webrtc/rtc_tools
Björn Terelius ff61273c01 Revert "Create new API for RtcEventLogParser."
This reverts commit 9e336ec0b8.

Reason for revert: Code can accidentally include the deprecated parser but link with the new one, or vice versa. Reverting to fix naming.

Original change's description:
> Create new API for RtcEventLogParser.
> 
> The new API stores events gathered by event type. For example, it is
> possible to ask fo a list of all incoming RTCP messages or all audio
> playout events.
> 
> The new API is experimental and may change over next few weeks. Once
> it has stabilized and all unit tests and existing tools have been
> ported to the new API, the old one will be removed.
> 
> This CL also updates the event_log_visualizer tool to use the new
> parser API. This is not a funcional change except for:
> - Incoming and outgoing audio level are now drawn in two separate plots.
> - Incoming and outgoing timstamps are now drawn in two separate plots.
> - RTCP count is no longer split into Video and Audio. It also counts
>   all RTCP packets rather than only specific message types.
> - Slight timing difference in sendside BWE simulation due to only
>   iterating over transport feedbacks and not over all RTCP packets.
>   This timing changes are not visible in the plots.
> 
> 
> Media type for RTCP messages might not be identified correctly by
> rtc_event_log2text anymore. On the other hand, assigning a specific
> media type to an RTCP packet was a bit hacky to begin with.
> 
> Bug: webrtc:8111
> Change-Id: I8e7168302beb69b2e163a097a2a142b86dd4a26b
> Reviewed-on: https://webrtc-review.googlesource.com/60865
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23015}

TBR=terelius@webrtc.org,srte@webrtc.org,minyue@webrtc.org

Change-Id: Ib4bbcf0563423675a3cc1dce59ebf665e0c5dae9
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8111
Reviewed-on: https://webrtc-review.googlesource.com/72500
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23026}
2018-04-25 14:23:14 +00:00
..
agc Remove dependencies on modules:module_api from AudioProcessing. 2018-04-12 22:05:27 +00:00
barcode_tools Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
converter Normalize main(..) routines for WinUWP 2017-12-12 14:32:56 +00:00
event_log_visualizer Revert "Create new API for RtcEventLogParser." 2018-04-25 14:23:14 +00:00
force_mic_volume_max Normalize main(..) routines for WinUWP 2017-12-12 14:32:56 +00:00
frame_analyzer Store video_quality_loopback_test perf results in Chart JSON format. 2018-01-30 16:48:59 +00:00
frame_editing Normalize main(..) routines for WinUWP 2017-12-12 14:32:56 +00:00
loopback_test Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
network_tester Add missing includes checks.h/array_view.h 2018-04-10 10:33:34 +00:00
psnr_ssim_analyzer Normalize main(..) routines for WinUWP 2017-12-12 14:32:56 +00:00
py_event_log_analyzer Fix path to proto in py_event_log_analyzer/pb_parse.py 2018-03-19 07:42:35 +00:00
rtcbot Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
testing Fix path to AppRTC/collider on Windows 2018-04-04 14:47:40 +00:00
unpack_aecdump Options and settings for the Pre-amplifier. 2018-04-16 12:25:48 +00:00
author_line_count.sh Fix awk gensub call in author_line_count.sh script. 2017-12-06 08:16:12 +00:00
BUILD.gn Revert "Create new API for RtcEventLogParser." 2018-04-25 14:23:14 +00:00
class_usage.sh Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
compare_videos.py Store video_quality_loopback_test perf results in Chart JSON format. 2018-01-30 16:48:59 +00:00
DEPS Including libyuv headers using fully qualified paths. 2017-12-11 15:51:26 +00:00
header_usage.sh header_usage.sh: Keep leading directory for grep. 2017-10-19 13:46:51 +00:00
OWNERS Expand ownership of rtc_tools. 2017-11-06 11:13:26 +00:00
sanitizers_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
simple_command_line_parser.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
simple_command_line_parser.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
simple_command_line_parser_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
video_analysis.py Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
video_analysis_test.py Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00