Constants kRTCVp8CodecName, kRTCVp9CodecName, kRTCH264CodecName declared
in RTCRtpCodecParameters.h were defined without the "RTC" part in their
names, thus using them in the app code resulted in linking errors. This
patch fixes the naming mismatch.
BUG=webrtc:7721
Review-Url: https://codereview.webrtc.org/2910863002
Cr-Commit-Position: refs/heads/master@{#18363}
Similar to the existing constants for the media constraint-keys intended
for media sources, add the constants for the standard keys to generate
offers and answers.
This patch also adds a few comments to RTCMediaConstraints.h, to give
a better clue on the intended usage scope of declared media-constraint
keys and values.
BUG=webrtc:7722
Review-Url: https://codereview.webrtc.org/2908013002
Cr-Commit-Position: refs/heads/master@{#18362}
Only will be used if WEBRTC_POSIX and WEBRTC_LINUX are both defined and
"epoll_create" doesn't return an error. Otherwise the default "select"-based
IO loop will be used.
BUG=webrtc:7585
Review-Url: https://codereview.webrtc.org/2880923002
Cr-Commit-Position: refs/heads/master@{#18359}
1. Create unit tests for RtpDemuxer.
2. Add an RTC_DCHECK in RtpDemuxer that makes sure that the sink<->ssrc multimap does not allow multiple instances of the same association.
BUG=None
Review-Url: https://codereview.webrtc.org/2902823004
Cr-Commit-Position: refs/heads/master@{#18357}
Delete unused member |rtp_receiver_|, and simplify a return statement.
BUG=webrtc:5565
Review-Url: https://codereview.webrtc.org/2912363002
Cr-Commit-Position: refs/heads/master@{#18354}
It seems it got lost in a rebase with this CL:
https://codereview.webrtc.org/2893843003/
Bug: None
Change-Id: Iaf4952593c1a1a9490d31c595b05ab155c0a809e
Reviewed-on: https://chromium-review.googlesource.com/519167
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Kári Tristan Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18353}
Use an overloaded version of UpdateNetworkMetrics() which does not require us to explicitly pass in an rtc::Optional.
BUG=None
Review-Url: https://codereview.webrtc.org/2899313004
Cr-Commit-Position: refs/heads/master@{#18347}
The old video send stream configs could contain multiple SSRCs and RTX SSRCs (in case of simulcast). To ensure that the RtcEventLog parser is backwards compatible, we have to return one config for every SSRC.
Also update the parsing functions for the other config types to return the config instead of passing in an output parameter.
BUG=webrtc:7731
Review-Url: https://codereview.webrtc.org/2912113002
Cr-Commit-Position: refs/heads/master@{#18343}
Also deletes a couple of includes of call.h, which seem
unnecessary.
BUG=None
Review-Url: https://codereview.webrtc.org/2907403003
Cr-Commit-Position: refs/heads/master@{#18340}
(This CL concerns both the PLR-based as well as the RPLR-based versions of FecController.)
1. Make FecController disable only when below the disabling-threshold, so as to prevent toggling when the enabling-curve and the disabling-curve are identical.
2. Extend unit-test coverage accordingly.
BUG=None
Review-Url: https://codereview.webrtc.org/2906663002
Cr-Commit-Position: refs/heads/master@{#18337}
Add test for vp8/vp9 qp parser in both videoprocessor_integrationtest.
Check the qp from parser equal to that from the encoder
on every frame in every test.
Add test for vp8/vp9 qp parser in vp8/vp9_impl_test.
Check the qp parser on a single key frame.
BUG=None
Review-Url: https://codereview.webrtc.org/2903163002
Cr-Commit-Position: refs/heads/master@{#18334}
The previous limit leaved no margin for RTT.
BUG=webrtc:4172
Review-Url: https://codereview.webrtc.org/2911243002
Cr-Commit-Position: refs/heads/master@{#18333}
This CL adds a way for external clients to inject their own OpenGL(ES)
shaders to RTCEAGLVideoView/RTCNSGLVideoView. The shader interface
takes textures as arguments, and not RTCVideoFrame, so that
implementations only has to deal with actual OpenGL rendering, and not
converting frames into textures.
This CL also moves the internal shader code around a bit. The current
RTCShader interface with the implementations RTCI420Shader and
RTCNativeNV12Shader are removed. RTCEAGLVideoView and RTCNSGLVideoView
will be responsible for uploading the frames to textures instead
using the helper classes RTCI420TextureCache and RTCNV12TextureCache.
They then call the shader implementation with these textures. The
rendering code that used to be in RTCI420Shader and RTCNativeNV12Shader
have been merged into one RTCDefaultShaderDelegate class.
BUG=webrtc:7473
Review-Url: https://codereview.webrtc.org/2869143002
Cr-Commit-Position: refs/heads/master@{#18326}
magjed is already an owner of the two subfolders webrtc/sdk/android and
webrtc/sdk/objc, but that misses webrtc/sdk/BUILD.gn (as well as
possible future changes to the webrtc/sdk folder).
BUG=None
NOTRY=True
Review-Url: https://codereview.webrtc.org/2910073004
Cr-Commit-Position: refs/heads/master@{#18325}
Biggest change is to Remove MediaType as argument to RtcEventLog::LogRtpHeader and RtcEventLog::LogRtcpHeader.
Since the type is used by tools, these tools are rewritten to figure out the media type from the configurations instead.
BUG=webrtc:7538
TBR=solenberg@webrtc.org // For call.cc and voiceengine.cc
Review-Url: https://codereview.webrtc.org/2855143002
Cr-Commit-Position: refs/heads/master@{#18324}
