Commit Graph

23 Commits

Author SHA1 Message Date
266021dfa2 Add support for DegradationPreference in Android SDK
This wires the current degradation preference in the SDK, it will later
be nullable in a follow up change once the native API supports it.

Bug: webrtc:11164
Change-Id: I8324e6e0af996dfddfa07e3aff4ba242d9533388
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161321
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30170}
2020-01-07 17:20:41 +00:00
e725fdbcc1 Encoding constructor should be public.
Bug: webrtc:11157
Change-Id: Ic921b39c0aa6b24deef8452ec28352c16c193a1e
TBR: sakal@webrtc.org
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161151
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29990}
2019-12-03 16:54:20 +00:00
9f9e20a3dc Fix errorprone issues preventing Chromium Roll.
Some ErrorProne warnings have been enabled by [1], that broke the
Chromium Roll into WebRTC, this CL should have taken care of all the
problems.

[1] - https://chromium-review.googlesource.com/c/chromium/src/+/1935889

Bug: None
Change-Id: I2670e948c320984a122fdb774b891c98e05f582e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160862
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29933}
2019-11-27 12:52:48 +00:00
ae4b62318d Reland "Add bindings for simulcast and RIDs in Android SDK."
This is a reland of 177670afd6d4aa414e4aa75983da538b7f350ee8
Fixing failing tests.

TBR=magjed@webrtc.org
Original change's description:
> Add bindings for simulcast and RIDs in Android SDK.
>
> This adds the bindings for rid in RtpParameters.Encoding and bindings
> for send_encodings in RtpTransceiverInit to allow creating a transceiver
> with multiple send encodings.
>
> Bug: webrtc:10464
> Change-Id: I4c205dc0f466768c63b7efcb3c68e93277236da0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128960
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Seth Hampson <shampson@webrtc.org>
> Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27323}

Bug: webrtc:10464
Change-Id: I95fac3967217c20a9fdddb490aea30eca2061ef0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130362
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27402}
2019-04-01 17:45:46 +00:00
e8bc3a0a5a Revert "Add bindings for simulcast and RIDs in Android SDK."
This reverts commit 177670afd6d4aa414e4aa75983da538b7f350ee8.

Reason for revert: Fails android_instrumentation_test_apk:
https://ci.chromium.org/p/webrtc/builders/ci/Android64%20(M%20Nexus5X)/11553

Original change's description:
> Add bindings for simulcast and RIDs in Android SDK.
>
> This adds the bindings for rid in RtpParameters.Encoding and bindings
> for send_encodings in RtpTransceiverInit to allow creating a transceiver
> with multiple send encodings.
>
> Bug: webrtc:10464
> Change-Id: I4c205dc0f466768c63b7efcb3c68e93277236da0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128960
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Seth Hampson <shampson@webrtc.org>
> Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27323}

TBR=magjed@webrtc.org,shampson@webrtc.org,amithi@webrtc.org

Change-Id: Id6c4e2d41c3c2fbfad31baed907cfa73d82ef14a
No-Tree-Checks: True
No-Try: True
Bug: webrtc:10464
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130466
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27354}
2019-03-29 15:51:07 +00:00
177670afd6 Add bindings for simulcast and RIDs in Android SDK.
This adds the bindings for rid in RtpParameters.Encoding and bindings
for send_encodings in RtpTransceiverInit to allow creating a transceiver
with multiple send encodings.

Bug: webrtc:10464
Change-Id: I4c205dc0f466768c63b7efcb3c68e93277236da0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128960
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27323}
2019-03-27 18:06:00 +00:00
d8b980464c Add scaleResolutionDownBy to RtpParameters.Encoding in Android SDK.
Bug: webrtc:10069
Change-Id: I8130836c495d5584ca3e11e9e3155916b871ab21
Reviewed-on: https://webrtc-review.googlesource.com/c/120926
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26535}
2019-02-04 16:50:24 +00:00
e1301a8b3a Revert "Implement read-only codecPayloadType in RtpParameters"
This reverts commit 806e06d1366b58878ced05cdd8d1d56394982fe6.

Reason for revert: Breaks WebRTC roll to Chromium. https://chromium-review.googlesource.com/c/chromium/src/+/1375538

02:52:35.346 7748   [6936:11248:1213/025234.206:ERROR:mediaengine.cc(80)] Attempted to set RtpParameters with modified codecPayloadType (INVALID_MODIFICATION)

Original change's description:
> Implement read-only codecPayloadType in RtpParameters
> 
> Bug: webrtc:7580
> Change-Id: I6d901afa97262b6c6d9fe6c7366df465ec77bfb3
> Reviewed-on: https://webrtc-review.googlesource.com/c/113944
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Seth Hampson <shampson@webrtc.org>
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25993}

TBR=steveanton@webrtc.org,sakal@webrtc.org,andersc@webrtc.org,shampson@webrtc.org,orphis@webrtc.org

Change-Id: I157f9a79ae7133395431891e15e2c053559d359b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7580
Reviewed-on: https://webrtc-review.googlesource.com/c/114300
Reviewed-by: Henrik Grunell <henrikg@webrtc.org>
Commit-Queue: Henrik Grunell <henrikg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26000}
2018-12-13 12:13:30 +00:00
806e06d136 Implement read-only codecPayloadType in RtpParameters
Bug: webrtc:7580
Change-Id: I6d901afa97262b6c6d9fe6c7366df465ec77bfb3
Reviewed-on: https://webrtc-review.googlesource.com/c/113944
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25993}
2018-12-12 16:24:29 +00:00
69540f4419 Use android Nullable instead of javax Nullable
This is a propagation of upstream chromium change needed to
resume DEPS autorolls into WebRTC.

Original comment from upstream change:

> This change is made in preparation for an ErrorProne
> check to catch this at compile time. See bug for details.

Bug: chromium:771683
Change-Id: I56aed15f73a633dcadae7ece6c645cd3596f9257
Reviewed-on: https://webrtc-review.googlesource.com/c/113505
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Artem Titarenko <artit@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25951}
2018-12-10 15:03:58 +00:00
23eba22424 Add support for RtpEncodingParameters num_temporal_layers.
Configuring different number of temporal layers per simulcast layer is not supported.

Bug: webrtc:9785
Change-Id: I5709b2235233420e22e68fb0ae512305ae87e36c
Reviewed-on: https://webrtc-review.googlesource.com/c/102120
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24942}
2018-10-03 07:22:51 +00:00
4e5342f06a Android: Add maxFramerate to RtpParameters.
Bug: webrtc:9597
Change-Id: I1049b66860abbd69c4822756dee452b0db459ed4
Reviewed-on: https://webrtc-review.googlesource.com/91440
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24789}
2018-09-24 09:18:39 +00:00
abe301fe6c Add HeaderExtensions to RtpParameters
Bug: webrtc:7580
Change-Id: I4fcf3e8bc4975a6b2baa6f24a17c254d2bf521d9
Reviewed-on: https://webrtc-review.googlesource.com/78288
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23584}
2018-06-12 17:01:40 +00:00
613591a5b8 Add min_bitrate_bps to RtpEncodingParameters.
This CL adds the field but does not implement any functionality using it.

Bug: webrtc:9341
Change-Id: I533fc7f8bc1e40207aa16b834e0d7daa60709614
Reviewed-on: https://webrtc-review.googlesource.com/78741
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23466}
2018-05-31 10:17:57 +00:00
dacec71b16 Add Rtcp parameters for PeerConnection senders
Bug: webrtc:7580
Change-Id: Ibcf5e849a1f11f21fa75f6d006fecf1cd54f8552
Reviewed-on: https://webrtc-review.googlesource.com/78063
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23407}
2018-05-28 09:28:59 +00:00
b7d9d8346f Implement RtpCodecParameters::parameters
This will return all the fmtp parameters for the codecs, except for
DTMF codes that don't fit the key=value pattern.

Bug: webrtc:7112
Change-Id: I06a203ff64df2c3bc9bc2082cd0f374718b23510
Reviewed-on: https://webrtc-review.googlesource.com/71801
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23250}
2018-05-15 17:12:02 +00:00
cebf50ff75 Reland "Implement RtpParameters.transaction_id for PC RtpSenderInterface"
This is a reland of 5faf36ef3c582350fba5ef97a3549e440d81a283
The issue in Chrome has been fixed and this should be safe to reland.

TBR=deadbeef

Original change's description:
> Implement RtpParameters.transaction_id for PC RtpSenderInterface
>
> The transaction_id field should be refreshed for every getParameters()
> call and checked at each setParameters() call.
> This also checks that getParameters() was ever called to return a proper
> error code.
>
> Bug: webrtc:7580
> Change-Id: I6c6fe289542e486fc422cdc61577982b0529d4c1
> Reviewed-on: https://webrtc-review.googlesource.com/70820
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23120}

Bug: webrtc:7580
Change-Id: Iabd41fb21afdf452c039d5513824ae334f8d1d3f
Reviewed-on: https://webrtc-review.googlesource.com/76980
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23247}
2018-05-15 15:51:02 +00:00
909338b027 Revert "Implement RtpParameters.transaction_id for PC RtpSenderInterface"
This reverts commit 5faf36ef3c582350fba5ef97a3549e440d81a283.

Reason for revert: fast/peerconnection/RTCRtpSender-setParameters.html
 failing in webrtc roll, probably this CL? https://chromium-review.googlesource.com/c/chromium/src/+/1045889.

Original change's description:
> Implement RtpParameters.transaction_id for PC RtpSenderInterface
> 
> The transaction_id field should be refreshed for every getParameters()
> call and checked at each setParameters() call.
> This also checks that getParameters() was ever called to return a proper
> error code.
> 
> Bug: webrtc:7580
> Change-Id: I6c6fe289542e486fc422cdc61577982b0529d4c1
> Reviewed-on: https://webrtc-review.googlesource.com/70820
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23120}

TBR=steveanton@webrtc.org,deadbeef@webrtc.org,sakal@webrtc.org,kthelgason@webrtc.org,orphis@webrtc.org

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

Bug: webrtc:7580
Change-Id: I86da108227f8fc8d235bb2e9559377c800595b8c
Reviewed-on: https://webrtc-review.googlesource.com/74740
Reviewed-by: Max Morin <maxmorin@webrtc.org>
Commit-Queue: Max Morin <maxmorin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23134}
2018-05-07 08:02:34 +00:00
5faf36ef3c Implement RtpParameters.transaction_id for PC RtpSenderInterface
The transaction_id field should be refreshed for every getParameters()
call and checked at each setParameters() call.
This also checks that getParameters() was ever called to return a proper
error code.

Bug: webrtc:7580
Change-Id: I6c6fe289542e486fc422cdc61577982b0529d4c1
Reviewed-on: https://webrtc-review.googlesource.com/70820
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23120}
2018-05-04 13:07:25 +00:00
e7592d8d5f Annotate libjingle_peerconnection_java with @Nullable.
Bug: webrtc:8881
Change-Id: Ida2ef6c003567d19529c21629c916ed40e8de3a6
Reviewed-on: https://webrtc-review.googlesource.com/63380
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22563}
2018-03-22 13:13:44 +00:00
9060eb1528 Android: Generate JNI code for remaining classes in sdk/android
Bug: webrtc:8278
Change-Id: I20a4388ab347d8745d0edde808f7a0b610f077f9
Reviewed-on: https://webrtc-review.googlesource.com/31484
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21227}
2017-12-12 12:36:17 +00:00
6062f372c7 Android: Fix warnings
TBR=sakal@webrtc.org

Bug: webrtc:6597,webrtc:8534
Change-Id: I39f96d9e0e6d604051b1cc13368dd44fc82b30b0
Reviewed-on: https://webrtc-review.googlesource.com/23622
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20719}
2017-11-16 18:25:33 +00:00
bb547203bf Moving src/webrtc into src/.
In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
2017-09-15 04:25:06 +00:00