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gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
abe301fe6c5008f9a421e5794907685beec52c75
/
pc
/
rtpsender.cc
abe301f
Add HeaderExtensions to RtpParameters
by Florent Castelli
· 6 years ago
2d2c888
Returns RTCError for setting unimplemented RtpParameters.
by Seth Hampson
· 6 years ago
cebf50f
Reland "Implement RtpParameters.transaction_id for PC RtpSenderInterface"
by Florent Castelli
· 6 years ago
909338b
Revert "Implement RtpParameters.transaction_id for PC RtpSenderInterface"
by Max Morin
· 6 years ago
5faf36e
Implement RtpParameters.transaction_id for PC RtpSenderInterface
by Florent Castelli
· 6 years ago
ff40b14
Delete obsolete enable argument to SetVideoSend.
by Niels Möller
· 6 years ago
5b4f075
Reland "Reland "Adds support for multiple or no media stream ids.""
by Seth Hampson
· 6 years ago
191bf5c
Revert "Reland "Adds support for multiple or no media stream ids.""
by Tomas Gunnarsson
· 6 years ago
f351c34
Reland "Adds support for multiple or no media stream ids."
by Seth Hampson
· 6 years ago
bc609ea
Revert "Adds support for multiple or no media stream ids."
by Emircan Uysaler
· 6 years ago
1550292
Adds support for multiple or no media stream ids.
by Seth Hampson
· 6 years ago
3d976f6
Discard link to media channel when audio sender stopped.
by Harald Alvestrand
· 6 years ago
845e878
Name change from stream label to stream id for spec compliance.
by Seth Hampson
· 6 years ago
45cc890
Assorted logging pedantry
by Jonas Olsson
· 7 years ago
ba37b4b
Change return type of RtpSenderInterface::SetParameters from bool to RTCError
by Zach Stein
· 7 years ago
2d8609c
Move internal PeerConnection methods to PeerConnectionInternal
by Steve Anton
· 7 years ago
47136dd
Change RtpSenders to interact with the media channel directly
by Steve Anton
· 7 years ago
c72af93
Reland "Move stats ID generation from SSRC to local ID"
by Harald Alvestrand
· 7 years ago
c0092c3
Revert "Move stats ID generation from SSRC to local ID"
by Erik Språng
· 7 years ago
e357a4d
Move stats ID generation from SSRC to local ID
by Harald Alvestrand
· 7 years ago
02ee47c
Signal track ID correctly when Unified Plan semantics selected
by Steve Anton
· 7 years ago
f9381f0
Implement PeerConnection::AddTrack/RemoveTrack for Unified Plan
by Steve Anton
· 7 years ago
36f8f3e
Optional: Use nullopt and implicit construction in /pc
by Oskar Sundbom
· 7 years ago
675513b
Stop using LOG macros in favor of RTC_ prefixed macros.
by Mirko Bonadei
· 7 years ago
36b29d1
Enable cpplint in pc/
by Steve Anton
· 7 years ago
92ea95e
Fixing WebRTC after moving from src/webrtc to src/
by Mirko Bonadei
· 7 years ago
bb54720
Moving src/webrtc into src/.
by Mirko Bonadei
· 7 years ago
[Renamed from webrtc/pc/rtpsender.cc]
8ffb9c3
Change RtpSender to have multiple stream_ids
by Steve Anton
· 7 years ago
ee89e78
Replace CHECK(x && y) with two separate CHECK() calls
by kwiberg
· 7 years ago
c20978e
Rename webrtc/base -> webrtc/rtc_base
by Edward Lemur
· 7 years ago
a80c16a
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
by Henrik Kjellander
· 7 years ago
c3771cc
Update includes for webrtc/{base => rtc_base} rename (2/3)
by kjellander
· 7 years ago
b11fb25
Protect APM in webkit builds.
by agouaillard
· 8 years ago
20cb0c1
Move DTMF sender to RtpSender (as opposed to WebRtcSession).
by deadbeef
· 8 years ago
7bb87ee
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
by ossu
· 8 years ago
[Renamed (99%) from webrtc/api/rtpsender.cc]
ede5da4
Replace ASSERT by RTC_DCHECK in all non-test code.
by nisse
· 8 years ago
eb4ca4e
Replace RTC_DCHECK(false) with RTC_NOTREACHED().
by nisse
· 8 years ago
5214a0a
Add support for content hints to VideoTrack.
by pbos
· 8 years ago
ba29c6a
Reland 2 of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.
by Taylor Brandstetter
· 8 years ago
3784b4a
Revert of Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. (patchset #3 id:40001 of https://codereview.webrtc.org/2046173002/ )
by tkchin
· 8 years ago
2d54917
Reland of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.
by Taylor Brandstetter
· 8 years ago
1a7162d
Revert of Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. (patchset #3 id:40001 of https://codereview.webrtc.org/2046173002/ )
by deadbeef
· 8 years ago
bc58319
Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.
by Taylor Brandstetter
· 8 years ago
a601f5c
Separating internal and external methods of RtpSender/RtpReceiver.
by deadbeef
· 8 years ago
5a4a75a
Combining SetVideoSend and SetSource into one method.
by deadbeef
· 8 years ago
db0cd9e
Adding getParameters/setParameters APIs to RtpReceiver.
by Taylor Brandstetter
· 8 years ago
5dd42fd
Fixing a segfault that can occur when changing the track of an RtpSender.
by deadbeef
· 8 years ago
dabc944
Add missing tracing to RtpSender objects.
by Peter Boström
· 8 years ago
2ded9b1
Replace SetCapturer and SetCaptureDevice by SetSource. Drop return value.
by nisse
· 8 years ago
c0d31e9
Change VideoSourceInterface::needs_denoising() to return rtc::Optional<bool>
by Per
· 8 years ago
9e083d2
Reland of Delete empty API files and cleaned up includes. (patchset #1 id:1 of https://codereview.webrtc.org/1813083002/ )
by perkj
· 8 years ago
246b527
Revert of Delete empty API files and cleaned up includes. (patchset #2 id:20001 of https://codereview.webrtc.org/1809053002/ )
by deadbeef
· 8 years ago
c9022f5
Delete empty API files and cleaned up includes.
by perkj
· 8 years ago
dc1c62c
Enable setting the maximum bitrate limit in RtpSender.
by skvlad
· 8 years ago
0d3eef2
Add implementation of VideoTrackSource and make VideoCapturerTrackSource inherit from it.
by perkj
· 8 years ago
1a018dc
Prevent a voice channel from sending data before a source is set.
by Taylor Brandstetter
· 8 years ago
a3ede6c
Renamed VideoSourceInterface to VideoTrackSourceInterface.
by perkj
· 8 years ago
b24317b
Fix license headers in webrtc/api.
by kjellander
· 9 years ago
15583c1
Move talk/app/webrtc to webrtc/api
by Henrik Kjellander
· 9 years ago
[Renamed (98%) from talk/app/webrtc/rtpsender.cc]
6a062bd
Deleted method AudioTrackInterface::GetRenderer.
by nisse
· 9 years ago
3c16978
Remove cast to LocalAudioSource from AudioRtpSender.
by Tommi
· 9 years ago
e1f9d83
Adding AddTrack/RemoveTrack to native PeerConnection API.
by deadbeef
· 9 years ago
6955870
Convert channel counts to size_t.
by Peter Kasting
· 9 years ago
6eca7e3
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 :(
by tommi
· 9 years ago
fac0655
Reland of Adding the ability to create an RtpSender without a track.
by deadbeef
· 9 years ago
5def7b9
Revert of Adding the ability to create an RtpSender without a track. (patchset #3 id:300001 of https://codereview.webrtc.org/1413983004/ )
by deadbeef
· 9 years ago
6834fa1
Reland of Adding the ability to create an RtpSender without a track. (patchset #1 id:1 of https://codereview.webrtc.org/1426443007/ )
by deadbeef
· 9 years ago
8f46c63
Revert of Adding the ability to create an RtpSender without a track. (patchset #8 id:140001 of https://codereview.webrtc.org/1413713003/ )
by deadbeef
· 9 years ago
ac9d92c
Adding the ability to create an RtpSender without a track.
by deadbeef
· 9 years ago
0c4e06b
Use suffixed {uint,int}{8,16,32,64}_t types.
by Peter Boström
· 9 years ago
70ab1a1
Exposing RtpSenders and RtpReceivers from PeerConnection.
by deadbeef
· 9 years ago
6979b02
Adding stub files for RtpSender/RtpReceiver.
by deadbeef
· 9 years ago