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Prefix flag macros with WEBRTC_.

Browse Source 
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}
This commit is contained in:
Mirko Bonadei
2018-10-18 11:35:32 +02:00
committed by Commit Bot
parent 8dc280d804
commit 5ccdc1331f
44 changed files with 1098 additions and 937 deletions
Download Patch File Download Diff File Expand all files Collapse all files

49
modules/audio_coding/neteq/tools/neteq_quality_test.cc
View File

@ -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: "
"<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