webrtc/modules/audio_coding/neteq/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
..
audio_checksum.h Use generic MessageDigest class instead of MD5 / SHA-1 specific classes. 2017-12-21 12:39:50 +00:00
audio_loop.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
audio_loop.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
audio_sink.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
audio_sink.h Remove dependencies on modules:module_api from AudioProcessing. 2018-04-12 22:05:27 +00:00
constant_pcm_packet_source.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
constant_pcm_packet_source.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
DEPS Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
encode_neteq_input.cc Move some numeric utility code from rtc_base/ to rtc_base/numerics/ 2017-11-22 11:21:47 +00:00
encode_neteq_input.h Remove dependencies on modules:module_api from AudioProcessing. 2018-04-12 22:05:27 +00:00
fake_decode_from_file.cc Move some numeric utility code from rtc_base/ to rtc_base/numerics/ 2017-11-22 11:21:47 +00:00
fake_decode_from_file.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
input_audio_file.cc Fix for crash when reading from audio file in NetEq. 2018-03-23 18:29:05 +00:00
input_audio_file.h Replacing the legacy tool RTPencode with a new rtp_encode 2017-11-24 09:05:48 +00:00
input_audio_file_unittest.cc Move some numeric utility code from rtc_base/ to rtc_base/numerics/ 2017-11-22 11:21:47 +00:00
neteq_delay_analyzer.cc Remove dependencies on modules:module_api from AudioProcessing. 2018-04-12 22:05:27 +00:00
neteq_delay_analyzer.h Add code to generate python visualization to neteq_rtpplay 2017-12-06 10:52:42 +00:00
neteq_external_decoder_test.cc Remove dependencies on modules:module_api from AudioProcessing. 2018-04-12 22:05:27 +00:00
neteq_external_decoder_test.h Remove dependencies on modules:module_api from AudioProcessing. 2018-04-12 22:05:27 +00:00
neteq_input.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
neteq_input.h Optional: Use nullopt and implicit construction in /modules/audio_coding 2017-11-17 11:58:37 +00:00
neteq_packet_source_input.cc Optional: Use nullopt and implicit construction in /modules/audio_coding 2017-11-17 11:58:37 +00:00
neteq_packet_source_input.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
neteq_performance_test.cc Remove dependencies on modules:module_api from AudioProcessing. 2018-04-12 22:05:27 +00:00
neteq_performance_test.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
neteq_quality_test.cc Adds fixed PL loss mode to neteq_quality_test. 2018-02-13 15:34:04 +00:00
neteq_quality_test.h Remove dependencies on modules:module_api from AudioProcessing. 2018-04-12 22:05:27 +00:00
neteq_replacement_input.cc Optional: Use nullopt and implicit construction in /modules/audio_coding 2017-11-17 11:58:37 +00:00
neteq_replacement_input.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
neteq_rtpplay.cc Seperate NetEq stats getter to use in other tools. 2018-04-23 08:49:06 +00:00
neteq_stats_getter.cc Let NetEq stats getter provide time for each stats query. 2018-04-23 12:53:26 +00:00
neteq_stats_getter.h Let NetEq stats getter provide time for each stats query. 2018-04-23 12:53:26 +00:00
neteq_test.cc Adds voice concealment periods reporting to neteq_rtpplay. 2018-02-07 18:41:42 +00:00
neteq_test.h Adds voice concealment periods reporting to neteq_rtpplay. 2018-02-07 18:41:42 +00:00
output_audio_file.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
output_wav_file.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
packet.cc Remove dependencies on modules:module_api from AudioProcessing. 2018-04-12 22:05:27 +00:00
packet.h Fix circular dependencies in webrtc_common. 2017-12-15 14:33:26 +00:00
packet_source.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
packet_source.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
packet_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
resample_input_audio_file.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
resample_input_audio_file.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
rtc_event_log_source.cc Revert "Create new API for RtcEventLogParser." 2018-04-25 14:23:14 +00:00
rtc_event_log_source.h Revert "Create new API for RtcEventLogParser." 2018-04-25 14:23:14 +00:00
rtp_analyze.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtp_encode.cc Remove dependencies on modules:module_api from AudioProcessing. 2018-04-12 22:05:27 +00:00
rtp_file_source.cc Remove a couple of unnecessary winsock2.h includes 2018-04-03 08:49:58 +00:00
rtp_file_source.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
rtp_generator.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtp_generator.h Fix circular dependencies in webrtc_common. 2017-12-15 14:33:26 +00:00
rtp_jitter.cc Replacing the legacy tool RTPjitter with a new rtp_jitter 2017-11-24 13:38:59 +00:00
rtpcat.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00