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}
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Mirko Bonadei
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2dfa998be2
@ -20,151 +20,173 @@
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#include "test/field_trial.h"
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#include "test/testsupport/fileutils.h"
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DEFINE_string(plot_profile,
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"default",
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"A profile that selects a certain subset of the plots. Currently "
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"defined profiles are \"all\", \"none\", \"sendside_bwe\","
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"\"receiveside_bwe\" and \"default\"");
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WEBRTC_DEFINE_string(
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plot_profile,
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"default",
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"A profile that selects a certain subset of the plots. Currently "
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"defined profiles are \"all\", \"none\", \"sendside_bwe\","
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"\"receiveside_bwe\" and \"default\"");
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DEFINE_bool(plot_incoming_packet_sizes,
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false,
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"Plot bar graph showing the size of each incoming packet.");
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DEFINE_bool(plot_outgoing_packet_sizes,
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false,
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"Plot bar graph showing the size of each outgoing packet.");
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DEFINE_bool(plot_incoming_packet_count,
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false,
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"Plot the accumulated number of packets for each incoming stream.");
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DEFINE_bool(plot_outgoing_packet_count,
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false,
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"Plot the accumulated number of packets for each outgoing stream.");
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DEFINE_bool(plot_audio_playout,
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false,
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"Plot bar graph showing the time between each audio playout.");
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DEFINE_bool(plot_audio_level,
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false,
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"Plot line graph showing the audio level of incoming audio.");
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DEFINE_bool(plot_incoming_sequence_number_delta,
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false,
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"Plot the sequence number difference between consecutive incoming "
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"packets.");
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DEFINE_bool(
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WEBRTC_DEFINE_bool(plot_incoming_packet_sizes,
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false,
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"Plot bar graph showing the size of each incoming packet.");
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WEBRTC_DEFINE_bool(plot_outgoing_packet_sizes,
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false,
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"Plot bar graph showing the size of each outgoing packet.");
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WEBRTC_DEFINE_bool(
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plot_incoming_packet_count,
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false,
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"Plot the accumulated number of packets for each incoming stream.");
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WEBRTC_DEFINE_bool(
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plot_outgoing_packet_count,
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false,
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"Plot the accumulated number of packets for each outgoing stream.");
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WEBRTC_DEFINE_bool(
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plot_audio_playout,
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false,
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"Plot bar graph showing the time between each audio playout.");
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WEBRTC_DEFINE_bool(
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plot_audio_level,
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false,
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"Plot line graph showing the audio level of incoming audio.");
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WEBRTC_DEFINE_bool(
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plot_incoming_sequence_number_delta,
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false,
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"Plot the sequence number difference between consecutive incoming "
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"packets.");
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WEBRTC_DEFINE_bool(
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plot_incoming_delay_delta,
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false,
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"Plot the difference in 1-way path delay between consecutive packets.");
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DEFINE_bool(plot_incoming_delay,
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true,
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"Plot the 1-way path delay for incoming packets, normalized so "
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"that the first packet has delay 0.");
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DEFINE_bool(plot_incoming_loss_rate,
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true,
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"Compute the loss rate for incoming packets using a method that's "
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"similar to the one used for RTCP SR and RR fraction lost. Note "
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"that the loss rate can be negative if packets are duplicated or "
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"reordered.");
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DEFINE_bool(plot_incoming_bitrate,
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true,
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"Plot the total bitrate used by all incoming streams.");
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DEFINE_bool(plot_outgoing_bitrate,
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true,
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"Plot the total bitrate used by all outgoing streams.");
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DEFINE_bool(plot_incoming_stream_bitrate,
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true,
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"Plot the bitrate used by each incoming stream.");
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DEFINE_bool(plot_outgoing_stream_bitrate,
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true,
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"Plot the bitrate used by each outgoing stream.");
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DEFINE_bool(plot_simulated_receiveside_bwe,
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false,
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"Run the receive-side bandwidth estimator with the incoming rtp "
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"packets and plot the resulting estimate.");
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DEFINE_bool(plot_simulated_sendside_bwe,
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false,
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"Run the send-side bandwidth estimator with the outgoing rtp and "
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"incoming rtcp and plot the resulting estimate.");
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DEFINE_bool(plot_network_delay_feedback,
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true,
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"Compute network delay based on sent packets and the received "
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"transport feedback.");
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DEFINE_bool(plot_fraction_loss_feedback,
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true,
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"Plot packet loss in percent for outgoing packets (as perceived by "
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"the send-side bandwidth estimator).");
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DEFINE_bool(plot_pacer_delay,
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false,
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"Plot the time each sent packet has spent in the pacer (based on "
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"the difference between the RTP timestamp and the send "
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"timestamp).");
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DEFINE_bool(plot_timestamps,
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false,
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"Plot the rtp timestamps of all rtp and rtcp packets over time.");
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DEFINE_bool(plot_rtcp_details,
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false,
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"Plot the contents of all report blocks in all sender and receiver "
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"reports. This includes fraction lost, cumulative number of lost "
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"packets, extended highest sequence number and time since last "
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"received SR.");
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DEFINE_bool(plot_audio_encoder_bitrate_bps,
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false,
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"Plot the audio encoder target bitrate.");
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DEFINE_bool(plot_audio_encoder_frame_length_ms,
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false,
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"Plot the audio encoder frame length.");
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DEFINE_bool(
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WEBRTC_DEFINE_bool(
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plot_incoming_delay,
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true,
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"Plot the 1-way path delay for incoming packets, normalized so "
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"that the first packet has delay 0.");
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WEBRTC_DEFINE_bool(
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plot_incoming_loss_rate,
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true,
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"Compute the loss rate for incoming packets using a method that's "
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"similar to the one used for RTCP SR and RR fraction lost. Note "
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"that the loss rate can be negative if packets are duplicated or "
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"reordered.");
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WEBRTC_DEFINE_bool(plot_incoming_bitrate,
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true,
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"Plot the total bitrate used by all incoming streams.");
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WEBRTC_DEFINE_bool(plot_outgoing_bitrate,
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true,
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"Plot the total bitrate used by all outgoing streams.");
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WEBRTC_DEFINE_bool(plot_incoming_stream_bitrate,
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true,
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"Plot the bitrate used by each incoming stream.");
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WEBRTC_DEFINE_bool(plot_outgoing_stream_bitrate,
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true,
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"Plot the bitrate used by each outgoing stream.");
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WEBRTC_DEFINE_bool(
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plot_simulated_receiveside_bwe,
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false,
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"Run the receive-side bandwidth estimator with the incoming rtp "
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"packets and plot the resulting estimate.");
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WEBRTC_DEFINE_bool(
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plot_simulated_sendside_bwe,
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false,
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"Run the send-side bandwidth estimator with the outgoing rtp and "
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"incoming rtcp and plot the resulting estimate.");
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WEBRTC_DEFINE_bool(
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plot_network_delay_feedback,
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true,
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"Compute network delay based on sent packets and the received "
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"transport feedback.");
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WEBRTC_DEFINE_bool(
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plot_fraction_loss_feedback,
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true,
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"Plot packet loss in percent for outgoing packets (as perceived by "
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"the send-side bandwidth estimator).");
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WEBRTC_DEFINE_bool(
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plot_pacer_delay,
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false,
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"Plot the time each sent packet has spent in the pacer (based on "
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"the difference between the RTP timestamp and the send "
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"timestamp).");
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WEBRTC_DEFINE_bool(
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plot_timestamps,
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false,
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"Plot the rtp timestamps of all rtp and rtcp packets over time.");
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WEBRTC_DEFINE_bool(
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plot_rtcp_details,
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false,
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"Plot the contents of all report blocks in all sender and receiver "
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"reports. This includes fraction lost, cumulative number of lost "
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"packets, extended highest sequence number and time since last "
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"received SR.");
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WEBRTC_DEFINE_bool(plot_audio_encoder_bitrate_bps,
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false,
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"Plot the audio encoder target bitrate.");
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WEBRTC_DEFINE_bool(plot_audio_encoder_frame_length_ms,
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false,
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"Plot the audio encoder frame length.");
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WEBRTC_DEFINE_bool(
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plot_audio_encoder_packet_loss,
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false,
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"Plot the uplink packet loss fraction which is sent to the audio encoder.");
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DEFINE_bool(plot_audio_encoder_fec, false, "Plot the audio encoder FEC.");
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DEFINE_bool(plot_audio_encoder_dtx, false, "Plot the audio encoder DTX.");
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DEFINE_bool(plot_audio_encoder_num_channels,
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false,
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"Plot the audio encoder number of channels.");
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DEFINE_bool(plot_neteq_stats, false, "Plot the NetEq statistics.");
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DEFINE_bool(plot_ice_candidate_pair_config,
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false,
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"Plot the ICE candidate pair config events.");
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DEFINE_bool(plot_ice_connectivity_check,
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false,
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"Plot the ICE candidate pair connectivity checks.");
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WEBRTC_DEFINE_bool(plot_audio_encoder_fec,
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false,
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"Plot the audio encoder FEC.");
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WEBRTC_DEFINE_bool(plot_audio_encoder_dtx,
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false,
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"Plot the audio encoder DTX.");
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WEBRTC_DEFINE_bool(plot_audio_encoder_num_channels,
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false,
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"Plot the audio encoder number of channels.");
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WEBRTC_DEFINE_bool(plot_neteq_stats, false, "Plot the NetEq statistics.");
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WEBRTC_DEFINE_bool(plot_ice_candidate_pair_config,
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false,
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"Plot the ICE candidate pair config events.");
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WEBRTC_DEFINE_bool(plot_ice_connectivity_check,
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false,
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"Plot the ICE candidate pair connectivity checks.");
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DEFINE_string(
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WEBRTC_DEFINE_string(
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force_fieldtrials,
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"",
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"Field trials control experimental feature code which can be forced. "
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"E.g. running with --force_fieldtrials=WebRTC-FooFeature/Enabled/"
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" will assign the group Enabled to field trial WebRTC-FooFeature. Multiple "
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"trials are separated by \"/\"");
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DEFINE_string(wav_filename,
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"",
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"Path to wav file used for simulation of jitter buffer");
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DEFINE_bool(help, false, "prints this message");
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WEBRTC_DEFINE_string(wav_filename,
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"",
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"Path to wav file used for simulation of jitter buffer");
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WEBRTC_DEFINE_bool(help, false, "prints this message");
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DEFINE_bool(show_detector_state,
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false,
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"Show the state of the delay based BWE detector on the total "
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"bitrate graph");
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WEBRTC_DEFINE_bool(
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show_detector_state,
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false,
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"Show the state of the delay based BWE detector on the total "
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"bitrate graph");
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DEFINE_bool(show_alr_state,
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false,
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"Show the state ALR state on the total bitrate graph");
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WEBRTC_DEFINE_bool(show_alr_state,
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false,
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"Show the state ALR state on the total bitrate graph");
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DEFINE_bool(parse_unconfigured_header_extensions,
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true,
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"Attempt to parse unconfigured header extensions using the default "
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"WebRTC mapping. This can give very misleading results if the "
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"application negotiates a different mapping.");
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WEBRTC_DEFINE_bool(
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parse_unconfigured_header_extensions,
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true,
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"Attempt to parse unconfigured header extensions using the default "
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"WebRTC mapping. This can give very misleading results if the "
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"application negotiates a different mapping.");
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DEFINE_bool(print_triage_alerts,
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false,
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"Print triage alerts, i.e. a list of potential problems.");
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WEBRTC_DEFINE_bool(print_triage_alerts,
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false,
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"Print triage alerts, i.e. a list of potential problems.");
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DEFINE_bool(normalize_time,
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true,
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"Normalize the log timestamps so that the call starts at time 0.");
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WEBRTC_DEFINE_bool(
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normalize_time,
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true,
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"Normalize the log timestamps so that the call starts at time 0.");
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DEFINE_bool(protobuf_output,
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false,
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"Output charts as protobuf instead of python code.");
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WEBRTC_DEFINE_bool(protobuf_output,
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false,
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"Output charts as protobuf instead of python code.");
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void SetAllPlotFlags(bool setting);
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