Reland "Prefix flag macros with WEBRTC_."
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}diff --git a/rtc_tools/event_log_visualizer/main.cc b/rtc_tools/event_log_visualizer/main.cc
index 2a52d06..ad90cfb 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);