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gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
01611ed3ac4102dc209e50bd618dd7933e11f314
/
pc
/
rtp_sender.cc
b05ca4b
Implement new specification for degradation preference
by Florent Castelli
· 4 years, 6 months ago
3f1aee3
Change network_priority from a double to an enum.
by Taylor Brandstetter
· 4 years, 6 months ago
e77912b
Insert frame transformer between Encoded and Packetizer.
by Marina Ciocea
· 4 years, 6 months ago
99d6d81
Adding absolute capture timestamp to AudioTrackSinkInterface.
by Minyue Li
· 4 years, 7 months ago
a8c2f51
Remove unused non-standard RtpEncodingParameters members
by Florent Castelli
· 4 years, 9 months ago
cb11a31
Guard GenerateUniqueId() against concurrent access.
by Yves Gerey
· 5 years ago
1ff16c8
Add RtpSenderInterface.SetStreams
by Guido Urdaneta
· 5 years ago
cc18917
Revert "Improve spec compliance of SetStreamIDs in RtpSenderInterface"
by Henrik Andreassson
· 5 years ago
df5731e
Improve spec compliance of SetStreamIDs in RtpSenderInterface
by Guido Urdaneta
· 5 years ago
619b294
RtpSender's RtpParameters were invalidated in a call to SLD/SRD.
by Amit Hilbuch
· 5 years ago
ea7ef2a
Refactoring RtpSenderInternal to share implementation for Audio & Video.
by Amit Hilbuch
· 6 years ago
2297d33
Rejected simulcast layers will no longer appear in GetParameters().
by Amit Hilbuch
· 6 years ago
aa58415
Reland "Enabling Simulcast use via AddTransceiver."
by Amit Hilbuch
· 6 years ago
7832343
Revert "Enabling Simulcast use via AddTransceiver."
by Emircan Uysaler
· 6 years ago
ce470aa
Enabling Simulcast use via AddTransceiver.
by Amit Hilbuch
· 6 years ago
c1a0bcb
Implement the encoding RtpParameter scaleResolutionDownBy
by Florent Castelli
· 6 years ago
10542f2
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
by Steve Anton
· 6 years ago
1c05765
(3) Rename files to snake_case: move the files
by Steve Anton
· 6 years ago
[Renamed from pc/rtpsender.cc]
e1301a8
Revert "Implement read-only codecPayloadType in RtpParameters"
by Henrik Grunell
· 6 years ago
806e06d
Implement read-only codecPayloadType in RtpParameters
by Florent Castelli
· 6 years ago
3e70781
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2.
by Yves Gerey
· 6 years ago
dd9390c
Prevent channels being set on stopped transceiver.
by Amit Hilbuch
· 6 years ago
95ca6e1
AudioSource allows implementations to return settings
by Piotr (Peter) Slatala
· 6 years ago
c462a6e
Prevent the frame decryptor being set if the channel is stopped.
by Benjamin Wright
· 6 years ago
648d28a
Media engine and channel support for per-channel dscp values, specified by RtpParameter
by Tim Haloun
· 6 years ago
6cc9cca
Don't reset streams for the FrameEncryptor / FrameDecryptor unless they changed.
by Benjamin Wright
· 6 years ago
84583f6
Enable End-to-End Encrypted Audio Payloads.
by Benjamin Wright
· 6 years ago
892acf0
Add support for send_encodings parameters in addTransceiver
by Florent Castelli
· 6 years ago
8c1bf95
Reland "Add initial support for RtpEncodingParameters max_framerate."
by Åsa Persson
· 6 years ago
bfd412e
Adds integration of the FrameEncryptor/FrameDecryptor into the MediaChannel.
by Benjamin Wright
· 6 years ago
d81ac95
Injects FrameEncryptorInterface into RtpSender.
by Benjamin Wright
· 6 years ago
948b7e3
Revert "Add initial support for RtpEncodingParameters max_framerate."
by Mirko Bonadei
· 6 years ago
ced5cfd
Add initial support for RtpEncodingParameters max_framerate.
by Åsa Persson
· 6 years ago
87b3c51
Implement changing degradation preference with setParameters()
by Florent Castelli
· 6 years ago
111fdfd
Refactor RtpSender to take the sender ID as a constructor argument
by Steve Anton
· 6 years ago
c19ab07
Add support for content-hint value "text"
by Harald Alvestrand
· 6 years ago
b983bae
Remove unused/deprecated DTMF methods
by Steve Anton
· 6 years ago
5565981
Add functionality to set min/max bitrate per simulcast layer through RtpEncodingParameters.
by Åsa Persson
· 6 years ago
665174f
Reformat the WebRTC code base
by Yves Gerey
· 6 years ago
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
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