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2018-08-11Add UTC time to init event in AEC debug dump.Minyue Li
Bug: webrtc:9616 Change-Id: I1350212f0b8835fb64427483269da96d51670c01 Reviewed-on: https://webrtc-review.googlesource.com/92620 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Commit-Queue: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24267}
2018-08-10Clean up code regarding jitter buffer plot in event log visualizer.Minyue Li
Bug: webrtc:9147 Change-Id: I2c1f0b383706ae9a788eb8b5d308d4c7fe612730 Reviewed-on: https://webrtc-review.googlesource.com/92390 Commit-Queue: Minyue Li <minyue@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24261}
2018-08-08Remove definition and usage of macro GTEST_RELATIVE_PATH.Mirko Bonadei
The macro GTEST_RELATIVE_PATH is obsolete and since it is always defined this CL just removes it. Bug: webrtc:9564 Change-Id: Ieafa5b77351c4df87864588ba6b3de8f60d54e89 Reviewed-on: https://webrtc-review.googlesource.com/92080 Reviewed-by: Oleh Prypin <oprypin@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24226}
2018-08-06Remove unnecessary //base:base_java dependencies.Sami Kalliomäki
WebRTC code shouldn't depend on Chromium Android base code. Bug: None Change-Id: Ie094f26e4ee855769c9c5276bbb47242aae9c217 Reviewed-on: https://webrtc-review.googlesource.com/92387 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Commit-Queue: Sami Kalliomäki <sakal@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24194}
2018-07-25Delete root header file typedef.h.Niels Möller
Usage replaced with stdint.h, rtc_base/system/arch.h and rtc_base/system/unused.h, as appropriate. Bug: webrtc:6854 Change-Id: I97225465d14b969903d92979e2df3c3c05d35f18 Reviewed-on: https://webrtc-review.googlesource.com/90249 Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24100}
2018-07-24Plot RTCP SR and RR contents in event_log_visualizer.Bjorn Terelius
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. Bug: None Change-Id: Ifbded689a666da140c468e11c33b6c6f99a3041e Reviewed-on: https://webrtc-review.googlesource.com/90247 Reviewed-by: Alex Narest <alexnarest@webrtc.org> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24083}
2018-07-20Enable clang::find_bad_constructs for rtc_tools (part 1/2).Mirko Bonadei
This CL removes //build/config/clang:find_bad_constructs from the suppressed_configs list, which means that clang:find_bad_constructs is now enabled on these translation units. Bug: webrtc:9251, webrtc:163 Change-Id: I9c26b6129db24263f1aada9561f477db64091049 Reviewed-on: https://webrtc-review.googlesource.com/89742 Reviewed-by: Oleh Prypin <oprypin@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24051}
2018-07-06Break out Agc code from audio_processing.Alex Loiko
Splits 'modules/audio_processing:audio_processing' target. The files in modules/audio_processing/agc now are in targets in that folder. Reason for doing this was to include modules/audio_processing/agc/agc.h from another target in the dependent CL https://webrtc-review.googlesource.com/c/src/+/86603 This could help reducing the binary size in the future. Bug: webrtc:7494 Change-Id: I61f50ab6d5ce24d19f4097e0f3fa8b0170010887 Reviewed-on: https://webrtc-review.googlesource.com/87422 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23873}
2018-07-05Use absl::make_unique and absl::WrapUnique directlyKarl Wiberg
Instead of going through our wrappers in ptr_util.h. This CL was generated by the following script: git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",' git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g' git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g' git checkout -- rtc_base/ptr_util{.h,_unittest.cc} git cl format Followed by manually adding dependencies on //third_party/abseil-cpp/absl/memory until `gn check` stopped complaining. Bug: webrtc:9473 Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c Reviewed-on: https://webrtc-review.googlesource.com/86600 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-02Removes redundant delay based bwe.Sebastian Jansson
This removes the legacy DelayBasedBwe to reduce code redundancy and avoid the risk of applying changes on only one version. Bug: webrtc:8415 Change-Id: I88aba03adbb77ee0ff0a97a8b3be6ddf028af48a Reviewed-on: https://webrtc-review.googlesource.com/85364 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23798}
2018-06-26Adding NetEq lifetime stats to event log visualizer.Minyue Li
Bug: webrtc:9147 Change-Id: I798f8ac41192182d50df6fe98fbe56c8cb7f294c Reviewed-on: https://webrtc-review.googlesource.com/85340 Commit-Queue: Minyue Li <minyue@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23738}
2018-06-25Adding jitter buffer plots for all SSRCs in event log visualizer.Minyue Li
Bug: webrtc:9147 Change-Id: I64291666d329c026f35ecf1c4245b192794441fe Reviewed-on: https://webrtc-review.googlesource.com/84745 Commit-Queue: Minyue Li <minyue@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23726}
2018-06-21Aligning time in audio jitter buffer plot to other plots in rtc event log ↵Minyue Li
visualizer. Bug: webrtc:9147 Change-Id: I4ddb3e93ea04a11a68e097ecad731d6d9d6842a9 Reviewed-on: https://webrtc-review.googlesource.com/75322 Reviewed-by: Björn Terelius <terelius@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23712}
2018-06-20Removing usage of //build/config/compiler:no_size_t_to_int_warning.Mirko Bonadei
Bug: webrtc:9251, webrtc:1348 Change-Id: I76e52abbfab5666cad73044b49172a9799539108 Reviewed-on: https://webrtc-review.googlesource.com/84144 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23686}
2018-06-19Reformat the WebRTC code baseYves Gerey
Running clang-format with chromium's style guide. The goal is n-fold: * providing consistency and readability (that's what code guidelines are for) * preventing noise with presubmit checks and git cl format * building on the previous point: making it easier to automatically fix format issues * you name it Please consider using git-hyper-blame to ignore this commit. Bug: webrtc:9340 Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87 Reviewed-on: https://webrtc-review.googlesource.com/81185 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23660}
2018-06-18Replace rtc::Optional with absl::optional in test and rtc_toolsDanil Chapovalov
This is a no-op change because rtc::Optional is an alias to absl::optional This CL generated by running script with parameters 'test rtc_tools' find $@ -type f \( -name \*.h -o -name \*.cc \) \ -exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \ -exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \ -exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+ find $@ -type f -name BUILD.gn \ -exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+; git cl format Bug: webrtc:9078 Change-Id: Ibb43c737f4c45fe300736382b0dd2d8ab32c6377 Reviewed-on: https://webrtc-review.googlesource.com/83944 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23642}
2018-06-12Remove manual references to exe_and_shlib_depsTom Anderson
After [1], a manual dependency on exe_and_shlib_deps is no longer necessary since it's automatically added. This CL removes all remaining manual references to exe_and_shlib_deps. [1] https://chromium.googlesource.com/chromium/src.git/+/d7ed1f0a9c28c932fddc834ca5de44f28266c7f5 BUG=chromium:845700 R=tommi@webrtc.org Change-Id: I92942bc08c0e34c5c39df3c71f56f89476f8d95c Reviewed-on: https://webrtc-review.googlesource.com/83061 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23573}
2018-05-29Split LoggedBweProbeResult into -Success and -Failure.Bjorn Terelius
Also change ParsedEventLog::EventType to enum class. Bug: webrtc:8111 Change-Id: I4747fb9cbcbdb963fa032770078218e5b416b3da Reviewed-on: https://webrtc-review.googlesource.com/79280 Commit-Queue: Björn Terelius <terelius@webrtc.org> Reviewed-by: Elad Alon <eladalon@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23432}
2018-05-25Add support for visualizing event logs without normalizing time.Stefan Holmer
Bug: webrtc:9299 Change-Id: Icdc4cba14f143cedb7c35347dd9711ab13f975d8 Reviewed-on: https://webrtc-review.googlesource.com/77820 Commit-Queue: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23392}
2018-05-24Replaces redundant congestion controller componentsSebastian Jansson
This CL replaces components in the congestion controller module that are identical to equivalent components in the rtp and goog_cc subfolder. Some redundant components are left as they were not trivial to replace. Bug: webrtc:8415 Change-Id: I86a1f164d7b100b8ec8ba7dbc1c9bda2128a4f37 Reviewed-on: https://webrtc-review.googlesource.com/78521 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23384}
2018-05-14Make unpack_aecdump optionally unpack render/capture call orderSam Zackrisson
It is stored in a text file as a stream of 'r' and 'c' characters - render and capture. This is the format output by APM with apm_debug_dump on, and it is readable by audioproc_f. Bug: webrtc:9252 Change-Id: I01e9e104ed7e3fb45e623730343a0c2addc81d1b Reviewed-on: https://webrtc-review.googlesource.com/75502 Reviewed-by: Oleh Prypin <oprypin@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23213}
2018-05-07Plot NetEq stats in RTC event log visualizer.Minyue Li
Bug: webrtc:9147 Change-Id: I61ec7bc5299201e25e1efc503b73b84d5be3ebbf Reviewed-on: https://webrtc-review.googlesource.com/71740 Commit-Queue: Minyue Li <minyue@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23151}
2018-05-04Split NetEq simulation and jitter buffer plot to be able to plot other ↵Minyue Li
metrics in the simulation. Bug: webrtc:9147 Change-Id: Ied37dedd19fc24a48700fb01645cee6288d3efa7 Reviewed-on: https://webrtc-review.googlesource.com/70160 Commit-Queue: Minyue Li <minyue@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23125}
2018-04-27Reland "Create new API for RtcEventLogParser."Bjorn Terelius
The new API stores events gathered by event type. For example, it is possible to ask for 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: Ib244338c86a2c1a010c668a7aba440482023b512 Reviewed-on: https://webrtc-review.googlesource.com/73140 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23056}
2018-04-25Revert "Create new API for RtcEventLogParser."Björn Terelius
This reverts commit 9e336ec0b8a77c3461d13677cff3563c11c88daa. 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}
2018-04-25Create new API for RtcEventLogParser.Bjorn Terelius
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}
2018-04-23GN hack to tag targets as poisonous (and use it with audio codecs)Karl Wiberg
Only specially taggged targets may transitively depend on poisonous targets. We first apply it to audio codecs. This makes it much clearer exactly what parts of the code still have dependencies on the audio codecs (and we want to eventually get rid of pretty much all of them). Bug: webrtc:8396, webrtc:9121 Change-Id: Iba5c2e806c702b5cfe881022674705f647896d43 Reviewed-on: https://webrtc-review.googlesource.com/69520 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22979}
2018-04-20Correcting payload size to NetEq simulator in RTC event log analyzer.Minyue Li
Bug: webrtc:9171, b/77841364 Change-Id: Ia56b61df1cb824d9d1bf9ec7d93770082803b642 Reviewed-on: https://webrtc-review.googlesource.com/71140 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Commit-Queue: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22948}
2018-04-16Options and settings for the Pre-amplifier.Alex Loiko
Add configuration fields for the pre-amplifier in the Audio Processing Module. Also add flags and settings for the pre-amplifier in audioproc_f. Also make the setting stored in Aec Dumps. And make the setting applied when playing back Aec Dumps in audioproc_f. Bug: webrtc:9138 Change-Id: I4e59df200e1ebc56f06fae74ebf17d85858958a3 Reviewed-on: https://webrtc-review.googlesource.com/69560 Reviewed-by: Oleh Prypin <oprypin@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22876}
2018-04-12Remove dependencies on modules:module_api from AudioProcessing.Fredrik Solenberg
- Directly include api/audio/audio_frame.h everywhere AudioFrame is used. - This *will* remove transient dependencies on libjpeg and a bunch of other things from the e.g. APM. - audio_frame.h still included from module_common_types.h for backwards compatibility with clients. Bug: webrtc:9139, webrtc:7504 Change-Id: Id96f9268c01667fbcc29a01f5c1dd25a37836897 Reviewed-on: https://webrtc-review.googlesource.com/62464 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22845}
2018-04-10Add missing includes checks.h/array_view.hDanil Chapovalov
instead of relying on optional.h to included these 2 headers. Bug: webrtc:9078 Change-Id: I7a4b3facd81690b8f107640487e129986c1f5ff6 Reviewed-on: https://webrtc-review.googlesource.com/68602 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22803}
2018-04-04Fix path to AppRTC/collider on WindowsOleh Prypin
Bug: webrtc:7602 No-Try: True Change-Id: I4d8f254e1316481f35638a1a2882275dfec2b5c1 Reviewed-on: https://webrtc-review.googlesource.com/66860 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Oleh Prypin <oprypin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22735}
2018-04-04Bypass browser join confirmation in prebuilt AppRTCOleh Prypin
This is still needed by Chromium tests. Copied from https://webrtc.googlesource.com/webrtc.DEPS/+/76533443ed95184aa45dc3b4af383fc301a53f80/copy_apprtc.py Bug: webrtc:7602 Change-Id: I17f0159fe43176df95ad2e27ff330650d6645d67 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/66680 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Oleh Prypin <oprypin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22718}
2018-04-04Use sys.executable to launch another Python scriptOleh Prypin
To make setup_apprtc.py work on Windows Bug: webrtc:7602 Change-Id: I17c19c1cb8b2b71dafd90ae5f8be80e50c3397e9 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/66660 Reviewed-by: Oleh Prypin <oprypin@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Oleh Prypin <oprypin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22715}
2018-03-19Fix path to proto in py_event_log_analyzer/pb_parse.pyOleh Prypin
Bug: chromium:611808 No-Try: True Change-Id: I173f0270a07896d9edddfef6b68592e6b404ecab Reviewed-on: https://webrtc-review.googlesource.com/62680 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Oleh Prypin <oprypin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22481}
2018-03-16Remove the public_deps to fileutils from test_support.Patrik Höglund
Bug: webrtc:8946 Change-Id: Ia01d8bb1b42485e29f26792b9266228743d7fd90 No-Presubmit: true Reviewed-on: https://webrtc-review.googlesource.com/62100 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22465}
2018-03-15Split up audio_device build targetPaulina Hensman
We currently have one build target containing everything for audio_device: the interfaces, the "fine" audio buffer, and the actual implementations for each platform. Since we are planning to move the Android implementation to the sdk/android folder, we only want to depend on the interfaces and the "fine" audio buffer, not the other platform specific implementations. This CL splits the audio_device target into three different targets: the interfaces, the fine audio buffer, and the platform specific implementations. The default audio_device target now points to the interfaces instead. Bug: webrtc:7452 Change-Id: I57e849cc6f4087d950fa02d969ecc682934839cd Reviewed-on: https://webrtc-review.googlesource.com/61321 Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22452}
2018-03-07Separate build target for rtc_base/numerics/safe_minmax.hKarl Wiberg
So that we can avoid dependency cycles. Bug: none Change-Id: I821d9f1319dff01403d6e4e310cbb2d4b2b125e8 Reviewed-on: https://webrtc-review.googlesource.com/60500 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22328}
2018-03-02Update prebuilt AppRTCOleh Prypin
Version cfb35d9212a06c1dfc31149c54f196708287a149 Google Cloud SDK 191.0.0 Node v8.9.4 Bug: None No-Try: True Change-Id: Ia488899e28bbee37db5cae0ada76729c349ba7dd Reviewed-on: https://webrtc-review.googlesource.com/59324 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Oleh Prypin <oprypin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22265}
2018-02-20Preparing for task queue in congenstion controllerSebastian Jansson
This cl prepares for a later CL introducing a new send side congestion controller that will run on a task queue. It mostly consists of minor fixes but adds some new interfaces that are unused in practice. Bug: webrtc:8415 Change-Id: I1b58d0180a18eb15320d18733dac0dfe2e0f902a Reviewed-on: https://webrtc-review.googlesource.com/53321 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22099}
2018-02-16Removing access to send side cc in rtp controller.Sebastian Jansson
This CL removes direct access to SendSideCongestionController (SSCC) via the RtpTransportControllerSend interface and replaces all usages with calls on RtpTransportControllerSend which will in turn calls SSCC. This prepares for later refactor of RtpTransportControllerSend. Bug: webrtc:8415 Change-Id: I68363a3ab0203b95579f747402a1e7f58a5eeeb5 Reviewed-on: https://webrtc-review.googlesource.com/53860 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22044}
2018-02-14Revert "Revert "Revert "Reland "Moved congestion controller to task queue.""""Sebastian Jansson
This reverts commit 65792c5a5c542201f7b9feefded505842692e6ed. Reason for revert: <INSERT REASONING HERE> Original change's description: > Revert "Revert "Reland "Moved congestion controller to task queue.""" > > This reverts commit 4e849f6925b2ac44b0957a228d7131fc391fca54. > > Reason for revert: <INSERT REASONING HERE> > > Original change's description: > > Revert "Reland "Moved congestion controller to task queue."" > > > > This reverts commit 57daeb7ac7f3d80992905b53fea500953fcfd793. > > > > Reason for revert: Cause increased congestion and deadlocks in downstream project > > > > Original change's description: > > > Reland "Moved congestion controller to task queue." > > > > > > This is a reland of 0cbcba7ea0dced1a7f353c64d6cf91d46ccb29f9. > > > > > > Original change's description: > > > > Moved congestion controller to task queue. > > > > > > > > The goal of this work is to make it easier to experiment with the > > > > bandwidth estimation implementation. For this reason network control > > > > functionality is moved from SendSideCongestionController(SSCC), > > > > PacedSender and BitrateController to the newly created > > > > GoogCcNetworkController which implements the newly created > > > > NetworkControllerInterface. This allows the implementation to be > > > > replaced at runtime in the future. > > > > > > > > This is the first part of a split of a larger CL, see: > > > > https://webrtc-review.googlesource.com/c/src/+/39788/8 > > > > For further explanations. > > > > > > > > Bug: webrtc:8415 > > > > Change-Id: I770189c04cc31b313bd4e57821acff55fbcb1ad3 > > > > Reviewed-on: https://webrtc-review.googlesource.com/43840 > > > > Commit-Queue: Sebastian Jansson <srte@webrtc.org> > > > > Reviewed-by: Björn Terelius <terelius@webrtc.org> > > > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > > > Cr-Commit-Position: refs/heads/master@{#21868} > > > > > > Bug: webrtc:8415 > > > Change-Id: I1d1756a30deed5b421b1c91c1918a13b6bb455da > > > Reviewed-on: https://webrtc-review.googlesource.com/48000 > > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > > Commit-Queue: Sebastian Jansson <srte@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#21899} > > > > TBR=terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org > > > > # Not skipping CQ checks because original CL landed > 1 day ago. > > > > Bug: webrtc:8415 > > Change-Id: Ida8074dcac2cc28b3629228eb22846d8a8e81b83 > > Reviewed-on: https://webrtc-review.googlesource.com/52980 > > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#22017} > > TBR=danilchap@webrtc.org,terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org > > Change-Id: I3393b74370c4f4d0955f50728005b2b925be169b > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8415 > Reviewed-on: https://webrtc-review.googlesource.com/53262 > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > Commit-Queue: Sebastian Jansson <srte@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22023} TBR=danilchap@webrtc.org,terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org Change-Id: Id68ad986ee51142b7be3381d0793709b4392fe2c No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8415 Reviewed-on: https://webrtc-review.googlesource.com/53360 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22024}
2018-02-14Revert "Revert "Reland "Moved congestion controller to task queue."""Sebastian Jansson
This reverts commit 4e849f6925b2ac44b0957a228d7131fc391fca54. Reason for revert: <INSERT REASONING HERE> Original change's description: > Revert "Reland "Moved congestion controller to task queue."" > > This reverts commit 57daeb7ac7f3d80992905b53fea500953fcfd793. > > Reason for revert: Cause increased congestion and deadlocks in downstream project > > Original change's description: > > Reland "Moved congestion controller to task queue." > > > > This is a reland of 0cbcba7ea0dced1a7f353c64d6cf91d46ccb29f9. > > > > Original change's description: > > > Moved congestion controller to task queue. > > > > > > The goal of this work is to make it easier to experiment with the > > > bandwidth estimation implementation. For this reason network control > > > functionality is moved from SendSideCongestionController(SSCC), > > > PacedSender and BitrateController to the newly created > > > GoogCcNetworkController which implements the newly created > > > NetworkControllerInterface. This allows the implementation to be > > > replaced at runtime in the future. > > > > > > This is the first part of a split of a larger CL, see: > > > https://webrtc-review.googlesource.com/c/src/+/39788/8 > > > For further explanations. > > > > > > Bug: webrtc:8415 > > > Change-Id: I770189c04cc31b313bd4e57821acff55fbcb1ad3 > > > Reviewed-on: https://webrtc-review.googlesource.com/43840 > > > Commit-Queue: Sebastian Jansson <srte@webrtc.org> > > > Reviewed-by: Björn Terelius <terelius@webrtc.org> > > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#21868} > > > > Bug: webrtc:8415 > > Change-Id: I1d1756a30deed5b421b1c91c1918a13b6bb455da > > Reviewed-on: https://webrtc-review.googlesource.com/48000 > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > Commit-Queue: Sebastian Jansson <srte@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21899} > > TBR=terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org > > # Not skipping CQ checks because original CL landed > 1 day ago. > > Bug: webrtc:8415 > Change-Id: Ida8074dcac2cc28b3629228eb22846d8a8e81b83 > Reviewed-on: https://webrtc-review.googlesource.com/52980 > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22017} TBR=danilchap@webrtc.org,terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org Change-Id: I3393b74370c4f4d0955f50728005b2b925be169b No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8415 Reviewed-on: https://webrtc-review.googlesource.com/53262 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22023}
2018-02-14Revert "Reland "Moved congestion controller to task queue.""Danil Chapovalov
This reverts commit 57daeb7ac7f3d80992905b53fea500953fcfd793. Reason for revert: Cause increased congestion and deadlocks in downstream project Original change's description: > Reland "Moved congestion controller to task queue." > > This is a reland of 0cbcba7ea0dced1a7f353c64d6cf91d46ccb29f9. > > Original change's description: > > Moved congestion controller to task queue. > > > > The goal of this work is to make it easier to experiment with the > > bandwidth estimation implementation. For this reason network control > > functionality is moved from SendSideCongestionController(SSCC), > > PacedSender and BitrateController to the newly created > > GoogCcNetworkController which implements the newly created > > NetworkControllerInterface. This allows the implementation to be > > replaced at runtime in the future. > > > > This is the first part of a split of a larger CL, see: > > https://webrtc-review.googlesource.com/c/src/+/39788/8 > > For further explanations. > > > > Bug: webrtc:8415 > > Change-Id: I770189c04cc31b313bd4e57821acff55fbcb1ad3 > > Reviewed-on: https://webrtc-review.googlesource.com/43840 > > Commit-Queue: Sebastian Jansson <srte@webrtc.org> > > Reviewed-by: Björn Terelius <terelius@webrtc.org> > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21868} > > Bug: webrtc:8415 > Change-Id: I1d1756a30deed5b421b1c91c1918a13b6bb455da > Reviewed-on: https://webrtc-review.googlesource.com/48000 > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > Commit-Queue: Sebastian Jansson <srte@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21899} TBR=terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org # Not skipping CQ checks because original CL landed > 1 day ago. Bug: webrtc:8415 Change-Id: Ida8074dcac2cc28b3629228eb22846d8a8e81b83 Reviewed-on: https://webrtc-review.googlesource.com/52980 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22017}
2018-02-14Revert "Moved congestion controller to goog_cc folder."Danil Chapovalov
This reverts commit e6cefdf9c572cdce55ff0497ad6e516c76132ee8. Reason for revert: conflicts with reverting https://webrtc-review.googlesource.com/c/src/+/52980 Original change's description: > Moved congestion controller to goog_cc folder. > > Bug: webrtc:8415 > Change-Id: I2070da0cacf1dbfc4b6a89285af3e68fd03497ab > Reviewed-on: https://webrtc-review.googlesource.com/43841 > Commit-Queue: Sebastian Jansson <srte@webrtc.org> > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21928} TBR=terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org # Not skipping CQ checks because original CL landed > 1 day ago. Bug: webrtc:8415 Change-Id: If8965e1e745e57694192b9ca2a69503c722658d9 Reviewed-on: https://webrtc-review.googlesource.com/53020 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22012}
2018-02-07Delete macro RTC_ACCESS_ON, replaced by RTC_GUARDED_BY.Niels Möller
Both macros do the same thing, as wrappers for __attribute__((guarded_by)), and more names for the same thing doesn't add to clarity. Bug: none Change-Id: Iaaf7b21dbf3345ee90fee22c39b636823d195eb0 Reviewed-on: https://webrtc-review.googlesource.com/48361 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21929}
2018-02-07Moved congestion controller to goog_cc folder.Sebastian Jansson
Bug: webrtc:8415 Change-Id: I2070da0cacf1dbfc4b6a89285af3e68fd03497ab Reviewed-on: https://webrtc-review.googlesource.com/43841 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21928}
2018-02-06Reland "Moved congestion controller to task queue."Sebastian Jansson
This is a reland of 0cbcba7ea0dced1a7f353c64d6cf91d46ccb29f9. Original change's description: > Moved congestion controller to task queue. > > The goal of this work is to make it easier to experiment with the > bandwidth estimation implementation. For this reason network control > functionality is moved from SendSideCongestionController(SSCC), > PacedSender and BitrateController to the newly created > GoogCcNetworkController which implements the newly created > NetworkControllerInterface. This allows the implementation to be > replaced at runtime in the future. > > This is the first part of a split of a larger CL, see: > https://webrtc-review.googlesource.com/c/src/+/39788/8 > For further explanations. > > Bug: webrtc:8415 > Change-Id: I770189c04cc31b313bd4e57821acff55fbcb1ad3 > Reviewed-on: https://webrtc-review.googlesource.com/43840 > Commit-Queue: Sebastian Jansson <srte@webrtc.org> > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21868} Bug: webrtc:8415 Change-Id: I1d1756a30deed5b421b1c91c1918a13b6bb455da Reviewed-on: https://webrtc-review.googlesource.com/48000 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21899}
2018-02-05Made modules/audio_processing/vad its own target.Alex Loiko
WHAT: made a BUILD.gn with library and tests in the Audio Processing Module Voice Activity Detector directory. Updated depending code. Fixed a Clang warning. WHY: to make it possible for a target to depend on just the VAD and not the whole APM. There are other benefits: * Sometimes faster compilation. * The VAD takes up 28000 bytes of libjingle_peerconnection_so.so. Making a peerconnection shared object file without the VAD has to be done in steps. The first step is a custom target for the VAD. Hence this Cl. Change-Id: Iea0207a0b5979db26baaf46b24beaefbb1c431af BUG: webrtc:5716, webrtc:7494 Reviewed-on: https://webrtc-review.googlesource.com/47521 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Oleh Prypin <oprypin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21893}
2018-02-03Bind the structured ICE logging with P2PTransportChannel.Qingsi Wang
This change list passes the instance of RtcEventLog from Peerconnection down to P2PTransportChannel, and binds the structured ICE logging with ICE layer objects. Logs of ICE connectivity checks are injected for candidate pairs. TBR=terelius@webrtc.org Bug: None Change-Id: Ia979dbbac6d31dcf0f8988da1065bdfc3e461821 Reviewed-on: https://webrtc-review.googlesource.com/34660 Commit-Queue: Qingsi Wang <qingsi@google.com> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21884}
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