Commit Graph

12 Commits

Author SHA1 Message Date
3c16978c84 Remove cast to LocalAudioSource from AudioRtpSender.
We can't assume that the audio source implementation will be our own internal one and we shouldn't apply local audio options to a remote audio track this way either.

BUG=5423
R=solenberg@webrtc.org

Review URL: https://codereview.webrtc.org/1576913002 .

Cr-Commit-Position: refs/heads/master@{#11341}
2016-01-21 15:12:28 +00:00
e1f9d837ae Adding AddTrack/RemoveTrack to native PeerConnection API.
Also, now creating the RtpSender/RtpReceiver proxy objects immediately,
rather than waiting until when GetSenders/GetReceivers is called.

Review URL: https://codereview.webrtc.org/1563403002

Cr-Commit-Position: refs/heads/master@{#11259}
2016-01-14 23:35:46 +00:00
6955870806 Convert channel counts to size_t.
IIRC, this was originally requested by ajm during review of the other size_t conversions I did over the past year, and I agreed it made sense, but wanted to do it separately since those changes were already gargantuan.

BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, henrika@webrtc.org, kjellander@webrtc.org, minyue@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://codereview.webrtc.org/1316523002 .

Cr-Commit-Position: refs/heads/master@{#11229}
2016-01-13 00:26:55 +00:00
6eca7e3c37 Add a 'remote' property to MediaSourceInterface. Also adding an implementation to the relevant sources we have (audio/video) and an extra check where we're casting a source into a local audio source :(
Additionally:
* Moving all implementation inside RemoteAudioTrack into AudioTrack and remove RemoteAudioTrack.
* AddSink/RemoveSink are now on all audio sources (like they are for video sources).

While doing this I found that some of our tests are broken :) and fixed them.  They were broken because AudioTrack didn't previously do much such as updating its state.

BUG=chromium:569526

Review URL: https://codereview.webrtc.org/1522903002

Cr-Commit-Position: refs/heads/master@{#11026}
2015-12-15 12:27:20 +00:00
fac0655fd7 Reland of Adding the ability to create an RtpSender without a track.
(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}
2015-11-25 19:26:08 +00:00
5def7b9fde Revert of Adding the ability to create an RtpSender without a track. (patchset #3 id:300001 of https://codereview.webrtc.org/1413983004/ )
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}
2015-11-20 19:43:27 +00:00
6834fa10f1 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

Review URL: https://codereview.webrtc.org/1413983004

Cr-Commit-Position: refs/heads/master@{#10730}
2015-11-20 17:50:02 +00:00
8f46c63f6f 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

Review URL: https://codereview.webrtc.org/1426443007

Cr-Commit-Position: refs/heads/master@{#10417}
2015-10-26 21:11:25 +00:00
ac9d92ccbe 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.

Review URL: https://codereview.webrtc.org/1413713003

Cr-Commit-Position: refs/heads/master@{#10414}
2015-10-26 18:48:26 +00:00
0c4e06b4c6 Use suffixed {uint,int}{8,16,32,64}_t types.
Removes the use of uint8, etc. in favor of uint8_t.

BUG=webrtc:5024
R=henrik.lundin@webrtc.org, henrikg@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://codereview.webrtc.org/1362503003 .

Cr-Commit-Position: refs/heads/master@{#10196}
2015-10-07 10:23:32 +00:00
70ab1a1ca8 Exposing RtpSenders and RtpReceivers from PeerConnection.
This CL essentially converts [Local|Remote]TrackHandler to
Rtp[Sender|Receiver], and adds a "SetTrack" method for RtpSender.

It also gets rid of MediaStreamHandler and MediaStreamHandlerContainer,
since these classes weren't really anything more than containers.
PeerConnection now manages the RtpSenders and RtpReceivers directly.

Review URL: https://codereview.webrtc.org/1351803002

Cr-Commit-Position: refs/heads/master@{#10100}
2015-09-28 23:54:02 +00:00
6979b024d7 Adding stub files for RtpSender/RtpReceiver.
This will allow Chromium's build files to be updated, so that when the
real RtpSender CL is submitted, it doesn't break the FYI bots.

Review URL: https://codereview.webrtc.org/1364813004

Cr-Commit-Position: refs/heads/master@{#10065}
2015-09-24 23:47:59 +00:00