This CL does the following:
* Split out MediaStream JNI code from peerconnection.cc to mediastream.h/mediastream.cc.
* Split out RtpSender JNI code from peerconnection.cc to rtpsender.h/rtpsender.cc.
* Split out TurnCustomizer JNI code from peerconnection.cc to turncustomizer.h/turncustomizer.cc.
* Add missing instanceof function to WrappedNativeVideoDecoder.java.
* Move some PeerConnectionFactory JNI declarations from pc/video.cc to peerconnectionfactory.cc.
* Add declaration to video.h for the JNI functions that depend on EglBase14_jni.h.
* Use a scoped object to store the global Java MediaStream objects that also call dispose.
Bug: webrtc:8278
Change-Id: I3c56a599b8bcbc8f34e5c5a7b9c9fe1d192ff3f3
Reviewed-on: https://webrtc-review.googlesource.com/34645
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21380}
This allows application to construct PeerConnection object in JNI and
pass that to Android API. API for wrapping Java PeerConnection Observers
is exposed for convenience.
Bug: webrtc:8662
Change-Id: Id110b92e6bb5ab00661cd50616d05c3e18a1697d
Reviewed-on: https://webrtc-review.googlesource.com/34520
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21379}
Bug: webrtc:8620
Change-Id: I9b0b6d8dc9bda366e925dda9a5b92fc4e3fd9f43
Reviewed-on: https://webrtc-review.googlesource.com/35003
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21378}
SDP is a detail of PeerConnection and is not used by anything in p2p, so
it belongs in the pc/ directory. This also allows
MediaContentDescription to be inlined in the future.
Bug: webrtc:8620
Change-Id: I38b65ede9942e29eb15035ab29f2be988da1e5ce
Reviewed-on: https://webrtc-review.googlesource.com/33781
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21376}
This change allows work to be done in parallel for lower level implementation and wiring/exposing multiple simulcast layer's encoding parameters at the api interface.
Bug: webrtc:8653
Change-Id: I89c9a6af0786134771d28526056759bd63213a0a
Reviewed-on: https://webrtc-review.googlesource.com/32902
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21375}
This is a reland of 4770fd935ac92400487bddd3b755753572e6d692
Original change's description:
> Move JsepTransport from p2p/base to pc/.
>
> The JsepTransport class is moved to pc/ and the utility methods and
> enums are moved to where they are used.
>
> With JsepTransport moved to pc/, JsepTransport can depend on objects in
> pc/ including RtpTranport, SrtpTransport etc.
>
> Forked from https://webrtc-review.googlesource.com/c/src/+/31762/7
>
> Bug: webrtc:8636
> Change-Id: I4e8569fe3012946e87deb280f6139f0fd98de34d
> Reviewed-on: https://webrtc-review.googlesource.com/33701
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21333}
Bug: webrtc:8636
Change-Id: Ibce42be898b96dd8e0266b595611d2ffc86581a8
Reviewed-on: https://webrtc-review.googlesource.com/34586
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21371}
Also gets rid of refs to bug 7504, which is now closed.
Bug: webrtc:7504
Change-Id: I105355a5372ad9c2ae8ef52ae275cb4037731c3d
Reviewed-on: https://webrtc-review.googlesource.com/34643
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21366}
Trivial change. Adding Alex as TBR. Same log exists for playout already.
This change makes is easier to compare logs.
NOTRY=TRUE
TBR=glaznev
Bug: NONE
Change-Id: I5dd23a4435d7816d8c171a0769132ac9d2f7f5aa
Reviewed-on: https://webrtc-review.googlesource.com/34654
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21361}
C++ API allows passing all configuration through RTCConfiguration
object. This adds all values previously passed through PC constraints
to Java RTCConfiguration object and deprecates API that takes PC
contraints.
Using the deprecated API overrides the values in RTCConfigration
object.
Bug: webrtc:8663, webrtc:8662
Change-Id: I128432c3caba74403513fb1347ff58830c643885
Reviewed-on: https://webrtc-review.googlesource.com/33460
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21357}
This reverts commit 28a06b16cc4daa9f380ad45af8acfd11b6057283.
Reason for revert: Causes some unexpected perf changes.
Original change's description:
> Smoother frame dropping when screenshare_layers limits fps
>
> Currently, when input fps is higher than the configured target fps in
> screenshare_layers, we drop frames with the help of a rate tracker using
> a one second sliding window. This is not optimal.
> For instance, given a 5fps limit and a 30fps capturer, the window will
> not be saturated until we have added the first 5 frames. Then we will
> drop all frames until the oldest one drops out, at which point we can
> immediately fill it's place. This results in quick 5 frame bursts every
> second.
>
> In addition to rate tracker, also set a limit on minimum interval
> required between input frames, based on target frame rate.
>
> Bug: webrtc:4172
> Change-Id: I49f0abf4d549673cc6b3fafe580b529ea3feaf57
> Reviewed-on: https://webrtc-review.googlesource.com/34380
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21325}
TBR=ilnik@webrtc.org,sprang@webrtc.org
Change-Id: I7ca5b0c847310dbb11dce28773082eac946c0ba4
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:4172
Reviewed-on: https://webrtc-review.googlesource.com/34780
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21354}
On some devices `android.hardware.Camera.open` returns null
instead of raising exception. It causes `NPE` inside
`Camera1Session.create` when method `setPreviewTexture` is
invoked on local variable `camera`, which is `null`.
Bug: webrtc:8658
Change-Id: Ic65b4aef2c0b8b65735a9db02433b536bfe92ddd
Reviewed-on: https://webrtc-review.googlesource.com/33620
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21352}
I realized I could use configs to fix some duplication that I
partially introduced.
Verified APM_DEBUG_DUMP is set appropriately by looking at the
compiler command line.
Bug: webrtc:6828
Change-Id: Ia990e2721546d65639567cd3ab788439e328c5da
Reviewed-on: https://webrtc-review.googlesource.com/34642
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21349}
Changing window type which improves the filter accuracy
at the cost of a slight reduction in convergence time.
Bug: webrtc:8661
Change-Id: Id0e5c66ec179f94471cbca0a2b8d1b94d8146ca6
Reviewed-on: https://webrtc-review.googlesource.com/34501
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21347}
https://webrtc.googlesource.com/src.git/+/26246cac660a95f439b7d1c593edec2929806d3f
that was reverted due to compile error on windows.
Changes since last is an addition of a cast to uint16_t in stun.cc:1018.
---
Add RelayPortFactoryInterface that allows for custom relay (e.g turn) ports
This patch adds a RelayPortFactoryInterface that allows
for custom relay ports. The factor is added as optional argument
to BasicPortAlloctor. If none is provided a default implementation
that mimics existing behavior is created.
The patch also adds 2 stun functions, namely to copy a
StunAttribute and to remove StunAttribute's from a StunMessage.
Bug: webrtc:8640
Change-Id: If23638317130060286f576c94401de55c60a1821
Reviewed-on: https://webrtc-review.googlesource.com/34181
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21345}
The stats collectors would only ever call this on the signaling
thread, so they might as well just ask the voice/video channel
their transport_name directly.
Bug: None
Change-Id: I8dd36210ff22d222b45b5f5e22c253f601cdc79e
Reviewed-on: https://webrtc-review.googlesource.com/34581
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21339}
The functionality is moved into AudioState.
TBR: henrika@webrtc.org
Bug: webrtc:4690
Change-Id: I015482ad18a39609634f6ead9e991d5210107f0f
Reviewed-on: https://webrtc-review.googlesource.com/34502
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21338}
This reverts commit 4770fd935ac92400487bddd3b755753572e6d692.
Reason for revert: breaks downstream projects
Original change's description:
> Move JsepTransport from p2p/base to pc/.
>
> The JsepTransport class is moved to pc/ and the utility methods and
> enums are moved to where they are used.
>
> With JsepTransport moved to pc/, JsepTransport can depend on objects in
> pc/ including RtpTranport, SrtpTransport etc.
>
> Forked from https://webrtc-review.googlesource.com/c/src/+/31762/7
>
> Bug: webrtc:8636
> Change-Id: I4e8569fe3012946e87deb280f6139f0fd98de34d
> Reviewed-on: https://webrtc-review.googlesource.com/33701
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21333}
TBR=steveanton@webrtc.org,deadbeef@webrtc.org,pthatcher@webrtc.org
Change-Id: Ia72c6d7f185a95b21fd0aec90e7fdc00cb1fb423
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8636
Reviewed-on: https://webrtc-review.googlesource.com/34600
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21335}
The JsepTransport class is moved to pc/ and the utility methods and
enums are moved to where they are used.
With JsepTransport moved to pc/, JsepTransport can depend on objects in
pc/ including RtpTranport, SrtpTransport etc.
Forked from https://webrtc-review.googlesource.com/c/src/+/31762/7
Bug: webrtc:8636
Change-Id: I4e8569fe3012946e87deb280f6139f0fd98de34d
Reviewed-on: https://webrtc-review.googlesource.com/33701
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21333}
Previously, the RTCStatsCollector needed to ask the voice/video
channel for its transport name in order to generate transport
level stats. That would happen on the networking thread which was
unsafe because the voice/video channel could have disappeared in
the duration of the asynchronous thread hop from the signaling
thread to the networking thread. This changes the networking stats
code to check a saved map that tracks the transport name for each
voice/video channel.
Bug: None
Change-Id: I1f03ba8c0526eaa4419f660f18b8b9da62c3f932
Reviewed-on: https://webrtc-review.googlesource.com/33660
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21332}
