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2018-06-26Revert "Add Timestamp accessor methods to the EncodedImage class."Björn Terelius
This reverts commit f34d467b03da4f20a1d036a20966fcad43d2433f. Reason for revert: Seems to break downstream project. Original change's description: > Add Timestamp accessor methods to the EncodedImage class. > > Bug: webrtc:9378 > Change-Id: I59bf14f631f92f0f4e05f60d4af25641a23a53f9 > Reviewed-on: https://webrtc-review.googlesource.com/82100 > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > Reviewed-by: Philip Eliasson <philipel@webrtc.org> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > Commit-Queue: Niels Moller <nisse@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#23734} TBR=brandtr@webrtc.org,nisse@webrtc.org,stefan@webrtc.org,philipel@webrtc.org Change-Id: I3aa0c0119426886bc583c918aae862eb7f4b6b63 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:9378 Reviewed-on: https://webrtc-review.googlesource.com/85600 Reviewed-by: Björn Terelius <terelius@webrtc.org> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23739}
2018-06-26Add Timestamp accessor methods to the EncodedImage class.Niels Möller
Bug: webrtc:9378 Change-Id: I59bf14f631f92f0f4e05f60d4af25641a23a53f9 Reviewed-on: https://webrtc-review.googlesource.com/82100 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23734}
2018-06-26Re-introduce a read of a bool in APM fuzzersSam Zackrisson
This slightly increases fuzzer coverage of the APM. (.25 % points more line coverage.) Bug: webrtc:9413 Change-Id: Ic992423f1dcf34fa0aa9649c8035a8e48b0ccdb2 Reviewed-on: https://webrtc-review.googlesource.com/85342 Reviewed-by: Alex Loiko <aleloi@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23732}
2018-06-21Fix for VP9 K-SVC video freeze frame when send bandwidth is restricted.“Michael
Added distinction between number of configured and number of actively encoded spatial layers and include number of actively encoded spatial layers in ssData. Modified layer_filtering_transport.cc test to parse from the RTP header and use the number of actively encoded spatial layers for filtering spatial video layers. Bug: webrtc:9425 Change-Id: Ic9f8895ab08b0626f9bb53a75ec33d8e7eb8706e Reviewed-on: https://webrtc-review.googlesource.com/84243 Commit-Queue: Michael Horowitz <mhoro@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23716}
2018-06-21Generalize SimulcastEncoderAdapter, use for H264 & VP8.Sergio Garcia Murillo
* Move SimulcastEncoderAdapter out under modules/video_coding * Move SimulcastRateAllocator back out to modules/video_coding/utility * Move TemporalLayers and ScreenshareLayers to modules/video_coding/utility * Move any VP8 specific code - such as temporal layer bitrate budgeting - under codec type dependent conditionals. * Plumb the simulcast index for H264 in the codec specific and RTP format data structures. TBR=sprang@webrtc.org,stefan@webrtc.org,titovartem@webrtc.org Bug: webrtc:5840 Change-Id: I2d3b130622dd7ceec5528f3ab6c46f109e6bafb8 Reviewed-on: https://webrtc-review.googlesource.com/84743 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23715}
2018-06-21Revert "Implement H264 simulcast support and generalize ↵Mirko Bonadei
SimulcastEncoderAdapter use for H264 & VP8." This reverts commit 07efe436c9002e139845f62486e3ee4e29f0d85b. Reason for revert: Breaks downstream project. cricket::GetSimulcastConfig method signature has been updated. I think you can get away with a default value for temporal_layers_supported (and then you can remove it after a few days when projects will be updated). Original change's description: > Implement H264 simulcast support and generalize SimulcastEncoderAdapter use for H264 & VP8. > > * Move SimulcastEncoderAdapter out under modules/video_coding > * Move SimulcastRateAllocator back out to modules/video_coding/utility > * Move TemporalLayers and ScreenshareLayers to modules/video_coding/utility > * Move any VP8 specific code - such as temporal layer bitrate budgeting - > under codec type dependent conditionals. > * Plumb the simulcast index for H264 in the codec specific and RTP format data structures. > > Bug: webrtc:5840 > Change-Id: Ieced8a00e38f273c1a6cfd0f5431a87d07b8f44e > Reviewed-on: https://webrtc-review.googlesource.com/64100 > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > Reviewed-by: Erik Språng <sprang@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#23705} TBR=sprang@webrtc.org,stefan@webrtc.org,mflodman@webrtc.org,hta@webrtc.org,sergio.garcia.murillo@gmail.com,titovartem@webrtc.org,agouaillard@gmail.com Change-Id: Ic9d3b1eeaf195bb5ec2063954421f5e77866d663 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:5840 Reviewed-on: https://webrtc-review.googlesource.com/84760 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23710}
2018-06-21Add AGC1 fuzzerSam Zackrisson
Fuzzes the config and audio inputs to GainControlImpl. Seems able to cover a few hundred lines of code that the APM fuzzer hasn't been able to reach. Bug: webrtc:9413 Change-Id: I32776505be9c416ec03113c12437a92dcfadd827 Reviewed-on: https://webrtc-review.googlesource.com/84589 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23709}
2018-06-21Revert "NetEq: Deprecate playout modes Fax, Off and Streaming"Henrik Lundin
This reverts commit 80c4cca4915dbc6094a5bfae749f85f7371eadd1. Reason for revert: Breaks downstream tests. Original change's description: > NetEq: Deprecate playout modes Fax, Off and Streaming > > The playout modes other than Normal have not been reachable for a long > time, other than through tests. It is time to deprecate them. > > The only meaningful use was that Fax mode was sometimes set from > tests, in order to avoid time-stretching operations (accelerate and > pre-emptive expand) from messing with the test results. With this CL, > a new config is added instead, which lets the user specify exactly > this: don't do time-stretching. > > As a result of Fax and Off modes being removed, the following code > clean-up was done: > - Fold DecisionLogicNormal into DecisionLogic. > - Remove AudioRepetition and AlternativePlc operations, since they can > no longer be reached. > > Bug: webrtc:9421 > Change-Id: I651458e9c1931a99f3b07e242817d303bac119df > Reviewed-on: https://webrtc-review.googlesource.com/84123 > Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> > Reviewed-by: Ivo Creusen <ivoc@webrtc.org> > Reviewed-by: Minyue Li <minyue@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#23704} TBR=henrik.lundin@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org Change-Id: I555aae8850fc4ac1ea919bfa72c11b5218066f30 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:9421 Reviewed-on: https://webrtc-review.googlesource.com/84680 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23706}
2018-06-21Implement H264 simulcast support and generalize SimulcastEncoderAdapter use ↵Sergio Garcia Murillo
for H264 & VP8. * Move SimulcastEncoderAdapter out under modules/video_coding * Move SimulcastRateAllocator back out to modules/video_coding/utility * Move TemporalLayers and ScreenshareLayers to modules/video_coding/utility * Move any VP8 specific code - such as temporal layer bitrate budgeting - under codec type dependent conditionals. * Plumb the simulcast index for H264 in the codec specific and RTP format data structures. Bug: webrtc:5840 Change-Id: Ieced8a00e38f273c1a6cfd0f5431a87d07b8f44e Reviewed-on: https://webrtc-review.googlesource.com/64100 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23705}
2018-06-21NetEq: Deprecate playout modes Fax, Off and StreamingHenrik Lundin
The playout modes other than Normal have not been reachable for a long time, other than through tests. It is time to deprecate them. The only meaningful use was that Fax mode was sometimes set from tests, in order to avoid time-stretching operations (accelerate and pre-emptive expand) from messing with the test results. With this CL, a new config is added instead, which lets the user specify exactly this: don't do time-stretching. As a result of Fax and Off modes being removed, the following code clean-up was done: - Fold DecisionLogicNormal into DecisionLogic. - Remove AudioRepetition and AlternativePlc operations, since they can no longer be reached. Bug: webrtc:9421 Change-Id: I651458e9c1931a99f3b07e242817d303bac119df Reviewed-on: https://webrtc-review.googlesource.com/84123 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23704}
2018-06-21Remove nonlinear beamformer API from APMSam Zackrisson
This CL removes the remaining beamformer parts from the APM. Bug: webrtc:9402 Change-Id: I9ab2795bd2813d17166ed0925125257b82d98a74 Reviewed-on: https://webrtc-review.googlesource.com/83340 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23694}
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}
2018-06-20Delete pre_decode_callback.Niels Möller
Only user was the replay.cc tool, when dumping frames to a file. It is changed to instead inject a special decoder. Bug: None Change-Id: I521fbba1a0ef440cff7d786f6f4c6397e33f764f Reviewed-on: https://webrtc-review.googlesource.com/83121 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23675}
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}
2018-06-19Cover AecDump calls in APM fuzzer.Alex Loiko
This is done by attaching a mocked AecDump to APM in the APM fuzzer. It gives higher fuzzer coverage. BEFORE: #4905 DONE cov: 7739 ft: 46097 corp: 4093/387Mb lim: 4 exec/s: 3 rss: 504Mb AFTER: #4905 DONE cov: 8130 ft: 47662 corp: 4099/386Mb lim: 4 exec/s: 3 rss: 524Mb Bug: webrtc:7820 Change-Id: If8bae9bfd7aca08f1873e2440ae65a2e74ba3a6b Reviewed-on: https://webrtc-review.googlesource.com/84127 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23656}
2018-06-19Always enable 'delay-agnostic' in APM fuzzer.Alex Loiko
This 'fixes' a bug in the non-delay-agnostic code by not fuzzing it. We plan to always enable the delay-agnostic feature. In Chrome, delay-agnostic mode is always on: https://cs.chromium.org/chromium/src/content/renderer/media/stream/media_stream_audio_processor.cc?l=579 Bug: chromium:824638 webrtc:9423 Change-Id: I3d9cac2bc11857fd55549d13c52db4c99dec956c Reviewed-on: https://webrtc-review.googlesource.com/83984 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23651}
2018-06-19Fuzz AEC field trial killswitchesSam Zackrisson
The fuzzer data is used to configure the field trials of the AEC. This increases fuzzer coverage of modules/audio_processing/aec3/ by roughly 500 lines of code, ~ 3 % points increase in APM coverage for desktop Chrome. Bug: webrtc:9413 Change-Id: Iea9059747a8492a7ca2091a359e7883750c45b27 Reviewed-on: https://webrtc-review.googlesource.com/83732 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23650}
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}
2018-06-15Fix a downstream test failure.Ying Wang
In rare case the packets number may loop around and in the same FEC-protected group the packet sequence number became out of order. Bug: chromium:850493 Change-Id: Ice82aafd537e0edc1dbdb8b934e11e7c42a4cf60 Reviewed-on: https://webrtc-review.googlesource.com/82802 Commit-Queue: Ying Wang <yinwa@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23633}
2018-06-15Delete an unneeded include of pathutils.h.Niels Möller
TBR: phoglund@webrtc.org Bug: webrtc:6424 Change-Id: Idc70ecf9093786307cccec624f1edf11542afa6b Reviewed-on: https://webrtc-review.googlesource.com/83724 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23626}
2018-06-14Change echo detector to scoped_refptrIvo Creusen
The echo detector is currently stored as a unique_ptr, but when injecting an echo detector, a scoped_refptr makes more sense since the ownership will be shared. Bug: webrtc:8732 Change-Id: I2180014acb84f1cd5c361864a444b7b6574520f5 Reviewed-on: https://webrtc-review.googlesource.com/83325 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23610}
2018-06-13Use enum class for VideoCodecMode and VideoCodecComplexity.Niels Möller
Bug: webrtc:7660 Change-Id: I6a8ef01f8abcc25c8efaf0af387408343a7c8ba3 Reviewed-on: https://webrtc-review.googlesource.com/81240 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23595}
2018-06-11Document that preferred VideoFrame constructor takes no RTP timestamp.Niels Möller
And update most internal calls to use it. Bug: webrtc:5740, webrtc:9372 Change-Id: Ib57d4ebfa7b0729af6d22981a792f0fdadf8a13f Reviewed-on: https://webrtc-review.googlesource.com/81743 Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23567}
2018-06-08Revert "Move class VideoCodec from common_types.h to its own api header file."Danil Chapovalov
This reverts commit efc71e565e9b36bcdfb4571f59e34bbd8fabd0cd. Reason for revert: probably breaks downstream test Original change's description: > Move class VideoCodec from common_types.h to its own api header file. > > Bug: webrtc:7660 > Change-Id: I91f19bfc2565461328f30081f8383e136419aefb > Reviewed-on: https://webrtc-review.googlesource.com/79881 > Commit-Queue: Niels Moller <nisse@webrtc.org> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#23544} TBR=danilchap@webrtc.org,brandtr@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org Change-Id: Id8bd37c79c2f8d09a4d88368765230103f1db2c8 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:7660 Reviewed-on: https://webrtc-review.googlesource.com/82101 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23547}
2018-06-08Move class VideoCodec from common_types.h to its own api header file.Niels Möller
Bug: webrtc:7660 Change-Id: I91f19bfc2565461328f30081f8383e136419aefb Reviewed-on: https://webrtc-review.googlesource.com/79881 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23544}
2018-06-07Removing warning suppression flags from test/.Mirko Bonadei
Bug: webrtc:9251 Change-Id: Ibe3971adcc13d30d4a3360ecfe6d525e02428b28 Reviewed-on: https://webrtc-review.googlesource.com/81188 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23533}
2018-06-07Delete RTP-specific values from the VideoCodecType enum.Niels Möller
Bug: None Change-Id: Icd6a03f4dc7cfe074ba1e0370ed40938f0f1d7ed Reviewed-on: https://webrtc-review.googlesource.com/80442 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23527}
2018-06-04Delete enum RtpVideoCodecTypes, replaced with VideoCodecType.Niels Möller
Bug: webrtc:8995 Change-Id: I0b44aa26f2f6a81aec7ca1281b8513d8e03228b8 Reviewed-on: https://webrtc-review.googlesource.com/79561 Reviewed-by: Sami Kalliomäki <sakal@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23507}
2018-06-01Add JoinFilename to testsupport code, replacing use of rtc::Pathname.Niels Möller
This is a partial revert of https://codereview.webrtc.org/2533213005, deleting rtc::File methods accepting an rtc::Pathname argument. Bug: webrtc:6424 Change-Id: Ib16bdc7294dbddfa12ba9ae206c024ff97e529a4 Reviewed-on: https://webrtc-review.googlesource.com/80180 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23489}
2018-06-01Update packet_buffer_fuzzer to fuzz full packets.philipel
Bug: webrtc:7728 Change-Id: I9d33404470c2ecf8d6f91c57c9dc9fd4dd821a18 Reviewed-on: https://webrtc-review.googlesource.com/77424 Commit-Queue: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23485}
2018-05-31Refactor SimulcastTestUtility into SimulcastTestFixture{,Impl}Rasmus Brandt
This will allow exposing the interface to downstream users that want to test VP8 simulcast. No functional changes to the tests themselves are expected. Bug: webrtc:9281 Change-Id: I4128b8f35a4412c5b330cf55c8dc0e173d4570da Reviewed-on: https://webrtc-review.googlesource.com/77361 Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23469}
2018-05-24Move socklen_t definition for windows to win32.h.Niels Möller
Bug: webrtc:6853 Change-Id: Ie73cd959707b32b928acdabd46329830b2bb2c27 Reviewed-on: https://webrtc-review.googlesource.com/78720 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23381}
2018-05-23Limit input length for SDP fuzzer.Patrik Höglund
This limits the SDP to 16KB, which sounds enough. Bug: chromium:813328 Change-Id: I58c7b3e073108fd7b3495e8182b5c632e9619fe7 Reviewed-on: https://webrtc-review.googlesource.com/78280 Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23360}
2018-05-21Move VideoStreamEncoderInterface to api/.Niels Möller
Bug: webrtc:8830 Change-Id: I17908b4ef6a043acf22e2110b9672012d5fa7fc0 Reviewed-on: https://webrtc-review.googlesource.com/74481 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23334}
2018-05-21Delete unneeded includes of basictypes.h.Niels Möller
This is a kitchen-sink header, some pieces should be moved to byteorder.h, the rest likely deleted. Delete most includes of basictypes.h. In leaf headers, include stddef.h and stdint.h explicitly where needed. Bug: webrtc:6853 Change-Id: Ibc809936a8f94d418e4eb650da1e89c1b9142073 Reviewed-on: https://webrtc-review.googlesource.com/77721 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23333}
2018-05-18Allows injection of network controller factory into peer connection factory.Sebastian Jansson
Bug: webrtc:9155 Change-Id: I0a17024042f154297aba20f5d2dc766feb27f3f7 Reviewed-on: https://webrtc-review.googlesource.com/73123 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23313}
2018-05-18Move VideoEncoderConfig from call/ to api/.Niels Möller
Bug: webrtc:8830 Change-Id: I42abd45bff9a70fe00733424b34874925c523dc8 Reviewed-on: https://webrtc-review.googlesource.com/77683 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23303}
2018-05-17Fuzzing for video_coding::FrameBuffer2.philipel
Bug: webrtc:7728 Change-Id: I712289a82d408dde1db73a1cc44f0c69a6b639ff Reviewed-on: https://webrtc-review.googlesource.com/31841 Commit-Queue: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23282}
2018-05-17Merge DegradationPreference enums.Taylor Brandstetter
This replaces webrtc::VideoSendStream::DegradationPreference with webrtc::DegradationPreference, and adds "DISABLED". It's still not wired up from RtpSenderInterface::SetParameters to the underlying video engine; that would be the next step. Bug: webrtc:8830 Change-Id: I582ffd04eaef33c73d9892e52e789804c933b864 Reviewed-on: https://webrtc-review.googlesource.com/77024 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23276}
2018-05-17RtpFrameReferenceFinder fuzzer.philipel
Bug: webrtc:7728 Change-Id: I641772837384a4d8070db2138b93f4157f997d03 Reviewed-on: https://webrtc-review.googlesource.com/74584 Commit-Queue: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23271}
2018-05-16Delete unused header file unittest_utils.h.Niels Möller
Became unused with cl https://webrtc-review.googlesource.com/40740, and last spurious reference dropped in cl https://webrtc-review.googlesource.com/43360. Bug: None Change-Id: Ib5f27a437c1ae8f7cc2df84d6d9eda8c297481d6 Reviewed-on: https://webrtc-review.googlesource.com/76981 Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23255}
2018-05-15Control inter-layer prediction mode in test apps.Sergey Silkin
Vp9 encoder supports several inter-layer prediction modes. This adds possibility to control and test them in video/ss/sv loopback. Filtering of sent packets has been modified. In addition to high spatial and temporal layers it now filters out packets of low spatial layers where non_ref_for_inter_layer_pred bit is set to true. Bug: none Change-Id: I17b1ee8f1ac1d70a6914eb86d153790ef2da9679 Reviewed-on: https://webrtc-review.googlesource.com/76540 Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23233}
2018-05-14Fix build errors when rtc_use_builtin_sw_codecs is set to false.Anders Carlsson
The previous effort of building WebRTC without SW codecs stopped when libjingle_peerconnection was possible to build. In order to make the group("default") target build, this basically updates a bunch of tests to explicitly depend on the built-in software video codecs. Bug: webrtc:7925 Change-Id: I2715414770c197fca01cb8dbde173a21f4434500 Reviewed-on: https://webrtc-review.googlesource.com/70503 Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Commit-Queue: Anders Carlsson <andersc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23216}
2018-05-14New file api/video/BUILD.gnNiels Möller
Build targets involving files under api/video/ are moved into this file, from api/BUILD.gn. In addition, drop "_api" part of target names, and move the header file api/videosinkinterface.h to api/video/video_sink_interface.h. Bug: webrtc:9253 Change-Id: I2896d3f063db8dff902bc29738578395b2fcc155 Reviewed-on: https://webrtc-review.googlesource.com/75500 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23207}
2018-05-09Create a fuzzer for the Opus encoderHenrik Lundin
The fuzzer is very simple. It only considers the default encoder configuration at this point. Bug: chromium:826914 Change-Id: Ifa248a1dba80efb231807750e40082ec5580636a Reviewed-on: https://webrtc-review.googlesource.com/75261 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23192}
2018-05-09Removing -Wno-comment.Mirko Bonadei
Chromium is suppressing this warning only on GCC [1], so WebRTC should not suppress it on clang and just rely on Chromium's defaults. [1] - https://cs.chromium.org/chromium/src/build/config/compiler/BUILD.gn?l=1356&rcl=027d7fa1c191f60f754985b9c235597f8c9a2081 Bug: webrtc:9251 Change-Id: I9316cbdda4083da7d859ff0b9c60579546ddbfcb Reviewed-on: https://webrtc-review.googlesource.com/75301 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23183}
2018-05-08Deprecate RTPFragmentationHeader argument to VideoDecoder::DecodeNiels Möller
Intend to delete in a later cl. Bug: webrtc:6471 Change-Id: Icf0fcd40e0d3287dc59b684fae6552b40b47204a Reviewed-on: https://webrtc-review.googlesource.com/39511 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23162}
2018-04-25Moving demux from FakeNetworkPipe to DirectTransport.Sebastian Jansson
This CL moves the responsibility for demuxing from FakeNetworkPipe to DirectTransport. This makes the interface for FakeNetworkPipe more consistent. It exposes fewer different interfaces for different usages. It also means that any time degradations applied to the packets due in FakeNetworkPipe in tests will now be propagated to Call in a more realistic manner. Previously the time was set to uninitialized which meant that Call filled in values based on the system clock. Bug: webrtc:9054 Change-Id: Ie534062f5ae9ad992c06b19e43804138a35702f0 Reviewed-on: https://webrtc-review.googlesource.com/64260 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23017}
2018-04-23Move BitrateAllocation to api/ and rename it VideoBitrateAllocationErik Språng
Since the webrtc_common build target does not have visibility set, we cannot easily use BitrateAllocation in other parts of Chromium. This is currently blocking parts of chromium:794608, and I know of other usage outside webrtc already, so moving it to api/ should be warranted. Also, since there's some naming confusion and this class is video specific rename it VideoBitrateAllocation. This also fits with the standard interface for producing these: VideoBitrateAllocator. Bug: chromium:794608 Change-Id: I4c0fae40f9365e860c605a76a4f67ecc9b9cf9fe Reviewed-on: https://webrtc-review.googlesource.com/70783 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22986}
2018-04-19Reland "Move creating encoder to VideoStreamEncoder."Niels Möller
This is a reland of fb82fcc7f9c414dc8ba1ddd314e9524fee54cb80 Original change's description: > Move creating encoder to VideoStreamEncoder. > > This used to be in WebRtcVideoChannel::WebRtcVideoSendStream. > One implication is that encoder is not created until the first > frame arrives, and some of the tests needed updates to emit a > frame or two. > > Bug: webrtc:8830 > Change-Id: I78169b2bb4dfa4197b4b4229af9fd69d0f747835 > Reviewed-on: https://webrtc-review.googlesource.com/64885 > Commit-Queue: Niels Moller <nisse@webrtc.org> > Reviewed-by: Erik Språng <sprang@webrtc.org> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22905} TBR=magjed@webrtc.org,kwiberg@webrtc.org Bug: webrtc:8830 Change-Id: I9565095ea1880fb49d15111198c08b2fcb84f18c Reviewed-on: https://webrtc-review.googlesource.com/70740 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22930}
generated by cgit v1.2.3 (git 2.25.1) at 2024-07-17 14:52:41 +0000