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Reland "Prefix flag macros with WEBRTC_."

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This is a reland of 5ccdc1331fcc3cd78eaa14408fe0c38d37a5a51d

Original change's description:
> 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}

TBR=kwiberg@webrtc.org

Bug: webrtc:9884
Change-Id: I5ba5368a231a334d135ed5e6fd7a279629ced8a3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/107161
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25277}
This commit is contained in:
Mirko Bonadei
2018-10-18 11:35:32 +02:00
committed by Commit Bot
parent c538fc77b0
commit 2dfa998be2
44 changed files with 1098 additions and 937 deletions
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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