(absl::GetFlag(FLAGS_abs_send_time) == -1 || // Default
+ (absl::GetFlag(FLAGS_abs_send_time) > 0 &&
+ absl::GetFlag(FLAGS_abs_send_time) <= 255)); // Extension ID
- printf("Input file: %s\n", argv[1]);
+ printf("Input file: %s\n", args[1]);
std::unique_ptr<webrtc::test::RtpFileSource> file_source(
- webrtc::test::RtpFileSource::Create(argv[1]));
+ webrtc::test::RtpFileSource::Create(args[1]));
assert(file_source.get());
// Set RTP extension IDs.
bool print_audio_level = false;
- if (FLAG_audio_level != -1) {
+ if (absl::GetFlag(FLAGS_audio_level) != -1) {
print_audio_level = true;
file_source->RegisterRtpHeaderExtension(webrtc::kRtpExtensionAudioLevel,
- FLAG_audio_level);
+ absl::GetFlag(FLAGS_audio_level));
}
bool print_abs_send_time = false;
- if (FLAG_abs_send_time != -1) {
+ if (absl::GetFlag(FLAGS_abs_send_time) != -1) {
print_abs_send_time = true;
file_source->RegisterRtpHeaderExtension(
- webrtc::kRtpExtensionAbsoluteSendTime, FLAG_abs_send_time);
+ webrtc::kRtpExtensionAbsoluteSendTime,
+ absl::GetFlag(FLAGS_abs_send_time));
}
FILE* out_file;
- if (argc == 3) {
- out_file = fopen(argv[2], "wt");
+ if (args.size() == 3) {
+ out_file = fopen(args[2], "wt");
if (!out_file) {
- printf("Cannot open output file %s\n", argv[2]);
+ printf("Cannot open output file %s\n", args[2]);
return -1;
}
- printf("Output file: %s\n\n", argv[2]);
+ printf("Output file: %s\n\n", args[2]);
} else {
out_file = stdout;
}
@@ -150,7 +147,7 @@
}
fprintf(out_file, "\n");
- if (packet->header().payloadType == FLAG_red) {
+ if (packet->header().payloadType == absl::GetFlag(FLAGS_red)) {
std::list<webrtc::RTPHeader*> red_headers;
packet->ExtractRedHeaders(&red_headers);
while (!red_headers.empty()) {
diff --git a/modules/audio_coding/neteq/tools/rtp_encode.cc b/modules/audio_coding/neteq/tools/rtp_encode.cc
index a75802b..0379d21 100644
--- a/modules/audio_coding/neteq/tools/rtp_encode.cc
+++ b/modules/audio_coding/neteq/tools/rtp_encode.cc
@@ -20,7 +20,10 @@
#include <iostream>
#include <map>
#include <string>
+#include <vector>
+#include "absl/flags/flag.h"
+#include "absl/flags/parse.h"
#include "absl/memory/memory.h"
#include "api/audio/audio_frame.h"
#include "api/audio_codecs/L16/audio_encoder_L16.h"
@@ -35,30 +38,29 @@
#include "rtc_base/flags.h"
#include "rtc_base/numerics/safe_conversions.h"
+ABSL_FLAG(bool, list_codecs, false, "Enumerate all codecs");
+ABSL_FLAG(std::string, codec, "opus", "Codec to use");
+ABSL_FLAG(int,
+ frame_len,
+ 0,
+ "Frame length in ms; 0 indicates codec default value");
+ABSL_FLAG(int, bitrate, 0, "Bitrate in kbps; 0 indicates codec default value");
+ABSL_FLAG(int,
+ payload_type,
+ -1,
+ "RTP payload type; -1 indicates codec default value");
+ABSL_FLAG(int,
+ cng_payload_type,
+ -1,
+ "RTP payload type for CNG; -1 indicates default value");
+ABSL_FLAG(int, ssrc, 0, "SSRC to write to the RTP header");
+ABSL_FLAG(bool, dtx, false, "Use DTX/CNG");
+ABSL_FLAG(int, sample_rate, 48000, "Sample rate of the input file");
+
namespace webrtc {
namespace test {
namespace {
-// Define command line flags.
-WEBRTC_DEFINE_bool(list_codecs, false, "Enumerate all codecs");
-WEBRTC_DEFINE_string(codec, "opus", "Codec to use");
-WEBRTC_DEFINE_int(frame_len,
- 0,
- "Frame length in ms; 0 indicates codec default value");
-WEBRTC_DEFINE_int(bitrate,
- 0,
- "Bitrate in kbps; 0 indicates codec default value");
-WEBRTC_DEFINE_int(payload_type,
- -1,
- "RTP payload type; -1 indicates codec default value");
-WEBRTC_DEFINE_int(cng_payload_type,
- -1,
- "RTP payload type for CNG; -1 indicates default value");
-WEBRTC_DEFINE_int(ssrc, 0, "SSRC to write to the RTP header");
-WEBRTC_DEFINE_bool(dtx, false, "Use DTX/CNG");
-WEBRTC_DEFINE_int(sample_rate, 48000, "Sample rate of the input file");
-WEBRTC_DEFINE_bool(help, false, "Print this message");
-
// Add new codecs here, and to the map below.
enum class CodecType {
kOpus,
@@ -160,8 +162,8 @@
};
void SetFrameLenIfFlagIsPositive(int* config_frame_len) {
- if (FLAG_frame_len > 0) {
- *config_frame_len = FLAG_frame_len;
+ if (absl::GetFlag(FLAGS_frame_len) > 0) {
+ *config_frame_len = absl::GetFlag(FLAGS_frame_len);
}
}
@@ -199,10 +201,10 @@
switch (codec_type) {
case CodecType::kOpus: {
AudioEncoderOpus::Config config = GetCodecConfig<AudioEncoderOpus>();
- if (FLAG_bitrate > 0) {
- config.bitrate_bps = FLAG_bitrate;
+ if (absl::GetFlag(FLAGS_bitrate) > 0) {
+ config.bitrate_bps = absl::GetFlag(FLAGS_bitrate);
}
- config.dtx_enabled = FLAG_dtx;
+ config.dtx_enabled = absl::GetFlag(FLAGS_dtx);
RTC_CHECK(config.IsOk());
return AudioEncoderOpus::MakeAudioEncoder(config, payload_type);
}
@@ -261,33 +263,25 @@
}
return 0;
};
- cng_config.payload_type = FLAG_cng_payload_type != -1
- ? FLAG_cng_payload_type
+ cng_config.payload_type = absl::GetFlag(FLAGS_cng_payload_type) != -1
+ ? absl::GetFlag(FLAGS_cng_payload_type)
: default_payload_type();
return cng_config;
}
int RunRtpEncode(int argc, char* argv[]) {
- const std::string program_name = argv[0];
+ std::vector<char*> args = absl::ParseCommandLine(argc, argv);
const std::string usage =
"Tool for generating an RTP dump file from audio input.\n"
- "Run " +
- program_name +
- " --help for usage.\n"
- "Example usage:\n" +
- program_name + " input.pcm output.rtp --codec=[codec] " +
+ "Example usage:\n"
+ "./rtp_encode input.pcm output.rtp --codec=[codec] "
"--frame_len=[frame_len] --bitrate=[bitrate]\n\n";
- if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) || FLAG_help ||
- (!FLAG_list_codecs && argc != 3)) {
+ if (!absl::GetFlag(FLAGS_list_codecs) && args.size() != 3) {
printf("%s", usage.c_str());
- if (FLAG_help) {
- rtc::FlagList::Print(nullptr, false);
- return 0;
- }
return 1;
}
- if (FLAG_list_codecs) {
+ if (absl::GetFlag(FLAGS_list_codecs)) {
printf("The following arguments are valid --codec parameters:\n");
for (const auto& c : CodecList()) {
printf(" %s\n", c.first.c_str());
@@ -295,22 +289,23 @@
return 0;
}
- const auto codec_it = CodecList().find(FLAG_codec);
+ const auto codec_it = CodecList().find(absl::GetFlag(FLAGS_codec));
if (codec_it == CodecList().end()) {
- printf("%s is not a valid codec name.\n", FLAG_codec);
+ printf("%s is not a valid codec name.\n",
+ absl::GetFlag(FLAGS_codec).c_str());
printf("Use argument --list_codecs to see all valid codec names.\n");
return 1;
}
// Create the codec.
- const int payload_type = FLAG_payload_type == -1
+ const int payload_type = absl::GetFlag(FLAGS_payload_type) == -1
? codec_it->second.default_payload_type
- : FLAG_payload_type;
+ : absl::GetFlag(FLAGS_payload_type);
std::unique_ptr<AudioEncoder> codec =
CreateEncoder(codec_it->second.type, payload_type);
// Create an external VAD/CNG encoder if needed.
- if (FLAG_dtx && !codec_it->second.internal_dtx) {
+ if (absl::GetFlag(FLAGS_dtx) && !codec_it->second.internal_dtx) {
AudioEncoderCngConfig cng_config = GetCngConfig(codec->SampleRateHz());
RTC_DCHECK(codec);
cng_config.speech_encoder = std::move(codec);
@@ -325,11 +320,11 @@
acm->SetEncoder(std::move(codec));
// Open files.
- printf("Input file: %s\n", argv[1]);
- InputAudioFile input_file(argv[1], false); // Open input in non-looping mode.
- FILE* out_file = fopen(argv[2], "wb");
- RTC_CHECK(out_file) << "Could not open file " << argv[2] << " for writing";
- printf("Output file: %s\n", argv[2]);
+ printf("Input file: %s\n", args[1]);
+ InputAudioFile input_file(args[1], false); // Open input in non-looping mode.
+ FILE* out_file = fopen(args[2], "wb");
+ RTC_CHECK(out_file) << "Could not open file " << args[2] << " for writing";
+ printf("Output file: %s\n", args[2]);
fprintf(out_file, "#!rtpplay1.0 \n"); //,
// Write 3 32-bit values followed by 2 16-bit values, all set to 0. This means
// a total of 16 bytes.
@@ -337,12 +332,13 @@
RTC_CHECK_EQ(fwrite(file_header, sizeof(file_header), 1, out_file), 1);
// Create and register the packetizer, which will write the packets to file.
- Packetizer packetizer(out_file, FLAG_ssrc, timestamp_rate_hz);
+ Packetizer packetizer(out_file, absl::GetFlag(FLAGS_ssrc), timestamp_rate_hz);
RTC_DCHECK_EQ(acm->RegisterTransportCallback(&packetizer), 0);
AudioFrame audio_frame;
- audio_frame.samples_per_channel_ = FLAG_sample_rate / 100; // 10 ms
- audio_frame.sample_rate_hz_ = FLAG_sample_rate;
+ audio_frame.samples_per_channel_ =
+ absl::GetFlag(FLAGS_sample_rate) / 100; // 10 ms
+ audio_frame.sample_rate_hz_ = absl::GetFlag(FLAGS_sample_rate);
audio_frame.num_channels_ = 1;
while (input_file.Read(audio_frame.samples_per_channel_,
diff --git a/modules/audio_coding/neteq/tools/rtp_jitter.cc b/modules/audio_coding/neteq/tools/rtp_jitter.cc
index 92a7a8d..3521145 100644
--- a/modules/audio_coding/neteq/tools/rtp_jitter.cc
+++ b/modules/audio_coding/neteq/tools/rtp_jitter.cc
@@ -17,14 +17,11 @@
#include "api/array_view.h"
#include "modules/rtp_rtcp/source/byte_io.h"
#include "rtc_base/buffer.h"
-#include "rtc_base/flags.h"
namespace webrtc {
namespace test {
namespace {
-WEBRTC_DEFINE_bool(help, false, "Print help message");
-
constexpr size_t kRtpDumpHeaderLength = 8;
// Returns the next packet or an empty buffer if end of file was encountered.
@@ -83,10 +80,9 @@
"Tool for alternating the arrival times in an RTP dump file.\n"
"Example usage:\n" +
program_name + " input.rtp arrival_times_ms.txt output.rtp\n\n";
- if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) || FLAG_help ||
- argc != 4) {
+ if (argc != 4) {
printf("%s", usage.c_str());
- return FLAG_help ? 0 : 1;
+ return 1;
}
printf("Input RTP file: %s\n", argv[1]);
