diff --git a/audio/test/low_bandwidth_audio_test.cc b/audio/test/low_bandwidth_audio_test.cc index 1009a5ed91..857f983d57 100644 --- a/audio/test/low_bandwidth_audio_test.cc +++ b/audio/test/low_bandwidth_audio_test.cc @@ -14,14 +14,15 @@ #include "system_wrappers/include/sleep.h" #include "test/testsupport/fileutils.h" -DEFINE_int(sample_rate_hz, - 16000, - "Sample rate (Hz) of the produced audio files."); +WEBRTC_DEFINE_int(sample_rate_hz, + 16000, + "Sample rate (Hz) of the produced audio files."); -DEFINE_bool(quick, - false, - "Don't do the full audio recording. " - "Used to quickly check that the test runs without crashing."); +WEBRTC_DEFINE_bool( + quick, + false, + "Don't do the full audio recording. " + "Used to quickly check that the test runs without crashing."); namespace webrtc { namespace test { diff --git a/call/rampup_tests.cc b/call/rampup_tests.cc index 115f968c7a..6ce17e9489 100644 --- a/call/rampup_tests.cc +++ b/call/rampup_tests.cc @@ -40,7 +40,9 @@ std::vector GenerateSsrcs(size_t num_streams, uint32_t ssrc_offset) { } } // namespace -DEFINE_string(ramp_dump_name, "", "Filename for dumped received RTP stream."); +WEBRTC_DEFINE_string(ramp_dump_name, + "", + "Filename for dumped received RTP stream."); RampUpTester::RampUpTester(size_t num_video_streams, size_t num_audio_streams, diff --git a/examples/peerconnection/client/flagdefs.h b/examples/peerconnection/client/flagdefs.h index 564e0e95f9..7bbc3839e4 100644 --- a/examples/peerconnection/client/flagdefs.h +++ b/examples/peerconnection/client/flagdefs.h @@ -19,16 +19,16 @@ extern const uint16_t kDefaultServerPort; // From defaults.[h|cc] // header file so that they can be shared across the different main.cc's // for each platform. -DEFINE_bool(help, false, "Prints this message"); -DEFINE_bool(autoconnect, - false, - "Connect to the server without user " - "intervention."); -DEFINE_string(server, "localhost", "The server to connect to."); -DEFINE_int(port, - kDefaultServerPort, - "The port on which the server is listening."); -DEFINE_bool( +WEBRTC_DEFINE_bool(help, false, "Prints this message"); +WEBRTC_DEFINE_bool(autoconnect, + false, + "Connect to the server without user " + "intervention."); +WEBRTC_DEFINE_string(server, "localhost", "The server to connect to."); +WEBRTC_DEFINE_int(port, + kDefaultServerPort, + "The port on which the server is listening."); +WEBRTC_DEFINE_bool( autocall, false, "Call the first available other client on " diff --git a/examples/stunprober/main.cc b/examples/stunprober/main.cc index 45a76f5237..f438fcf64d 100644 --- a/examples/stunprober/main.cc +++ b/examples/stunprober/main.cc @@ -33,16 +33,21 @@ using stunprober::StunProber; using stunprober::AsyncCallback; -DEFINE_bool(help, false, "Prints this message"); -DEFINE_int(interval, 10, "Interval of consecutive stun pings in milliseconds"); -DEFINE_bool(shared_socket, false, "Share socket mode for different remote IPs"); -DEFINE_int(pings_per_ip, - 10, - "Number of consecutive stun pings to send for each IP"); -DEFINE_int(timeout, - 1000, - "Milliseconds of wait after the last ping sent before exiting"); -DEFINE_string( +WEBRTC_DEFINE_bool(help, false, "Prints this message"); +WEBRTC_DEFINE_int(interval, + 10, + "Interval of consecutive stun pings in milliseconds"); +WEBRTC_DEFINE_bool(shared_socket, + false, + "Share socket mode for different remote IPs"); +WEBRTC_DEFINE_int(pings_per_ip, + 10, + "Number of consecutive stun pings to send for each IP"); +WEBRTC_DEFINE_int( + timeout, + 1000, + "Milliseconds of wait after the last ping sent before exiting"); +WEBRTC_DEFINE_string( servers, "stun.l.google.com:19302,stun1.l.google.com:19302,stun2.l.google.com:19302", "Comma separated STUN server addresses with ports"); diff --git a/logging/rtc_event_log/rtc_event_log2rtp_dump.cc b/logging/rtc_event_log/rtc_event_log2rtp_dump.cc index 46ce248ff6..094af1f82d 100644 --- a/logging/rtc_event_log/rtc_event_log2rtp_dump.cc +++ b/logging/rtc_event_log/rtc_event_log2rtp_dump.cc @@ -32,31 +32,32 @@ namespace { using MediaType = webrtc::ParsedRtcEventLogNew::MediaType; -DEFINE_bool( +WEBRTC_DEFINE_bool( audio, true, "Use --noaudio to exclude audio packets from the converted RTPdump file."); -DEFINE_bool( +WEBRTC_DEFINE_bool( video, true, "Use --novideo to exclude video packets from the converted RTPdump file."); -DEFINE_bool( +WEBRTC_DEFINE_bool( data, true, "Use --nodata to exclude data packets from the converted RTPdump file."); -DEFINE_bool( +WEBRTC_DEFINE_bool( rtp, true, "Use --nortp to exclude RTP packets from the converted RTPdump file."); -DEFINE_bool( +WEBRTC_DEFINE_bool( rtcp, true, "Use --nortcp to exclude RTCP packets from the converted RTPdump file."); -DEFINE_string(ssrc, - "", - "Store only packets with this SSRC (decimal or hex, the latter " - "starting with 0x)."); -DEFINE_bool(help, false, "Prints this message."); +WEBRTC_DEFINE_string( + ssrc, + "", + "Store only packets with this SSRC (decimal or hex, the latter " + "starting with 0x)."); +WEBRTC_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, diff --git a/logging/rtc_event_log/rtc_event_log2stats.cc b/logging/rtc_event_log/rtc_event_log2stats.cc index 928c829f49..da8f23a5ec 100644 --- a/logging/rtc_event_log/rtc_event_log2stats.cc +++ b/logging/rtc_event_log/rtc_event_log2stats.cc @@ -36,7 +36,7 @@ RTC_POP_IGNORING_WUNDEF() namespace { -DEFINE_bool(help, false, "Prints this message."); +WEBRTC_DEFINE_bool(help, false, "Prints this message."); struct Stats { int count = 0; diff --git a/logging/rtc_event_log/rtc_event_log2text.cc b/logging/rtc_event_log/rtc_event_log2text.cc index 67cb6ecdf4..4cdde7dbe6 100644 --- a/logging/rtc_event_log/rtc_event_log2text.cc +++ b/logging/rtc_event_log/rtc_event_log2text.cc @@ -42,35 +42,44 @@ namespace { -DEFINE_bool(unknown, true, "Use --nounknown to exclude unknown events."); -DEFINE_bool(startstop, true, "Use --nostartstop to exclude start/stop events."); -DEFINE_bool(config, true, "Use --noconfig to exclude stream configurations."); -DEFINE_bool(bwe, true, "Use --nobwe to exclude BWE events."); -DEFINE_bool(incoming, true, "Use --noincoming to exclude incoming packets."); -DEFINE_bool(outgoing, true, "Use --nooutgoing to exclude packets."); +WEBRTC_DEFINE_bool(unknown, true, "Use --nounknown to exclude unknown events."); +WEBRTC_DEFINE_bool(startstop, + true, + "Use --nostartstop to exclude start/stop events."); +WEBRTC_DEFINE_bool(config, + true, + "Use --noconfig to exclude stream configurations."); +WEBRTC_DEFINE_bool(bwe, true, "Use --nobwe to exclude BWE events."); +WEBRTC_DEFINE_bool(incoming, + true, + "Use --noincoming to exclude incoming packets."); +WEBRTC_DEFINE_bool(outgoing, true, "Use --nooutgoing to exclude packets."); // TODO(terelius): Note that the media type doesn't work with outgoing packets. -DEFINE_bool(audio, true, "Use --noaudio to exclude audio packets."); +WEBRTC_DEFINE_bool(audio, true, "Use --noaudio to exclude audio packets."); // TODO(terelius): Note that the media type doesn't work with outgoing packets. -DEFINE_bool(video, true, "Use --novideo to exclude video packets."); +WEBRTC_DEFINE_bool(video, true, "Use --novideo to exclude video packets."); // TODO(terelius): Note that the media type doesn't work with outgoing packets. -DEFINE_bool(data, true, "Use --nodata to exclude data packets."); -DEFINE_bool(rtp, true, "Use --nortp to exclude RTP packets."); -DEFINE_bool(rtcp, true, "Use --nortcp to exclude RTCP packets."); -DEFINE_bool(playout, true, "Use --noplayout to exclude audio playout events."); -DEFINE_bool(ana, true, "Use --noana to exclude ANA events."); -DEFINE_bool(probe, true, "Use --noprobe to exclude probe events."); -DEFINE_bool(ice, true, "Use --noice to exclude ICE events."); +WEBRTC_DEFINE_bool(data, true, "Use --nodata to exclude data packets."); +WEBRTC_DEFINE_bool(rtp, true, "Use --nortp to exclude RTP packets."); +WEBRTC_DEFINE_bool(rtcp, true, "Use --nortcp to exclude RTCP packets."); +WEBRTC_DEFINE_bool(playout, + true, + "Use --noplayout to exclude audio playout events."); +WEBRTC_DEFINE_bool(ana, true, "Use --noana to exclude ANA events."); +WEBRTC_DEFINE_bool(probe, true, "Use --noprobe to exclude probe events."); +WEBRTC_DEFINE_bool(ice, true, "Use --noice to exclude ICE events."); -DEFINE_bool(print_full_packets, - false, - "Print the full RTP headers and RTCP packets in hex."); +WEBRTC_DEFINE_bool(print_full_packets, + false, + "Print the full RTP headers and RTCP packets in hex."); // TODO(terelius): Allow a list of SSRCs. -DEFINE_string(ssrc, - "", - "Print only packets with this SSRC (decimal or hex, the latter " - "starting with 0x)."); -DEFINE_bool(help, false, "Prints this message."); +WEBRTC_DEFINE_string( + ssrc, + "", + "Print only packets with this SSRC (decimal or hex, the latter " + "starting with 0x)."); +WEBRTC_DEFINE_bool(help, false, "Prints this message."); using MediaType = webrtc::ParsedRtcEventLogNew::MediaType; diff --git a/modules/audio_coding/neteq/neteq_unittest.cc b/modules/audio_coding/neteq/neteq_unittest.cc index 3c63aa736a..1c9b9e7123 100644 --- a/modules/audio_coding/neteq/neteq_unittest.cc +++ b/modules/audio_coding/neteq/neteq_unittest.cc @@ -52,7 +52,7 @@ RTC_PUSH_IGNORING_WUNDEF() RTC_POP_IGNORING_WUNDEF() #endif -DEFINE_bool(gen_ref, false, "Generate reference files."); +WEBRTC_DEFINE_bool(gen_ref, false, "Generate reference files."); namespace webrtc { diff --git a/modules/audio_coding/neteq/test/neteq_ilbc_quality_test.cc b/modules/audio_coding/neteq/test/neteq_ilbc_quality_test.cc index ad61235cf2..6f10345298 100644 --- a/modules/audio_coding/neteq/test/neteq_ilbc_quality_test.cc +++ b/modules/audio_coding/neteq/test/neteq_ilbc_quality_test.cc @@ -25,7 +25,7 @@ namespace { static const int kInputSampleRateKhz = 8; static const int kOutputSampleRateKhz = 8; -DEFINE_int(frame_size_ms, 20, "Codec frame size (milliseconds)."); +WEBRTC_DEFINE_int(frame_size_ms, 20, "Codec frame size (milliseconds)."); } // namespace diff --git a/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc b/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc index 94984b87e6..651b0ca71a 100644 --- a/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc +++ b/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc @@ -21,7 +21,7 @@ static const int kIsacBlockDurationMs = 30; static const int kIsacInputSamplingKhz = 16; static const int kIsacOutputSamplingKhz = 16; -DEFINE_int(bit_rate_kbps, 32, "Target bit rate (kbps)."); +WEBRTC_DEFINE_int(bit_rate_kbps, 32, "Target bit rate (kbps)."); } // namespace diff --git a/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc b/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc index 6861e4c55a..f4a36363ba 100644 --- a/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc +++ b/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc @@ -22,24 +22,26 @@ namespace { static const int kOpusBlockDurationMs = 20; static const int kOpusSamplingKhz = 48; -DEFINE_int(bit_rate_kbps, 32, "Target bit rate (kbps)."); +WEBRTC_DEFINE_int(bit_rate_kbps, 32, "Target bit rate (kbps)."); -DEFINE_int(complexity, - 10, - "Complexity: 0 ~ 10 -- defined as in Opus" - "specification."); +WEBRTC_DEFINE_int(complexity, + 10, + "Complexity: 0 ~ 10 -- defined as in Opus" + "specification."); -DEFINE_int(maxplaybackrate, 48000, "Maximum playback rate (Hz)."); +WEBRTC_DEFINE_int(maxplaybackrate, 48000, "Maximum playback rate (Hz)."); -DEFINE_int(application, 0, "Application mode: 0 -- VOIP, 1 -- Audio."); +WEBRTC_DEFINE_int(application, 0, "Application mode: 0 -- VOIP, 1 -- Audio."); -DEFINE_int(reported_loss_rate, 10, "Reported percentile of packet loss."); +WEBRTC_DEFINE_int(reported_loss_rate, + 10, + "Reported percentile of packet loss."); -DEFINE_bool(fec, false, "Enable FEC for encoding (-nofec to disable)."); +WEBRTC_DEFINE_bool(fec, false, "Enable FEC for encoding (-nofec to disable)."); -DEFINE_bool(dtx, false, "Enable DTX for encoding (-nodtx to disable)."); +WEBRTC_DEFINE_bool(dtx, false, "Enable DTX for encoding (-nodtx to disable)."); -DEFINE_int(sub_packets, 1, "Number of sub packets to repacketize."); +WEBRTC_DEFINE_int(sub_packets, 1, "Number of sub packets to repacketize."); } // namespace diff --git a/modules/audio_coding/neteq/test/neteq_pcm16b_quality_test.cc b/modules/audio_coding/neteq/test/neteq_pcm16b_quality_test.cc index 8872b94a06..9c53919d54 100644 --- a/modules/audio_coding/neteq/test/neteq_pcm16b_quality_test.cc +++ b/modules/audio_coding/neteq/test/neteq_pcm16b_quality_test.cc @@ -26,7 +26,7 @@ namespace { static const int kInputSampleRateKhz = 48; static const int kOutputSampleRateKhz = 48; -DEFINE_int(frame_size_ms, 20, "Codec frame size (milliseconds)."); +WEBRTC_DEFINE_int(frame_size_ms, 20, "Codec frame size (milliseconds)."); } // namespace diff --git a/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc b/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc index 54ff849739..85f22671ea 100644 --- a/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc +++ b/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc @@ -25,7 +25,7 @@ namespace { static const int kInputSampleRateKhz = 8; static const int kOutputSampleRateKhz = 8; -DEFINE_int(frame_size_ms, 20, "Codec frame size (milliseconds)."); +WEBRTC_DEFINE_int(frame_size_ms, 20, "Codec frame size (milliseconds)."); } // namespace diff --git a/modules/audio_coding/neteq/test/neteq_speed_test.cc b/modules/audio_coding/neteq/test/neteq_speed_test.cc index c1d78c5730..70777a2d02 100644 --- a/modules/audio_coding/neteq/test/neteq_speed_test.cc +++ b/modules/audio_coding/neteq/test/neteq_speed_test.cc @@ -16,10 +16,10 @@ #include "rtc_base/flags.h" // Define command line flags. -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."); -DEFINE_bool(help, false, "Print this message."); +WEBRTC_DEFINE_int(runtime_ms, 10000, "Simulated runtime in ms."); +WEBRTC_DEFINE_int(lossrate, 10, "Packet lossrate; drop every N packets."); +WEBRTC_DEFINE_float(drift, 0.1f, "Clockdrift factor."); +WEBRTC_DEFINE_bool(help, false, "Print this message."); int main(int argc, char* argv[]) { std::string program_name = argv[0]; diff --git a/modules/audio_coding/neteq/tools/neteq_quality_test.cc b/modules/audio_coding/neteq/tools/neteq_quality_test.cc index faca8959d4..2ee6779c53 100644 --- a/modules/audio_coding/neteq/tools/neteq_quality_test.cc +++ b/modules/audio_coding/neteq/tools/neteq_quality_test.cc @@ -47,42 +47,47 @@ static bool ValidateFilename(const std::string& value, bool write) { return true; } -DEFINE_string( +WEBRTC_DEFINE_string( in_filename, DefaultInFilename().c_str(), "Filename for input audio (specify sample rate with --input_sample_rate, " "and channels with --channels)."); -DEFINE_int(input_sample_rate, 16000, "Sample rate of input file in Hz."); +WEBRTC_DEFINE_int(input_sample_rate, 16000, "Sample rate of input file in Hz."); -DEFINE_int(channels, 1, "Number of channels in input audio."); +WEBRTC_DEFINE_int(channels, 1, "Number of channels in input audio."); -DEFINE_string(out_filename, - DefaultOutFilename().c_str(), - "Name of output audio file."); +WEBRTC_DEFINE_string(out_filename, + DefaultOutFilename().c_str(), + "Name of output audio file."); -DEFINE_int(runtime_ms, 10000, "Simulated runtime (milliseconds)."); +WEBRTC_DEFINE_int(runtime_ms, 10000, "Simulated runtime (milliseconds)."); -DEFINE_int(packet_loss_rate, 10, "Percentile of packet loss."); +WEBRTC_DEFINE_int(packet_loss_rate, 10, "Percentile of packet loss."); -DEFINE_int(random_loss_mode, - kUniformLoss, - "Random loss mode: 0--no loss, 1--uniform loss, 2--Gilbert Elliot " - "loss, 3--fixed loss."); +WEBRTC_DEFINE_int( + random_loss_mode, + kUniformLoss, + "Random loss mode: 0--no loss, 1--uniform loss, 2--Gilbert Elliot " + "loss, 3--fixed loss."); -DEFINE_int(burst_length, - 30, - "Burst length in milliseconds, only valid for Gilbert Elliot loss."); +WEBRTC_DEFINE_int( + burst_length, + 30, + "Burst length in milliseconds, only valid for Gilbert Elliot loss."); -DEFINE_float(drift_factor, 0.0, "Time drift factor."); +WEBRTC_DEFINE_float(drift_factor, 0.0, "Time drift factor."); -DEFINE_int(preload_packets, 0, "Preload the buffer with this many packets."); +WEBRTC_DEFINE_int(preload_packets, + 0, + "Preload the buffer with this many packets."); -DEFINE_string(loss_events, - "", - "List of loss events time and duration separated by comma: " - " , " - ", ..."); +WEBRTC_DEFINE_string( + loss_events, + "", + "List of loss events time and duration separated by comma: " + " , " + ", ..."); // ProbTrans00Solver() is to calculate the transition probability from no-loss // state to itself in a modified Gilbert Elliot packet loss model. The result is diff --git a/modules/audio_coding/neteq/tools/neteq_rtpplay.cc b/modules/audio_coding/neteq/tools/neteq_rtpplay.cc index 25e8cd8816..c2726eb470 100644 --- a/modules/audio_coding/neteq/tools/neteq_rtpplay.cc +++ b/modules/audio_coding/neteq/tools/neteq_rtpplay.cc @@ -17,17 +17,17 @@ #include "system_wrappers/include/field_trial.h" #include "test/field_trial.h" -DEFINE_bool(codec_map, - false, - "Prints the mapping between RTP payload type and " - "codec"); -DEFINE_string( +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."); -DEFINE_bool(help, false, "Prints this message"); +WEBRTC_DEFINE_bool(help, false, "Prints this message"); int main(int argc, char* argv[]) { webrtc::test::NetEqTestFactory factory; diff --git a/modules/audio_coding/neteq/tools/neteq_test_factory.cc b/modules/audio_coding/neteq/tools/neteq_test_factory.cc index df3a9f0c9c..93da54c8af 100644 --- a/modules/audio_coding/neteq/tools/neteq_test_factory.cc +++ b/modules/audio_coding/neteq/tools/neteq_test_factory.cc @@ -91,50 +91,57 @@ static bool ValidateExtensionId(int value) { } // Define command line flags. -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_string(replacement_audio_file, - "", - "A PCM file that will be used to populate " - "dummy" - " RTP packets"); -DEFINE_string(ssrc, - "", - "Only use packets with this SSRC (decimal or hex, the latter " - "starting with 0x)"); -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_int(video_content_type, 7, "Extension ID for video content type"); -DEFINE_int(video_timing, 8, "Extension ID for video timing"); -DEFINE_bool(matlabplot, - false, - "Generates a matlab script for plotting the delay profile"); -DEFINE_bool(pythonplot, - false, - "Generates a python script for plotting the delay profile"); -DEFINE_bool(concealment_events, false, "Prints concealment events"); -DEFINE_int(max_nr_packets_in_buffer, - 50, - "Maximum allowed number of packets in the buffer"); +WEBRTC_DEFINE_int(pcmu, 0, "RTP payload type for PCM-u"); +WEBRTC_DEFINE_int(pcma, 8, "RTP payload type for PCM-a"); +WEBRTC_DEFINE_int(ilbc, 102, "RTP payload type for iLBC"); +WEBRTC_DEFINE_int(isac, 103, "RTP payload type for iSAC"); +WEBRTC_DEFINE_int(isac_swb, 104, "RTP payload type for iSAC-swb (32 kHz)"); +WEBRTC_DEFINE_int(opus, 111, "RTP payload type for Opus"); +WEBRTC_DEFINE_int(pcm16b, 93, "RTP payload type for PCM16b-nb (8 kHz)"); +WEBRTC_DEFINE_int(pcm16b_wb, 94, "RTP payload type for PCM16b-wb (16 kHz)"); +WEBRTC_DEFINE_int(pcm16b_swb32, + 95, + "RTP payload type for PCM16b-swb32 (32 kHz)"); +WEBRTC_DEFINE_int(pcm16b_swb48, + 96, + "RTP payload type for PCM16b-swb48 (48 kHz)"); +WEBRTC_DEFINE_int(g722, 9, "RTP payload type for G.722"); +WEBRTC_DEFINE_int(avt, 106, "RTP payload type for AVT/DTMF (8 kHz)"); +WEBRTC_DEFINE_int(avt_16, 114, "RTP payload type for AVT/DTMF (16 kHz)"); +WEBRTC_DEFINE_int(avt_32, 115, "RTP payload type for AVT/DTMF (32 kHz)"); +WEBRTC_DEFINE_int(avt_48, 116, "RTP payload type for AVT/DTMF (48 kHz)"); +WEBRTC_DEFINE_int(red, 117, "RTP payload type for redundant audio (RED)"); +WEBRTC_DEFINE_int(cn_nb, 13, "RTP payload type for comfort noise (8 kHz)"); +WEBRTC_DEFINE_int(cn_wb, 98, "RTP payload type for comfort noise (16 kHz)"); +WEBRTC_DEFINE_int(cn_swb32, 99, "RTP payload type for comfort noise (32 kHz)"); +WEBRTC_DEFINE_int(cn_swb48, 100, "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, 1, "Extension ID for audio level (RFC 6464)"); +WEBRTC_DEFINE_int(abs_send_time, 3, "Extension ID for absolute sender time"); +WEBRTC_DEFINE_int(transport_seq_no, + 5, + "Extension ID for transport sequence number"); +WEBRTC_DEFINE_int(video_content_type, 7, "Extension ID for video content type"); +WEBRTC_DEFINE_int(video_timing, 8, "Extension ID for video timing"); +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(concealment_events, false, "Prints concealment events"); +WEBRTC_DEFINE_int(max_nr_packets_in_buffer, + 50, + "Maximum allowed number of packets in the buffer"); // Maps a codec type to a printable name string. std::string CodecName(NetEqDecoder codec) { diff --git a/modules/audio_coding/neteq/tools/rtp_analyze.cc b/modules/audio_coding/neteq/tools/rtp_analyze.cc index f9390381cd..9d3041ea66 100644 --- a/modules/audio_coding/neteq/tools/rtp_analyze.cc +++ b/modules/audio_coding/neteq/tools/rtp_analyze.cc @@ -19,16 +19,16 @@ #include "rtc_base/flags.h" // Define command line flags. -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"); +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"); int main(int argc, char* argv[]) { std::string program_name = argv[0]; diff --git a/modules/audio_coding/neteq/tools/rtp_encode.cc b/modules/audio_coding/neteq/tools/rtp_encode.cc index 5065ca1675..f48b04d113 100644 --- a/modules/audio_coding/neteq/tools/rtp_encode.cc +++ b/modules/audio_coding/neteq/tools/rtp_encode.cc @@ -40,20 +40,24 @@ namespace test { namespace { // Define command line flags. -DEFINE_bool(list_codecs, false, "Enumerate all codecs"); -DEFINE_string(codec, "opus", "Codec to use"); -DEFINE_int(frame_len, 0, "Frame length in ms; 0 indicates codec default value"); -DEFINE_int(bitrate, 0, "Bitrate in kbps; 0 indicates codec default value"); -DEFINE_int(payload_type, - -1, - "RTP payload type; -1 indicates codec default value"); -DEFINE_int(cng_payload_type, - -1, - "RTP payload type for CNG; -1 indicates default value"); -DEFINE_int(ssrc, 0, "SSRC to write to the RTP header"); -DEFINE_bool(dtx, false, "Use DTX/CNG"); -DEFINE_int(sample_rate, 48000, "Sample rate of the input file"); -DEFINE_bool(help, false, "Print this message"); +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 { diff --git a/modules/audio_coding/neteq/tools/rtp_jitter.cc b/modules/audio_coding/neteq/tools/rtp_jitter.cc index 3c49443036..92a7a8d215 100644 --- a/modules/audio_coding/neteq/tools/rtp_jitter.cc +++ b/modules/audio_coding/neteq/tools/rtp_jitter.cc @@ -23,7 +23,7 @@ namespace webrtc { namespace test { namespace { -DEFINE_bool(help, false, "Print help message"); +WEBRTC_DEFINE_bool(help, false, "Print help message"); constexpr size_t kRtpDumpHeaderLength = 8; diff --git a/modules/audio_mixer/audio_mixer_test.cc b/modules/audio_mixer/audio_mixer_test.cc index 2004aeef46..b60787755d 100644 --- a/modules/audio_mixer/audio_mixer_test.cc +++ b/modules/audio_mixer/audio_mixer_test.cc @@ -19,24 +19,25 @@ #include "rtc_base/flags.h" #include "rtc_base/strings/string_builder.h" -DEFINE_bool(help, false, "Prints this message"); -DEFINE_int(sampling_rate, - 16000, - "Rate at which to mix (all input streams must have this rate)"); +WEBRTC_DEFINE_bool(help, false, "Prints this message"); +WEBRTC_DEFINE_int( + sampling_rate, + 16000, + "Rate at which to mix (all input streams must have this rate)"); -DEFINE_bool( +WEBRTC_DEFINE_bool( stereo, false, "Enable stereo (interleaved). Inputs need not be as this parameter."); -DEFINE_bool(limiter, true, "Enable limiter."); -DEFINE_string(output_file, - "mixed_file.wav", - "File in which to store the mixed result."); -DEFINE_string(input_file_1, "", "First input. Default none."); -DEFINE_string(input_file_2, "", "Second input. Default none."); -DEFINE_string(input_file_3, "", "Third input. Default none."); -DEFINE_string(input_file_4, "", "Fourth input. Default none."); +WEBRTC_DEFINE_bool(limiter, true, "Enable limiter."); +WEBRTC_DEFINE_string(output_file, + "mixed_file.wav", + "File in which to store the mixed result."); +WEBRTC_DEFINE_string(input_file_1, "", "First input. Default none."); +WEBRTC_DEFINE_string(input_file_2, "", "Second input. Default none."); +WEBRTC_DEFINE_string(input_file_3, "", "Third input. Default none."); +WEBRTC_DEFINE_string(input_file_4, "", "Fourth input. Default none."); namespace webrtc { namespace test { diff --git a/modules/audio_processing/agc2/rnn_vad/rnn_vad_tool.cc b/modules/audio_processing/agc2/rnn_vad/rnn_vad_tool.cc index 5fba0bfac8..b66dfd684d 100644 --- a/modules/audio_processing/agc2/rnn_vad/rnn_vad_tool.cc +++ b/modules/audio_processing/agc2/rnn_vad/rnn_vad_tool.cc @@ -25,22 +25,22 @@ namespace rnn_vad { namespace test { namespace { -DEFINE_string(i, "", "Path to the input wav file"); +WEBRTC_DEFINE_string(i, "", "Path to the input wav file"); std::string InputWavFile() { return static_cast(FLAG_i); } -DEFINE_string(f, "", "Path to the output features file"); +WEBRTC_DEFINE_string(f, "", "Path to the output features file"); std::string OutputFeaturesFile() { return static_cast(FLAG_f); } -DEFINE_string(o, "", "Path to the output VAD probabilities file"); +WEBRTC_DEFINE_string(o, "", "Path to the output VAD probabilities file"); std::string OutputVadProbsFile() { return static_cast(FLAG_o); } -DEFINE_bool(help, false, "Prints this message"); +WEBRTC_DEFINE_bool(help, false, "Prints this message"); } // namespace diff --git a/modules/audio_processing/test/audioproc_float_impl.cc b/modules/audio_processing/test/audioproc_float_impl.cc index 3a448877d8..e3f5004637 100644 --- a/modules/audio_processing/test/audioproc_float_impl.cc +++ b/modules/audio_processing/test/audioproc_float_impl.cc @@ -37,155 +37,170 @@ const char kUsageDescription[] = "processing module, either based on wav files or " "protobuf debug dump recordings.\n"; -DEFINE_string(dump_input, "", "Aec dump input filename"); -DEFINE_string(dump_output, "", "Aec dump output filename"); -DEFINE_string(i, "", "Forward stream input wav filename"); -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_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_bool(fixed_interface, - false, - "Use the fixed interface when operating on wav files"); -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_int(agc, - kParameterNotSpecifiedValue, - "Activate (1) or deactivate(0) the AGC"); -DEFINE_int(agc2, - kParameterNotSpecifiedValue, - "Activate (1) or deactivate(0) the AGC2"); -DEFINE_int(pre_amplifier, - kParameterNotSpecifiedValue, - "Activate (1) or deactivate(0) the pre amplifier"); -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(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_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(aec3, - kParameterNotSpecifiedValue, - "Activate (1) or deactivate(0) the experimental AEC mode AEC3"); -DEFINE_int(experimental_agc, - kParameterNotSpecifiedValue, - "Activate (1) or deactivate(0) the experimental AGC"); -DEFINE_int(experimental_agc_disable_digital_adaptive, - kParameterNotSpecifiedValue, - "Force-deactivate (1) digital adaptation in " - "experimental AGC. Digital adaptation is active by default (0)."); -DEFINE_int(experimental_agc_analyze_before_aec, - kParameterNotSpecifiedValue, - "Make level estimation happen before AEC" - " in the experimental AGC. After AEC is the default (0)"); -DEFINE_int( +WEBRTC_DEFINE_string(dump_input, "", "Aec dump input filename"); +WEBRTC_DEFINE_string(dump_output, "", "Aec dump output filename"); +WEBRTC_DEFINE_string(i, "", "Forward stream input wav filename"); +WEBRTC_DEFINE_string(o, "", "Forward stream output wav filename"); +WEBRTC_DEFINE_string(ri, "", "Reverse stream input wav filename"); +WEBRTC_DEFINE_string(ro, "", "Reverse stream output wav filename"); +WEBRTC_DEFINE_string(artificial_nearend, "", "Artificial nearend wav filename"); +WEBRTC_DEFINE_int(output_num_channels, + kParameterNotSpecifiedValue, + "Number of forward stream output channels"); +WEBRTC_DEFINE_int(reverse_output_num_channels, + kParameterNotSpecifiedValue, + "Number of Reverse stream output channels"); +WEBRTC_DEFINE_int(output_sample_rate_hz, + kParameterNotSpecifiedValue, + "Forward stream output sample rate in Hz"); +WEBRTC_DEFINE_int(reverse_output_sample_rate_hz, + kParameterNotSpecifiedValue, + "Reverse stream output sample rate in Hz"); +WEBRTC_DEFINE_bool(fixed_interface, + false, + "Use the fixed interface when operating on wav files"); +WEBRTC_DEFINE_int(aec, + kParameterNotSpecifiedValue, + "Activate (1) or deactivate(0) the echo canceller"); +WEBRTC_DEFINE_int(aecm, + kParameterNotSpecifiedValue, + "Activate (1) or deactivate(0) the mobile echo controller"); +WEBRTC_DEFINE_int(ed, + kParameterNotSpecifiedValue, + "Activate (1) or deactivate (0) the residual echo detector"); +WEBRTC_DEFINE_string(ed_graph, + "", + "Output filename for graph of echo likelihood"); +WEBRTC_DEFINE_int(agc, + kParameterNotSpecifiedValue, + "Activate (1) or deactivate(0) the AGC"); +WEBRTC_DEFINE_int(agc2, + kParameterNotSpecifiedValue, + "Activate (1) or deactivate(0) the AGC2"); +WEBRTC_DEFINE_int(pre_amplifier, + kParameterNotSpecifiedValue, + "Activate (1) or deactivate(0) the pre amplifier"); +WEBRTC_DEFINE_int(hpf, + kParameterNotSpecifiedValue, + "Activate (1) or deactivate(0) the high-pass filter"); +WEBRTC_DEFINE_int(ns, + kParameterNotSpecifiedValue, + "Activate (1) or deactivate(0) the noise suppressor"); +WEBRTC_DEFINE_int(ts, + kParameterNotSpecifiedValue, + "Activate (1) or deactivate(0) the transient suppressor"); +WEBRTC_DEFINE_int(vad, + kParameterNotSpecifiedValue, + "Activate (1) or deactivate(0) the voice activity detector"); +WEBRTC_DEFINE_int(le, + kParameterNotSpecifiedValue, + "Activate (1) or deactivate(0) the level estimator"); +WEBRTC_DEFINE_bool( + all_default, + false, + "Activate all of the default components (will be overridden by any " + "other settings)"); +WEBRTC_DEFINE_int(aec_suppression_level, + kParameterNotSpecifiedValue, + "Set the aec suppression level (0-2)"); +WEBRTC_DEFINE_int(delay_agnostic, + kParameterNotSpecifiedValue, + "Activate (1) or deactivate(0) the AEC delay agnostic mode"); +WEBRTC_DEFINE_int(extended_filter, + kParameterNotSpecifiedValue, + "Activate (1) or deactivate(0) the AEC extended filter mode"); +WEBRTC_DEFINE_int( + aec3, + kParameterNotSpecifiedValue, + "Activate (1) or deactivate(0) the experimental AEC mode AEC3"); +WEBRTC_DEFINE_int(experimental_agc, + kParameterNotSpecifiedValue, + "Activate (1) or deactivate(0) the experimental AGC"); +WEBRTC_DEFINE_int( + experimental_agc_disable_digital_adaptive, + kParameterNotSpecifiedValue, + "Force-deactivate (1) digital adaptation in " + "experimental AGC. Digital adaptation is active by default (0)."); +WEBRTC_DEFINE_int(experimental_agc_analyze_before_aec, + kParameterNotSpecifiedValue, + "Make level estimation happen before AEC" + " in the experimental AGC. After AEC is the default (0)"); +WEBRTC_DEFINE_int( experimental_agc_agc2_level_estimator, kParameterNotSpecifiedValue, "AGC2 level estimation" " in the experimental AGC. AGC1 level estimation is the default (0)"); -DEFINE_int( +WEBRTC_DEFINE_int( refined_adaptive_filter, kParameterNotSpecifiedValue, "Activate (1) or deactivate(0) the refined adaptive filter functionality"); -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_float(agc2_enable_adaptive_gain, - kParameterNotSpecifiedValue, - "Activate (1) or deactivate(0) the AGC2 adaptive gain"); -DEFINE_float(agc2_fixed_gain_db, 0.f, "AGC2 fixed gain (dB) to apply"); -DEFINE_float(pre_amplifier_gain_factor, - 1.f, - "Pre-amplifier gain factor (linear) to apply"); -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(use_stream_delay, - kParameterNotSpecifiedValue, - "Activate (1) or deactivate(0) reporting the stream delay"); -DEFINE_int(stream_drift_samples, - kParameterNotSpecifiedValue, - "Specify the number of stream drift samples to use"); -DEFINE_int(initial_mic_level, 100, "Initial mic level (0-255)"); -DEFINE_int(simulate_mic_gain, - 0, - "Activate (1) or deactivate(0) the analog mic gain simulation"); -DEFINE_int(simulated_mic_kind, - kParameterNotSpecifiedValue, - "Specify which microphone kind to use for microphone simulation"); -DEFINE_bool(performance_report, false, "Report the APM performance "); -DEFINE_bool(verbose, false, "Produce verbose output"); -DEFINE_bool(quiet, false, "Avoid producing information about the progress."); -DEFINE_bool(bitexactness_report, - false, - "Report bitexactness for aec dump result reproduction"); -DEFINE_bool(discard_settings_in_aecdump, - false, - "Discard any config settings specified in the aec dump"); -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(print_aec3_parameter_values, - false, - "Print parameter values used in AEC3 in JSON-format"); -DEFINE_string(aec3_settings, - "", - "File in JSON-format with custom AEC3 settings"); -DEFINE_bool(help, false, "Print this message"); +WEBRTC_DEFINE_int(agc_mode, + kParameterNotSpecifiedValue, + "Specify the AGC mode (0-2)"); +WEBRTC_DEFINE_int(agc_target_level, + kParameterNotSpecifiedValue, + "Specify the AGC target level (0-31)"); +WEBRTC_DEFINE_int(agc_limiter, + kParameterNotSpecifiedValue, + "Activate (1) or deactivate(0) the level estimator"); +WEBRTC_DEFINE_int(agc_compression_gain, + kParameterNotSpecifiedValue, + "Specify the AGC compression gain (0-90)"); +WEBRTC_DEFINE_float(agc2_enable_adaptive_gain, + kParameterNotSpecifiedValue, + "Activate (1) or deactivate(0) the AGC2 adaptive gain"); +WEBRTC_DEFINE_float(agc2_fixed_gain_db, 0.f, "AGC2 fixed gain (dB) to apply"); +WEBRTC_DEFINE_float(pre_amplifier_gain_factor, + 1.f, + "Pre-amplifier gain factor (linear) to apply"); +WEBRTC_DEFINE_int(vad_likelihood, + kParameterNotSpecifiedValue, + "Specify the VAD likelihood (0-3)"); +WEBRTC_DEFINE_int(ns_level, + kParameterNotSpecifiedValue, + "Specify the NS level (0-3)"); +WEBRTC_DEFINE_int(stream_delay, + kParameterNotSpecifiedValue, + "Specify the stream delay in ms to use"); +WEBRTC_DEFINE_int(use_stream_delay, + kParameterNotSpecifiedValue, + "Activate (1) or deactivate(0) reporting the stream delay"); +WEBRTC_DEFINE_int(stream_drift_samples, + kParameterNotSpecifiedValue, + "Specify the number of stream drift samples to use"); +WEBRTC_DEFINE_int(initial_mic_level, 100, "Initial mic level (0-255)"); +WEBRTC_DEFINE_int( + simulate_mic_gain, + 0, + "Activate (1) or deactivate(0) the analog mic gain simulation"); +WEBRTC_DEFINE_int( + simulated_mic_kind, + kParameterNotSpecifiedValue, + "Specify which microphone kind to use for microphone simulation"); +WEBRTC_DEFINE_bool(performance_report, false, "Report the APM performance "); +WEBRTC_DEFINE_bool(verbose, false, "Produce verbose output"); +WEBRTC_DEFINE_bool(quiet, + false, + "Avoid producing information about the progress."); +WEBRTC_DEFINE_bool(bitexactness_report, + false, + "Report bitexactness for aec dump result reproduction"); +WEBRTC_DEFINE_bool(discard_settings_in_aecdump, + false, + "Discard any config settings specified in the aec dump"); +WEBRTC_DEFINE_bool(store_intermediate_output, + false, + "Creates new output files after each init"); +WEBRTC_DEFINE_string(custom_call_order_file, + "", + "Custom process API call order file"); +WEBRTC_DEFINE_bool(print_aec3_parameter_values, + false, + "Print parameter values used in AEC3 in JSON-format"); +WEBRTC_DEFINE_string(aec3_settings, + "", + "File in JSON-format with custom AEC3 settings"); +WEBRTC_DEFINE_bool(help, false, "Print this message"); void SetSettingIfSpecified(const std::string& value, absl::optional* parameter) { diff --git a/modules/audio_processing/test/conversational_speech/generator.cc b/modules/audio_processing/test/conversational_speech/generator.cc index 741a1ca86b..fa561cf8e0 100644 --- a/modules/audio_processing/test/conversational_speech/generator.cc +++ b/modules/audio_processing/test/conversational_speech/generator.cc @@ -32,10 +32,10 @@ const char kUsageDescription[] = "Command-line tool to generate multiple-end audio tracks to simulate " "conversational speech with two or more participants.\n"; -DEFINE_string(i, "", "Directory containing the speech turn wav files"); -DEFINE_string(t, "", "Path to the timing text file"); -DEFINE_string(o, "", "Output wav files destination path"); -DEFINE_bool(help, false, "Prints this message"); +WEBRTC_DEFINE_string(i, "", "Directory containing the speech turn wav files"); +WEBRTC_DEFINE_string(t, "", "Path to the timing text file"); +WEBRTC_DEFINE_string(o, "", "Output wav files destination path"); +WEBRTC_DEFINE_bool(help, false, "Prints this message"); } // namespace diff --git a/modules/audio_processing/test/py_quality_assessment/quality_assessment/apm_vad.cc b/modules/audio_processing/test/py_quality_assessment/quality_assessment/apm_vad.cc index a6184b5f9f..4b6ada27b6 100644 --- a/modules/audio_processing/test/py_quality_assessment/quality_assessment/apm_vad.cc +++ b/modules/audio_processing/test/py_quality_assessment/quality_assessment/apm_vad.cc @@ -24,9 +24,9 @@ constexpr int kMaxSampleRate = 48000; constexpr size_t kMaxFrameLen = kAudioFrameLengthMilliseconds * kMaxSampleRate / 1000; -DEFINE_string(i, "", "Input wav file"); -DEFINE_string(o_probs, "", "VAD probabilities output file"); -DEFINE_string(o_rms, "", "VAD output file"); +WEBRTC_DEFINE_string(i, "", "Input wav file"); +WEBRTC_DEFINE_string(o_probs, "", "VAD probabilities output file"); +WEBRTC_DEFINE_string(o_rms, "", "VAD output file"); int main(int argc, char* argv[]) { if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true)) diff --git a/modules/audio_processing/test/py_quality_assessment/quality_assessment/sound_level.cc b/modules/audio_processing/test/py_quality_assessment/quality_assessment/sound_level.cc index 98cf84c888..35a2c11aeb 100644 --- a/modules/audio_processing/test/py_quality_assessment/quality_assessment/sound_level.cc +++ b/modules/audio_processing/test/py_quality_assessment/quality_assessment/sound_level.cc @@ -26,13 +26,13 @@ constexpr size_t kMaxFrameLen = kMaxFrameLenMs * kMaxSampleRate / 1000; const double kOneDbReduction = DbToRatio(-1.0); -DEFINE_string(i, "", "Input wav file"); -DEFINE_string(oc, "", "Config output file"); -DEFINE_string(ol, "", "Levels output file"); -DEFINE_float(a, 5.f, "Attack (ms)"); -DEFINE_float(d, 20.f, "Decay (ms)"); -DEFINE_int(f, 10, "Frame length (ms)"); -DEFINE_bool(help, false, "prints this message"); +WEBRTC_DEFINE_string(i, "", "Input wav file"); +WEBRTC_DEFINE_string(oc, "", "Config output file"); +WEBRTC_DEFINE_string(ol, "", "Levels output file"); +WEBRTC_DEFINE_float(a, 5.f, "Attack (ms)"); +WEBRTC_DEFINE_float(d, 20.f, "Decay (ms)"); +WEBRTC_DEFINE_int(f, 10, "Frame length (ms)"); +WEBRTC_DEFINE_bool(help, false, "prints this message"); int main(int argc, char* argv[]) { if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true)) { diff --git a/modules/audio_processing/test/py_quality_assessment/quality_assessment/vad.cc b/modules/audio_processing/test/py_quality_assessment/quality_assessment/vad.cc index 191cb1e9fc..8a134ed185 100644 --- a/modules/audio_processing/test/py_quality_assessment/quality_assessment/vad.cc +++ b/modules/audio_processing/test/py_quality_assessment/quality_assessment/vad.cc @@ -27,8 +27,8 @@ constexpr size_t kMaxFrameLen = constexpr uint8_t kBitmaskBuffSize = 8; -DEFINE_string(i, "", "Input wav file"); -DEFINE_string(o, "", "VAD output file"); +WEBRTC_DEFINE_string(i, "", "Input wav file"); +WEBRTC_DEFINE_string(o, "", "VAD output file"); int main(int argc, char* argv[]) { if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true)) diff --git a/modules/audio_processing/transient/transient_suppression_test.cc b/modules/audio_processing/transient/transient_suppression_test.cc index 9e7ecd5786..e15f69c042 100644 --- a/modules/audio_processing/transient/transient_suppression_test.cc +++ b/modules/audio_processing/transient/transient_suppression_test.cc @@ -23,26 +23,28 @@ #include "test/gtest.h" #include "test/testsupport/fileutils.h" -DEFINE_string(in_file_name, "", "PCM file that contains the signal."); -DEFINE_string(detection_file_name, - "", - "PCM file that contains the detection signal."); -DEFINE_string(reference_file_name, - "", - "PCM file that contains the reference signal."); +WEBRTC_DEFINE_string(in_file_name, "", "PCM file that contains the signal."); +WEBRTC_DEFINE_string(detection_file_name, + "", + "PCM file that contains the detection signal."); +WEBRTC_DEFINE_string(reference_file_name, + "", + "PCM file that contains the reference signal."); -DEFINE_int(chunk_size_ms, - 10, - "Time between each chunk of samples in milliseconds."); +WEBRTC_DEFINE_int(chunk_size_ms, + 10, + "Time between each chunk of samples in milliseconds."); -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."); +WEBRTC_DEFINE_int(sample_rate_hz, + 16000, + "Sampling frequency of the signal in Hertz."); +WEBRTC_DEFINE_int(detection_rate_hz, + 0, + "Sampling frequency of the detection signal in Hertz."); -DEFINE_int(num_channels, 1, "Number of channels."); +WEBRTC_DEFINE_int(num_channels, 1, "Number of channels."); -DEFINE_bool(help, false, "Print this message."); +WEBRTC_DEFINE_bool(help, false, "Print this message."); namespace webrtc { diff --git a/modules/remote_bitrate_estimator/tools/bwe_rtp.cc b/modules/remote_bitrate_estimator/tools/bwe_rtp.cc index 0230db17dc..d7cdb54d92 100644 --- a/modules/remote_bitrate_estimator/tools/bwe_rtp.cc +++ b/modules/remote_bitrate_estimator/tools/bwe_rtp.cc @@ -24,28 +24,30 @@ namespace flags { -DEFINE_string(extension_type, - "abs", - "Extension type, either abs for absolute send time or tsoffset " - "for timestamp offset."); +WEBRTC_DEFINE_string( + extension_type, + "abs", + "Extension type, either abs for absolute send time or tsoffset " + "for timestamp offset."); std::string ExtensionType() { return static_cast(FLAG_extension_type); } -DEFINE_int(extension_id, 3, "Extension id."); +WEBRTC_DEFINE_int(extension_id, 3, "Extension id."); int ExtensionId() { return static_cast(FLAG_extension_id); } -DEFINE_string(input_file, "", "Input file."); +WEBRTC_DEFINE_string(input_file, "", "Input file."); std::string InputFile() { return static_cast(FLAG_input_file); } -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)."); +WEBRTC_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 SsrcFilter() { std::string ssrc_filter_string = static_cast(FLAG_ssrc_filter); if (ssrc_filter_string.empty()) @@ -64,7 +66,7 @@ std::set SsrcFilter() { return ssrcs; } -DEFINE_bool(help, false, "Print this message."); +WEBRTC_DEFINE_bool(help, false, "Print this message."); } // namespace flags bool ParseArgsAndSetupEstimator(int argc, diff --git a/rtc_base/flags.h b/rtc_base/flags.h index 1c476d80be..a08bfd29b8 100644 --- a/rtc_base/flags.h +++ b/rtc_base/flags.h @@ -36,7 +36,7 @@ union FlagValue { // bool values ('bool b = "false";' results in b == true!), we pass // and int argument to New_BOOL as this appears to be safer - sigh. // In particular, it prevents the (not uncommon!) bug where a bool - // flag is defined via: DEFINE_bool(flag, "false", "some comment");. + // flag is defined via: WEBRTC_DEFINE_bool(flag, "false", "some comment");. static FlagValue New_BOOL(int b) { FlagValue v; v.b = (b != 0); @@ -155,7 +155,7 @@ class Flag { }; // Internal use only. -#define DEFINE_FLAG(type, c_type, name, default, comment) \ +#define WEBRTC_DEFINE_FLAG(type, c_type, name, default, comment) \ /* define and initialize the flag */ \ c_type FLAG_##name = (default); \ /* register the flag */ \ @@ -164,25 +164,25 @@ class Flag { rtc::FlagValue::New_##type(default)) // Internal use only. -#define DECLARE_FLAG(c_type, name) \ - /* declare the external flag */ \ +#define WEBRTC_DECLARE_FLAG(c_type, name) \ + /* declare the external flag */ \ extern c_type FLAG_##name // Use the following macros to define a new flag: -#define DEFINE_bool(name, default, comment) \ - DEFINE_FLAG(BOOL, bool, name, default, comment) -#define DEFINE_int(name, default, comment) \ - DEFINE_FLAG(INT, int, name, default, comment) -#define DEFINE_float(name, default, comment) \ - DEFINE_FLAG(FLOAT, double, name, default, comment) -#define DEFINE_string(name, default, comment) \ - DEFINE_FLAG(STRING, const char*, name, default, comment) +#define WEBRTC_DEFINE_bool(name, default, comment) \ + WEBRTC_DEFINE_FLAG(BOOL, bool, name, default, comment) +#define WEBRTC_DEFINE_int(name, default, comment) \ + WEBRTC_DEFINE_FLAG(INT, int, name, default, comment) +#define WEBRTC_DEFINE_float(name, default, comment) \ + WEBRTC_DEFINE_FLAG(FLOAT, double, name, default, comment) +#define WEBRTC_DEFINE_string(name, default, comment) \ + WEBRTC_DEFINE_FLAG(STRING, const char*, name, default, comment) // Use the following macros to declare a flag defined elsewhere: -#define DECLARE_bool(name) DECLARE_FLAG(bool, name) -#define DECLARE_int(name) DECLARE_FLAG(int, name) -#define DECLARE_float(name) DECLARE_FLAG(double, name) -#define DECLARE_string(name) DECLARE_FLAG(const char*, name) +#define WEBRTC_DECLARE_bool(name) WEBRTC_DECLARE_FLAG(bool, name) +#define WEBRTC_DECLARE_int(name) WEBRTC_DECLARE_FLAG(int, name) +#define WEBRTC_DECLARE_float(name) WEBRTC_DECLARE_FLAG(double, name) +#define WEBRTC_DECLARE_string(name) WEBRTC_DECLARE_FLAG(const char*, name) // The global list of all flags. class FlagList { diff --git a/rtc_base/unittest_main.cc b/rtc_base/unittest_main.cc index 98b8a4879f..28a1bbbc09 100644 --- a/rtc_base/unittest_main.cc +++ b/rtc_base/unittest_main.cc @@ -31,19 +31,20 @@ #include "test/ios/test_support.h" #endif -DEFINE_bool(help, false, "prints this message"); -DEFINE_string(log, "", "logging options to use"); -DEFINE_string( +WEBRTC_DEFINE_bool(help, false, "prints this message"); +WEBRTC_DEFINE_string(log, "", "logging options to use"); +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."); #if defined(WEBRTC_WIN) -DEFINE_int(crt_break_alloc, -1, "memory allocation to break on"); -DEFINE_bool(default_error_handlers, - false, - "leave the default exception/dbg handler functions in place"); +WEBRTC_DEFINE_int(crt_break_alloc, -1, "memory allocation to break on"); +WEBRTC_DEFINE_bool( + default_error_handlers, + false, + "leave the default exception/dbg handler functions in place"); void TestInvalidParameterHandler(const wchar_t* expression, const wchar_t* function, diff --git a/rtc_tools/agc/activity_metric.cc b/rtc_tools/agc/activity_metric.cc index b4ed3facb3..9b2276f16f 100644 --- a/rtc_tools/agc/activity_metric.cc +++ b/rtc_tools/agc/activity_metric.cc @@ -30,32 +30,34 @@ static const int kAgcAnalWindowSamples = 100; static const float kDefaultActivityThreshold = 0.3f; -DEFINE_bool(standalone_vad, true, "enable stand-alone VAD"); -DEFINE_string(true_vad, - "", - "name of a file containing true VAD in 'int'" - " format"); -DEFINE_string(video_vad, - "", - "name of a file containing video VAD (activity" - " probabilities) in double format. One activity per 10ms is" - " required. If no file is given the video information is not" - " incorporated. Negative activity is interpreted as video is" - " not adapted and the statistics are not computed during" - " the learning phase. Note that the negative video activities" - " are ONLY allowed at the beginning."); -DEFINE_string(result, - "", - "name of a file to write the results. The results" - " will be appended to the end of the file. This is optional."); -DEFINE_string(audio_content, - "", - "name of a file where audio content is written" - " to, in double format."); -DEFINE_float(activity_threshold, - kDefaultActivityThreshold, - "Activity threshold"); -DEFINE_bool(help, false, "prints this message"); +WEBRTC_DEFINE_bool(standalone_vad, true, "enable stand-alone VAD"); +WEBRTC_DEFINE_string(true_vad, + "", + "name of a file containing true VAD in 'int'" + " format"); +WEBRTC_DEFINE_string( + video_vad, + "", + "name of a file containing video VAD (activity" + " probabilities) in double format. One activity per 10ms is" + " required. If no file is given the video information is not" + " incorporated. Negative activity is interpreted as video is" + " not adapted and the statistics are not computed during" + " the learning phase. Note that the negative video activities" + " are ONLY allowed at the beginning."); +WEBRTC_DEFINE_string( + result, + "", + "name of a file to write the results. The results" + " will be appended to the end of the file. This is optional."); +WEBRTC_DEFINE_string(audio_content, + "", + "name of a file where audio content is written" + " to, in double format."); +WEBRTC_DEFINE_float(activity_threshold, + kDefaultActivityThreshold, + "Activity threshold"); +WEBRTC_DEFINE_bool(help, false, "prints this message"); namespace webrtc { diff --git a/rtc_tools/event_log_visualizer/main.cc b/rtc_tools/event_log_visualizer/main.cc index 2a52d064bb..ad90cfbacb 100644 --- a/rtc_tools/event_log_visualizer/main.cc +++ b/rtc_tools/event_log_visualizer/main.cc @@ -20,151 +20,173 @@ #include "test/field_trial.h" #include "test/testsupport/fileutils.h" -DEFINE_string(plot_profile, - "default", - "A profile that selects a certain subset of the plots. Currently " - "defined profiles are \"all\", \"none\", \"sendside_bwe\"," - "\"receiveside_bwe\" and \"default\""); +WEBRTC_DEFINE_string( + plot_profile, + "default", + "A profile that selects a certain subset of the plots. Currently " + "defined profiles are \"all\", \"none\", \"sendside_bwe\"," + "\"receiveside_bwe\" and \"default\""); -DEFINE_bool(plot_incoming_packet_sizes, - false, - "Plot bar graph showing the size of each incoming packet."); -DEFINE_bool(plot_outgoing_packet_sizes, - false, - "Plot bar graph showing the size of each outgoing packet."); -DEFINE_bool(plot_incoming_packet_count, - false, - "Plot the accumulated number of packets for each incoming stream."); -DEFINE_bool(plot_outgoing_packet_count, - false, - "Plot the accumulated number of packets for each outgoing stream."); -DEFINE_bool(plot_audio_playout, - false, - "Plot bar graph showing the time between each audio playout."); -DEFINE_bool(plot_audio_level, - false, - "Plot line graph showing the audio level of incoming audio."); -DEFINE_bool(plot_incoming_sequence_number_delta, - false, - "Plot the sequence number difference between consecutive incoming " - "packets."); -DEFINE_bool( +WEBRTC_DEFINE_bool(plot_incoming_packet_sizes, + false, + "Plot bar graph showing the size of each incoming packet."); +WEBRTC_DEFINE_bool(plot_outgoing_packet_sizes, + false, + "Plot bar graph showing the size of each outgoing packet."); +WEBRTC_DEFINE_bool( + plot_incoming_packet_count, + false, + "Plot the accumulated number of packets for each incoming stream."); +WEBRTC_DEFINE_bool( + plot_outgoing_packet_count, + false, + "Plot the accumulated number of packets for each outgoing stream."); +WEBRTC_DEFINE_bool( + plot_audio_playout, + false, + "Plot bar graph showing the time between each audio playout."); +WEBRTC_DEFINE_bool( + plot_audio_level, + false, + "Plot line graph showing the audio level of incoming audio."); +WEBRTC_DEFINE_bool( + plot_incoming_sequence_number_delta, + false, + "Plot the sequence number difference between consecutive incoming " + "packets."); +WEBRTC_DEFINE_bool( plot_incoming_delay_delta, false, "Plot the difference in 1-way path delay between consecutive packets."); -DEFINE_bool(plot_incoming_delay, - true, - "Plot the 1-way path delay for incoming packets, normalized so " - "that the first packet has delay 0."); -DEFINE_bool(plot_incoming_loss_rate, - true, - "Compute the loss rate for incoming packets using a method that's " - "similar to the one used for RTCP SR and RR fraction lost. Note " - "that the loss rate can be negative if packets are duplicated or " - "reordered."); -DEFINE_bool(plot_incoming_bitrate, - true, - "Plot the total bitrate used by all incoming streams."); -DEFINE_bool(plot_outgoing_bitrate, - true, - "Plot the total bitrate used by all outgoing streams."); -DEFINE_bool(plot_incoming_stream_bitrate, - true, - "Plot the bitrate used by each incoming stream."); -DEFINE_bool(plot_outgoing_stream_bitrate, - true, - "Plot the bitrate used by each outgoing stream."); -DEFINE_bool(plot_simulated_receiveside_bwe, - false, - "Run the receive-side bandwidth estimator with the incoming rtp " - "packets and plot the resulting estimate."); -DEFINE_bool(plot_simulated_sendside_bwe, - false, - "Run the send-side bandwidth estimator with the outgoing rtp and " - "incoming rtcp and plot the resulting estimate."); -DEFINE_bool(plot_network_delay_feedback, - true, - "Compute network delay based on sent packets and the received " - "transport feedback."); -DEFINE_bool(plot_fraction_loss_feedback, - true, - "Plot packet loss in percent for outgoing packets (as perceived by " - "the send-side bandwidth estimator)."); -DEFINE_bool(plot_pacer_delay, - false, - "Plot the time each sent packet has spent in the pacer (based on " - "the difference between the RTP timestamp and the send " - "timestamp)."); -DEFINE_bool(plot_timestamps, - false, - "Plot the rtp timestamps of all rtp and rtcp packets over time."); -DEFINE_bool(plot_rtcp_details, - false, - "Plot the contents of all report blocks in all sender and receiver " - "reports. This includes fraction lost, cumulative number of lost " - "packets, extended highest sequence number and time since last " - "received SR."); -DEFINE_bool(plot_audio_encoder_bitrate_bps, - false, - "Plot the audio encoder target bitrate."); -DEFINE_bool(plot_audio_encoder_frame_length_ms, - false, - "Plot the audio encoder frame length."); -DEFINE_bool( +WEBRTC_DEFINE_bool( + plot_incoming_delay, + true, + "Plot the 1-way path delay for incoming packets, normalized so " + "that the first packet has delay 0."); +WEBRTC_DEFINE_bool( + plot_incoming_loss_rate, + true, + "Compute the loss rate for incoming packets using a method that's " + "similar to the one used for RTCP SR and RR fraction lost. Note " + "that the loss rate can be negative if packets are duplicated or " + "reordered."); +WEBRTC_DEFINE_bool(plot_incoming_bitrate, + true, + "Plot the total bitrate used by all incoming streams."); +WEBRTC_DEFINE_bool(plot_outgoing_bitrate, + true, + "Plot the total bitrate used by all outgoing streams."); +WEBRTC_DEFINE_bool(plot_incoming_stream_bitrate, + true, + "Plot the bitrate used by each incoming stream."); +WEBRTC_DEFINE_bool(plot_outgoing_stream_bitrate, + true, + "Plot the bitrate used by each outgoing stream."); +WEBRTC_DEFINE_bool( + plot_simulated_receiveside_bwe, + false, + "Run the receive-side bandwidth estimator with the incoming rtp " + "packets and plot the resulting estimate."); +WEBRTC_DEFINE_bool( + plot_simulated_sendside_bwe, + false, + "Run the send-side bandwidth estimator with the outgoing rtp and " + "incoming rtcp and plot the resulting estimate."); +WEBRTC_DEFINE_bool( + plot_network_delay_feedback, + true, + "Compute network delay based on sent packets and the received " + "transport feedback."); +WEBRTC_DEFINE_bool( + plot_fraction_loss_feedback, + true, + "Plot packet loss in percent for outgoing packets (as perceived by " + "the send-side bandwidth estimator)."); +WEBRTC_DEFINE_bool( + plot_pacer_delay, + false, + "Plot the time each sent packet has spent in the pacer (based on " + "the difference between the RTP timestamp and the send " + "timestamp)."); +WEBRTC_DEFINE_bool( + plot_timestamps, + false, + "Plot the rtp timestamps of all rtp and rtcp packets over time."); +WEBRTC_DEFINE_bool( + plot_rtcp_details, + false, + "Plot the contents of all report blocks in all sender and receiver " + "reports. This includes fraction lost, cumulative number of lost " + "packets, extended highest sequence number and time since last " + "received SR."); +WEBRTC_DEFINE_bool(plot_audio_encoder_bitrate_bps, + false, + "Plot the audio encoder target bitrate."); +WEBRTC_DEFINE_bool(plot_audio_encoder_frame_length_ms, + false, + "Plot the audio encoder frame length."); +WEBRTC_DEFINE_bool( plot_audio_encoder_packet_loss, false, "Plot the uplink packet loss fraction which is sent to the audio encoder."); -DEFINE_bool(plot_audio_encoder_fec, false, "Plot the audio encoder FEC."); -DEFINE_bool(plot_audio_encoder_dtx, false, "Plot the audio encoder DTX."); -DEFINE_bool(plot_audio_encoder_num_channels, - false, - "Plot the audio encoder number of channels."); -DEFINE_bool(plot_neteq_stats, false, "Plot the NetEq statistics."); -DEFINE_bool(plot_ice_candidate_pair_config, - false, - "Plot the ICE candidate pair config events."); -DEFINE_bool(plot_ice_connectivity_check, - false, - "Plot the ICE candidate pair connectivity checks."); +WEBRTC_DEFINE_bool(plot_audio_encoder_fec, + false, + "Plot the audio encoder FEC."); +WEBRTC_DEFINE_bool(plot_audio_encoder_dtx, + false, + "Plot the audio encoder DTX."); +WEBRTC_DEFINE_bool(plot_audio_encoder_num_channels, + false, + "Plot the audio encoder number of channels."); +WEBRTC_DEFINE_bool(plot_neteq_stats, false, "Plot the NetEq statistics."); +WEBRTC_DEFINE_bool(plot_ice_candidate_pair_config, + false, + "Plot the ICE candidate pair config events."); +WEBRTC_DEFINE_bool(plot_ice_connectivity_check, + false, + "Plot the ICE candidate pair connectivity checks."); -DEFINE_string( +WEBRTC_DEFINE_string( force_fieldtrials, "", "Field trials control experimental feature code which can be forced. " "E.g. running with --force_fieldtrials=WebRTC-FooFeature/Enabled/" " will assign the group Enabled to field trial WebRTC-FooFeature. Multiple " "trials are separated by \"/\""); -DEFINE_string(wav_filename, - "", - "Path to wav file used for simulation of jitter buffer"); -DEFINE_bool(help, false, "prints this message"); +WEBRTC_DEFINE_string(wav_filename, + "", + "Path to wav file used for simulation of jitter buffer"); +WEBRTC_DEFINE_bool(help, false, "prints this message"); -DEFINE_bool(show_detector_state, - false, - "Show the state of the delay based BWE detector on the total " - "bitrate graph"); +WEBRTC_DEFINE_bool( + show_detector_state, + false, + "Show the state of the delay based BWE detector on the total " + "bitrate graph"); -DEFINE_bool(show_alr_state, - false, - "Show the state ALR state on the total bitrate graph"); +WEBRTC_DEFINE_bool(show_alr_state, + false, + "Show the state ALR state on the total bitrate graph"); -DEFINE_bool(parse_unconfigured_header_extensions, - true, - "Attempt to parse unconfigured header extensions using the default " - "WebRTC mapping. This can give very misleading results if the " - "application negotiates a different mapping."); +WEBRTC_DEFINE_bool( + parse_unconfigured_header_extensions, + true, + "Attempt to parse unconfigured header extensions using the default " + "WebRTC mapping. This can give very misleading results if the " + "application negotiates a different mapping."); -DEFINE_bool(print_triage_alerts, - false, - "Print triage alerts, i.e. a list of potential problems."); +WEBRTC_DEFINE_bool(print_triage_alerts, + false, + "Print triage alerts, i.e. a list of potential problems."); -DEFINE_bool(normalize_time, - true, - "Normalize the log timestamps so that the call starts at time 0."); +WEBRTC_DEFINE_bool( + normalize_time, + true, + "Normalize the log timestamps so that the call starts at time 0."); -DEFINE_bool(protobuf_output, - false, - "Output charts as protobuf instead of python code."); +WEBRTC_DEFINE_bool(protobuf_output, + false, + "Output charts as protobuf instead of python code."); void SetAllPlotFlags(bool setting); diff --git a/rtc_tools/unpack_aecdump/unpack.cc b/rtc_tools/unpack_aecdump/unpack.cc index af8bf3187a..142b49730a 100644 --- a/rtc_tools/unpack_aecdump/unpack.cc +++ b/rtc_tools/unpack_aecdump/unpack.cc @@ -29,27 +29,34 @@ RTC_PUSH_IGNORING_WUNDEF() RTC_POP_IGNORING_WUNDEF() // TODO(andrew): unpack more of the data. -DEFINE_string(input_file, "input", "The name of the input stream file."); -DEFINE_string(output_file, - "ref_out", - "The name of the reference output stream file."); -DEFINE_string(reverse_file, - "reverse", - "The name of the reverse input stream file."); -DEFINE_string(delay_file, "delay.int32", "The name of the delay file."); -DEFINE_string(drift_file, "drift.int32", "The name of the drift file."); -DEFINE_string(level_file, "level.int32", "The name of the level file."); -DEFINE_string(keypress_file, "keypress.bool", "The name of the keypress file."); -DEFINE_string(callorder_file, - "callorder", - "The name of the render/capture call order file."); -DEFINE_string(settings_file, "settings.txt", "The name of the settings file."); -DEFINE_bool(full, false, "Unpack the full set of files (normally not needed)."); -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."); +WEBRTC_DEFINE_string(input_file, "input", "The name of the input stream file."); +WEBRTC_DEFINE_string(output_file, + "ref_out", + "The name of the reference output stream file."); +WEBRTC_DEFINE_string(reverse_file, + "reverse", + "The name of the reverse input stream file."); +WEBRTC_DEFINE_string(delay_file, "delay.int32", "The name of the delay file."); +WEBRTC_DEFINE_string(drift_file, "drift.int32", "The name of the drift file."); +WEBRTC_DEFINE_string(level_file, "level.int32", "The name of the level file."); +WEBRTC_DEFINE_string(keypress_file, + "keypress.bool", + "The name of the keypress file."); +WEBRTC_DEFINE_string(callorder_file, + "callorder", + "The name of the render/capture call order file."); +WEBRTC_DEFINE_string(settings_file, + "settings.txt", + "The name of the settings file."); +WEBRTC_DEFINE_bool(full, + false, + "Unpack the full set of files (normally not needed)."); +WEBRTC_DEFINE_bool(raw, false, "Write raw data instead of a WAV file."); +WEBRTC_DEFINE_bool( + text, + false, + "Write non-audio files as text files instead of binary files."); +WEBRTC_DEFINE_bool(help, false, "Print this message."); #define PRINT_CONFIG(field_name) \ if (msg.has_##field_name()) { \ diff --git a/test/scenario/scenario.cc b/test/scenario/scenario.cc index 7eb09e8bb7..2395a7a41e 100644 --- a/test/scenario/scenario.cc +++ b/test/scenario/scenario.cc @@ -16,7 +16,7 @@ #include "rtc_base/flags.h" #include "test/testsupport/fileutils.h" -DEFINE_bool(scenario_logs, false, "Save logs from scenario framework."); +WEBRTC_DEFINE_bool(scenario_logs, false, "Save logs from scenario framework."); namespace webrtc { namespace test { diff --git a/test/test_main_lib.cc b/test/test_main_lib.cc index 8e1b213eb2..95144c98fc 100644 --- a/test/test_main_lib.cc +++ b/test/test_main_lib.cc @@ -32,13 +32,13 @@ #if defined(WEBRTC_IOS) #include "test/ios/test_support.h" -DEFINE_string(NSTreatUnknownArgumentsAsOpen, - "", - "Intentionally ignored flag intended for iOS simulator."); -DEFINE_string(ApplePersistenceIgnoreState, - "", - "Intentionally ignored flag intended for iOS simulator."); -DEFINE_bool( +WEBRTC_DEFINE_string(NSTreatUnknownArgumentsAsOpen, + "", + "Intentionally ignored flag intended for iOS simulator."); +WEBRTC_DEFINE_string(ApplePersistenceIgnoreState, + "", + "Intentionally ignored flag intended for iOS simulator."); +WEBRTC_DEFINE_bool( save_chartjson_result, false, "Store the perf results in Documents/perf_result.json in the format " @@ -48,12 +48,12 @@ DEFINE_bool( #else -DEFINE_string( +WEBRTC_DEFINE_string( isolated_script_test_output, "", "Path to output an empty JSON file which Chromium infra requires."); -DEFINE_string( +WEBRTC_DEFINE_string( isolated_script_test_perf_output, "", "Path where the perf results should be stored in the JSON format described " @@ -63,16 +63,16 @@ DEFINE_string( #endif -DEFINE_bool(logs, false, "print logs to stderr"); +WEBRTC_DEFINE_bool(logs, false, "print logs to stderr"); -DEFINE_string( +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."); -DEFINE_bool(help, false, "Print this message."); +WEBRTC_DEFINE_bool(help, false, "Print this message."); namespace webrtc { diff --git a/test/testsupport/test_artifacts.cc b/test/testsupport/test_artifacts.cc index 0f7e0cd4ff..9438cefbc4 100644 --- a/test/testsupport/test_artifacts.cc +++ b/test/testsupport/test_artifacts.cc @@ -24,9 +24,9 @@ const std::string& DefaultArtifactPath() { } } // namespace -DEFINE_string(test_artifacts_dir, - DefaultArtifactPath().c_str(), - "The output folder where test output should be saved."); +WEBRTC_DEFINE_string(test_artifacts_dir, + DefaultArtifactPath().c_str(), + "The output folder where test output should be saved."); namespace webrtc { namespace test { diff --git a/test/testsupport/test_artifacts_unittest.cc b/test/testsupport/test_artifacts_unittest.cc index c423cd90ea..267ea93676 100644 --- a/test/testsupport/test_artifacts_unittest.cc +++ b/test/testsupport/test_artifacts_unittest.cc @@ -21,7 +21,7 @@ #include "test/gtest.h" #include "test/testsupport/fileutils.h" -DECLARE_string(test_artifacts_dir); +WEBRTC_DECLARE_string(test_artifacts_dir); namespace webrtc { namespace test { diff --git a/video/full_stack_tests.cc b/video/full_stack_tests.cc index 21a8e621bc..6562173e87 100644 --- a/video/full_stack_tests.cc +++ b/video/full_stack_tests.cc @@ -22,21 +22,24 @@ namespace webrtc { namespace flags { -DEFINE_string(rtc_event_log_name, - "", - "Filename for rtc event log. Two files " - "with \"_send\" and \"_recv\" suffixes will be created."); +WEBRTC_DEFINE_string(rtc_event_log_name, + "", + "Filename for rtc event log. Two files " + "with \"_send\" and \"_recv\" suffixes will be created."); std::string RtcEventLogName() { return static_cast(FLAG_rtc_event_log_name); } -DEFINE_string(rtp_dump_name, "", "Filename for dumped received RTP stream."); +WEBRTC_DEFINE_string(rtp_dump_name, + "", + "Filename for dumped received RTP stream."); std::string RtpDumpName() { return static_cast(FLAG_rtp_dump_name); } -DEFINE_string(encoded_frame_path, - "", - "The base path for encoded frame logs. Created files will have " - "the form ..(recv|send.).ivf"); +WEBRTC_DEFINE_string( + encoded_frame_path, + "", + "The base path for encoded frame logs. Created files will have " + "the form ..(recv|send.).ivf"); std::string EncodedFramePath() { return static_cast(FLAG_encoded_frame_path); } diff --git a/video/replay.cc b/video/replay.cc index 5c17e5b3b7..a3fd1753b5 100644 --- a/video/replay.cc +++ b/video/replay.cc @@ -72,39 +72,39 @@ namespace flags { // TODO(pbos): Multiple receivers. // Flag for payload type. -DEFINE_int(media_payload_type, - test::CallTest::kPayloadTypeVP8, - "Media payload type"); +WEBRTC_DEFINE_int(media_payload_type, + test::CallTest::kPayloadTypeVP8, + "Media payload type"); static int MediaPayloadType() { return static_cast(FLAG_media_payload_type); } // Flag for RED payload type. -DEFINE_int(red_payload_type, - test::CallTest::kRedPayloadType, - "RED payload type"); +WEBRTC_DEFINE_int(red_payload_type, + test::CallTest::kRedPayloadType, + "RED payload type"); static int RedPayloadType() { return static_cast(FLAG_red_payload_type); } // Flag for ULPFEC payload type. -DEFINE_int(ulpfec_payload_type, - test::CallTest::kUlpfecPayloadType, - "ULPFEC payload type"); +WEBRTC_DEFINE_int(ulpfec_payload_type, + test::CallTest::kUlpfecPayloadType, + "ULPFEC payload type"); static int UlpfecPayloadType() { return static_cast(FLAG_ulpfec_payload_type); } -DEFINE_int(media_payload_type_rtx, - test::CallTest::kSendRtxPayloadType, - "Media over RTX payload type"); +WEBRTC_DEFINE_int(media_payload_type_rtx, + test::CallTest::kSendRtxPayloadType, + "Media over RTX payload type"); static int MediaPayloadTypeRtx() { return static_cast(FLAG_media_payload_type_rtx); } -DEFINE_int(red_payload_type_rtx, - test::CallTest::kRtxRedPayloadType, - "RED over RTX payload type"); +WEBRTC_DEFINE_int(red_payload_type_rtx, + test::CallTest::kRtxRedPayloadType, + "RED over RTX payload type"); static int RedPayloadTypeRtx() { return static_cast(FLAG_red_payload_type_rtx); } @@ -115,7 +115,7 @@ const std::string& DefaultSsrc() { std::to_string(test::CallTest::kVideoSendSsrcs[0]); return ssrc; } -DEFINE_string(ssrc, DefaultSsrc().c_str(), "Incoming SSRC"); +WEBRTC_DEFINE_string(ssrc, DefaultSsrc().c_str(), "Incoming SSRC"); static uint32_t Ssrc() { return rtc::StringToNumber(FLAG_ssrc).value(); } @@ -125,54 +125,56 @@ const std::string& DefaultSsrcRtx() { std::to_string(test::CallTest::kSendRtxSsrcs[0]); return ssrc_rtx; } -DEFINE_string(ssrc_rtx, DefaultSsrcRtx().c_str(), "Incoming RTX SSRC"); +WEBRTC_DEFINE_string(ssrc_rtx, DefaultSsrcRtx().c_str(), "Incoming RTX SSRC"); static uint32_t SsrcRtx() { return rtc::StringToNumber(FLAG_ssrc_rtx).value(); } // Flag for abs-send-time id. -DEFINE_int(abs_send_time_id, -1, "RTP extension ID for abs-send-time"); +WEBRTC_DEFINE_int(abs_send_time_id, -1, "RTP extension ID for abs-send-time"); static int AbsSendTimeId() { return static_cast(FLAG_abs_send_time_id); } // Flag for transmission-offset id. -DEFINE_int(transmission_offset_id, - -1, - "RTP extension ID for transmission-offset"); +WEBRTC_DEFINE_int(transmission_offset_id, + -1, + "RTP extension ID for transmission-offset"); static int TransmissionOffsetId() { return static_cast(FLAG_transmission_offset_id); } // Flag for rtpdump input file. -DEFINE_string(input_file, "", "input file"); +WEBRTC_DEFINE_string(input_file, "", "input file"); static std::string InputFile() { return static_cast(FLAG_input_file); } -DEFINE_string(config_file, "", "config file"); +WEBRTC_DEFINE_string(config_file, "", "config file"); static std::string ConfigFile() { return static_cast(FLAG_config_file); } // Flag for raw output files. -DEFINE_string(out_base, "", "Basename (excluding .jpg) for raw output"); +WEBRTC_DEFINE_string(out_base, "", "Basename (excluding .jpg) for raw output"); static std::string OutBase() { return static_cast(FLAG_out_base); } -DEFINE_string(decoder_bitstream_filename, "", "Decoder bitstream output file"); +WEBRTC_DEFINE_string(decoder_bitstream_filename, + "", + "Decoder bitstream output file"); static std::string DecoderBitstreamFilename() { return static_cast(FLAG_decoder_bitstream_filename); } // Flag for video codec. -DEFINE_string(codec, "VP8", "Video codec"); +WEBRTC_DEFINE_string(codec, "VP8", "Video codec"); static std::string Codec() { return static_cast(FLAG_codec); } -DEFINE_bool(help, false, "Print this message."); +WEBRTC_DEFINE_bool(help, false, "Print this message."); } // namespace flags static const uint32_t kReceiverLocalSsrc = 0x123456; diff --git a/video/screenshare_loopback.cc b/video/screenshare_loopback.cc index 245d2636b0..4696666908 100644 --- a/video/screenshare_loopback.cc +++ b/video/screenshare_loopback.cc @@ -22,73 +22,76 @@ namespace webrtc { namespace flags { // Flags common with video loopback, with different default values. -DEFINE_int(width, 1850, "Video width (crops source)."); +WEBRTC_DEFINE_int(width, 1850, "Video width (crops source)."); size_t Width() { return static_cast(FLAG_width); } -DEFINE_int(height, 1110, "Video height (crops source)."); +WEBRTC_DEFINE_int(height, 1110, "Video height (crops source)."); size_t Height() { return static_cast(FLAG_height); } -DEFINE_int(fps, 5, "Frames per second."); +WEBRTC_DEFINE_int(fps, 5, "Frames per second."); int Fps() { return static_cast(FLAG_fps); } -DEFINE_int(min_bitrate, 50, "Call and stream min bitrate in kbps."); +WEBRTC_DEFINE_int(min_bitrate, 50, "Call and stream min bitrate in kbps."); int MinBitrateKbps() { return static_cast(FLAG_min_bitrate); } -DEFINE_int(start_bitrate, 300, "Call start bitrate in kbps."); +WEBRTC_DEFINE_int(start_bitrate, 300, "Call start bitrate in kbps."); int StartBitrateKbps() { return static_cast(FLAG_start_bitrate); } -DEFINE_int(target_bitrate, 200, "Stream target bitrate in kbps."); +WEBRTC_DEFINE_int(target_bitrate, 200, "Stream target bitrate in kbps."); int TargetBitrateKbps() { return static_cast(FLAG_target_bitrate); } -DEFINE_int(max_bitrate, 1000, "Call and stream max bitrate in kbps."); +WEBRTC_DEFINE_int(max_bitrate, 1000, "Call and stream max bitrate in kbps."); int MaxBitrateKbps() { return static_cast(FLAG_max_bitrate); } -DEFINE_int(num_temporal_layers, 2, "Number of temporal layers to use."); +WEBRTC_DEFINE_int(num_temporal_layers, 2, "Number of temporal layers to use."); int NumTemporalLayers() { return static_cast(FLAG_num_temporal_layers); } // Flags common with video loopback, with equal default values. -DEFINE_string(codec, "VP8", "Video codec to use."); +WEBRTC_DEFINE_string(codec, "VP8", "Video codec to use."); std::string Codec() { return static_cast(FLAG_codec); } -DEFINE_string(rtc_event_log_name, - "", - "Filename for rtc event log. Two files " - "with \"_send\" and \"_recv\" suffixes will be created."); +WEBRTC_DEFINE_string(rtc_event_log_name, + "", + "Filename for rtc event log. Two files " + "with \"_send\" and \"_recv\" suffixes will be created."); std::string RtcEventLogName() { return static_cast(FLAG_rtc_event_log_name); } -DEFINE_string(rtp_dump_name, "", "Filename for dumped received RTP stream."); +WEBRTC_DEFINE_string(rtp_dump_name, + "", + "Filename for dumped received RTP stream."); std::string RtpDumpName() { return static_cast(FLAG_rtp_dump_name); } -DEFINE_int(selected_tl, - -1, - "Temporal layer to show or analyze. -1 to disable filtering."); +WEBRTC_DEFINE_int( + selected_tl, + -1, + "Temporal layer to show or analyze. -1 to disable filtering."); int SelectedTL() { return static_cast(FLAG_selected_tl); } -DEFINE_int( +WEBRTC_DEFINE_int( duration, 0, "Duration of the test in seconds. If 0, rendered will be shown instead."); @@ -96,71 +99,74 @@ int DurationSecs() { return static_cast(FLAG_duration); } -DEFINE_string(output_filename, "", "Target graph data filename."); +WEBRTC_DEFINE_string(output_filename, "", "Target graph data filename."); std::string OutputFilename() { return static_cast(FLAG_output_filename); } -DEFINE_string(graph_title, - "", - "If empty, title will be generated automatically."); +WEBRTC_DEFINE_string(graph_title, + "", + "If empty, title will be generated automatically."); std::string GraphTitle() { return static_cast(FLAG_graph_title); } -DEFINE_int(loss_percent, 0, "Percentage of packets randomly lost."); +WEBRTC_DEFINE_int(loss_percent, 0, "Percentage of packets randomly lost."); int LossPercent() { return static_cast(FLAG_loss_percent); } -DEFINE_int(link_capacity, - 0, - "Capacity (kbps) of the fake link. 0 means infinite."); +WEBRTC_DEFINE_int(link_capacity, + 0, + "Capacity (kbps) of the fake link. 0 means infinite."); int LinkCapacityKbps() { return static_cast(FLAG_link_capacity); } -DEFINE_int(queue_size, 0, "Size of the bottleneck link queue in packets."); +WEBRTC_DEFINE_int(queue_size, + 0, + "Size of the bottleneck link queue in packets."); int QueueSize() { return static_cast(FLAG_queue_size); } -DEFINE_int(avg_propagation_delay_ms, - 0, - "Average link propagation delay in ms."); +WEBRTC_DEFINE_int(avg_propagation_delay_ms, + 0, + "Average link propagation delay in ms."); int AvgPropagationDelayMs() { return static_cast(FLAG_avg_propagation_delay_ms); } -DEFINE_int(std_propagation_delay_ms, - 0, - "Link propagation delay standard deviation in ms."); +WEBRTC_DEFINE_int(std_propagation_delay_ms, + 0, + "Link propagation delay standard deviation in ms."); int StdPropagationDelayMs() { return static_cast(FLAG_std_propagation_delay_ms); } -DEFINE_int(num_streams, 0, "Number of streams to show or analyze."); +WEBRTC_DEFINE_int(num_streams, 0, "Number of streams to show or analyze."); int NumStreams() { return static_cast(FLAG_num_streams); } -DEFINE_int(selected_stream, - 0, - "ID of the stream to show or analyze. " - "Set to the number of streams to show them all."); +WEBRTC_DEFINE_int(selected_stream, + 0, + "ID of the stream to show or analyze. " + "Set to the number of streams to show them all."); int SelectedStream() { return static_cast(FLAG_selected_stream); } -DEFINE_int(num_spatial_layers, 1, "Number of spatial layers to use."); +WEBRTC_DEFINE_int(num_spatial_layers, 1, "Number of spatial layers to use."); int NumSpatialLayers() { return static_cast(FLAG_num_spatial_layers); } -DEFINE_int(inter_layer_pred, - 0, - "Inter-layer prediction mode. " - "0 - enabled, 1 - disabled, 2 - enabled only for key pictures."); +WEBRTC_DEFINE_int( + inter_layer_pred, + 0, + "Inter-layer prediction mode. " + "0 - enabled, 1 - disabled, 2 - enabled only for key pictures."); InterLayerPredMode InterLayerPred() { if (FLAG_inter_layer_pred == 0) { return InterLayerPredMode::kOn; @@ -172,58 +178,65 @@ InterLayerPredMode InterLayerPred() { } } -DEFINE_int(selected_sl, - -1, - "Spatial layer to show or analyze. -1 to disable filtering."); +WEBRTC_DEFINE_int(selected_sl, + -1, + "Spatial layer to show or analyze. -1 to disable filtering."); int SelectedSL() { return static_cast(FLAG_selected_sl); } -DEFINE_string(stream0, - "", - "Comma separated values describing VideoStream for stream #0."); +WEBRTC_DEFINE_string( + stream0, + "", + "Comma separated values describing VideoStream for stream #0."); std::string Stream0() { return static_cast(FLAG_stream0); } -DEFINE_string(stream1, - "", - "Comma separated values describing VideoStream for stream #1."); +WEBRTC_DEFINE_string( + stream1, + "", + "Comma separated values describing VideoStream for stream #1."); std::string Stream1() { return static_cast(FLAG_stream1); } -DEFINE_string(sl0, - "", - "Comma separated values describing SpatialLayer for layer #0."); +WEBRTC_DEFINE_string( + sl0, + "", + "Comma separated values describing SpatialLayer for layer #0."); std::string SL0() { return static_cast(FLAG_sl0); } -DEFINE_string(sl1, - "", - "Comma separated values describing SpatialLayer for layer #1."); +WEBRTC_DEFINE_string( + sl1, + "", + "Comma separated values describing SpatialLayer for layer #1."); std::string SL1() { return static_cast(FLAG_sl1); } -DEFINE_string(encoded_frame_path, - "", - "The base path for encoded frame logs. Created files will have " - "the form ..(recv|send.).ivf"); +WEBRTC_DEFINE_string( + encoded_frame_path, + "", + "The base path for encoded frame logs. Created files will have " + "the form ..(recv|send.).ivf"); std::string EncodedFramePath() { return static_cast(FLAG_encoded_frame_path); } -DEFINE_bool(logs, false, "print logs to stderr"); +WEBRTC_DEFINE_bool(logs, false, "print logs to stderr"); -DEFINE_bool(send_side_bwe, true, "Use send-side bandwidth estimation"); +WEBRTC_DEFINE_bool(send_side_bwe, true, "Use send-side bandwidth estimation"); -DEFINE_bool(generic_descriptor, false, "Use the generic frame descriptor."); +WEBRTC_DEFINE_bool(generic_descriptor, + false, + "Use the generic frame descriptor."); -DEFINE_bool(allow_reordering, false, "Allow packet reordering to occur"); +WEBRTC_DEFINE_bool(allow_reordering, false, "Allow packet reordering to occur"); -DEFINE_string( +WEBRTC_DEFINE_string( force_fieldtrials, "", "Field trials control experimental feature code which can be forced. " @@ -232,12 +245,14 @@ DEFINE_string( "trials are separated by \"/\""); // Screenshare-specific flags. -DEFINE_int(min_transmit_bitrate, 400, "Min transmit bitrate incl. padding."); +WEBRTC_DEFINE_int(min_transmit_bitrate, + 400, + "Min transmit bitrate incl. padding."); int MinTransmitBitrateKbps() { return FLAG_min_transmit_bitrate; } -DEFINE_bool( +WEBRTC_DEFINE_bool( generate_slides, false, "Whether to use randomly generated slides or read them from files."); @@ -245,14 +260,14 @@ bool GenerateSlides() { return static_cast(FLAG_generate_slides); } -DEFINE_int(slide_change_interval, - 10, - "Interval (in seconds) between simulated slide changes."); +WEBRTC_DEFINE_int(slide_change_interval, + 10, + "Interval (in seconds) between simulated slide changes."); int SlideChangeInterval() { return static_cast(FLAG_slide_change_interval); } -DEFINE_int( +WEBRTC_DEFINE_int( scroll_duration, 0, "Duration (in seconds) during which a slide will be scrolled into place."); @@ -260,9 +275,10 @@ int ScrollDuration() { return static_cast(FLAG_scroll_duration); } -DEFINE_string(slides, - "", - "Comma-separated list of *.yuv files to display as slides."); +WEBRTC_DEFINE_string( + slides, + "", + "Comma-separated list of *.yuv files to display as slides."); std::vector Slides() { std::vector slides; std::string slides_list = FLAG_slides; @@ -270,7 +286,7 @@ std::vector Slides() { return slides; } -DEFINE_bool(help, false, "prints this message"); +WEBRTC_DEFINE_bool(help, false, "prints this message"); } // namespace flags diff --git a/video/sv_loopback.cc b/video/sv_loopback.cc index 864fc634ef..1ff4956ca2 100644 --- a/video/sv_loopback.cc +++ b/video/sv_loopback.cc @@ -33,73 +33,79 @@ InterLayerPredMode IntToInterLayerPredMode(int inter_layer_pred) { } // Flags for video. -DEFINE_int(vwidth, 640, "Video width."); +WEBRTC_DEFINE_int(vwidth, 640, "Video width."); size_t VideoWidth() { return static_cast(FLAG_vwidth); } -DEFINE_int(vheight, 480, "Video height."); +WEBRTC_DEFINE_int(vheight, 480, "Video height."); size_t VideoHeight() { return static_cast(FLAG_vheight); } -DEFINE_int(vfps, 30, "Video frames per second."); +WEBRTC_DEFINE_int(vfps, 30, "Video frames per second."); int VideoFps() { return static_cast(FLAG_vfps); } -DEFINE_int(capture_device_index, - 0, - "Capture device to select for video stream"); +WEBRTC_DEFINE_int(capture_device_index, + 0, + "Capture device to select for video stream"); size_t GetCaptureDevice() { return static_cast(FLAG_capture_device_index); } -DEFINE_int(vtarget_bitrate, 400, "Video stream target bitrate in kbps."); +WEBRTC_DEFINE_int(vtarget_bitrate, 400, "Video stream target bitrate in kbps."); int VideoTargetBitrateKbps() { return static_cast(FLAG_vtarget_bitrate); } -DEFINE_int(vmin_bitrate, 100, "Video stream min bitrate in kbps."); +WEBRTC_DEFINE_int(vmin_bitrate, 100, "Video stream min bitrate in kbps."); int VideoMinBitrateKbps() { return static_cast(FLAG_vmin_bitrate); } -DEFINE_int(vmax_bitrate, 2000, "Video stream max bitrate in kbps."); +WEBRTC_DEFINE_int(vmax_bitrate, 2000, "Video stream max bitrate in kbps."); int VideoMaxBitrateKbps() { return static_cast(FLAG_vmax_bitrate); } -DEFINE_bool(suspend_below_min_bitrate, - false, - "Suspends video below the configured min bitrate."); +WEBRTC_DEFINE_bool(suspend_below_min_bitrate, + false, + "Suspends video below the configured min bitrate."); -DEFINE_int(vnum_temporal_layers, - 1, - "Number of temporal layers for video. Set to 1-4 to override."); +WEBRTC_DEFINE_int( + vnum_temporal_layers, + 1, + "Number of temporal layers for video. Set to 1-4 to override."); int VideoNumTemporalLayers() { return static_cast(FLAG_vnum_temporal_layers); } -DEFINE_int(vnum_streams, 0, "Number of video streams to show or analyze."); +WEBRTC_DEFINE_int(vnum_streams, + 0, + "Number of video streams to show or analyze."); int VideoNumStreams() { return static_cast(FLAG_vnum_streams); } -DEFINE_int(vnum_spatial_layers, 1, "Number of video spatial layers to use."); +WEBRTC_DEFINE_int(vnum_spatial_layers, + 1, + "Number of video spatial layers to use."); int VideoNumSpatialLayers() { return static_cast(FLAG_vnum_spatial_layers); } -DEFINE_int(vinter_layer_pred, - 2, - "Video inter-layer prediction mode. " - "0 - enabled, 1 - disabled, 2 - enabled only for key pictures."); +WEBRTC_DEFINE_int( + vinter_layer_pred, + 2, + "Video inter-layer prediction mode. " + "0 - enabled, 1 - disabled, 2 - enabled only for key pictures."); InterLayerPredMode VideoInterLayerPred() { return IntToInterLayerPredMode(FLAG_vinter_layer_pred); } -DEFINE_string( +WEBRTC_DEFINE_string( vstream0, "", "Comma separated values describing VideoStream for video stream #0."); @@ -107,7 +113,7 @@ std::string VideoStream0() { return static_cast(FLAG_vstream0); } -DEFINE_string( +WEBRTC_DEFINE_string( vstream1, "", "Comma separated values describing VideoStream for video stream #1."); @@ -115,7 +121,7 @@ std::string VideoStream1() { return static_cast(FLAG_vstream1); } -DEFINE_string( +WEBRTC_DEFINE_string( vsl0, "", "Comma separated values describing SpatialLayer for video layer #0."); @@ -123,7 +129,7 @@ std::string VideoSL0() { return static_cast(FLAG_vsl0); } -DEFINE_string( +WEBRTC_DEFINE_string( vsl1, "", "Comma separated values describing SpatialLayer for video layer #1."); @@ -131,98 +137,105 @@ std::string VideoSL1() { return static_cast(FLAG_vsl1); } -DEFINE_int(vselected_tl, - -1, - "Temporal layer to show or analyze for screenshare. -1 to disable " - "filtering."); +WEBRTC_DEFINE_int( + vselected_tl, + -1, + "Temporal layer to show or analyze for screenshare. -1 to disable " + "filtering."); int VideoSelectedTL() { return static_cast(FLAG_vselected_tl); } -DEFINE_int(vselected_stream, - 0, - "ID of the stream to show or analyze for screenshare." - "Set to the number of streams to show them all."); +WEBRTC_DEFINE_int(vselected_stream, + 0, + "ID of the stream to show or analyze for screenshare." + "Set to the number of streams to show them all."); int VideoSelectedStream() { return static_cast(FLAG_vselected_stream); } -DEFINE_int(vselected_sl, - -1, - "Spatial layer to show or analyze for screenshare. -1 to disable " - "filtering."); +WEBRTC_DEFINE_int( + vselected_sl, + -1, + "Spatial layer to show or analyze for screenshare. -1 to disable " + "filtering."); int VideoSelectedSL() { return static_cast(FLAG_vselected_sl); } // Flags for screenshare. -DEFINE_int(min_transmit_bitrate, - 400, - "Min transmit bitrate incl. padding for screenshare."); +WEBRTC_DEFINE_int(min_transmit_bitrate, + 400, + "Min transmit bitrate incl. padding for screenshare."); int ScreenshareMinTransmitBitrateKbps() { return FLAG_min_transmit_bitrate; } -DEFINE_int(swidth, 1850, "Screenshare width (crops source)."); +WEBRTC_DEFINE_int(swidth, 1850, "Screenshare width (crops source)."); size_t ScreenshareWidth() { return static_cast(FLAG_swidth); } -DEFINE_int(sheight, 1110, "Screenshare height (crops source)."); +WEBRTC_DEFINE_int(sheight, 1110, "Screenshare height (crops source)."); size_t ScreenshareHeight() { return static_cast(FLAG_sheight); } -DEFINE_int(sfps, 5, "Frames per second for screenshare."); +WEBRTC_DEFINE_int(sfps, 5, "Frames per second for screenshare."); int ScreenshareFps() { return static_cast(FLAG_sfps); } -DEFINE_int(starget_bitrate, 100, "Screenshare stream target bitrate in kbps."); +WEBRTC_DEFINE_int(starget_bitrate, + 100, + "Screenshare stream target bitrate in kbps."); int ScreenshareTargetBitrateKbps() { return static_cast(FLAG_starget_bitrate); } -DEFINE_int(smin_bitrate, 100, "Screenshare stream min bitrate in kbps."); +WEBRTC_DEFINE_int(smin_bitrate, 100, "Screenshare stream min bitrate in kbps."); int ScreenshareMinBitrateKbps() { return static_cast(FLAG_smin_bitrate); } -DEFINE_int(smax_bitrate, 2000, "Screenshare stream max bitrate in kbps."); +WEBRTC_DEFINE_int(smax_bitrate, + 2000, + "Screenshare stream max bitrate in kbps."); int ScreenshareMaxBitrateKbps() { return static_cast(FLAG_smax_bitrate); } -DEFINE_int(snum_temporal_layers, - 2, - "Number of temporal layers to use in screenshare."); +WEBRTC_DEFINE_int(snum_temporal_layers, + 2, + "Number of temporal layers to use in screenshare."); int ScreenshareNumTemporalLayers() { return static_cast(FLAG_snum_temporal_layers); } -DEFINE_int(snum_streams, - 0, - "Number of screenshare streams to show or analyze."); +WEBRTC_DEFINE_int(snum_streams, + 0, + "Number of screenshare streams to show or analyze."); int ScreenshareNumStreams() { return static_cast(FLAG_snum_streams); } -DEFINE_int(snum_spatial_layers, - 1, - "Number of screenshare spatial layers to use."); +WEBRTC_DEFINE_int(snum_spatial_layers, + 1, + "Number of screenshare spatial layers to use."); int ScreenshareNumSpatialLayers() { return static_cast(FLAG_snum_spatial_layers); } -DEFINE_int(sinter_layer_pred, - 0, - "Screenshare inter-layer prediction mode. " - "0 - enabled, 1 - disabled, 2 - enabled only for key pictures."); +WEBRTC_DEFINE_int( + sinter_layer_pred, + 0, + "Screenshare inter-layer prediction mode. " + "0 - enabled, 1 - disabled, 2 - enabled only for key pictures."); InterLayerPredMode ScreenshareInterLayerPred() { return IntToInterLayerPredMode(FLAG_sinter_layer_pred); } -DEFINE_string( +WEBRTC_DEFINE_string( sstream0, "", "Comma separated values describing VideoStream for screenshare stream #0."); @@ -230,7 +243,7 @@ std::string ScreenshareStream0() { return static_cast(FLAG_sstream0); } -DEFINE_string( +WEBRTC_DEFINE_string( sstream1, "", "Comma separated values describing VideoStream for screenshare stream #1."); @@ -238,7 +251,7 @@ std::string ScreenshareStream1() { return static_cast(FLAG_sstream1); } -DEFINE_string( +WEBRTC_DEFINE_string( ssl0, "", "Comma separated values describing SpatialLayer for screenshare layer #0."); @@ -246,7 +259,7 @@ std::string ScreenshareSL0() { return static_cast(FLAG_ssl0); } -DEFINE_string( +WEBRTC_DEFINE_string( ssl1, "", "Comma separated values describing SpatialLayer for screenshare layer #1."); @@ -254,31 +267,33 @@ std::string ScreenshareSL1() { return static_cast(FLAG_ssl1); } -DEFINE_int(sselected_tl, - -1, - "Temporal layer to show or analyze for screenshare. -1 to disable " - "filtering."); +WEBRTC_DEFINE_int( + sselected_tl, + -1, + "Temporal layer to show or analyze for screenshare. -1 to disable " + "filtering."); int ScreenshareSelectedTL() { return static_cast(FLAG_sselected_tl); } -DEFINE_int(sselected_stream, - 0, - "ID of the stream to show or analyze for screenshare." - "Set to the number of streams to show them all."); +WEBRTC_DEFINE_int(sselected_stream, + 0, + "ID of the stream to show or analyze for screenshare." + "Set to the number of streams to show them all."); int ScreenshareSelectedStream() { return static_cast(FLAG_sselected_stream); } -DEFINE_int(sselected_sl, - -1, - "Spatial layer to show or analyze for screenshare. -1 to disable " - "filtering."); +WEBRTC_DEFINE_int( + sselected_sl, + -1, + "Spatial layer to show or analyze for screenshare. -1 to disable " + "filtering."); int ScreenshareSelectedSL() { return static_cast(FLAG_sselected_sl); } -DEFINE_bool( +WEBRTC_DEFINE_bool( generate_slides, false, "Whether to use randomly generated slides or read them from files."); @@ -286,14 +301,14 @@ bool GenerateSlides() { return static_cast(FLAG_generate_slides); } -DEFINE_int(slide_change_interval, - 10, - "Interval (in seconds) between simulated slide changes."); +WEBRTC_DEFINE_int(slide_change_interval, + 10, + "Interval (in seconds) between simulated slide changes."); int SlideChangeInterval() { return static_cast(FLAG_slide_change_interval); } -DEFINE_int( +WEBRTC_DEFINE_int( scroll_duration, 0, "Duration (in seconds) during which a slide will be scrolled into place."); @@ -301,9 +316,10 @@ int ScrollDuration() { return static_cast(FLAG_scroll_duration); } -DEFINE_string(slides, - "", - "Comma-separated list of *.yuv files to display as slides."); +WEBRTC_DEFINE_string( + slides, + "", + "Comma-separated list of *.yuv files to display as slides."); std::vector Slides() { std::vector slides; std::string slides_list = FLAG_slides; @@ -312,31 +328,31 @@ std::vector Slides() { } // Flags common with screenshare and video loopback, with equal default values. -DEFINE_int(start_bitrate, 600, "Call start bitrate in kbps."); +WEBRTC_DEFINE_int(start_bitrate, 600, "Call start bitrate in kbps."); int StartBitrateKbps() { return static_cast(FLAG_start_bitrate); } -DEFINE_string(codec, "VP8", "Video codec to use."); +WEBRTC_DEFINE_string(codec, "VP8", "Video codec to use."); std::string Codec() { return static_cast(FLAG_codec); } -DEFINE_bool(analyze_video, - false, - "Analyze video stream (if --duration is present)"); +WEBRTC_DEFINE_bool(analyze_video, + false, + "Analyze video stream (if --duration is present)"); bool AnalyzeVideo() { return static_cast(FLAG_analyze_video); } -DEFINE_bool(analyze_screenshare, - false, - "Analyze screenshare stream (if --duration is present)"); +WEBRTC_DEFINE_bool(analyze_screenshare, + false, + "Analyze screenshare stream (if --duration is present)"); bool AnalyzeScreenshare() { return static_cast(FLAG_analyze_screenshare); } -DEFINE_int( +WEBRTC_DEFINE_int( duration, 0, "Duration of the test in seconds. If 0, rendered will be shown instead."); @@ -344,100 +360,113 @@ int DurationSecs() { return static_cast(FLAG_duration); } -DEFINE_string(output_filename, "", "Target graph data filename."); +WEBRTC_DEFINE_string(output_filename, "", "Target graph data filename."); std::string OutputFilename() { return static_cast(FLAG_output_filename); } -DEFINE_string(graph_title, - "", - "If empty, title will be generated automatically."); +WEBRTC_DEFINE_string(graph_title, + "", + "If empty, title will be generated automatically."); std::string GraphTitle() { return static_cast(FLAG_graph_title); } -DEFINE_int(loss_percent, 0, "Percentage of packets randomly lost."); +WEBRTC_DEFINE_int(loss_percent, 0, "Percentage of packets randomly lost."); int LossPercent() { return static_cast(FLAG_loss_percent); } -DEFINE_int(avg_burst_loss_length, -1, "Average burst length of lost packets."); +WEBRTC_DEFINE_int(avg_burst_loss_length, + -1, + "Average burst length of lost packets."); int AvgBurstLossLength() { return static_cast(FLAG_avg_burst_loss_length); } -DEFINE_int(link_capacity, - 0, - "Capacity (kbps) of the fake link. 0 means infinite."); +WEBRTC_DEFINE_int(link_capacity, + 0, + "Capacity (kbps) of the fake link. 0 means infinite."); int LinkCapacityKbps() { return static_cast(FLAG_link_capacity); } -DEFINE_int(queue_size, 0, "Size of the bottleneck link queue in packets."); +WEBRTC_DEFINE_int(queue_size, + 0, + "Size of the bottleneck link queue in packets."); int QueueSize() { return static_cast(FLAG_queue_size); } -DEFINE_int(avg_propagation_delay_ms, - 0, - "Average link propagation delay in ms."); +WEBRTC_DEFINE_int(avg_propagation_delay_ms, + 0, + "Average link propagation delay in ms."); int AvgPropagationDelayMs() { return static_cast(FLAG_avg_propagation_delay_ms); } -DEFINE_string(rtc_event_log_name, - "", - "Filename for rtc event log. Two files " - "with \"_send\" and \"_recv\" suffixes will be created. " - "Works only when --duration is set."); +WEBRTC_DEFINE_string(rtc_event_log_name, + "", + "Filename for rtc event log. Two files " + "with \"_send\" and \"_recv\" suffixes will be created. " + "Works only when --duration is set."); std::string RtcEventLogName() { return static_cast(FLAG_rtc_event_log_name); } -DEFINE_string(rtp_dump_name, "", "Filename for dumped received RTP stream."); +WEBRTC_DEFINE_string(rtp_dump_name, + "", + "Filename for dumped received RTP stream."); std::string RtpDumpName() { return static_cast(FLAG_rtp_dump_name); } -DEFINE_int(std_propagation_delay_ms, - 0, - "Link propagation delay standard deviation in ms."); +WEBRTC_DEFINE_int(std_propagation_delay_ms, + 0, + "Link propagation delay standard deviation in ms."); int StdPropagationDelayMs() { return static_cast(FLAG_std_propagation_delay_ms); } -DEFINE_string(encoded_frame_path, - "", - "The base path for encoded frame logs. Created files will have " - "the form ..(recv|send.).ivf"); +WEBRTC_DEFINE_string( + encoded_frame_path, + "", + "The base path for encoded frame logs. Created files will have " + "the form ..(recv|send.).ivf"); std::string EncodedFramePath() { return static_cast(FLAG_encoded_frame_path); } -DEFINE_bool(logs, false, "print logs to stderr"); +WEBRTC_DEFINE_bool(logs, false, "print logs to stderr"); -DEFINE_bool(send_side_bwe, true, "Use send-side bandwidth estimation"); +WEBRTC_DEFINE_bool(send_side_bwe, true, "Use send-side bandwidth estimation"); -DEFINE_bool(generic_descriptor, false, "Use the generic frame descriptor."); +WEBRTC_DEFINE_bool(generic_descriptor, + false, + "Use the generic frame descriptor."); -DEFINE_bool(allow_reordering, false, "Allow packet reordering to occur"); +WEBRTC_DEFINE_bool(allow_reordering, false, "Allow packet reordering to occur"); -DEFINE_bool(use_ulpfec, false, "Use RED+ULPFEC forward error correction."); +WEBRTC_DEFINE_bool(use_ulpfec, + false, + "Use RED+ULPFEC forward error correction."); -DEFINE_bool(use_flexfec, false, "Use FlexFEC forward error correction."); +WEBRTC_DEFINE_bool(use_flexfec, false, "Use FlexFEC forward error correction."); -DEFINE_bool(audio, false, "Add audio stream"); +WEBRTC_DEFINE_bool(audio, false, "Add audio stream"); -DEFINE_bool(audio_video_sync, - false, - "Sync audio and video stream (no effect if" - " audio is false)"); +WEBRTC_DEFINE_bool(audio_video_sync, + false, + "Sync audio and video stream (no effect if" + " audio is false)"); -DEFINE_bool(audio_dtx, false, "Enable audio DTX (no effect if audio is false)"); +WEBRTC_DEFINE_bool(audio_dtx, + false, + "Enable audio DTX (no effect if audio is false)"); -DEFINE_bool(video, true, "Add video stream"); +WEBRTC_DEFINE_bool(video, true, "Add video stream"); -DEFINE_string( +WEBRTC_DEFINE_string( force_fieldtrials, "", "Field trials control experimental feature code which can be forced. " @@ -446,15 +475,16 @@ DEFINE_string( "trials are separated by \"/\""); // Video-specific flags. -DEFINE_string(vclip, - "", - "Name of the clip to show. If empty, the camera is used. Use " - "\"Generator\" for chroma generator."); +WEBRTC_DEFINE_string( + vclip, + "", + "Name of the clip to show. If empty, the camera is used. Use " + "\"Generator\" for chroma generator."); std::string VideoClip() { return static_cast(FLAG_vclip); } -DEFINE_bool(help, false, "prints this message"); +WEBRTC_DEFINE_bool(help, false, "prints this message"); } // namespace flags diff --git a/video/video_analyzer.cc b/video/video_analyzer.cc index 1b1c557e60..6d16b1a055 100644 --- a/video/video_analyzer.cc +++ b/video/video_analyzer.cc @@ -25,10 +25,11 @@ #include "test/testsupport/perf_test.h" #include "test/testsupport/test_artifacts.h" -DEFINE_bool(save_worst_frame, - false, - "Enable saving a frame with the lowest PSNR to a jpeg file in the " - "test_artifacts_dir"); +WEBRTC_DEFINE_bool( + save_worst_frame, + false, + "Enable saving a frame with the lowest PSNR to a jpeg file in the " + "test_artifacts_dir"); namespace webrtc { namespace { diff --git a/video/video_loopback.cc b/video/video_loopback.cc index b6715ac4ad..4cdddb94e5 100644 --- a/video/video_loopback.cc +++ b/video/video_loopback.cc @@ -22,61 +22,62 @@ namespace webrtc { namespace flags { // Flags common with screenshare loopback, with different default values. -DEFINE_int(width, 640, "Video width."); +WEBRTC_DEFINE_int(width, 640, "Video width."); size_t Width() { return static_cast(FLAG_width); } -DEFINE_int(height, 480, "Video height."); +WEBRTC_DEFINE_int(height, 480, "Video height."); size_t Height() { return static_cast(FLAG_height); } -DEFINE_int(fps, 30, "Frames per second."); +WEBRTC_DEFINE_int(fps, 30, "Frames per second."); int Fps() { return static_cast(FLAG_fps); } -DEFINE_int(capture_device_index, 0, "Capture device to select"); +WEBRTC_DEFINE_int(capture_device_index, 0, "Capture device to select"); size_t GetCaptureDevice() { return static_cast(FLAG_capture_device_index); } -DEFINE_int(min_bitrate, 50, "Call and stream min bitrate in kbps."); +WEBRTC_DEFINE_int(min_bitrate, 50, "Call and stream min bitrate in kbps."); int MinBitrateKbps() { return static_cast(FLAG_min_bitrate); } -DEFINE_int(start_bitrate, 300, "Call start bitrate in kbps."); +WEBRTC_DEFINE_int(start_bitrate, 300, "Call start bitrate in kbps."); int StartBitrateKbps() { return static_cast(FLAG_start_bitrate); } -DEFINE_int(target_bitrate, 800, "Stream target bitrate in kbps."); +WEBRTC_DEFINE_int(target_bitrate, 800, "Stream target bitrate in kbps."); int TargetBitrateKbps() { return static_cast(FLAG_target_bitrate); } -DEFINE_int(max_bitrate, 800, "Call and stream max bitrate in kbps."); +WEBRTC_DEFINE_int(max_bitrate, 800, "Call and stream max bitrate in kbps."); int MaxBitrateKbps() { return static_cast(FLAG_max_bitrate); } -DEFINE_bool(suspend_below_min_bitrate, - false, - "Suspends video below the configured min bitrate."); +WEBRTC_DEFINE_bool(suspend_below_min_bitrate, + false, + "Suspends video below the configured min bitrate."); -DEFINE_int(num_temporal_layers, - 1, - "Number of temporal layers. Set to 1-4 to override."); +WEBRTC_DEFINE_int(num_temporal_layers, + 1, + "Number of temporal layers. Set to 1-4 to override."); int NumTemporalLayers() { return static_cast(FLAG_num_temporal_layers); } -DEFINE_int(inter_layer_pred, - 2, - "Inter-layer prediction mode. " - "0 - enabled, 1 - disabled, 2 - enabled only for key pictures."); +WEBRTC_DEFINE_int( + inter_layer_pred, + 2, + "Inter-layer prediction mode. " + "0 - enabled, 1 - disabled, 2 - enabled only for key pictures."); InterLayerPredMode InterLayerPred() { if (FLAG_inter_layer_pred == 0) { return InterLayerPredMode::kOn; @@ -89,19 +90,20 @@ InterLayerPredMode InterLayerPred() { } // Flags common with screenshare loopback, with equal default values. -DEFINE_string(codec, "VP8", "Video codec to use."); +WEBRTC_DEFINE_string(codec, "VP8", "Video codec to use."); std::string Codec() { return static_cast(FLAG_codec); } -DEFINE_int(selected_tl, - -1, - "Temporal layer to show or analyze. -1 to disable filtering."); +WEBRTC_DEFINE_int( + selected_tl, + -1, + "Temporal layer to show or analyze. -1 to disable filtering."); int SelectedTL() { return static_cast(FLAG_selected_tl); } -DEFINE_int( +WEBRTC_DEFINE_int( duration, 0, "Duration of the test in seconds. If 0, rendered will be shown instead."); @@ -109,156 +111,174 @@ int DurationSecs() { return static_cast(FLAG_duration); } -DEFINE_string(output_filename, "", "Target graph data filename."); +WEBRTC_DEFINE_string(output_filename, "", "Target graph data filename."); std::string OutputFilename() { return static_cast(FLAG_output_filename); } -DEFINE_string(graph_title, - "", - "If empty, title will be generated automatically."); +WEBRTC_DEFINE_string(graph_title, + "", + "If empty, title will be generated automatically."); std::string GraphTitle() { return static_cast(FLAG_graph_title); } -DEFINE_int(loss_percent, 0, "Percentage of packets randomly lost."); +WEBRTC_DEFINE_int(loss_percent, 0, "Percentage of packets randomly lost."); int LossPercent() { return static_cast(FLAG_loss_percent); } -DEFINE_int(avg_burst_loss_length, -1, "Average burst length of lost packets."); +WEBRTC_DEFINE_int(avg_burst_loss_length, + -1, + "Average burst length of lost packets."); int AvgBurstLossLength() { return static_cast(FLAG_avg_burst_loss_length); } -DEFINE_int(link_capacity, - 0, - "Capacity (kbps) of the fake link. 0 means infinite."); +WEBRTC_DEFINE_int(link_capacity, + 0, + "Capacity (kbps) of the fake link. 0 means infinite."); int LinkCapacityKbps() { return static_cast(FLAG_link_capacity); } -DEFINE_int(queue_size, 0, "Size of the bottleneck link queue in packets."); +WEBRTC_DEFINE_int(queue_size, + 0, + "Size of the bottleneck link queue in packets."); int QueueSize() { return static_cast(FLAG_queue_size); } -DEFINE_int(avg_propagation_delay_ms, - 0, - "Average link propagation delay in ms."); +WEBRTC_DEFINE_int(avg_propagation_delay_ms, + 0, + "Average link propagation delay in ms."); int AvgPropagationDelayMs() { return static_cast(FLAG_avg_propagation_delay_ms); } -DEFINE_string(rtc_event_log_name, - "", - "Filename for rtc event log. Two files " - "with \"_send\" and \"_recv\" suffixes will be created."); +WEBRTC_DEFINE_string(rtc_event_log_name, + "", + "Filename for rtc event log. Two files " + "with \"_send\" and \"_recv\" suffixes will be created."); std::string RtcEventLogName() { return static_cast(FLAG_rtc_event_log_name); } -DEFINE_string(rtp_dump_name, "", "Filename for dumped received RTP stream."); +WEBRTC_DEFINE_string(rtp_dump_name, + "", + "Filename for dumped received RTP stream."); std::string RtpDumpName() { return static_cast(FLAG_rtp_dump_name); } -DEFINE_int(std_propagation_delay_ms, - 0, - "Link propagation delay standard deviation in ms."); +WEBRTC_DEFINE_int(std_propagation_delay_ms, + 0, + "Link propagation delay standard deviation in ms."); int StdPropagationDelayMs() { return static_cast(FLAG_std_propagation_delay_ms); } -DEFINE_int(num_streams, 0, "Number of streams to show or analyze."); +WEBRTC_DEFINE_int(num_streams, 0, "Number of streams to show or analyze."); int NumStreams() { return static_cast(FLAG_num_streams); } -DEFINE_int(selected_stream, - 0, - "ID of the stream to show or analyze. " - "Set to the number of streams to show them all."); +WEBRTC_DEFINE_int(selected_stream, + 0, + "ID of the stream to show or analyze. " + "Set to the number of streams to show them all."); int SelectedStream() { return static_cast(FLAG_selected_stream); } -DEFINE_int(num_spatial_layers, 1, "Number of spatial layers to use."); +WEBRTC_DEFINE_int(num_spatial_layers, 1, "Number of spatial layers to use."); int NumSpatialLayers() { return static_cast(FLAG_num_spatial_layers); } -DEFINE_int(selected_sl, - -1, - "Spatial layer to show or analyze. -1 to disable filtering."); +WEBRTC_DEFINE_int(selected_sl, + -1, + "Spatial layer to show or analyze. -1 to disable filtering."); int SelectedSL() { return static_cast(FLAG_selected_sl); } -DEFINE_string(stream0, - "", - "Comma separated values describing VideoStream for stream #0."); +WEBRTC_DEFINE_string( + stream0, + "", + "Comma separated values describing VideoStream for stream #0."); std::string Stream0() { return static_cast(FLAG_stream0); } -DEFINE_string(stream1, - "", - "Comma separated values describing VideoStream for stream #1."); +WEBRTC_DEFINE_string( + stream1, + "", + "Comma separated values describing VideoStream for stream #1."); std::string Stream1() { return static_cast(FLAG_stream1); } -DEFINE_string(sl0, - "", - "Comma separated values describing SpatialLayer for layer #0."); +WEBRTC_DEFINE_string( + sl0, + "", + "Comma separated values describing SpatialLayer for layer #0."); std::string SL0() { return static_cast(FLAG_sl0); } -DEFINE_string(sl1, - "", - "Comma separated values describing SpatialLayer for layer #1."); +WEBRTC_DEFINE_string( + sl1, + "", + "Comma separated values describing SpatialLayer for layer #1."); std::string SL1() { return static_cast(FLAG_sl1); } -DEFINE_string(encoded_frame_path, - "", - "The base path for encoded frame logs. Created files will have " - "the form ..(recv|send.).ivf"); +WEBRTC_DEFINE_string( + encoded_frame_path, + "", + "The base path for encoded frame logs. Created files will have " + "the form ..(recv|send.).ivf"); std::string EncodedFramePath() { return static_cast(FLAG_encoded_frame_path); } -DEFINE_bool(logs, false, "print logs to stderr"); +WEBRTC_DEFINE_bool(logs, false, "print logs to stderr"); -DEFINE_bool(send_side_bwe, true, "Use send-side bandwidth estimation"); +WEBRTC_DEFINE_bool(send_side_bwe, true, "Use send-side bandwidth estimation"); -DEFINE_bool(generic_descriptor, false, "Use the generic frame descriptor."); +WEBRTC_DEFINE_bool(generic_descriptor, + false, + "Use the generic frame descriptor."); -DEFINE_bool(allow_reordering, false, "Allow packet reordering to occur"); +WEBRTC_DEFINE_bool(allow_reordering, false, "Allow packet reordering to occur"); -DEFINE_bool(use_ulpfec, false, "Use RED+ULPFEC forward error correction."); +WEBRTC_DEFINE_bool(use_ulpfec, + false, + "Use RED+ULPFEC forward error correction."); -DEFINE_bool(use_flexfec, false, "Use FlexFEC forward error correction."); +WEBRTC_DEFINE_bool(use_flexfec, false, "Use FlexFEC forward error correction."); -DEFINE_bool(audio, false, "Add audio stream"); +WEBRTC_DEFINE_bool(audio, false, "Add audio stream"); -DEFINE_bool(use_real_adm, - false, - "Use real ADM instead of fake (no effect if audio is false)"); +WEBRTC_DEFINE_bool( + use_real_adm, + false, + "Use real ADM instead of fake (no effect if audio is false)"); -DEFINE_bool(audio_video_sync, - false, - "Sync audio and video stream (no effect if" - " audio is false)"); +WEBRTC_DEFINE_bool(audio_video_sync, + false, + "Sync audio and video stream (no effect if" + " audio is false)"); -DEFINE_bool(audio_dtx, false, "Enable audio DTX (no effect if audio is false)"); +WEBRTC_DEFINE_bool(audio_dtx, + false, + "Enable audio DTX (no effect if audio is false)"); -DEFINE_bool(video, true, "Add video stream"); +WEBRTC_DEFINE_bool(video, true, "Add video stream"); -DEFINE_string( +WEBRTC_DEFINE_string( force_fieldtrials, "", "Field trials control experimental feature code which can be forced. " @@ -267,14 +287,15 @@ DEFINE_string( "trials are separated by \"/\""); // Video-specific flags. -DEFINE_string(clip, - "", - "Name of the clip to show. If empty, using chroma generator."); +WEBRTC_DEFINE_string( + clip, + "", + "Name of the clip to show. If empty, using chroma generator."); std::string Clip() { return static_cast(FLAG_clip); } -DEFINE_bool(help, false, "prints this message"); +WEBRTC_DEFINE_bool(help, false, "prints this message"); } // namespace flags