Prefix flag macros with WEBRTC_.
Macros defined in rtc_base/flags.h are intended to be used to define
flags in WebRTC's binaries (e.g. tests).
They are currently not prefixed and this could cause problems with
downstream clients since these names are quite common.
This CL adds the 'WEBRTC_' prefix to them.
Generated with:
for x in DECLARE DEFINE; do
for y in bool int float string FLAG; do
git grep -l "\b$x\_$y\b" | \
xargs sed -i "s/\b$x\_$y\b/WEBRTC_$x\_$y/g"
done
done
git cl format
Bug: webrtc:9884
Change-Id: I7b524762b6a3e5aa5b2fc2395edd3e1a0fe72591
Reviewed-on: https://webrtc-review.googlesource.com/c/106682
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25270}diff --git a/modules/audio_coding/neteq/neteq_unittest.cc b/modules/audio_coding/neteq/neteq_unittest.cc
index 3c63aa7..1c9b9e7 100644
--- a/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_unittest.cc
@@ -52,7 +52,7 @@
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 ad61235..6f10345 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 @@
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 94984b87..651b0ca 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 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 6861e4c..f4a3636 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 @@
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 8872b94..9c53919 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 @@
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 54ff849..85f2267 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 @@
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 c1d78c5..70777a2 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 faca895..2ee6779 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 @@
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: "
- "<first_event_time> <first_event_duration>, <second_event_time> "
- "<second_event_duration>, ...");
+WEBRTC_DEFINE_string(
+ loss_events,
+ "",
+ "List of loss events time and duration separated by comma: "
+ "<first_event_time> <first_event_duration>, <second_event_time> "
+ "<second_event_duration>, ...");
// ProbTrans00Solver() is to calculate the transition probability from no-loss
// state to itself in a modified Gilbert Elliot packet loss model. The result is
diff --git a/modules/audio_coding/neteq/tools/neteq_rtpplay.cc b/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
index 25e8cd8..c2726eb 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 df3a9f0..93da54c 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 @@
}
// 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 f939038..9d3041e 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 5065ca1..f48b04d 100644
--- a/modules/audio_coding/neteq/tools/rtp_encode.cc
+++ b/modules/audio_coding/neteq/tools/rtp_encode.cc
@@ -40,20 +40,24 @@
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 3c49443..92a7a8d 100644
--- a/modules/audio_coding/neteq/tools/rtp_jitter.cc
+++ b/modules/audio_coding/neteq/tools/rtp_jitter.cc
@@ -23,7 +23,7 @@
namespace test {
namespace {
-DEFINE_bool(help, false, "Print help message");
+WEBRTC_DEFINE_bool(help, false, "Print help message");
constexpr size_t kRtpDumpHeaderLength = 8;