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webrtc
/
src
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webrtc
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3cd5dff85d13f08aa1109e1cd2f23c01454bb0e3
/
pc
/
rtpsender.cc
0abc754
Change RtpSender to have multiple stream_ids
by Steve Anton
· 7 years ago
ea15d68
Replace CHECK(x && y) with two separate CHECK() calls
by kwiberg
· 7 years ago
76de83e
Rename webrtc/base -> webrtc/rtc_base
by Edward Lemur
· 8 years ago
bc32410
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
by Henrik Kjellander
· 8 years ago
60154fd
Update includes for webrtc/{base => rtc_base} rename (2/3)
by kjellander
· 8 years ago
633cb75
Protect APM in webkit builds.
by agouaillard
· 8 years ago
d4996cd
Move DTMF sender to RtpSender (as opposed to WebRtcSession).
by deadbeef
· 8 years ago
25e4ac7
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
by ossu
· 8 years ago
[Renamed (98%) from api/rtpsender.cc]
a875ff8
Replace ASSERT by RTC_DCHECK in all non-test code.
by nisse
· 8 years ago
1709986
Replace RTC_DCHECK(false) with RTC_NOTREACHED().
by nisse
· 8 years ago
37d7b3f
Add support for content hints to VideoTrack.
by pbos
· 8 years ago
bc1f6bd
Reland 2 of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.
by Taylor Brandstetter
· 9 years ago
00dbf61
Revert of Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. (patchset #3 id:40001 of https://codereview.webrtc.org/2046173002/ )
by tkchin
· 9 years ago
23492b9
Reland of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.
by Taylor Brandstetter
· 9 years ago
a7ba978
Revert of Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. (patchset #3 id:40001 of https://codereview.webrtc.org/2046173002/ )
by deadbeef
· 9 years ago
f29b3f1
Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.
by Taylor Brandstetter
· 9 years ago
0fec214
Separating internal and external methods of RtpSender/RtpReceiver.
by deadbeef
· 9 years ago
4e20ddd
Combining SetVideoSend and SetSource into one method.
by deadbeef
· 9 years ago
80b957b
Adding getParameters/setParameters APIs to RtpReceiver.
by Taylor Brandstetter
· 9 years ago
f6eaeac
Fixing a segfault that can occur when changing the track of an RtpSender.
by deadbeef
· 9 years ago
e38f5d6
Add missing tracing to RtpSender objects.
by Peter Boström
· 9 years ago
cc4f458
Replace SetCapturer and SetCaptureDevice by SetSource. Drop return value.
by nisse
· 9 years ago
3c4cb59
Change VideoSourceInterface::needs_denoising() to return rtc::Optional<bool>
by Per
· 9 years ago
5f44bf3
Reland of Delete empty API files and cleaned up includes. (patchset #1 id:1 of https://codereview.webrtc.org/1813083002/ )
by perkj
· 9 years ago
8396577
Revert of Delete empty API files and cleaned up includes. (patchset #2 id:20001 of https://codereview.webrtc.org/1809053002/ )
by deadbeef
· 9 years ago
2350721
Delete empty API files and cleaned up includes.
by perkj
· 9 years ago
4ac195e
Enable setting the maximum bitrate limit in RtpSender.
by skvlad
· 9 years ago
f4524c4
Add implementation of VideoTrackSource and make VideoCapturerTrackSource inherit from it.
by perkj
· 9 years ago
e38b09a
Prevent a voice channel from sending data before a source is set.
by Taylor Brandstetter
· 9 years ago
2a4ff98
Renamed VideoSourceInterface to VideoTrackSourceInterface.
by perkj
· 9 years ago
95ed4e6
Fix license headers in webrtc/api.
by kjellander
· 9 years ago
bd0ae45
Move talk/app/webrtc to webrtc/api
by Henrik Kjellander
· 9 years ago