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webrtc/modules/audio_coding/neteq
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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
..
include NetEq: Remove background noise fill during long expansions 2018-04-23 06:59:46 +00:00
mock Pass a real audio codec pair ID to decoders that we create 2018-03-21 13:55:18 +00:00
test Add pcm16b quality test supporting 48khz. 2018-02-02 17:18:06 +00:00
tools Revert "Create new API for RtcEventLogParser." 2018-04-25 14:23:14 +00:00
accelerate.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
accelerate.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
audio_decoder_unittest.cc Optional: Use nullopt and implicit construction in /modules/audio_coding 2017-11-17 11:58:37 +00:00
audio_multi_vector.cc Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
audio_multi_vector.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
audio_multi_vector_unittest.cc Move some numeric utility code from rtc_base/ to rtc_base/numerics/ 2017-11-22 11:21:47 +00:00
audio_vector.cc Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
audio_vector.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
audio_vector_unittest.cc Move some numeric utility code from rtc_base/ to rtc_base/numerics/ 2017-11-22 11:21:47 +00:00
background_noise.cc NetEq: Remove background noise fill during long expansions 2018-04-23 06:59:46 +00:00
background_noise.h NetEq: Remove background noise fill during long expansions 2018-04-23 06:59:46 +00:00
background_noise_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
buffer_level_filter.cc NetEq: Fix an UBSan error 2017-10-23 11:56:47 +00:00
buffer_level_filter.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
buffer_level_filter_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
comfort_noise.cc Stop using LOG macros in favor of RTC_ prefixed macros. 2017-11-09 11:56:32 +00:00
comfort_noise.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
comfort_noise_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
cross_correlation.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
cross_correlation.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
decision_logic.cc Remove dependencies on modules:module_api from AudioProcessing. 2018-04-12 22:05:27 +00:00
decision_logic.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
decision_logic_fax.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
decision_logic_fax.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
decision_logic_normal.cc Remove dependencies on modules:module_api from AudioProcessing. 2018-04-12 22:05:27 +00:00
decision_logic_normal.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
decision_logic_unittest.cc Pass a real audio codec pair ID to decoders that we create 2018-03-21 13:55:18 +00:00
decoder_database.cc Remove incompatiblities with absl::optional in audio_coding 2018-04-17 12:05:13 +00:00
decoder_database.h Pass a real audio codec pair ID to decoders that we create 2018-03-21 13:55:18 +00:00
decoder_database_unittest.cc Pass a real audio codec pair ID to decoders that we create 2018-03-21 13:55:18 +00:00
defines.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
delay_manager.cc Fix for overflow bug in histogram scaling function in NetEq. 2017-12-11 17:01:36 +00:00
delay_manager.h NetEq: Drop unnecessary dependency on the audio decoder implementations 2017-10-16 12:57:47 +00:00
delay_manager_unittest.cc Fix for overflow bug in histogram scaling function in NetEq. 2017-12-11 17:01:36 +00:00
delay_peak_detector.cc Move some numeric utility code from rtc_base/ to rtc_base/numerics/ 2017-11-22 11:21:47 +00:00
delay_peak_detector.h Added experiment to improve handling of frame length changes in NetEq. 2017-10-13 13:26:57 +00:00
delay_peak_detector_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
dsp_helper.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
dsp_helper.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
dsp_helper_unittest.cc Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
dtmf_buffer.cc Stop using LOG macros in favor of RTC_ prefixed macros. 2017-11-09 11:56:32 +00:00
dtmf_buffer.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
dtmf_buffer_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
dtmf_tone_generator.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
dtmf_tone_generator.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
dtmf_tone_generator_unittest.cc Add decibel conversion functions to //common_audio:common_audio 2018-02-16 10:46:48 +00:00
expand.cc NetEq: Remove background noise fill during long expansions 2018-04-23 06:59:46 +00:00
expand.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
expand_uma_logger.cc Add new histograms WebRTC.Audio.(Speech)ExpandRatePercent 2018-04-10 21:32:55 +00:00
expand_uma_logger.h Add new histograms WebRTC.Audio.(Speech)ExpandRatePercent 2018-04-10 21:32:55 +00:00
expand_unittest.cc Move some numeric utility code from rtc_base/ to rtc_base/numerics/ 2017-11-22 11:21:47 +00:00
merge.cc NetEq: Guarding against reading outside of memory 2018-02-26 09:30:00 +00:00
merge.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
merge_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
nack_tracker.cc Remove dependencies on modules:module_api from AudioProcessing. 2018-04-12 22:05:27 +00:00
nack_tracker.h Remove dependencies on modules:module_api from AudioProcessing. 2018-04-12 22:05:27 +00:00
nack_tracker_unittest.cc Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
neteq.cc NetEq: Remove background noise fill during long expansions 2018-04-23 06:59:46 +00:00
neteq_decoder_enum.cc Optional: Use nullopt and implicit construction in /modules/audio_coding 2017-11-17 11:58:37 +00:00
neteq_decoder_enum.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
neteq_external_decoder_unittest.cc Remove dependencies on modules:module_api from AudioProcessing. 2018-04-12 22:05:27 +00:00
neteq_impl.cc NetEq: Remove background noise fill during long expansions 2018-04-23 06:59:46 +00:00
neteq_impl.h NetEq: Remove background noise fill during long expansions 2018-04-23 06:59:46 +00:00
neteq_impl_unittest.cc Remove dependencies on modules:module_api from AudioProcessing. 2018-04-12 22:05:27 +00:00
neteq_network_stats_unittest.cc Remove dependencies on modules:module_api from AudioProcessing. 2018-04-12 22:05:27 +00:00
neteq_stereo_unittest.cc Remove dependencies on modules:module_api from AudioProcessing. 2018-04-12 22:05:27 +00:00
neteq_unittest.cc NetEq: Remove background noise fill during long expansions 2018-04-23 06:59:46 +00:00
neteq_unittest.proto Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
normal.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
normal.h Move some numeric utility code from rtc_base/ to rtc_base/numerics/ 2017-11-22 11:21:47 +00:00
normal_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
packet.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
packet.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
packet_buffer.cc Optional: Use nullopt and implicit construction in /modules/audio_coding 2017-11-17 11:58:37 +00:00
packet_buffer.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
packet_buffer_unittest.cc Pass a real audio codec pair ID to decoders that we create 2018-03-21 13:55:18 +00:00
post_decode_vad.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
post_decode_vad.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
post_decode_vad_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
preemptive_expand.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
preemptive_expand.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
random_vector.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
random_vector.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
random_vector_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
red_payload_splitter.cc Move some numeric utility code from rtc_base/ to rtc_base/numerics/ 2017-11-22 11:21:47 +00:00
red_payload_splitter.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
red_payload_splitter_unittest.cc Pass a real audio codec pair ID to decoders that we create 2018-03-21 13:55:18 +00:00
rtcp.cc Remove dependencies on modules:module_api from AudioProcessing. 2018-04-12 22:05:27 +00:00
rtcp.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
statistics_calculator.cc Adds voice concealment periods reporting to neteq_rtpplay. 2018-02-07 18:41:42 +00:00
statistics_calculator.h Adds voice concealment periods reporting to neteq_rtpplay. 2018-02-07 18:41:42 +00:00
statistics_calculator_unittest.cc NetEq: Simplify the dependencies of GetNetworkStatistics 2017-09-25 20:32:12 +00:00
sync_buffer.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
sync_buffer.h Remove dependencies on modules:module_api from AudioProcessing. 2018-04-12 22:05:27 +00:00
sync_buffer_unittest.cc Move some numeric utility code from rtc_base/ to rtc_base/numerics/ 2017-11-22 11:21:47 +00:00
tick_timer.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
tick_timer.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
tick_timer_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
time_stretch.cc Move some numeric utility code from rtc_base/ to rtc_base/numerics/ 2017-11-22 11:21:47 +00:00
time_stretch.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
time_stretch_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
timestamp_scaler.cc NetEq: Drop unnecessary dependency on the audio decoder implementations 2017-10-16 12:57:47 +00:00
timestamp_scaler.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
timestamp_scaler_unittest.cc Pass a real audio codec pair ID to decoders that we create 2018-03-21 13:55:18 +00:00