Testing the nack module by implementing it into the current jitter buffer
under the experiment WebRTC-NewVideoJitterBuffer.
BUG=webrtc:5514
Review URL: https://codereview.webrtc.org/1778503002
Cr-Commit-Position: refs/heads/master@{#11969}
Removes per-extension functions in ViEChannel/ViEReceiver and instead
register extensions directly on the RTP module by mapping extension
string to RTP-header-extension type.
BUG=webrtc:5494
R=danilchap@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1740133002 .
Cr-Commit-Position: refs/heads/master@{#11786}
Adds negotiation of rtx codecs for red and vp9. To keep backwards
compatibility with older Chrome versions, this change includes two
hacks:
1. Red packets will be retransmitted over the rtx codec associated with
vp8 if no rtx codec is associated with red. This is how Chrome does
it today and ensures that we still can send red over rtx to older
versions.
2. If rtx packets associated with the media codec (vp8/vp9 etc) are
received and red has been negotiated, we will assume that the sender
incorrectly has packetized red inside the rtx header associated with
media. We will therefore restore it with the red payload type
instead, which ensures that we can still receive rtx associated with
red from old versions.
Offering multiple rtx codecs to older versions should not be a problem
since old versions themselves only try to negotiate rtx for vp8.
R=pbos@webrtc.orgTBR=mflodman@webrtc.org
BUG=webrtc:4024
TEST=Verified by running apprtc and emulating packet loss between Chrome with and without the patch.
Review URL: https://codereview.webrtc.org/1649493004 .
Cr-Commit-Position: refs/heads/master@{#11472}
Adds logging to RTPSender and RTCPSender, pushing an event log pointer from Channel through ModuleRtpRtcpImpl to the Sender objects.
BUG=webrtc:4741
Review URL: https://codereview.webrtc.org/1571283002
Cr-Commit-Position: refs/heads/master@{#11336}
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=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 -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
