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Author SHA1 Message Date
Bjorn A Mellem
88db835278 Change DataChannelTransportInterface/Sink methods to pure virtual. 
These methods are implemented everywhere, so they no longer need to
provide default implementations.

Bug: webrtc:9719
Change-Id: Idf67a78010a55f545d882793d0d6edbccfae525b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154002
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29262}
 2019-09-20 20:06:14 +00:00
Bjorn A Mellem
d702231268 Cleanup deprecated monitoring of MediaTransport state. 
PeerConnection now watches when data channels become ready to send
through its implementation of DataChannelSink, and no longer needs to
monitor the MediaTransport state.

Bug: webrtc:9719
Change-Id: I3e17747eb03926a3791c204bf5a1d2dc67855c09
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154001
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29261}
 2019-09-20 19:44:20 +00:00
Patrik Höglund
5ac329c8aa Cap h264 fuzzer input to 200k. 
Verified it no longer times out on the input that spawned the bug.

Bug: chromium:1005853
Change-Id: I5b0ab25aaefdc8b451b4d976b1c3b8f8d38f13e9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153840
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29260}
 2019-09-20 18:15:00 +00:00
Johannes Kron
03bbef5e1f Fix accidental change of transport time metric 
The transport time metric was accidentially changed by the CL
https://webrtc-review.googlesource.com/c/src/+/153660

This CL restore the transport time metric to how it has been
measured before, that is, time from encoder output to decoder input.

Bug: webrtc:10975
Change-Id: I66f022f26976451d28c0374b22849f14f9c02378
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153886
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29259}
 2019-09-20 15:56:55 +00:00
chromium-webrtc-autoroll
c2e9d84822 Roll chromium_revision 303c57cf17..230cc8f7e4 (698351:698466) 
Change log: 303c57cf17..230cc8f7e4
Full diff: 303c57cf17..230cc8f7e4

Changed dependencies
* src/base: 8185ff85d6..b70d15bab8
* src/build: 9417d7b115..ceed4a086a
* src/ios: 5173d293d5..1ccb60e1f4
* src/testing: f461f416c2..bf89d559c0
* src/third_party: 3f8d8226d0..e6a4066287
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/54ac3634e8..fa002e6324
* src/third_party/freetype/src: 04ebb2a000..5327092bb2
* src/tools: 2be0437563..9ee4313e8c
DEPS diff: 303c57cf17..230cc8f7e4/DEPS

Clang version changed b4160cb94c54f0b31d0ce14694950dac7b6cd83f:13bdae8541c3fc5acf6ee7de78ec5ab8446848e4
Details: 303c57cf17..230cc8f7e4/tools/clang/scripts/update.py

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: Ibd10426a9a77cea093ee8fca635348e320257a6c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153960
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#29258}
 2019-09-20 14:48:00 +00:00
Åsa Persson
27b0e0d6b3 Remove obsolete todo comment in simulcast.h 
Bug: none
Change-Id: I1c51919564a8b8bae842fa6421054a2b1faba42a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153885
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29257}
 2019-09-20 14:17:20 +00:00
Niels Möller
544dfb5a97 Delete isac GetBandwidthInfo/SetBandwidthInfo 
Bug: webrtc:10098
Change-Id: I4a56cdc6d081b15a1fc52cba2051783daf4e5ae3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153881
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29256}
 2019-09-20 13:53:52 +00:00
Danil Chapovalov
ef83cc5458 Add fuzzer testing for Dependency Descriptor rtp header extension 
Bug: webrtc:10342
Change-Id: I46c61b9a137a7148ed80ad38da62132dacb270f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153662
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29255}
 2019-09-20 12:40:24 +00:00
Danil Chapovalov
04fd21513b Cleanup passing rtp packet to ulpfec receiver. 
Pass RtpPacket class of header and raw packet separately

Bug: None
Change-Id: Id6d107db0e3751ff3dec87321ce6f850da0ee33a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153740
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29254}
 2019-09-20 11:09:11 +00:00
philipel
0cff4fce55 Removed unused frame_size param from RtpFrameObject ctor. 
Bug: webrtc:10979
Change-Id: Idde493dc7f5165e3ca173d5a38861b444b5904a4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153668
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29253}
 2019-09-20 10:56:01 +00:00
Niels Möller
48b32b748e Delete support for enabling adaptive isac mode 
This appears unused. If deleted, other code related to isac bandwidth
estimation becomes unused and may be deleted in followup cls.

Bug: webrtc:10098
Change-Id: Ifeac2e90de895b12c337ea28cc33704350b9abf4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153667
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29252}
 2019-09-20 10:41:09 +00:00
philipel
b5e4785464 RtpFrameObject now takes an EncodedImageBuffer in its ctor. 
Bug: webrtc:10979
Change-Id: Ibc8b4a524ca95b5faa8850a41df8f2f0136a2969
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153666
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29251}
 2019-09-20 10:15:01 +00:00
Mirko Bonadei
fb59a6aa3f Return const char* from ToString(RTCErrorType error). 
Returning absl::string_view causes problems to the Chromium/WebRTC
component build because absl::operator<< needs to be exported.

This CL switches to `const char*` which should be enough to avoid
to generate temporaries.

Bug: webrtc:9419
Change-Id: If169a6f95c7efd21ac8ce108c7f2f80a76ff2313
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153842
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29250}
 2019-09-20 08:27:52 +00:00
Niels Möller
e0b31677b6 Delete dead code inside #ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED 
Bug: None
Change-Id: If31f2a5e4a2536b3c7fda596f2c251e8074a18d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153671
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29249}
 2019-09-20 07:40:10 +00:00
Sam Zackrisson
feee1e4c36 Add flag to APM to force multichannel even with AEC3 
Currently, APM fakes multichannel in two ways:
 - With injected AECs, capture processing is only performed on the left
channel. The result is copied into the other channels.
 - With multichannel render audio, all channels are mixed into one
before analysing.

This CL adds a flag to disable these behaviors, ensuring proper
multichannel processing happens throughout the APM pipeline.

Adds killswitches to separately disable render / capture multichannel.

Additionally - AEC3 currently crashes when running with multichannel.
This CL adds the missing pieces to at least have it run without
triggering any DCHECKS, including making the high pass filter properly
handle multichannel.

Bug: webrtc:10913, webrtc:10907
Change-Id: I38795bf8f312b959fcc816a056fba2c68d4e424d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152483
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29248}
 2019-09-20 06:36:12 +00:00
Niels Möller
e24557f51e Declare api:libjingle_peerconnection_api dependency on media:media_base 
Deletes one of the two // nogncheck comments in peerconnection_interface.h.

Bug: webrtc:8733
Change-Id: I7cda4957206c4c5f36261517718b526734bad68a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150798
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29247}
 2019-09-20 06:29:22 +00:00
chromium-webrtc-autoroll
2051b8bf05 Roll chromium_revision a536fa4a4a..303c57cf17 (698214:698351) 
Change log: a536fa4a4a..303c57cf17
Full diff: a536fa4a4a..303c57cf17

Changed dependencies
* src/base: 2670e49cfb..8185ff85d6
* src/build: df75aae7da..9417d7b115
* src/ios: 425a78aea9..5173d293d5
* src/testing: 6099b3ff92..f461f416c2
* src/third_party: 8f7ec0c439..3f8d8226d0
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/4f96a26c2f..54ac3634e8
* src/third_party/depot_tools: 2c80d01be6..7735f52c6d
* src/tools: 4357bb3217..2be0437563
DEPS diff: a536fa4a4a..303c57cf17/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: Id5610d7f41fbe939e0b205f994761c1b16b1ca8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153820
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#29246}
 2019-09-20 04:41:10 +00:00
chromium-webrtc-autoroll
95c538f17b Roll chromium_revision fc1e948f93..a536fa4a4a (698112:698214) 
Change log: fc1e948f93..a536fa4a4a
Full diff: fc1e948f93..a536fa4a4a

Changed dependencies
* src/base: e766beee3d..2670e49cfb
* src/build: e2f0a70047..df75aae7da
* src/ios: 3efb3bffed..425a78aea9
* src/testing: 0b2bff4e23..6099b3ff92
* src/third_party: 81ec316b3b..8f7ec0c439
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/2ad535693f..4f96a26c2f
* src/third_party/depot_tools: c6be56eced..2c80d01be6
* src/tools: 142e5efe9a..4357bb3217
DEPS diff: fc1e948f93..a536fa4a4a/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: Ic80366d287c04b7f4b261ec5d58b5de84f7359a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153761
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#29245}
 2019-09-19 22:43:37 +00:00
chromium-webrtc-autoroll
f288c8ee6e Roll chromium_revision cf1a2beb4b..fc1e948f93 (697976:698112) 
Change log: cf1a2beb4b..fc1e948f93
Full diff: cf1a2beb4b..fc1e948f93

Changed dependencies
* src/base: 3fe4a418db..e766beee3d
* src/build: 76a4ca1e1b..e2f0a70047
* src/ios: 8de41b2ec5..3efb3bffed
* src/testing: 3f13136080..0b2bff4e23
* src/third_party: 90b037fc6f..81ec316b3b
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/033994d4a3..2ad535693f
* src/tools: 97ef1887ae..142e5efe9a
DEPS diff: cf1a2beb4b..fc1e948f93/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: Ibf08e9d45f2f11894b088c8a18781a10b64d8523
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153722
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#29244}
 2019-09-19 18:34:57 +00:00
Johannes Kron
c12db81e79 Add frame receive to frame rendered metric to video_quality_analyzer 
Bug: webrtc:10975
Change-Id: I6b36566efbbb52d27ca6cb44cb3b40aaf0cacb7b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153660
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29243}
 2019-09-19 14:43:04 +00:00
philipel
f0be5b5380 Make GetBitstream non-virtual since it is no longer needed for testing. 
Bug: webrtc:10979
Change-Id: Id313c7fddbec40b9f19dae95f736379b872e3082
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153663
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29242}
 2019-09-19 14:04:09 +00:00
Sebastian Jansson
40de3cc5ef Propagating TargetRate struct to BitrateAllocator. 
Bug: webrtc:9883
Change-Id: I443ac7f1ef0f933e2165fdb2f912d314acc7f2f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153345
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29241}
 2019-09-19 14:03:04 +00:00
Johannes Kron
ac315b283c Add support for max/min encode bitrate to peer connection quality test 
Bug: webrtc:10975
Change-Id: I9be551040936d2e9b5e41dd1bbaea2ad4afd36ed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153481
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29240}
 2019-09-19 13:47:29 +00:00
Niels Möller
6a092637f0 Delete obsolete isac "assign" api 
Bug: None
Change-Id: I116e3f4b89e2c1e1f0d06e2ff5d58d2a50e2aadb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153665
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29239}
 2019-09-19 13:26:26 +00:00
chromium-webrtc-autoroll
d8ffbb0bc4 Roll chromium_revision afdb2e7a8b..cf1a2beb4b (697871:697976) 
Change log: afdb2e7a8b..cf1a2beb4b
Full diff: afdb2e7a8b..cf1a2beb4b

Changed dependencies
* src/build: 050608ea95..76a4ca1e1b
* src/ios: 797d97f55d..8de41b2ec5
* src/testing: c223ceca68..3f13136080
* src/third_party: d7037a0728..90b037fc6f
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/6a17fd7fb7..033994d4a3
* src/tools: efcee32728..97ef1887ae
DEPS diff: afdb2e7a8b..cf1a2beb4b/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: I8653cfade8885b6166110d4d72819e30e7d21ea4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153700
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#29238}
 2019-09-19 12:32:23 +00:00
philipel
76161f7446 Move the call to GetBitstream out of the RtpFrameObject ctor. 
Bug: webrtc:10979
Change-Id: I9159eb04d4a371e8ed8f932a989d6b884faa7be7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153661
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29237}
 2019-09-19 12:08:04 +00:00
henrika
14137a1064 Adds logging of audio sessions status on the recording side in ADM for Android. 
Goal is to be able to retrieve more details about possible microphone conflicts in
cases where Init/Start of audio recording fails.

Only supported on Android N and higher.

Also adds new boolean UMA histogram called WebRTC.Audio.SourceMatchesRecordingSession.
Its value is stored after the recording session has been stopped.

Does not affect the media flow or functionality of the ADM. Time to start audio should
not be affected either since the new check and logging takes place on a separate
ExecutorService thread.

See go/webrtc-adm-android for more details and examples.

Bug: webrtc:10971
Change-Id: Ia80c1534e326907a1582824225d5f58caa016922
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150793
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29236}
 2019-09-19 11:35:10 +00:00
Björn Terelius
86873f0cd3 Improve field trial error message. 
Bug: None
Change-Id: I112cda6fead3d68136fd7be551686e40191fa87f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153482
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29235}
 2019-09-19 09:38:49 +00:00
Niels Möller
e942b141d8 New build target api:media_interface 
Bug: webrtc:8733
Change-Id: I84bbefb1a5ef8e592db29b79499d60ac80c23464
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153180
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29234}
 2019-09-19 09:32:27 +00:00
Sebastian Jansson
0a5ed896e2 Adds remote estimates to rtc event log. 
Bug: webrtc:10742
Change-Id: I0db998a05492603fcdeedca780d9ee3d64aa00d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151651
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29233}
 2019-09-19 09:22:37 +00:00
Danil Chapovalov
6ed60e39dc Implement Dependency Descriptor writer 
Bug: webrtc:10342
Change-Id: I561825265c0990864e1d16aeed4afbdd98871940
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153350
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29232}
 2019-09-19 08:51:40 +00:00
Björn Terelius
489843f1b1 Improve trendline estimator logging. 
Bug: None
Change-Id: I7cc6dc7f45ddb7325252516490436bea1ec8d250
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153521
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29231}
 2019-09-19 08:33:11 +00:00
Danil Chapovalov
693bf1eea1 Delete modules/rtp_rtcp local DivideRoundToNearest in favor on one in rtc_base 
To resolve a TODO

Bug: None
Change-Id: I90e10af24718e1aafd7e72076731b34c1110bb4b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153524
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29230}
 2019-09-19 08:10:12 +00:00
chromium-webrtc-autoroll
bd24260791 Roll chromium_revision eae7ecf757..afdb2e7a8b (697744:697871) 
Change log: eae7ecf757..afdb2e7a8b
Full diff: eae7ecf757..afdb2e7a8b

Changed dependencies
* src/base: 01355d2940..3fe4a418db
* src/build: 71ed08eea6..050608ea95
* src/ios: e12f797b06..797d97f55d
* src/testing: 74e29479c8..c223ceca68
* src/third_party: a939fa9d82..d7037a0728
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/cbc960899a..6a17fd7fb7
* src/third_party/depot_tools: 3f79763629..c6be56eced
* src/tools: 8c0c370d66..efcee32728
DEPS diff: eae7ecf757..afdb2e7a8b/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: Ia0e447ae02cf1a0a9a1e1a49deca4762b486fd81
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153620
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#29229}
 2019-09-19 02:33:30 +00:00
chromium-webrtc-autoroll
efa04efaaa Roll chromium_revision 65274319fc..eae7ecf757 (697640:697744) 
Change log: 65274319fc..eae7ecf757
Full diff: 65274319fc..eae7ecf757

Changed dependencies
* src/base: 0b98e1163e..01355d2940
* src/build: dd9f9cd163..71ed08eea6
* src/ios: c4c50839b0..e12f797b06
* src/testing: c684ea24f6..74e29479c8
* src/third_party: a9fa316c88..a939fa9d82
* src/third_party/depot_tools: 6f9a0238ce..3f79763629
* src/tools: db6a8ea870..8c0c370d66
DEPS diff: 65274319fc..eae7ecf757/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: Ic88b3d2dd39fd73f9f5db6ef22c0aac3d6f40b3b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153580
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#29228}
 2019-09-18 20:28:20 +00:00
Sebastian Jansson
93b1ea2168 Using struct for bitrate allocation limits. 
Bug: webrtc:9883
Change-Id: I855c29808ffa14626d78842491fdf81cd00589e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153344
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29227}
 2019-09-18 17:03:59 +00:00
Sebastian Jansson
1b83a9e400 Only handle each RTCP once. 
Previously, each RTCP packet was handled several times in a row, once
per m-section. This caused various weirdness and log warning spam, in
particular when using unified plan.

The cause was that the packets were wired trough each BaseChannel
instance up to the Call class. With this fix, the RTCP packets are wired
once per RtpTransportInternal via the common peer connection class.

Bug: chromium:1002875
Change-Id: I41c4eb3b68e215ebe0f2c6fb93ae0ee73335b89a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152668
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29226}
 2019-09-18 16:54:39 +00:00
chromium-webrtc-autoroll
4bad650ba7 Roll chromium_revision 2bd75c72c1..65274319fc (697505:697640) 
Change log: 2bd75c72c1..65274319fc
Full diff: 2bd75c72c1..65274319fc

Changed dependencies
* src/base: 107a963c0b..0b98e1163e
* src/build: aae0a7b1db..dd9f9cd163
* src/ios: 809c1d07f6..c4c50839b0
* src/testing: ae66c6e30c..c684ea24f6
* src/third_party: 0e5ec1d6e5..a9fa316c88
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/5ce7022394..a7d9ac2af4
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/b4e53c4a99..cbc960899a
* src/third_party/icu: faee8bc705..2ecd66c696
* src/tools: ee770715c2..db6a8ea870
DEPS diff: 2bd75c72c1..65274319fc/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: I5602dbb9d7c7e4e208d5efadf6e764cfbb63c41b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153540
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#29225}
 2019-09-18 16:37:39 +00:00
Niels Möller
7b04a91f4a Delete almost all default methods on PeerConnectionInterface 
Keeping default implementations only for methods involved in
ongoing transitions.

Intended to catch inconsistencies between the interface and the
PeerConnectionProxy class, at compile time.

Bug: webrtc:10716
Change-Id: I4cb126c353855f7288ba09273fa6f87aaa0f32eb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140860
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29224}
 2019-09-18 16:27:44 +00:00
philipel
e607a06338 Removed unused include from PacketBuffer. 
Bug: none
Change-Id: I502f634e85421e38a02cd31d8ae5446cbe32d138
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153520
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29223}
 2019-09-18 15:35:59 +00:00
Danil Chapovalov
33b83fdc95 Introduce integer division helpers with non-default rounding 
There are multiple places in webrtc code where alternative than
default rounding is desired. Typically this rounding is inlined.
e.g. as (<x> + <y>/2) / <y> making code more clumpsy (<y> might be long expression)
and unsafe for large values of <x>

This change introduce small helpers to address both concerns.

Bug: None
Change-Id: Icd8dcee80a697b7c50ba0b2e50295087d2be8670
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153354
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29222}
 2019-09-18 15:08:39 +00:00
Evan Shrubsole
b6a45dda4c Revert "Fix minor regression caused by a8336d3" 
This reverts commit 809198edfff416fce8d75b574a43afab5e67b1cd.

Reason for revert: Performance regressions that need to be addressed.

Original change's description:
> Fix minor regression caused by a8336d3
> 
> VideoEncoder::SetRates was being called unnessesarily when the fields
> appended to RateControlParameters were changed. Since SetRates only
> cares about RateControlParameters, it should have only been called if
> the RateControlParameters themselves were actually changed.
> 
> Bug: webrtc:10126
> Change-Id: Ic47d67e642a3043307fec950e5fba970d9f95167
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152829
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Evan Shrubsole <eshr@google.com>
> Cr-Commit-Position: refs/heads/master@{#29208}

TBR=sprang@webrtc.org,eshr@google.com

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:10126
Change-Id: I133cbe5d8cb894ed944ae8a2d0f63a78bbed72ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153484
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29221}
 2019-09-18 14:38:15 +00:00
Mirko Bonadei
53227ccba9 Remove webrtc::MinPositive from api/. 
Follow-up of https://webrtc-review.googlesource.com/c/src/+/153220,
where during code review it was suggested to move webrtc::MinPositive
out of the api/ directory.

Bug: None
Change-Id: I0c3b87a9ffd1cd205a85dddd9f44cfd95eb02206
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153480
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29220}
 2019-09-18 12:52:09 +00:00
Johannes Kron
1162ba285d Add max/min encode bitrates to video config of peer connection tests 
Extend PeerConnectionE2EQualityTestFixture::VideoConfig with
min_encode_bitrate_bps and max_encode_bitrate_bps.

These are needed to be able to specify the bitrate to be used in tests.

Bug: None
Change-Id: I8af88020e9b364d924e2cecb2bdcc12bf287394d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153352
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29219}
 2019-09-18 09:15:03 +00:00
chromium-webrtc-autoroll
7cfde54849 Roll chromium_revision 51a0808947..2bd75c72c1 (697405:697505) 
Change log: 51a0808947..2bd75c72c1
Full diff: 51a0808947..2bd75c72c1

Changed dependencies
* src/base: 8b3a663d3b..107a963c0b
* src/ios: 2b489af222..809c1d07f6
* src/third_party: 04016a4f18..0e5ec1d6e5
* src/tools: 5ed44acf45..ee770715c2
DEPS diff: 51a0808947..2bd75c72c1/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: I524f06254bc0287f538f92aa34a48107c0c7a24b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153463
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#29218}
 2019-09-18 08:42:34 +00:00
Mirko Bonadei
738bfa7bab Remove api/bitrate_constraints.h. 
Bug: webrtc:8733
Change-Id: Iaeb26e07d399f25dc18b0c4af38ed400577a5d3a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153220
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29217}
 2019-09-18 06:37:58 +00:00
Mirko Bonadei
c128df14ee Update style guide for absl::make_unique. 
No-Try: True
Bug: webrtc:10945
Change-Id: I707aefda5d5b224d78b97ce3122e095c7b9b1f1c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153356
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29216}
 2019-09-18 06:10:58 +00:00
chromium-webrtc-autoroll
95c4b916ce Roll chromium_revision 31d9542abc..51a0808947 (697288:697405) 
Change log: 31d9542abc..51a0808947
Full diff: 31d9542abc..51a0808947

Changed dependencies
* src/base: 9d1bb9a333..8b3a663d3b
* src/build: 48ea8b8e18..aae0a7b1db
* src/ios: fb0b52197c..2b489af222
* src/testing: 24e33d5203..ae66c6e30c
* src/third_party: cf41eae8a8..04016a4f18
* src/third_party/depot_tools: 2c210a4908..6f9a0238ce
* src/third_party/freetype/src: 99f23d6ff2..04ebb2a000
* src/tools: f7c3756749..5ed44acf45
DEPS diff: 31d9542abc..51a0808947/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: Ibb815b17d9359814b3a4b364b58e4f61c08f0bd1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153440
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#29215}
 2019-09-18 00:37:04 +00:00
Sebastian Jansson
ee5ec9a93a Replacing local closure classes with C++14 moving capture lambdas. 
Bug: webrtc:10945
Change-Id: I569b9495cae98f204065911e13c37c31f35da372
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153241
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29214}
 2019-09-17 19:43:05 +00:00
Sebastian Jansson
4d461ba298 Reusing MediaStreamAllocationConfig struct in ObserverConfig. 
This makes it easier to follow the code and reduces the risk of
accidents in the mapping of fields.

Also renaming the ObserverConfig struct to AllocatableTrack to better
reflect what it represents.

Bug: webrtc:9883
Change-Id: Ia320363813db2b4bf7b37852882a1ccb7644ae0b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153342
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29213}
 2019-09-17 19:31:25 +00:00