文件最后提交记录最后更新时间
Delete SendDelayObserver interface send delay is now measured through SendPacketObserver interface Bug: None Change-Id: I0dc3de1522e2824d9431d7e3a3dc524588687dda Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319500 Reviewed-by: Åsa Persson <asapersson@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40755} 2 年前
Fix usages of RTC_DCHECK to GTEST macros to ensure tests pass in release builds as well Using RTC_DCHECK for test validation is wrong to begin with (gets compiled out in non-debug builds, which measn we may miss validations), but becomes extra problematic when we include code with side-effects inside the DCHECK, which results in release-build tests having a different flow than intended Bug: webrtc:15572 Change-Id: I89d5b55f903b9d93fe4a929548d1b9fcde8941be Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323182 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41005} 2 年前
Refactor AbsoluteCaptureTimeSender and AbsoluteCaptureTimeInterpolator Removed thread safety: for a low level helper it adds overhead that users may not need. In particular RtpSenderVideo doesn't need it because calls to SendVideo are already synchronized. Added a feature to force producing extension as requested by downstream. Cleanup and document api: Changed rtp_frequency type to int as it has no reason to use uint32_t per style guide Changed absolute_capture_time to NtpTime to clarify both units and offset of the time. NtpTime has trivial conversion to/from uint64_t Documented all the parameters. Cleanup tests. Bug: b/307553606 Change-Id: I0922ca4d3c89f124eeb561742dca79ed5c2327fd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325022 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Chen Xing <chxg@google.com> Cr-Commit-Position: refs/heads/main@{#41023} 2 年前
Refactor AbsoluteCaptureTimeSender and AbsoluteCaptureTimeInterpolator Removed thread safety: for a low level helper it adds overhead that users may not need. In particular RtpSenderVideo doesn't need it because calls to SendVideo are already synchronized. Added a feature to force producing extension as requested by downstream. Cleanup and document api: Changed rtp_frequency type to int as it has no reason to use uint32_t per style guide Changed absolute_capture_time to NtpTime to clarify both units and offset of the time. NtpTime has trivial conversion to/from uint64_t Documented all the parameters. Cleanup tests. Bug: b/307553606 Change-Id: I0922ca4d3c89f124eeb561742dca79ed5c2327fd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325022 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Chen Xing <chxg@google.com> Cr-Commit-Position: refs/heads/main@{#41023} 2 年前
Refactor AbsoluteCaptureTimeSender and AbsoluteCaptureTimeInterpolator Removed thread safety: for a low level helper it adds overhead that users may not need. In particular RtpSenderVideo doesn't need it because calls to SendVideo are already synchronized. Added a feature to force producing extension as requested by downstream. Cleanup and document api: Changed rtp_frequency type to int as it has no reason to use uint32_t per style guide Changed absolute_capture_time to NtpTime to clarify both units and offset of the time. NtpTime has trivial conversion to/from uint64_t Documented all the parameters. Cleanup tests. Bug: b/307553606 Change-Id: I0922ca4d3c89f124eeb561742dca79ed5c2327fd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325022 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Chen Xing <chxg@google.com> Cr-Commit-Position: refs/heads/main@{#41023} 2 年前
Refactor AbsoluteCaptureTimeSender and AbsoluteCaptureTimeInterpolator Removed thread safety: for a low level helper it adds overhead that users may not need. In particular RtpSenderVideo doesn't need it because calls to SendVideo are already synchronized. Added a feature to force producing extension as requested by downstream. Cleanup and document api: Changed rtp_frequency type to int as it has no reason to use uint32_t per style guide Changed absolute_capture_time to NtpTime to clarify both units and offset of the time. NtpTime has trivial conversion to/from uint64_t Documented all the parameters. Cleanup tests. Bug: b/307553606 Change-Id: I0922ca4d3c89f124eeb561742dca79ed5c2327fd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325022 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Chen Xing <chxg@google.com> Cr-Commit-Position: refs/heads/main@{#41023} 2 年前
Refactor AbsoluteCaptureTimeSender and AbsoluteCaptureTimeInterpolator Removed thread safety: for a low level helper it adds overhead that users may not need. In particular RtpSenderVideo doesn't need it because calls to SendVideo are already synchronized. Added a feature to force producing extension as requested by downstream. Cleanup and document api: Changed rtp_frequency type to int as it has no reason to use uint32_t per style guide Changed absolute_capture_time to NtpTime to clarify both units and offset of the time. NtpTime has trivial conversion to/from uint64_t Documented all the parameters. Cleanup tests. Bug: b/307553606 Change-Id: I0922ca4d3c89f124eeb561742dca79ed5c2327fd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325022 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Chen Xing <chxg@google.com> Cr-Commit-Position: refs/heads/main@{#41023} 2 年前
Refactor AbsoluteCaptureTimeSender and AbsoluteCaptureTimeInterpolator Removed thread safety: for a low level helper it adds overhead that users may not need. In particular RtpSenderVideo doesn't need it because calls to SendVideo are already synchronized. Added a feature to force producing extension as requested by downstream. Cleanup and document api: Changed rtp_frequency type to int as it has no reason to use uint32_t per style guide Changed absolute_capture_time to NtpTime to clarify both units and offset of the time. NtpTime has trivial conversion to/from uint64_t Documented all the parameters. Cleanup tests. Bug: b/307553606 Change-Id: I0922ca4d3c89f124eeb561742dca79ed5c2327fd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325022 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Chen Xing <chxg@google.com> Cr-Commit-Position: refs/heads/main@{#41023} 2 年前
In DependencyDescriptor rtp header extension drop partial chain support i.e. when chain are used, require each decode target to be protected by some chain. where previously it was allowed to mark decode target as unprotected. See https://github.com/AOMediaCodec/av1-rtp-spec/pull/125 Bug: webrtc:10342 Change-Id: Ia2800036e890db44bb1162abfa1a497ff68f3b24 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178807 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31772} 5 年前
Propagate active decode targets bitmask into DependencyDescriptor Bug: webrtc:10342 Change-Id: I5e8a204881b94fe5786b14e27cefce2fe056e91b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178140 Reviewed-by: Björn Terelius <terelius@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31579} 5 年前
In DependencyDescriptor rtp header extension drop partial chain support i.e. when chain are used, require each decode target to be protected by some chain. where previously it was allowed to mark decode target as unprotected. See https://github.com/AOMediaCodec/av1-rtp-spec/pull/125 Bug: webrtc:10342 Change-Id: Ia2800036e890db44bb1162abfa1a497ff68f3b24 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178807 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31772} 5 年前
rtp_rtcp/source: fix some minor typos Bug: None Change-Id: Iedc6e3b7e0cb92256255afc4cd76c66b01099c1e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/310080 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40362} 2 年前
Format almost everything. This CL was generated by running git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \ grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \ grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \ grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \ grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \ | xargs clang-format -i ; git cl format Most of these changes are clang-format grouping and reordering includes differently. Bug: webrtc:9340 Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28505} 6 年前
Method for converting q32 to TimeDelta in capture clock offset updater In change https://webrtc-review.googlesource.com/c/src/+/319961, I changed a error. Also the same code will be added for video to enable Glass 2 Glass metric for Android. To me it make sense to add this method, and then change the audio code and video code to use it. Bug: None Change-Id: Id5d38c3bb8266213a93e67ceb82e88d65f29de53 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320080 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Olov Brändström <brandstrom@google.com> Cr-Commit-Position: refs/heads/main@{#40745} 2 年前
Method for converting q32 to TimeDelta in capture clock offset updater In change https://webrtc-review.googlesource.com/c/src/+/319961, I changed a error. Also the same code will be added for video to enable Glass 2 Glass metric for Android. To me it make sense to add this method, and then change the audio code and video code to use it. Bug: None Change-Id: Id5d38c3bb8266213a93e67ceb82e88d65f29de53 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320080 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Olov Brändström <brandstrom@google.com> Cr-Commit-Position: refs/heads/main@{#40745} 2 年前
Method for converting q32 to TimeDelta in capture clock offset updater In change https://webrtc-review.googlesource.com/c/src/+/319961, I changed a error. Also the same code will be added for video to enable Glass 2 Glass metric for Android. To me it make sense to add this method, and then change the audio code and video code to use it. Bug: None Change-Id: Id5d38c3bb8266213a93e67ceb82e88d65f29de53 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320080 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Olov Brändström <brandstrom@google.com> Cr-Commit-Position: refs/heads/main@{#40745} 2 年前
Add codec name H265 to support H265 in WebRTC Bug: webrtc:13485 Change-Id: I352b15a65867f0d56fc8e9a9e03081bd3258108e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316283 Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40773} 2 年前
Add implementations of the VideoRtpDepacketizer interface while suboptimal, these implementions are complete and allow to swap code from using RtpDepacketizer interface to VideoRtpDepacketizer Bug: webrtc:11152 Change-Id: Ie7823feeb5b0563b74754255aaedfada9d446ac5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161380 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30031} 6 年前
Migrate modules/rtp_rtcp to webrtc::Mutex. Bug: webrtc:11567 Change-Id: I4c71f3a28ef875af2c232b1b553840d6e21649d3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178804 Commit-Queue: Markus Handell <handellm@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31645} 5 年前
Migrate modules/rtp_rtcp to webrtc::Mutex. Bug: webrtc:11567 Change-Id: I4c71f3a28ef875af2c232b1b553840d6e21649d3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178804 Commit-Queue: Markus Handell <handellm@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31645} 5 年前
Reformat the WebRTC code base 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}7 年前
Use backticks not vertical bars to denote variables in comments for /modules/rtp_rtcp Bug: webrtc:12338 Change-Id: I52eb3b6675c4705e22f51b70799ed6139a3b46bc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227164 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34686} 4 年前
Format almost everything. This CL was generated by running git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \ grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \ grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \ grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \ grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \ | xargs clang-format -i ; git cl format Most of these changes are clang-format grouping and reordering includes differently. Bug: webrtc:9340 Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28505} 6 年前
Remove part of the FEC code table that covers FEC code for group of 13-48 media packets, instead generate interleaved FEC code at run time. FEC code masks for protection of group of 1 - 12 media packets is not changed. Bug: webrtc:9165 Change-Id: I57c8fd032c7a5192d0da8dfde96550b328cf6620 Reviewed-on: https://webrtc-review.googlesource.com/69680 Commit-Queue: Ying Wang <yinwa@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22921}7 年前
Delete root header file typedef.h. 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}7 年前
Move Word32Align helper next to the only place it is used in Bug: None Change-Id: I99b34b78c6a560afa3638e2ba2f403e25602b12e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226862 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34583} 4 年前
Use backticks not vertical bars to denote variables in comments Bug: webrtc:12338 Change-Id: I89c8b3a328d04203177522cbdfd9e606fd4bce4c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228246 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34696} 4 年前
Cleanup flexfec03 TODOs and logs to indicate there is no intent to implement additional features there Bug: None Change-Id: I774c2356439ee52e73cd70802f28fa5e5b560b8e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316925 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40594} 2 年前
Change FinalizeFecHeader to receive list of streams Changed FinalizeFecHeader to recieve a list of ProtectedStream struct, in order to prepare for receiving multiple ssrcs to protect in the same FEC packet header. Implementation of the multistream case will follow in next CL. Change-Id: I697ef9172a07797a6f500b9ec3a9916f8f45bc04 Bug: webrtc:15002 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307620 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40269} 2 年前
Change FinalizeFecHeader to receive list of streams Changed FinalizeFecHeader to recieve a list of ProtectedStream struct, in order to prepare for receiving multiple ssrcs to protect in the same FEC packet header. Implementation of the multistream case will follow in next CL. Change-Id: I697ef9172a07797a6f500b9ec3a9916f8f45bc04 Bug: webrtc:15002 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307620 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40269} 2 年前
rtp_rtcp/source: fix some minor typos Bug: None Change-Id: Iedc6e3b7e0cb92256255afc4cd76c66b01099c1e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/310080 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40362} 2 年前
Change FinalizeFecHeader to receive list of streams Changed FinalizeFecHeader to recieve a list of ProtectedStream struct, in order to prepare for receiving multiple ssrcs to protect in the same FEC packet header. Implementation of the multistream case will follow in next CL. Change-Id: I697ef9172a07797a6f500b9ec3a9916f8f45bc04 Bug: webrtc:15002 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307620 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40269} 2 年前
Change flexfec header writer to finalize header according to updated RFC Change implementation of FinalizeFecHeader to write the FEC header for multiple ssrcs according to the updated RFC. Change-Id: I280964b2e53c3579f348fbd42815c966840375ac Bug: webrtc:15002 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307601 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40270} 2 年前
fec: Skip traversal the list of recovered packets if possible Do not traverse the list of recovered media packets if none of them was recovered through FEC recovery procedure. Bug: None Change-Id: Ib3aa59c946919fab08f0e20fcf279b1b8032d0e3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315320 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Auto-Submit: Andrei Volykhin <andrey.volykhin@lge.com> Cr-Commit-Position: refs/heads/main@{#40546} 2 年前
Remove deprecated RecoveredPacketReceiver::OnRecoveredPacket signature Bug: webrtc:7135, webrtc:14795 Change-Id: Ib2f434b59542d6d8a2b8a287047417b784187602 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290567 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Auto-Submit: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39049} 3 年前
In RtpRtcp use BitrateTracker instead of RateStatistics to measure bitrate BitrateTracker uses RateStatistics underneath, thus algorithm is the same, but it provides Timestamp/TimeDelta friendly interface Bug: webrtc:13757 Change-Id: I9f2fcb3d498b2a137b531b94b660d15aa273c4bf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/312600 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40465} 2 年前
Move Word32Align helper next to the only place it is used in Bug: None Change-Id: I99b34b78c6a560afa3638e2ba2f403e25602b12e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226862 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34583} 4 年前
fec: Skip traversal the list of recovered packets if possible Do not traverse the list of recovered media packets if none of them was recovered through FEC recovery procedure. Bug: None Change-Id: Ib3aa59c946919fab08f0e20fcf279b1b8032d0e3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315320 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Auto-Submit: Andrei Volykhin <andrey.volykhin@lge.com> Cr-Commit-Position: refs/heads/main@{#40546} 2 年前
fec: Skip traversal the list of recovered packets if possible Do not traverse the list of recovered media packets if none of them was recovered through FEC recovery procedure. Bug: None Change-Id: Ib3aa59c946919fab08f0e20fcf279b1b8032d0e3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315320 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Auto-Submit: Andrei Volykhin <andrey.volykhin@lge.com> Cr-Commit-Position: refs/heads/main@{#40546} 2 年前
rtp_rtcp/source: fix some minor typos Bug: None Change-Id: Iedc6e3b7e0cb92256255afc4cd76c66b01099c1e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/310080 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40362} 2 年前
Use backticks not vertical bars to denote variables in comments for /modules/rtp_rtcp Bug: webrtc:12338 Change-Id: I52eb3b6675c4705e22f51b70799ed6139a3b46bc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227164 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34686} 4 年前
Rename EncodedImage property Timetamp to RtpTimestamp To avoid name collision with Timestamp type, To avoid confusion with capture time represented as Timestamp Bug: webrtc:9378 Change-Id: I8438a9cf4316e5f81d98c2af9dc9454c21c78e70 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320601 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40796} 2 年前
Implement setMetadata for receiver encoded video frames This change adds a new function to RTPFrameObject to allow setting the RTPVideoHeader from VideoFrameMetadata. The setMetadata function in TransformableVideoReceiverFrame disallows changing anything other than frameID and dependencies. Change-Id: I74e55ffbe1f426b660c2e243b20358c6a6cc2ffd Bug: chromium:1464853 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/314963 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Palak Agarwal <agpalak@google.com> Reviewed-by: Tony Herre <herre@google.com> Cr-Commit-Position: refs/heads/main@{#40530} 2 年前
Remove deprecated TransformableAudioFrameInterface::getHeader() method Fixed: chromium:1456628 Change-Id: I12ea08070578de846f042c64f2808b55de1603a8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315960 Auto-Submit: Tony Herre <herre@google.com> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40555} 2 年前
Reland "Move leb128 helper functions into own build target" This is a reland of commit fa962ffc698bda5bc7306ac5c3fd626eef737775 Original change's description: > Move leb128 helper functions into own build target > > to remove duplicated implementation of these functions between av1 packetizer, av1 depacketizer and video allocation rtp header extension > > Bug: None > Change-Id: I30049f31c289bdb9e0aad6520f5145d1f999e635 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290731 > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> > Reviewed-by: Philip Eliasson <philipel@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#39069} Bug: None Change-Id: I091276868599a6716407db2972457507ddd46a8c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290961 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39135} 3 年前
Reland "Move leb128 helper functions into own build target" This is a reland of commit fa962ffc698bda5bc7306ac5c3fd626eef737775 Original change's description: > Move leb128 helper functions into own build target > > to remove duplicated implementation of these functions between av1 packetizer, av1 depacketizer and video allocation rtp header extension > > Bug: None > Change-Id: I30049f31c289bdb9e0aad6520f5145d1f999e635 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290731 > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> > Reviewed-by: Philip Eliasson <philipel@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#39069} Bug: None Change-Id: I091276868599a6716407db2972457507ddd46a8c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290961 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39135} 3 年前
Reland "Move leb128 helper functions into own build target" This is a reland of commit fa962ffc698bda5bc7306ac5c3fd626eef737775 Original change's description: > Move leb128 helper functions into own build target > > to remove duplicated implementation of these functions between av1 packetizer, av1 depacketizer and video allocation rtp header extension > > Bug: None > Change-Id: I30049f31c289bdb9e0aad6520f5145d1f999e635 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290731 > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> > Reviewed-by: Philip Eliasson <philipel@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#39069} Bug: None Change-Id: I091276868599a6716407db2972457507ddd46a8c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290961 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39135} 3 年前
Remove internal overrides using old SendRtp and SendRtcp interfaces. This CL takes away all usages except for Android code. Low-Coverage-Reason: Refactoring old code Bug: webrtc:15410 Change-Id: I66bed6a4a2787b4177a82e599b48623ca67cd235 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315940 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40554} 2 年前
[Cleanup] Add missing #include. Remove useless ones. This CL is the result of running include-what-you-use tool on part of the code base (audio target and dependencies) plus manual fixes. bug: webrtc:8311 Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604 Reviewed-on: https://webrtc-review.googlesource.com/c/106280 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25311}7 年前
Format almost everything. This CL was generated by running git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \ grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \ grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \ grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \ grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \ | xargs clang-format -i ; git cl format Most of these changes are clang-format grouping and reordering includes differently. Bug: webrtc:9340 Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28505} 6 年前
Format almost everything. This CL was generated by running git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \ grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \ grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \ grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \ grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \ | xargs clang-format -i ; git cl format Most of these changes are clang-format grouping and reordering includes differently. Bug: webrtc:9340 Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28505} 6 年前
Use Timestamp type in RtpState struct Bug: webrtc:13757 Change-Id: I7f8fc1a9c4cbf464b3969c4754ce5aa9c5b5f076 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303500 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39960} 3 年前
Use Timestamp type in RtpState struct Bug: webrtc:13757 Change-Id: I7f8fc1a9c4cbf464b3969c4754ce5aa9c5b5f076 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303500 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39960} 3 年前
Reland "Represent RtpPacketToSend::capture_time with Timestamp" This reverts commit 56db8d09529d5ba92d24954a1d78a90c8ea2cd85. Reason for revert: downstream problem addressed Original change's description: > Revert "Represent RtpPacketToSend::capture_time with Timestamp" > > This reverts commit 385eb9714daa80306d2f92d36678c42892dab555. > > Reason for revert: Causes problems downstream: > > # > # Fatal error in: rtc_base/units/unit_base.h, line 122 > # last system error: 0 > # Check failed: value >= 0 (-234 vs. 0) > > Original change's description: > > Represent RtpPacketToSend::capture_time with Timestamp > > > > Bug: webrtc:13757 > > Change-Id: I0ede22cd34e3a59afe1477d8edd495dce64e3242 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252586 > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#36083} > > Bug: webrtc:13757 > Change-Id: I8442abd438be8726cf671d0f372d50ecfac6847e > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252720 > Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#36087} Bug: webrtc:13757 Change-Id: I1fa852757480116f35deb2b6c3c27800bdf5e197 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252781 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36093} 4 年前
Ensure payload type frequency does not cause divide-by-zero This CL does 2 things: - Change the DCHECK for payload_type_frequency to a CHECK (so that this error will be a crash not a divide-by-zero) - Change the replay helper that was used by the fuzzer to set the frequency of the packets to the video value (90K). Bug: chromium:1466826 Change-Id: I39941f250b1782b36a3bcddfd347a016591466ec Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/312700 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40468} 2 年前
In RtpRtcp use BitrateTracker instead of RateStatistics to measure bitrate BitrateTracker uses RateStatistics underneath, thus algorithm is the same, but it provides Timestamp/TimeDelta friendly interface Bug: webrtc:13757 Change-Id: I9f2fcb3d498b2a137b531b94b660d15aa273c4bf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/312600 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40465} 2 年前
Propagate time of the last received packet with Timestamp type Bug: webrtc:13757 Change-Id: I446fc10ad6a90ab9ecaac337b9f2ad4ccad37cbd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307020 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40211} 2 年前
Migrate RemoteNtpTimeEstimator to more precise time representations Reland of https://webrtc-review.googlesource.com/c/src/+/261311 Bug: webrtc:13757 Change-Id: I34a58100b8fadfe3dbea9ffce71829b7670daad8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261726 Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36838} 3 年前
Replace RTC_DCHECK with EXPECT_TRUE in time estimator UT code Replacing RTC_DCHECK code with EXPECT_TRUE in the remote ntp time estimator unittest code. This to prevent test failures when building and testing in non-debug mode. Bug: webrtc:15572 Change-Id: I372fcd6ee29a4ddc07d6b27ddd492dcea13d399f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323181 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Stefan Holmer <stefan@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40936} 2 年前
[Cleanup] Add missing #include. Remove useless ones. This CL is the result of running include-what-you-use tool on part of the code base (audio target and dependencies) plus manual fixes. bug: webrtc:8311 Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604 Reviewed-on: https://webrtc-review.googlesource.com/c/106280 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25311}7 年前
Reformat the WebRTC code base 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}7 年前
Format almost everything. This CL was generated by running git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \ grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \ grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \ grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \ grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \ | xargs clang-format -i ; git cl format Most of these changes are clang-format grouping and reordering includes differently. Bug: webrtc:9340 Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28505} 6 年前
Enable padding bit in TransportFeedback packets Set padding bit if the TransportFeedback packet contains zero padding. Also write number of padding elements at the last position of the packet. Bug: webrtc:10263 Change-Id: I8d17bc0e889f658ac383dec64ddcb95d438c9702 Reviewed-on: https://webrtc-review.googlesource.com/c/122240 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Johannes Kron <kron@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26646}7 年前
Replace virtual RtcpPacket::SetSenderSsrc with base member to slightly improve binary size. Bug: None Change-Id: I894c7d67a72f4a8077963d2ba0a7bb471a2e7e4d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156300 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29428} 6 年前
Format almost everything. This CL was generated by running git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \ grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \ grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \ grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \ grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \ | xargs clang-format -i ; git cl format Most of these changes are clang-format grouping and reordering includes differently. Bug: webrtc:9340 Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28505} 6 年前
Update RtcpReceiver to use Timesetamp/TimeDelta types instead of raw ints Bug: webrtc:13757 Change-Id: Ie0317a584406bec3c34403a7bc8059e4272b339f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311674 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40429} 2 年前
Update RtcpReceiver to use Timesetamp/TimeDelta types instead of raw ints Bug: webrtc:13757 Change-Id: Ie0317a584406bec3c34403a7bc8059e4272b339f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311674 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40429} 2 年前
Change RTCPReceiver::GetAndResetXrRrRtt to return TimeDelta Bug: webrtc:13757 Change-Id: Iaf3a540fbab51990fb6b983912e3b8c1bb1aaa81 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308940 Auto-Submit: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40302} 2 年前
Reland "rtp sender: don't send BYE on deactivating streams" This is a reland of commit a22c2a0c581cbe3f612f7a7d9fb9840186cc1e06 after systems depending on this have been fixed. Original change's description: > rtp sender: don't send BYE on deactivating streams > > as this breaks RTCP assumptions about SSRCs being no longer > active as defined in > https://www.rfc-editor.org/rfc/rfc3550#section-6.6 > > This should not be sent in reaction to temporarily disabling > a stream via RTCRtpParameters.active as this does not mean that > the participant is leaving the session as defined in > https://www.rfc-editor.org/rfc/rfc3550#section-6.3.7 > and does not indicate end of participation as defined in > https://www.rfc-editor.org/rfc/rfc3550#section-6.1 > which stipulates BYE should be the last packet sent from this SSRC. > > BUG=webrtc:11082 > > Change-Id: Ia5144857f85303643146b0759184f0f3f50b66e4 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273348 > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Philipp Hancke <phancke@microsoft.com> > Cr-Commit-Position: refs/heads/main@{#38059} Bug: webrtc:11082 Change-Id: Iad8b503b3101d1e684a4da2d1547b879e77b85dd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293861 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Cr-Commit-Position: refs/heads/main@{#40716} 2 年前
Cleanup calculating time between RTCP reports Move that calculation into dedicated function, move comment why it is calculated the way it is into the same function. Cleanup that comment - remove parts unused by current code, in particular remove description of code that was deleted a while ago Use more strict types for the calculation to make it clearer. Replace DCHECK result can't be zero with a clamp to ensure it can't be zero, because with large bitrates it may. Reland of https://webrtc-review.googlesource.com/c/src/+/315143 Bug: None Change-Id: I41ce383a2f19d489e4cae0b1bf1f720e0ffdd17a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315460 Reviewed-by: Åsa Persson <asapersson@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40538} 2 年前
Reland "rtp sender: don't send BYE on deactivating streams" This is a reland of commit a22c2a0c581cbe3f612f7a7d9fb9840186cc1e06 after systems depending on this have been fixed. Original change's description: > rtp sender: don't send BYE on deactivating streams > > as this breaks RTCP assumptions about SSRCs being no longer > active as defined in > https://www.rfc-editor.org/rfc/rfc3550#section-6.6 > > This should not be sent in reaction to temporarily disabling > a stream via RTCRtpParameters.active as this does not mean that > the participant is leaving the session as defined in > https://www.rfc-editor.org/rfc/rfc3550#section-6.3.7 > and does not indicate end of participation as defined in > https://www.rfc-editor.org/rfc/rfc3550#section-6.1 > which stipulates BYE should be the last packet sent from this SSRC. > > BUG=webrtc:11082 > > Change-Id: Ia5144857f85303643146b0759184f0f3f50b66e4 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273348 > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Philipp Hancke <phancke@microsoft.com> > Cr-Commit-Position: refs/heads/main@{#38059} Bug: webrtc:11082 Change-Id: Iad8b503b3101d1e684a4da2d1547b879e77b85dd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293861 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Cr-Commit-Position: refs/heads/main@{#40716} 2 年前
Migrate rtp_rtcp to absl::AnyInvocable based TaskQueueBase interface Bug: webrtc:14245 Change-Id: I037f964130648caf0bd1de86611f8681d475b078 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268146 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37481} 3 年前
Migrate rtp_rtcp to absl::AnyInvocable based TaskQueueBase interface Bug: webrtc:14245 Change-Id: I037f964130648caf0bd1de86611f8681d475b078 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268146 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37481} 3 年前
In RtcpTransciever refactor outgoing transport interface Replace Transport* interface with since std::function to stress this class doesn't produce RTP packets Repesent outgoing packet as ArrayView instead of pointer + length. Make outgoing transport optional, thus allowing to use RtcpTransciever as an rtcp parser. Bug: webrtc:8239, webrtc:14870 Change-Id: Ia582d9a980786df8e295adcebe27081258b80dc0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306280 Reviewed-by: Emil Lundmark <lndmrk@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40134} 2 年前
Cleanup RtcpTransceiver dependency on webrtc::Transport Bug: webrtc:8239 Change-Id: I5740935044ba422a32b571eb9f559e83b915fe15 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306522 Auto-Submit: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Emil Lundmark <lndmrk@webrtc.org> Commit-Queue: Emil Lundmark <lndmrk@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40152} 2 年前
Use absl::optional instead of std::optional We haven't switched to the std spelling in WebRTC yet. Change-Id: If21a6ee9ac19be8ce959b3192eb8de044048f157 Bug: None Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/310501 Auto-Submit: Emil Lundmark <lndmrk@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40356} 2 年前
In RtcpTransciever refactor outgoing transport interface Replace Transport* interface with since std::function to stress this class doesn't produce RTP packets Repesent outgoing packet as ArrayView instead of pointer + length. Make outgoing transport optional, thus allowing to use RtcpTransciever as an rtcp parser. Bug: webrtc:8239, webrtc:14870 Change-Id: Ia582d9a980786df8e295adcebe27081258b80dc0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306280 Reviewed-by: Emil Lundmark <lndmrk@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40134} 2 年前
Remove internal overrides using old SendRtp and SendRtcp interfaces. This CL takes away all usages except for Android code. Low-Coverage-Reason: Refactoring old code Bug: webrtc:15410 Change-Id: I66bed6a4a2787b4177a82e599b48623ca67cd235 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315940 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40554} 2 年前
Remove internal overrides using old SendRtp and SendRtcp interfaces. This CL takes away all usages except for Android code. Low-Coverage-Reason: Refactoring old code Bug: webrtc:15410 Change-Id: I66bed6a4a2787b4177a82e599b48623ca67cd235 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315940 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40554} 2 年前
Delete deprecated kUri constants for rtp header extensions Bug: webrtc:7472 Change-Id: Ib1af94cc434d93be254370f0d9b6ebaafe8817d4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232902 Reviewed-by: Åsa Persson <asapersson@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35085} 4 年前
Introduce new enum name for the dependency descriptor extension Dependency descriptor has finalized spec and thus deserve a dedicated name. Bug: webrtc:10342 Change-Id: I2c2f1d52c82cfff8372cd4092dfcc47a083a6009 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290402 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38973} 3 年前
Format /modules git ls-files | grep -e "\(\.h\|\.cc\)$" | grep -e "^modules/" | xargs clang-format -i ; git cl format after landing: add to .git-blame-ignore-revs Bug: webrtc:15082 Change-Id: I2c3cd28740062794f8c10e39d8406aadb9e9a35a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301620 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Auto-Submit: Jared Siskin <jtsiskin@meta.com> Cr-Commit-Position: refs/heads/main@{#39901} 3 年前
Migrate Dependency Descriptor parser from BitBuffer to BitstreamReader BitstreamReader itself uses idea of Read function that always succeed, and a separate function to check for errors. Thus extra layer in the DependencyDescriptorReader is not needed. Bug: None Change-Id: Ie58861f2cbecc02a5a1a9538232494b4442c9afd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231226 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#34940} 4 年前
Migrate Dependency Descriptor parser from BitBuffer to BitstreamReader BitstreamReader itself uses idea of Read function that always succeed, and a separate function to check for errors. Thus extra layer in the DependencyDescriptorReader is not needed. Bug: None Change-Id: Ie58861f2cbecc02a5a1a9538232494b4442c9afd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231226 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#34940} 4 年前
Fail instead of crashing while writing invalid dependency descriptor Bug: webrtc:10342 Change-Id: Ic9af7913aa9835450877940fc5cf29bebf774484 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/224082 Reviewed-by: Emil Lundmark <lndmrk@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34379} 4 年前
Use backticks not vertical bars to denote variables in comments for /modules/rtp_rtcp Bug: webrtc:12338 Change-Id: I52eb3b6675c4705e22f51b70799ed6139a3b46bc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227164 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34686} 4 年前
Do not propagate generic descriptor on receiving frame It was used only for the frame decryptor. Decryptor needs only raw representation that it can recreate in a way compatible with the new version of the descriptor. This relands commit abf73de8eae90e9ac7e88ce1d52728e8102e824f. with adjustments. Change-Id: I935977179bef31d8e1023964b967658e9a7db92d Bug: webrtc:10342 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168489 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30532} 6 年前
Do not propagate generic descriptor on receiving frame It was used only for the frame decryptor. Decryptor needs only raw representation that it can recreate in a way compatible with the new version of the descriptor. This relands commit abf73de8eae90e9ac7e88ce1d52728e8102e824f. with adjustments. Change-Id: I935977179bef31d8e1023964b967658e9a7db92d Bug: webrtc:10342 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168489 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30532} 6 年前
Rename current flexfec implementation flexfec_03 As per the comment in https://webrtc-review.googlesource.com/c/src/+/303240 on the flexfec_header_reader_writer2.h, renaming this file to flexfec_header_reader_writer.h and renaming the current implementation to flexfec_03_header_reader_writer.h as it is based on the 03 draft of the RFC. Change-Id: I80cb2aba6225ec7cd989a134c3204d1db0ac6f7c Bug: webrtc:15002 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307600 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40231} 2 年前
Add codec name H265 to support H265 in WebRTC Bug: webrtc:13485 Change-Id: I352b15a65867f0d56fc8e9a9e03081bd3258108e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316283 Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40773} 2 年前
Use backticks not vertical bars to denote variables in comments for /modules/rtp_rtcp Bug: webrtc:12338 Change-Id: I52eb3b6675c4705e22f51b70799ed6139a3b46bc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227164 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34686} 4 年前
Use single packet limit when all fragments end up in one H.264 packet Update RtpPacketizerH264::PacketizeStapA to use single_packet_reduction_len when all fragments end up in one H.264 packet. Previous code was using first_packet_reduction_len + last_packet_reduction_len for this case, which can cause an occasional RTC_CHECK crash in RtpPacketizerH264::NextAggregatePacket due to exceeding the available payload capacity of an RTP packet. Bug: webrtc:15477 Change-Id: Iba1564a6a29618bef22f19d82aba938420994b23 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319645 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40737} 2 年前
Fix math involving enums in C++20 (-Wdeprecated-anon-enum-enum-conversion) - Replace enum with constexpr if necessary. - Merge multiple definitions for H.264 NalDefs and FuDefs and apply constexpr. Bug: chromium:1284275 Change-Id: I4a4d95ed6aba258e7c19c3ae6251c8b78caf84ec Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276561 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Auto-Submit: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com> Cr-Commit-Position: refs/heads/main@{#38215} 3 年前
Use single packet limit when all fragments end up in one H.264 packet Update RtpPacketizerH264::PacketizeStapA to use single_packet_reduction_len when all fragments end up in one H.264 packet. Previous code was using first_packet_reduction_len + last_packet_reduction_len for this case, which can cause an occasional RTC_CHECK crash in RtpPacketizerH264::NextAggregatePacket due to exceeding the available payload capacity of an RTP packet. Bug: webrtc:15477 Change-Id: Iba1564a6a29618bef22f19d82aba938420994b23 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319645 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40737} 2 年前
Format almost everything. This CL was generated by running git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \ grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \ grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \ grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \ grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \ | xargs clang-format -i ; git cl format Most of these changes are clang-format grouping and reordering includes differently. Bug: webrtc:9340 Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28505} 6 年前
Remove all #include <assert.h>/<cassert> and usage in Obj-C code. This CL completes the removal of assert() and relative headers from the codebase (excluded //examples/objc/AppRTCMobile/third_party/SocketRocket which is in a third_party sub-directory). Bug: webrtc:6779 Change-Id: I93ed57168d2c0e011626873d66529488c5f484f2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225546 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34528} 4 年前
Remove RTC_DISALLOW_COPY_AND_ASSIGN from modules/ Bug: webrtc:13555, webrtc:13082 Change-Id: I2c2cbcbd918f0cfa970c1a964893220ba11d4b41 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247960 Reviewed-by: Artem Titov <titovartem@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: (Daniel.L) Byoungchan Lee <daniel.l@hpcnt.com> Cr-Commit-Position: refs/heads/main@{#35771} 4 年前
Remove RTPFragmentationHeader creation and propagation through webrtc Bug: webrtc:6471 Change-Id: I5cb1e10088aaecb5981888082b87ae9957bbaaef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181541 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31955} 5 年前
Tweak VP8 payload to comply with RFC 7741 This updates the VP8 payload diagrams to be compliant with RFC 7741. It also fixes some minor inconsistencies with PID, previously referred to as PartID. Bug: None Change-Id: I33eb57d96f3d95b01ef5f0afa21a9dc54b41db2d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230243 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Emil Lundmark <lndmrk@webrtc.org> Cr-Commit-Position: refs/heads/main@{#34859} 4 年前
Remove RTC_DISALLOW_COPY_AND_ASSIGN from modules/ Bug: webrtc:13555, webrtc:13082 Change-Id: I2c2cbcbd918f0cfa970c1a964893220ba11d4b41 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247960 Reviewed-by: Artem Titov <titovartem@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: (Daniel.L) Byoungchan Lee <daniel.l@hpcnt.com> Cr-Commit-Position: refs/heads/main@{#35771} 4 年前