Mirko Bonadei
f090bf70ed
Roll chromium_revision 566136c383..0d298af41c (1274799:1277684)
...
* Fix jni_zero integration following https://chromium-review.googlesource.com/c/chromium/src/+/5370266 .
* Add infra team as owner of jni_generator_helper.
Change log: 566136c383..0d298af41c566136c383..0d298af41c669f5c90a1..c76e0b4d4e68409c6133..1104ba151b68fce43789..608975a0c58c8e35d7e8..17d9539a97c0afd10e6e..903dfae8ffd0ea1392a4..a421ed20aahttps://boringssl.googlesource.com/boringssl.git/+log/4fa4804c8a..368d0d87d0
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/fb6c0bb480..9128ec6d34
* src/third_party/depot_tools: ca4cfdaf00..1cba129f31bddcd9f77b..d7c63cd216https://aomedia.googlesource.com/aom.git/+log/158761dfb4..80123cb352
* src/third_party/libc++/src: 80307e66e7..0c90b8212c19832b1702..cab4f31e1d13fb5d53a1..1d622188fc2fb44b3615..5e1e0c8f88566136c383..0d298af41chttps://webrtc-review.googlesource.com/c/src/+/344500
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41979}
2024-03-30 16:33:59 +00:00
Danil Chapovalov
007a62d83d
Cleanup sdk android code from using VideoEncoderFactory::CreateVideoEncoder
...
To force webrtc::Environment propagation through android VideoEncoderFactory wrappers
Bug: webrtc:15860
Change-Id: Icad7249b47a5c738b089e09714c729f91eddaf5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342960
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41936}
2024-03-21 10:18:37 +00:00
Bjorn Terelius
a0672c5f32
Roll chromium_revision
...
(from b9338390dfcef481083d44d44c243c0c73196342
to 247e4de55550191706972861e586831a0d94d524)
Manually remove third_party/accessibility_test_framework from DEPS
Bug: b/329245293
Change-Id: Ie977314daf636f5208948ae74b7958ef3e296ffb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342762
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41902}
2024-03-15 07:58:00 +00:00
Per K
0fa90887c5
Deprecate VideoFrame::timestamp() and set_timestamp
...
Instead, add rtp_timestamp and set_rtp_timestamp.
Bug: webrtc:13756
Change-Id: Ic4266394003e0d49e525d71f4d830f5e518299cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342781
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41894}
2024-03-13 11:08:37 +00:00
Danil Chapovalov
f3096afd48
Propagate Environment to create VideoEncoder through java wrappers
...
Bug: webrtc:15860
Change-Id: If1a2873a899e1b839822a4b56aa87d4bae70c581
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342740
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41887}
2024-03-12 15:34:12 +00:00
Mirko Bonadei
8adb080624
Roll chromium_revision a4279f2842..1f0d2a10bd (1259805:1264465)
...
Change log: a4279f2842..1f0d2a10bda4279f2842..1f0d2a10bdfd5eca261f..4edcfa650aa3566ffdee..e36f984f6ff35a7d885a..88acf0de9937d33be47e..0f9045d95ea7e90605df..c863d4783f121de111a9..9338c47087https://boringssl.googlesource.com/boringssl.git/+log/10a2132f50..23824fa0fe
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/c712e9cc34..189b13f92e
* src/third_party/dav1d/libdav1d: 47107e384b..7b15ca1375f76550541c..9d64acedea7c1b0b524c..79a88d3393a622de35ac..1112fa6b3bhttps://aomedia.googlesource.com/aom.git/+log/0cee19cfc8..a2d599c975
* src/third_party/libc++/src: 9d119c1f4a..1506720cb396b64eaac5..3316c112402f2c04c157..a6a2ec654be01c38d714..4183dabcacab7c5918b4..f9550c3f722b9f1d699f..2b7d7f5046a4279f2842..1f0d2a10bda4279f2842..1f0d2a10bdhttps://webrtc-review.googlesource.com/c/src/+/340402
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41796}
2024-02-23 23:57:11 +00:00
Zoé Lepaul
dc6a001b6b
Update and open WebRtcAudioUtils
...
Opening the visibility of a few methods from this utils class to allow
it to be used by other implementations of `AudioDeviceModule`.
Also updating a few methods, like adding new audio device types from
recent Android SDKs, and updating the definition of an emulator.
Bug: b/287409066
Change-Id: I1473fa0342252347ce92ee2319380ebb14e9885b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339905
Commit-Queue: Zoé Lepaul <xalep@webrtc.org>
Reviewed-by: Ranveer Aggarwal <ranvr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41778}
2024-02-21 11:18:40 +00:00
Danil Chapovalov
0355f455a4
Use Environment propagated through android sdk
...
This way VP8Decoder and DecoderFallback would use propagated instead of global field trials.
Bug: webrtc:15791, webrtc:10335
Change-Id: I5ad5fae38f5b9379bc6376334562c154fbc56e39
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/340040
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41763}
2024-02-19 15:54:39 +00:00
Danil Chapovalov
2eee89e904
Cleanup webrtc::Environment propagation through java wrappers
...
Force and thus guarantee VideoDecoder created through java wrappers get access to the webrtc::Environment
Bug: webrtc:15791
Change-Id: I3f145937c0b914c8e34b24e1ecc55da756551069
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338441
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41745}
2024-02-15 13:33:48 +00:00
Danil Chapovalov
22b6564d77
Propagate webrtc::Environment to create VideoDecoder through java wrappers
...
Bug: webrtc:15791
Change-Id: I25976158ef780211355deb934e83d275d4f881e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336980
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41686}
2024-02-07 11:58:20 +00:00
Henrik Boström
0bf7a6080f
[Stats] Migrate Java to Attributes().
...
To unblock code cleanup, uses of Members() are replaced by
Attributes(). This will allow replacing RTCStatsMember<T> with
absl::optional<T> in a future CL.
Bug: webrtc:15164
Change-Id: Ic4ca0d333a9cdaa78739bd8dd4515d2ecfacffbc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334162
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41512}
2024-01-12 10:33:31 +00:00
Bjorn Terelius
3a20023719
Roll chromium_revision 49983e8c01..93bbe3fbc0 (1242712:1244582)
...
Change log: 49983e8c01..93bbe3fbc049983e8c01..93bbe3fbc01a6ee27f96..44d8736657b4c3d1df5f..e920e5bbda5e016b7d32..f5d99b3266566a1bd097..6103905d1b5d325c28aa..ee8b405f3268c88f4ea9..dd54978044f49c2f1a20..22f54f197fhttps://chromium.googlesource.com/catapult.git/+log/65730c4295..f582f5bb51
* src/third_party/depot_tools: 259976c748..6953ebe3c1a07ca46cd0..7bac4d146a920c40cd43..155015f4be15db46be4e..8c2468e9dd7451ba4b85..f80f02a81e2602aecdf4..42293b96f8805d611c93..8650986e8c1be790cb8e..708b76cf0749983e8c01..93bbe3fbc049983e8c01..93bbe3fbc0https://webrtc-review.googlesource.com/c/src/+/333880
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41488}
2024-01-09 14:06:11 +00:00
Bjorn Terelius
9702f6c9fb
Roll chromium_revision fd0452ac3a..49983e8c01 (1242677:1242712)
...
Change log: fd0452ac3a..49983e8c01fd0452ac3a..49983e8c0161c7041d9f..566a1bd0979eafcc3b22..5d325c28aad012611f35..68c88f4ea9https://chromium.googlesource.com/catapult.git/+log/6d3a4756c7..65730c4295
* src/third_party/depot_tools: 0e40b92d9e..259976c748ca76683b78..a07ca46cd0https://aomedia.googlesource.com/aom.git/+log/4f632e5b6d..0eeb62d344
* src/third_party/libc++abi/src: 8806fb8bb2..7451ba4b857fc833e889..f477acb1042d7122e93a..805d611c939db0dc9c4a..1be790cb8efd0452ac3a..49983e8c01https://webrtc-review.googlesource.com/c/src/+/333401
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41481}
2024-01-08 14:42:12 +00:00
Danil Chapovalov
151003d341
Deprecate RtcEventLogFactory constructor taking unused parameter
...
Bug: webrtc:15656
Change-Id: I22ed4cca4c0ce7ebf9c533ed7434617bf0a0f4a3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330120
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41338}
2023-12-07 21:46:56 +00:00
Tomas Gunnarsson
3a15ba6fbf
Reland^2 "Reland: Remove unsupported configuration value, allow_codec_switching"
...
This reverts commit 117d847901ea231cd86ca152b359b88619b9de20.
Reason for revert: Downstream error has been corrected.
Original change's description:
> Revert "Reland: Remove unsupported configuration value, `allow_codec_switching`"
>
> This reverts commit 23501a2aa656b94e26d4c67b8b9393258551560f.
>
> Reason for revert: Breaks downstream features
>
> Original change's description:
> > Reland: Remove unsupported configuration value, `allow_codec_switching`
> >
> > This reverts commit 6b0c5babe0700f12493cf659e1b35c58d2327995.
> >
> > Reason for revert: Relanding once downstream issues have been addressed
> >
> > Original change's description:
> > > Revert "Remove unsupported configuration value, `allow_codec_switching`"
> > >
> > > This reverts commit 8f7a17f80f43a47ce3801a3cfd2afda3575c8023.
> > >
> > > Reason for revert: breaks downstream
> > >
> > > Original change's description:
> > > > Remove unsupported configuration value, `allow_codec_switching`
> > > >
> > > > Bug: webrtc:11341
> > > > Change-Id: I8ff598848996bd63ccc572e11f8f69c892a4a459
> > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324284
> > > > Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> > > > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> > > > Cr-Commit-Position: refs/heads/main@{#40995}
> > >
> > > Bug: webrtc:11341
> > > Change-Id: I784fd95062fc71f8dcc139b05121985f60709004
> > > No-Presubmit: true
> > > No-Tree-Checks: true
> > > No-Try: true
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324780
> > > Owners-Override: Philip Eliasson <philipel@webrtc.org>
> > > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> > > Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> > > Cr-Commit-Position: refs/heads/main@{#40998}
> >
> > Bug: webrtc:11341
> > Change-Id: I3cb3e699fd76942c51f0f42a99bcb19ac607632e
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324782
> > 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@{#41032}
>
> Bug: webrtc:11341
> Change-Id: I0eb8e6a464a8a51e6359caf8f43231dc275c4f20
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327382
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41161}
Bug: webrtc:11341
Change-Id: I4a5390a3b8c5e665b742fc564709847ad8853ba9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328160
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#41213}
2023-11-22 13:22:08 +00:00
Tomas Gunnarsson
117d847901
Revert "Reland: Remove unsupported configuration value, allow_codec_switching"
...
This reverts commit 23501a2aa656b94e26d4c67b8b9393258551560f.
Reason for revert: Breaks downstream features
Original change's description:
> Reland: Remove unsupported configuration value, `allow_codec_switching`
>
> This reverts commit 6b0c5babe0700f12493cf659e1b35c58d2327995.
>
> Reason for revert: Relanding once downstream issues have been addressed
>
> Original change's description:
> > Revert "Remove unsupported configuration value, `allow_codec_switching`"
> >
> > This reverts commit 8f7a17f80f43a47ce3801a3cfd2afda3575c8023.
> >
> > Reason for revert: breaks downstream
> >
> > Original change's description:
> > > Remove unsupported configuration value, `allow_codec_switching`
> > >
> > > Bug: webrtc:11341
> > > Change-Id: I8ff598848996bd63ccc572e11f8f69c892a4a459
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324284
> > > Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> > > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> > > Cr-Commit-Position: refs/heads/main@{#40995}
> >
> > Bug: webrtc:11341
> > Change-Id: I784fd95062fc71f8dcc139b05121985f60709004
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324780
> > Owners-Override: Philip Eliasson <philipel@webrtc.org>
> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> > Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#40998}
>
> Bug: webrtc:11341
> Change-Id: I3cb3e699fd76942c51f0f42a99bcb19ac607632e
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324782
> 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@{#41032}
Bug: webrtc:11341
Change-Id: I0eb8e6a464a8a51e6359caf8f43231dc275c4f20
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327382
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41161}
2023-11-15 08:10:28 +00:00
Raman Budny
36ed560339
Fall back to software encoding on wrong HW video encoder configuration
...
Catch one more IllegalArgumentException to avoid crashes on some devices.
Bug: webrtc:15636
Change-Id: I396473b409a1ceba8f4a91d5e4aa66d5fe3b0f44
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326002
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41113}
2023-11-09 12:21:11 +00:00
Danil Chapovalov
779c9dede9
Migrate CreatePeerConnectionFactory implementation to EnableMedia api
...
Bug: webrtc:15574
Change-Id: I2e109a62a9069f37a580fa64cacdd5a86a293203
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325860
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41069}
2023-11-02 23:01:31 +00:00
philipel
3ee8117856
H265 build fix for Android.
...
Build fix for H265 on Android so that https://webrtc-review.googlesource.com/c/src/+/325482 can land.
gn args:
target_os = "android"
proprietary_codecs = true
Bug: webrtc:15620
Change-Id: I8a134afbc50137ac17ce9a4a57d68dd3f3c6d52f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325483
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41053}
2023-10-31 16:21:02 +00:00
Tomas Gunnarsson
23501a2aa6
Reland: Remove unsupported configuration value, allow_codec_switching
...
This reverts commit 6b0c5babe0700f12493cf659e1b35c58d2327995.
Reason for revert: Relanding once downstream issues have been addressed
Original change's description:
> Revert "Remove unsupported configuration value, `allow_codec_switching`"
>
> This reverts commit 8f7a17f80f43a47ce3801a3cfd2afda3575c8023.
>
> Reason for revert: breaks downstream
>
> Original change's description:
> > Remove unsupported configuration value, `allow_codec_switching`
> >
> > Bug: webrtc:11341
> > Change-Id: I8ff598848996bd63ccc572e11f8f69c892a4a459
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324284
> > Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#40995}
>
> Bug: webrtc:11341
> Change-Id: I784fd95062fc71f8dcc139b05121985f60709004
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324780
> Owners-Override: Philip Eliasson <philipel@webrtc.org>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#40998}
Bug: webrtc:11341
Change-Id: I3cb3e699fd76942c51f0f42a99bcb19ac607632e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324782
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@{#41032}
2023-10-28 16:07:41 +00:00
Jakob Ivarsson
9efd080fa2
Implement GetStats in Android ADM.
...
Calls the AudioOutput implementation of GetStats, which is currently
not implemented.
Bug: webrtc:14653
Change-Id: Ieaf0e0c030a95d23c8950ff9038a64426142a789
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324800
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41003}
2023-10-25 07:50:16 +00:00
Philip Eliasson
6b0c5babe0
Revert "Remove unsupported configuration value, allow_codec_switching"
...
This reverts commit 8f7a17f80f43a47ce3801a3cfd2afda3575c8023.
Reason for revert: breaks downstream
Original change's description:
> Remove unsupported configuration value, `allow_codec_switching`
>
> Bug: webrtc:11341
> Change-Id: I8ff598848996bd63ccc572e11f8f69c892a4a459
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324284
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#40995}
Bug: webrtc:11341
Change-Id: I784fd95062fc71f8dcc139b05121985f60709004
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324780
Owners-Override: Philip Eliasson <philipel@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40998}
2023-10-24 08:19:46 +00:00
Tommi
8f7a17f80f
Remove unsupported configuration value, allow_codec_switching
...
Bug: webrtc:11341
Change-Id: I8ff598848996bd63ccc572e11f8f69c892a4a459
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324284
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40995}
2023-10-24 05:07:25 +00:00
Linus Nilsson
f166fe245c
Avoid excessive eglMakeCurrent calls
...
With shared EglConnections each client must ensure their EGLSurface
is made current every time they access the thread. This will lead to
unnecessary eglMakeCurrent calls when the EGLSurface is in fact already
current, such as when the EglConnection only has one client or when one
client accesses the thread without interruption.
Bug: b/217863437
Change-Id: I1b03daec4d5cd43af21fe9c168e3637f676b6fec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322322
Reviewed-by: Ranveer Aggarwal <ranvr@webrtc.org>
Commit-Queue: Linus Nilsson <lnilsson@webrtc.org>
Reviewed-by: Zoé Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40883}
2023-10-06 14:20:28 +00:00
Danil Chapovalov
2d508f10d3
Deprecate old names for EncodedImage::RtpTimestamp accessor and setter
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Replace remaining webrtc usage of the deprecated names.
Bug: webrtc:9378
Change-Id: Ie5bd2d3eaf68316e7c827fc35c7c7d8e2eadeb9f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321584
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40824}
2023-09-28 07:29:22 +00:00
qwu16
ae82df718c
Add codec name H265 to support H265 in WebRTC
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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}
2023-09-20 09:25:32 +00:00
Per Åhgren
9acda0b8ac
Add support for toggling noise suppression effect on Android
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Change-Id: I4868bd6531bde08c4108b0941086add210660dcb
Bug: b/279738239
Change-Id: I4868bd6531bde08c4108b0941086add210660dcb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318320
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40672}
2023-08-31 13:16:58 +00:00
Mirko Bonadei
c59be6d261
Roll chromium_revision 10080947c0..6b95b8aa08 (1174188:1175249)
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Change log: 10080947c0..6b95b8aa0810080947c0..6b95b8aa083de7d110cb..dcfe245ca13dd34519f9..b74cdc4550ca163845c7..16be42a9ffa265a85ace..51a637843f85b0f51488..b946312a8953a08ec089..93b5f4c408https://chromium.googlesource.com/catapult.git/+log/b119dc414e..84e3795d98
* src/third_party/depot_tools: d411904b84..54e86436a85769f13a6b..9e3c5d7e18e568f2855d..ab16995d921a0f13f46a..4057b9894310080947c0..6b95b8aa08https://webrtc-review.googlesource.com/c/src/+/313200
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Christoffer Jansson <jansson@webrtc.org>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40520}
2023-08-07 13:57:06 +00:00
Yaowen Guo
6fc700ec3d
Rland "Revert "Reland "Reland "Delete old Android ADM.""""
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This reverts commit 7534ebd2bf59212cce5e010dd6ed9b3bc944818e.
Reason for revert: Downstream projects have been updated, try it again.
R=perkj@webrtc.org
Bug: webrtc:7452
Change-Id: Ice48a563a6da499b6050b6f6e21bb0a18cd34f57
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271841
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40386}
2023-06-30 13:10:12 +00:00
Yury Yarashevich
87e74f9fb7
Remove unused combined_audio_video_bwe.
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Bug: None
Change-Id: Ie539351f98b7a0ebb5f08e0df5c5759a2bcb5588
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306520
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Yury Yarashevich <yura.yaroshevich@gmail.com>
Cr-Commit-Position: refs/heads/main@{#40160}
2023-05-26 15:56:00 +00:00
David Liu
784c339f34
Expose setCodecPreferences/getCapabilities for android
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Bug: webrtc:15177
Change-Id: If61ef9a87bc4f68d73cef6e681461682ca48f034
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304880
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40071}
2023-05-15 19:24:15 +00:00
Linus Nilsson
df4bc33e11
Allow EglBase instances to share EGLConnection.
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This enables clients of EglBase to keep using it but
share underlying EGLContext with other clients.
go/meet-android-eglcontext-reduction
Bug: b/225229697
Change-Id: I42719f25be7db169c39878b57a5f1487e3c1894e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301941
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Linus Nilsson <lnilsson@webrtc.org>
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39961}
2023-04-27 19:59:05 +00:00
Sergey Silkin
0421294df0
Enable RTC mode in Google HW AV1 encoder
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Bug: b/274179852
Change-Id: Id987fa8b73468532304f856bc97d40238f93b266
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302180
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39921}
2023-04-21 16:38:06 +00:00
Jared Siskin
6f86f6af00
Format /sdk
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git ls-files | grep -e "\(\.h\|\.cc\)$" | grep -e "^sdk/" | xargs clang-format -i ; git cl format
after landing: add to .git-blame-ignore-revs
Bug: webrtc:15082
Change-Id: If05d3c7555c4f2bf25e387249932787a93aa39c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302060
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39913}
2023-04-21 04:30:57 +00:00
Sergey Silkin
88429d572b
Account for stride when calculating buffer size
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https://webrtc-review.googlesource.com/c/src/+/240680 made encoder aware of stride and slice height of input buffer but calculation of buffer size passed to queueInputBuffer() was not updated.
Bug: webrtc:13427
Change-Id: Iba8687f56eda148ac67b331d35c45317a4ec5c59
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301321
Reviewed-by: Mirta Dvornicic <mirtad@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39895}
2023-04-19 10:10:32 +00:00
Sergey Silkin
6cf12bbe32
Fetch encoded QP before releasing output buffer
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Before this change we first released output frame buffer in the code path which prepends config buffer to a keyframe and then called getOutputFormat(index). getOutputFormat(index) throws an exception if index points to a released buffer. This change rearranges calls such that getOutputFormat(index) always executed before releaseOutputBuffer(index).
Bug: webrtc:15015
Change-Id: Ia64f5d8e7483aeeb316d1a71c0cb79233f4bbee9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301364
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39874}
2023-04-17 12:31:32 +00:00
Sergey Silkin
89facfc421
Fix potential null pointer access in HardwareVideoEncoder
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There was no check for null in the code that prepends config buffer to key frame buffer. It is not a requirement for codec to deliver config buffer. Some codecs probably do not do that.
Bug: none
Change-Id: Id9c57efc5d1de5f569fa773313da4db3cd8b074c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299900
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39807}
2023-04-11 13:50:41 +00:00
Philip Eliasson
7454fdd12b
Revert "Replace BuiltinVideo{Encoder,Decoder}Factory with Video{Encoder,Decoder}FactoryTemplate."
...
This reverts commit d2535a53cf014e4973b92bb6f00d7a2b87cd02c2.
Reason for revert: breaks downstream
Original change's description:
> Replace BuiltinVideo{Encoder,Decoder}Factory with Video{Encoder,Decoder}FactoryTemplate.
>
> Bug: webrtc:13573
> Change-Id: I07e4fe9be938ba2540351b73ff22a090c68afa00
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299663
> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39777}
Bug: webrtc:13573
Change-Id: I4d60383a46db4fdddd61e58b53c4ed07773434b2
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300543
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Owners-Override: Philip Eliasson <philipel@webrtc.org>
Auto-Submit: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39783}
2023-04-06 14:21:30 +00:00
philipel
d2535a53cf
Replace BuiltinVideo{Encoder,Decoder}Factory with Video{Encoder,Decoder}FactoryTemplate.
...
Bug: webrtc:13573
Change-Id: I07e4fe9be938ba2540351b73ff22a090c68afa00
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299663
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39777}
2023-04-06 09:04:11 +00:00
Sergey Silkin
910b225d82
Fetch encoded QP from MediaCodec encoders
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It is a part of "encoding statistics" feature [1] available in Android SDK 33. Local testing revealed that for HW VP8/9 encoders we get QP in range [0,64] which is not what WebRTC quality scaler expects. Exclude VP8/9 encoders for now.
[1] https://developer.android.com/reference/android/media/MediaFormat#VIDEO_ENCODING_STATISTICS_LEVEL_1
Bug: webrtc:15015
Change-Id: I8af2fd96afb34e18cb3e2cc3562b10149324c16e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298306
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39722}
2023-03-30 10:19:22 +00:00
Sergey Silkin
2d1fa4713f
Use MediaCodec API keys and values directly
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Replace locally-defined keys and values with constants from MediaCodec API (MediaFormat.KEY_..., etc). Value of a constant field is resolved at compile time according to 13.1.1 [1]. I.e., it is safe to reference a constant field not available in older API (MediaCodec API ignores unrecognized MediaFormat.KEY_ values).
[1] https://docs.oracle.com/javase/specs/jls/se20/html/jls-13.html#jls-13.1
Bug: none
Change-Id: I3c63cfd67cc22db1b957f908508b434d36d88806
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298940
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Linus Nilsson <lnilsson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39668}
2023-03-24 13:59:54 +00:00
Harald Alvestrand
041ecb87f5
New PeerConnectionFactory::CreateVideoTrack with refcounted source
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Bug: webrtc:15017
Change-Id: I04c794d8959583bb4cc5c3898f4175783ec49f16
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249363
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39635}
2023-03-22 09:10:27 +00:00
Sergey Silkin
0af2bc639a
Add H265 to VideoCodecMimeType
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This enables testing HW H265 codecs on devices where the support is available.
Bug: b/261160916, webrtc:14852
Change-Id: I32d102fcf483ea4ba46d6f5161342dbb584e7cc9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298040
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39591}
2023-03-17 15:28:11 +00:00
Sergey Silkin
a02f90691e
Remove resolution alignment requirement (part 2)
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https://webrtc-review.googlesource.com/c/src/+/296340 removed hard resolution alignment requirement from HardwareVideoEncoder.initEncode(). This CL removes the hard resolution alignment requirement from HardwareVideoEncoder.resetCodec().
Bug: webrtc:13089
Change-Id: Ia9fcd4d6a7ea16509ec3f12c3c78a76d1eb6c6f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296520
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39498}
2023-03-07 17:18:55 +00:00
Sergey Silkin
5c978cb262
Remove resolution alignment requirement
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This partially reverts https://webrtc-review.googlesource.com/c/src/+/229460 . HW encoder wrapper still requires 16x16 alignment but InitEncode() doesn't fail if input resolution doesn't satisfy this requirement.
If encoder can't handle 16x16 then it should return error from MediaCodec.configure(). Many HW encoder can handle resolutions not multiple of 16. Having strict requirement for resolution alignment in InitEncode() blocks usage and testing of these encoders.
This change doesn't affect WebRTC in Chrome Android since RTC encoder wrapper requires 16x16 alignment: https://source.chromium.org/chromium/chromium/src/+/main:media/gpu/android/android_video_encode_accelerator.cc;drc=4abbb981443d7403566ef8aff05fdaca1e837da3;l=494
Bug: webrtc:13089
Change-Id: I0c39908bbcd7d0740a9ee8afa022ba443ffdb4de
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296340
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39487}
2023-03-06 16:51:03 +00:00
Sergey Silkin
c494846bdf
Set is_hardware_accelerated=true in JNI decoder wrapper
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Assume that all Java decoders are hardware-accelerated.
Bug: b/261160916, webrtc:14852
Change-Id: I4c61839258c86ec9322b82d291542e2df6fd2ff1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295863
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39457}
2023-03-02 15:12:20 +00:00
Sergey Silkin
2bdf79ac91
Handle frame_types=null in VideoEncoderWrapper::Encode()
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frame_types=null is a valid input for Encoder(). VP8/VP9/H264 software encoder wrappers can handle frame_types=null.
Bug: b/261160916, webrtc:14852
Change-Id: I96ed98d553546ecbe70ff0356f314496f838c535
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293260
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39325}
2023-02-16 11:09:51 +00:00
Sergey Silkin
831664294c
Allow getScalingSettings to be called from any thread
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Scaling settings is a static property of the encoder wrapper. It has no any dependencies on underlaying HW codec and can be fetched from any thread.
getScalingSettings is called from VideoEncoderWrapper ctor [1]. Presence of checkIsOnValidThread() in getScalingSettings caused capturing of thread [2]. That required all following codec calls to be done from the thread that created the codec.
[1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/sdk/android/src/jni/video_encoder_wrapper.cc;drc=c05a1be5b49f5c03b6955b05bcbf47609e1b0381;l=41
[2] https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/rtc_base/java/src/org/webrtc/ThreadUtils.java;l=30
Bug: b/261160916, webrtc:14852
Change-Id: I7bb18bc8e3b371d83ccd44d4a5a096f716c0a0e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291807
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39303}
2023-02-13 15:14:42 +00:00
Philipp Hancke
b81823a5f0
stats: use Timestamp instead of uint64_t
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making it clear what unit is being used.
BUG=webrtc:13756
Change-Id: I6354d35a8e02bb93a905ccf32cb0b294b4813e41
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/289460
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39008}
2023-01-05 08:37:31 +00:00
Byoungchan Lee
4a680f11ae
Removed outdated comment of Notifier
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Notifier is thread-hostile, and we have added a SequenceChecker
on https://webrtc-review.googlesource.com/c/src/+/252520 ,
so this comment is no longer needed.
Bug: None
Change-Id: I39f7f75a786dd27d2f6299d85676e7182d9032eb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288180
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38899}
2022-12-15 13:17:41 +00:00
