Mostly moved code around in WebRtcVoiceEngine:
- Added new internal class WebRtcVoiceCodecs for static codec functions and the CodecPrefs.
- ConstructCodecs() -> WebRtcVoiceCodecs::SupportedCodecs().
- FindWebRtcCodec -> WebRtcVoiceCodecs::ToCodecInst().
- WebRtcVoiceMediaChannel::SetRecvCodecsInternal() folded into WebRtcVoiceMediaChannel::SetRecvCodecs() (slight logic change).
- Change to how SetRecPayloadType() is implemented in fakewebrtcvoiceengine.h (lines 460-470).
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1461333002
Cr-Commit-Position: refs/heads/master@{#10819}
Insted of using a fixed frame rate, we allow the camera to use a lower frame rate. The camera will choose depending on lightning condition.
TESTED= In a room with low light on N5, N6 N7, Galaxy 4.
BUG=webrtc:5262
R=magjed@webrtc.org
Review URL: https://codereview.webrtc.org/1479563004 .
Cr-Commit-Position: refs/heads/master@{#10807}
This cl make it possible for the hw video encoder to downscale a texture image before encoding. The purpose is to allow downscaling if the quality is too bad at the current resolution.
BUG=webrtc:4993
R=magjed@webrtc.org, pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1470043002 .
Cr-Commit-Position: refs/heads/master@{#10804}
(patchset #8 id:140001 of https://codereview.webrtc.org/1413713003/ )
Relanding after fixing CallAndModifyStream to account for new
procedures for adding/removing a track from a stream.
Original issue's description:
> Adding the ability to create an RtpSender without a track.
>
> This CL also changes AddStream to immediately create a sender, rather
> than waiting until the track is seen in SDP. And the PeerConnection now
> builds the list of "send streams" from the list of senders, rather than
> the collection of local media streams.
>
> Committed: https://crrev.com/ac9d92ccbe2b29590c53f702e11dc625820480d5
> Cr-Commit-Position: refs/heads/master@{#10414}
Review URL: https://codereview.webrtc.org/1468113002
Cr-Commit-Position: refs/heads/master@{#10790}
This is part of the project that makes RTC rendering more
smooth. We've already finished the developement of the
frame selection algorithm in WebMediaPlayerMS, where we
managed a frame pool, and based on the vsync interval, we
actively select the best frame to render in order to
maximize the rendering smoothness.
Thus the frame timeline control in IncomingVideoStream is
no longer needed, because with sophisticated frame
selection algorithm in WebMediaPlayerMS, the time control
in IncomingVideoStream will do nothing but add some extra
delay.
BUG=514873
Review URL: https://codereview.webrtc.org/1419673014
Cr-Commit-Position: refs/heads/master@{#10781}
SurfaceViewRenderer currently stores widthSpec/heightSpec internally, and triggers requestLayout() from renderFrameOnRenderThread()->checkConsistentLayout() when it detects a change using widthSpec/heightSpec. This is not reliable, because onMeasure() might be called several times during the layout process negotiation. For example it might look like this:
-> onMeasure(at most 1920, at most 1080)
<- setMeasuredDimension(1080, 1080)
-> onMeasure(exactly 1080, exactly 1080)
<- setMeasuredDimension(1080, 1080)
Then we store (exactly 1080, exactly 1080) even though we are allowed to be bigger than this, and requestLayout() will never be triggered.
This CL moves the requestLayout() trigger to updateFrameDimensionsAndReportEvents() when the frame size changes.
Other small changes in this CL are:
* Replace with/height variables with Point.
* Add logging in updateFrameDimensionsAndReportEvents() even when rendererEvents is null.
* Use Math.round() in RendererCommon.getDisplaySize() instead of integer cast.
R=hbos@webrtc.org
Review URL: https://codereview.webrtc.org/1453413005 .
Cr-Commit-Position: refs/heads/master@{#10774}
eglCreateSurface() calls are posted to the render thread from both init() and surfaceCreated(). If the render thread does not process the eglCreateSurface() message from init() before surfaceCreated() is called, eglCreateSurface() will be called twice resulting in a crash.
This CL makes sure eglCreateSurface() is only called once.
BUG=b/25815604
R=hbos@webrtc.org
Review URL: https://codereview.webrtc.org/1466133002 .
Cr-Commit-Position: refs/heads/master@{#10769}
This method should be used when the SurfaceTextureHelper is created to use a specific handler.
This now guarantee that the looper used by handler is destroyed after a frame has been returned.
Review URL: https://codereview.webrtc.org/1465163003
Cr-Commit-Position: refs/heads/master@{#10767}
"non-RTP protocols" refers to SCTP data channels. Because
there are no streams for SCTP data channels, the answer was being
set to RECVONLY.
BUG=webrtc:5228
Review URL: https://codereview.webrtc.org/1473013002
Cr-Commit-Position: refs/heads/master@{#10762}
Replace armv7 by arm and arm64 in documentation for iOS build
instructions.
BUG=5125
Review URL: https://codereview.webrtc.org/1418513014
Cr-Commit-Position: refs/heads/master@{#10761}
Reason for revert:
Still breaking CallAndModifyStream. Chromium CL intended to fix it (https://codereview.chromium.org/1435713002/) wasn't sufficient, because I forgot to call addStream/removeStream on the second connection.
Original issue's description:
> Reland of Adding the ability to create an RtpSender without a track. (patchset #1 id:1 of https://codereview.webrtc.org/1426443007/ )
>
> Reason for revert:
> Relanding, after changing the expectations of WebRtcBrowserTest.CallAndModifyStream.
>
> Original issue's description:
> > Revert of Adding the ability to create an RtpSender without a track. (patchset #8 id:140001 of https://codereview.webrtc.org/1413713003/ )
> >
> > Reason for revert:
> > Causing a compiler warning, and causing WebRtcBrowserTest.CallAndModifyStream to fail.
> >
> > Original issue's description:
> > > Adding the ability to create an RtpSender without a track.
> > >
> > > This CL also changes AddStream to immediately create a sender, rather
> > > than waiting until the track is seen in SDP. And the PeerConnection now
> > > builds the list of "send streams" from the list of senders, rather than
> > > the collection of local media streams.
> > >
> > > Committed: https://crrev.com/ac9d92ccbe2b29590c53f702e11dc625820480d5
> > > Cr-Commit-Position: refs/heads/master@{#10414}
> >
> > TBR=pthatcher@webrtc.org,pthatcher@chromium.org
> > NOPRESUBMIT=true
> > NOTREECHECKS=true
> > NOTRY=true
> >
> > Committed: https://crrev.com/8f46c63f6f764254892f4111b54aa1cc8f32eeeb
> > Cr-Commit-Position: refs/heads/master@{#10417}
>
> TBR=pthatcher@webrtc.org,pthatcher@chromium.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
>
> Committed: https://crrev.com/6834fa10f142bf5e2275142acb834898911d09ae
> Cr-Commit-Position: refs/heads/master@{#10730}
TBR=pthatcher@webrtc.org,pthatcher@chromium.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1460323002
Cr-Commit-Position: refs/heads/master@{#10732}
Simplify creation of VoE channels and Call streams in WVoMC.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1454073002
Cr-Commit-Position: refs/heads/master@{#10731}
Reason for revert:
Relanding, after changing the expectations of WebRtcBrowserTest.CallAndModifyStream.
Original issue's description:
> Revert of Adding the ability to create an RtpSender without a track. (patchset #8 id:140001 of https://codereview.webrtc.org/1413713003/ )
>
> Reason for revert:
> Causing a compiler warning, and causing WebRtcBrowserTest.CallAndModifyStream to fail.
>
> Original issue's description:
> > Adding the ability to create an RtpSender without a track.
> >
> > This CL also changes AddStream to immediately create a sender, rather
> > than waiting until the track is seen in SDP. And the PeerConnection now
> > builds the list of "send streams" from the list of senders, rather than
> > the collection of local media streams.
> >
> > Committed: https://crrev.com/ac9d92ccbe2b29590c53f702e11dc625820480d5
> > Cr-Commit-Position: refs/heads/master@{#10414}
>
> TBR=pthatcher@webrtc.org,pthatcher@chromium.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
>
> Committed: https://crrev.com/8f46c63f6f764254892f4111b54aa1cc8f32eeeb
> Cr-Commit-Position: refs/heads/master@{#10417}
TBR=pthatcher@webrtc.org,pthatcher@chromium.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1413983004
Cr-Commit-Position: refs/heads/master@{#10730}
It do the following:
The SurfaceTexture.updateTexImage() calls are moved from the video renderers into MediaCodecVideoDecoder, and the destructor of the texture frames will signal MediaCodecVideoDecoder that the frame has returned. This CL also removes the SurfaceTexture from the native handle and only exposes the texture matrix instead, because only the video source should access the SurfaceTexture.
It moves the responsibility of calculating the decode time to Java.
Patchset2 Refactor MediaCodecVideoDecoder to drop frames if a texture is not released.
R=magjed@webrtc.org
Review URL: https://codereview.webrtc.org/1440343002 .
Cr-Commit-Position: refs/heads/master@{#10706}
Reason for revert:
Broke chromium fyi build.
Original issue's description:
> Convert internal representation of Srtp cryptos from string to int.
>
> Note that the coversion from int to string happens in 3 places
> 1) SDP layer from int to external names. mediasession.cc GetSupportedSuiteNames.
> 2) for SSL_CTX_set_tlsext_use_srtp(), converting from int to internal names.
> 3) stats collection also needs external names.
>
> External names are like AES_CM_128_HMAC_SHA1_80, specified in sslstreamadapter.cc.
> Internal names are like SRTP_AES128_CM_SHA1_80, specified in opensslstreamadapter.cc.
>
> The conversion from string to int happens in one place only, SDP layer, SrtpFilter::ApplyParams().
>
> BUG=webrtc:5043
>
> Committed: https://crrev.com/2764e1027a08a5543e04b854a27a520801faf6eb
> Cr-Commit-Position: refs/heads/master@{#10701}
TBR=juberti@webrtc.org,pthatcher@webrtc.org,juberti@google.com
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5043
Review URL: https://codereview.webrtc.org/1455233005
Cr-Commit-Position: refs/heads/master@{#10702}
Note that the coversion from int to string happens in 3 places
1) SDP layer from int to external names. mediasession.cc GetSupportedSuiteNames.
2) for SSL_CTX_set_tlsext_use_srtp(), converting from int to internal names.
3) stats collection also needs external names.
External names are like AES_CM_128_HMAC_SHA1_80, specified in sslstreamadapter.cc.
Internal names are like SRTP_AES128_CM_SHA1_80, specified in opensslstreamadapter.cc.
The conversion from string to int happens in one place only, SDP layer, SrtpFilter::ApplyParams().
BUG=webrtc:5043
Review URL: https://codereview.webrtc.org/1416673006
Cr-Commit-Position: refs/heads/master@{#10701}
This makes it clearer this code not meant to be used as an API.
I could not find any use of this in downstream code.
BUG=webrtc:5095
TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
R=stefan@webrtc.orgTBR=magjed@webrtc.org
Review URL: https://codereview.webrtc.org/1440873005 .
Cr-Commit-Position: refs/heads/master@{#10699}
Not fully tested yet. Verified in test loopback application
with fake VP9 codec factory.
Assume that encoder generates bitstream in non flexible mode with
one temporal and one spatial layers.
R=magjed@webrtc.org
Review URL: https://codereview.webrtc.org/1451953002 .
Cr-Commit-Position: refs/heads/master@{#10695}
The main purpose was the interface-> include rename, but other files
were also moved, eliminating the "main" dir.
To avoid breaking downstream, the "interface" directories were copied
into a new "video_coding/include" dir. The old headers got pragma
warnings added about deprecation (a very short deprecation since I plan
to remove them as soon downstream is updated).
Other files also moved:
video_coding/main/source -> video_coding
video_coding/main/test -> video_coding/test
BUG=webrtc:5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417283007 .
Cr-Commit-Position: refs/heads/master@{#10694}
This breaks some client apps in annoying ways, so disable for now.
BUG=webrtc:4776
Review URL: https://codereview.webrtc.org/1461513003
Cr-Commit-Position: refs/heads/master@{#10693}
Since BundleFilter doesn't filter RTCP anymore we can have incoming
RTCPs for audio delivered to video, that delivery will fail when there
are no video receivers causing the log to be spammed.
BUG=webrtc:5223
R=henrika@webrtc.org
Review URL: https://codereview.webrtc.org/1458853002 .
Cr-Commit-Position: refs/heads/master@{#10687}
To avoid breaking downstream, the "interface" directories were copied
into a new "common_video/include" dir. The old headers got pragma
warnings added about deprecation (a very short deprecation since I plan
to remove them as soon downstream is updated).
The header guards are also identical to avoid mixing them up in the transition.
BUG=webrtc:5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
Review URL: https://codereview.webrtc.org/1418913006
Cr-Commit-Position: refs/heads/master@{#10659}
The original purpose with keeping one pending frame in SurfaceViewRenderer was to reduce latency for the first rendered frame when we are waiting for the Surface to be created. However, it is very dangerous to hold a pending frame indefinitely when used with a SurfaceTexture, because the SurfaceTexture only has one frame and thus holding a frame in the renderer will freeze everything and typically cause timeout crashes.
Review URL: https://codereview.webrtc.org/1435413006
Cr-Commit-Position: refs/heads/master@{#10638}
Reason for revert:
Causes fallback to SW decoder if a renderer is put in the background.
Original issue's description:
> Patchset 1 is a pure
> revert of "Revert of "Android MediaCodecVideoDecoder: Manage lifetime of texture frames" https://codereview.webrtc.org/1378033003/
>
> Following patchsets move the responsibility of calculating the decode time to Java.
>
> TESTED= Apprtc loopback using H264 and VP8 on N5, N6, N7, S5
>
> Committed: https://crrev.com/9cb8982e64f08d3d630bf7c3d2bcc78c10db88e2
> Cr-Commit-Position: refs/heads/master@{#10597}
TBR=magjed@webrtc.org,glaznev@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
Review URL: https://codereview.webrtc.org/1441363002 .
Cr-Commit-Position: refs/heads/master@{#10637}
Reason for revert:
Unfortunately this breaks an internal downstream project since we have an ancient libsrtp. Reverting until we can figure out how to update our libsrtp.
Original issue's description:
> Remove global list of SRTP sessions.
> Instead save a reference to the SrtpSession inside the srtp_ctx_t.
>
> BUG=webrtc:5133
>
> Committed: https://crrev.com/9cafd972779ed7b25886ab276e0ede7b7a8b76a1
> Cr-Commit-Position: refs/heads/master@{#10591}
TBR=juberti@google.com,juberti@webrtc.org,jbauch@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5133
Review URL: https://codereview.webrtc.org/1442863003
Cr-Commit-Position: refs/heads/master@{#10635}
When this is detected, we'll now "reconfigure" the senders and
receivers, which will reconnect the capturers/renderers to the
underlying streams which have been recreated.
BUG=webrtc:2136
Review URL: https://codereview.webrtc.org/1428243005
Cr-Commit-Position: refs/heads/master@{#10628}
This CL attempts to annotate accesses on >16 API levels using as
small scopes as possible. The TargetApi notations mean "yes, I know
I'm accessing a higher API and I take responsibility for gating the
call on Android API level". The Encoder/Decoder classes are annotated
on the whole class, but they're only accessed through JNI; we should
annotate on method level otherwise and preferably on private methods.
This patch also fixes some compiler-level deprecation warnings (i.e.
-Xlint:deprecation), but probably not all of them.
BUG=webrtc:5063
R=henrika@webrtc.org, kjellander@webrtc.org, magjed@webrtc.org
Review URL: https://codereview.webrtc.org/1412673008 .
Cr-Commit-Position: refs/heads/master@{#10624}
