This CL introduces a way for the VideoReceiveStreams to check whether
they are protected by any FlexfecReceiveStreams. This is done in the
VideoReceiveStream::Start() method, which then propagates this information
down to the jitter buffer adaptation logic.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2649973005
Cr-Commit-Position: refs/heads/master@{#16328}
Reason for revert:
Downstream project relied on changed struct.
Transition made possible by https://codereview.webrtc.org/2655243006/.
Original issue's description:
> Revert of Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. (patchset #7 id:160001 of https://codereview.webrtc.org/2646073004/ )
>
> Reason for revert:
> Breaks internal downstream project.
>
> Original issue's description:
> > Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters.
> >
> > Prior to this CL, received RTX (associated) payload types were only configured
> > when WebRtcVideoChannel2::AddRecvStream was called. In the same method, the RTX
> > SSRC was set up.
> >
> > After this CL, the RTX (associated) payload types are set in
> > WebRtcVideoChannel2::SetRecvParameters, which is the appropriate place to set
> > them. The RTX SSRC is still set in WebRtcVideoChannel2::AddRecvStream, since
> > that is the code path that sets other SSRCs.
> >
> > As part of this fix, the VideoReceiveStream::Config::Rtp struct is changed.
> > We remove the possibility for each video payload type to have an associated
> > specific RTX SSRC. Although the config previously allowed for this, all payload
> > types always had the same RTX SSRC set, and the underlying RtpPayloadRegistry
> > did not support multiple SSRCs. This change to the config struct should thus not
> > have any functional impact. The change does however affect the RtcEventLog, since
> > that is used for storing the VideoReceiveStream::Configs. For simplicity,
> > this CL does not change the event log proto definitions, instead duplicating
> > the serialized RTX SSRCs such that they fit in the existing proto definition.
> >
> > BUG=webrtc:7011
> >
> > Review-Url: https://codereview.webrtc.org/2646073004
> > Cr-Commit-Position: refs/heads/master@{#16302}
> > Committed: fe2bef39cd
>
> TBR=stefan@webrtc.org,magjed@webrtc.org,terelius@webrtc.org,brandtr@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7011
>
> Review-Url: https://codereview.webrtc.org/2649323010
> Cr-Commit-Position: refs/heads/master@{#16307}
> Committed: e4974953ceTBR=stefan@webrtc.org,magjed@webrtc.org,terelius@webrtc.org,kjellander@webrtc.org,kjellander@google.com
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
# NOTREECHECKS=true
# NOTRY=true
BUG=webrtc:7011
Review-Url: https://codereview.webrtc.org/2654163006
Cr-Commit-Position: refs/heads/master@{#16322}
Reason for revert:
Breaks internal downstream project.
Original issue's description:
> Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters.
>
> Prior to this CL, received RTX (associated) payload types were only configured
> when WebRtcVideoChannel2::AddRecvStream was called. In the same method, the RTX
> SSRC was set up.
>
> After this CL, the RTX (associated) payload types are set in
> WebRtcVideoChannel2::SetRecvParameters, which is the appropriate place to set
> them. The RTX SSRC is still set in WebRtcVideoChannel2::AddRecvStream, since
> that is the code path that sets other SSRCs.
>
> As part of this fix, the VideoReceiveStream::Config::Rtp struct is changed.
> We remove the possibility for each video payload type to have an associated
> specific RTX SSRC. Although the config previously allowed for this, all payload
> types always had the same RTX SSRC set, and the underlying RtpPayloadRegistry
> did not support multiple SSRCs. This change to the config struct should thus not
> have any functional impact. The change does however affect the RtcEventLog, since
> that is used for storing the VideoReceiveStream::Configs. For simplicity,
> this CL does not change the event log proto definitions, instead duplicating
> the serialized RTX SSRCs such that they fit in the existing proto definition.
>
> BUG=webrtc:7011
>
> Review-Url: https://codereview.webrtc.org/2646073004
> Cr-Commit-Position: refs/heads/master@{#16302}
> Committed: fe2bef39cdTBR=stefan@webrtc.org,magjed@webrtc.org,terelius@webrtc.org,brandtr@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7011
Review-Url: https://codereview.webrtc.org/2649323010
Cr-Commit-Position: refs/heads/master@{#16307}
Prior to this CL, received RTX (associated) payload types were only configured
when WebRtcVideoChannel2::AddRecvStream was called. In the same method, the RTX
SSRC was set up.
After this CL, the RTX (associated) payload types are set in
WebRtcVideoChannel2::SetRecvParameters, which is the appropriate place to set
them. The RTX SSRC is still set in WebRtcVideoChannel2::AddRecvStream, since
that is the code path that sets other SSRCs.
As part of this fix, the VideoReceiveStream::Config::Rtp struct is changed.
We remove the possibility for each video payload type to have an associated
specific RTX SSRC. Although the config previously allowed for this, all payload
types always had the same RTX SSRC set, and the underlying RtpPayloadRegistry
did not support multiple SSRCs. This change to the config struct should thus not
have any functional impact. The change does however affect the RtcEventLog, since
that is used for storing the VideoReceiveStream::Configs. For simplicity,
this CL does not change the event log proto definitions, instead duplicating
the serialized RTX SSRCs such that they fit in the existing proto definition.
BUG=webrtc:7011
Review-Url: https://codereview.webrtc.org/2646073004
Cr-Commit-Position: refs/heads/master@{#16302}
The original CL was reverted because of a bug discovered by the
chromium bots. Description of that CL:
> Review-Url: https://codereview.webrtc.org/2636443002
> Cr-Commit-Position: refs/heads/master@{#16135}
> Committed: a28e971e3b
The first patch set of this CL is the same as r16135.
Subsequence patch sets are the fixes applied.
Some new test cases have been added, which reveal a few more bugs that
have also been fixed.
BUG=webrtc:4172
Review-Url: https://codereview.webrtc.org/2641133002
Cr-Commit-Position: refs/heads/master@{#16299}
Bulk of the changes were done using
git grep -l '#include "webrtc/base/common.h"' | \
xargs sed -i '\,^#include.*webrtc/base/common\.h,d'
followed by adding back the include in the few places where it is
still needed, and in one case (pseudotcp.cc) instead deleting its use
of RTC_UNUSED.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2644103002
Cr-Commit-Position: refs/heads/master@{#16263}
... As opposed to DtlsTransportInternal.
The code is suboptimal right now, storing two pointers to the different
interfaces. This will all be cleaned up when we have an "RtpTransport"
abstraction that BaseChannel can use.
This CL also cleans up the "fake transport" classes a bit, and gives
them their own header files.
BUG=None
Review-Url: https://codereview.webrtc.org/2648233003
Cr-Commit-Position: refs/heads/master@{#16258}
webrtcvoiceengine.cc ensured that if the bitrate set for ISAC was 0,
it was changed to -1 so that the codec could manage the bitrate
itself.
webrtcsdp.cc ensured that if the bitrate set for ISAC was 0, it was
explicitly set to default values to avoid the codec's built in bitrate
management.
Eventually, there'll be no codec specific code like this in these
layers. This is one step towards that goal.
BUG=webrtc:5806
Review-Url: https://codereview.webrtc.org/2642923003
Cr-Commit-Position: refs/heads/master@{#16220}
Without this, every time WebRtcVideoEngine2 calls supported_codecs(),
the codec list grows.
BUG=webrtc:7020
Review-Url: https://codereview.webrtc.org/2639423006
Cr-Commit-Position: refs/heads/master@{#16178}
This CL adds an RTP module to FlexfecReceiveStreamImpl, and wires it up
to send RTCP RRs. It further makes some methods take const refs instead
of values, to make it more clear where packet copies are made. This
change reduces the number of copies by one, for the case when media
packets are added to the FlexFEC receiver.
The end-to-end test is modified to check for RTCP RRs being sent.
Part of this modification involves some indentation changes, and the
diff thus looks bigger than it logically is.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2625633003
Cr-Commit-Position: refs/heads/master@{#16106}
The existence of FlexfecConfig is due to a naive design. Now when it
is not used on the receiving side (see https://codereview.webrtc.org/2542413002),
it is time to remove it from the sending side as well.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2621573002
Cr-Commit-Position: refs/heads/master@{#16097}
That object will be used when we enable RTCP reporting from FlexfecReceiveStream.
Other related changes:
- Stop using FlexfecConfig (from config.h) at receive side in WebRtcVideoEngine2.
- Add a IsCompleteAndEnabled() method to FlexfecReceiveStream::Config, to be
used in WebRtcVideoEngine2.
- Centralize the construction of the FlexfecReceiveStream::Config in unit tests.
This will make future additions to the unit tests cleaner.
- Simplify setup for receiving FlexFEC in VideoQualityTest.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2589713003
Cr-Commit-Position: refs/heads/master@{#16059}
Earlier, the FlexFEC codec would receive the same default RTCP feedback
params as the media codecs. Since most of these are not used, there is
no point negotiating them.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2623513002
Cr-Commit-Position: refs/heads/master@{#16057}
Bulk of the changes were produced using
git grep -l ' ASSERT(' | grep -v test | grep -v 'common\.h' |\
xargs -n1 sed -i 's/ ASSERT(/ RTC_DCHECK(/'
followed by additional includes of base/checks.h in affected files,
and git cl format.
Also had to do some tweaks to #if !defined(NDEBUG) logic in the
taskrunner code (webrtc/base/task.cc, webrtc/base/taskparent.cc,
webrtc/base/taskparent.h, webrtc/base/taskrunner.cc), replaced to
consistently use RTC_DCHECK_IS_ON, and some of the checks needed
additional #if protection.
Test code was excluded, because it should probably use RTC_CHECK
rather than RTC_DCHECK.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2620303003
Cr-Commit-Position: refs/heads/master@{#16030}
Bulk of changes done using
git grep -l 'RTC_DCHECK(false)' | \
xargs sed -i 's/RTC_DCHECK(false)/RTC_NOTREACHED()/'
peerconnection.cc also used RTC_DCHECK(false && "msg") in two places,
which were updated manually.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2623313004
Cr-Commit-Position: refs/heads/master@{#16026}
Moves webrtc/common_video/rotation.h and parts of
webrtc/common_video/include/video_frame_buffer.h and
webrtc/video_frame.h, and adds to a new GN target api:video_frame_api.
BUG=webrtc:5880
Review-Url: https://codereview.webrtc.org/2517173004
Cr-Commit-Position: refs/heads/master@{#15993}
webrtcvideoengine2.cc uses a field for parameters_, and doesn't empty
out the current state in functions like SetCodec. In the case of
internal_source, SetCodec only set it for external encoders, which
means that in a switch from an internal-source external encoder to an
internal encoder, the internal_source bit would stay set.
(It's plausible that there are other places that are also unsafe and we
just don't notice because codec switches are uncommon in most usage)
In combination with https://codereview.webrtc.org/2574183002/,
generic_encoder.cc now creates 1x1 uninitialized frames as fake frames
for internal_source keyframe requests. The vp8 software encoder doesn't
deal correctly with frames of resolutions that don't match the
configured resolution (besides a DCHECK) and no longer throws these
away (they used to be 0x0 frames), so this results in the VP8
encoder creating a keyframe of the configured send codec size by reading
random memory off the end of the fake I420 frame. This could either
cause crashes or encoding junk data, depending on where the allocation
was.
BUG=webrtc:6957
Review-Url: https://codereview.webrtc.org/2617003003
Cr-Commit-Position: refs/heads/master@{#15969}
With this, RtpSender and RtpReceiver will always return an SSRC if one
is available. Also, attempts to change the SSRC with SetParameters will
fail, rather than silently doing nothing.
BUG=webrtc:6888
Review-Url: https://codereview.webrtc.org/2567333004
Cr-Commit-Position: refs/heads/master@{#15939}
Files only making use of size_t from basictypes.h are replaced with
stddef.h, except in cases where they already for instance use stdio.h or
stdlib.h that already provide size_t.
BUG=webrtc:6853
R=tommi@webrtc.org
Review-Url: https://codereview.webrtc.org/2605123002
Cr-Commit-Position: refs/heads/master@{#15865}
Problem fixed: RTP header extensions were not properly set in tests.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2593963003
Cr-Commit-Position: refs/heads/master@{#15741}
No need to pass a whole struct around, when only one member is used.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2589833002
Cr-Commit-Position: refs/heads/master@{#15687}
Also rename internal::FlexfecReceiveStream to FlexfecReceiveStreamImpl.
BUG=webrtc:6849
Review-Url: https://codereview.webrtc.org/2561123002
Cr-Commit-Position: refs/heads/master@{#15666}
Reason for revert:
Crashes perf tests, e.g.,
./out/Debug/webrtc_perf_tests --gtest_filter='FullStackTest.ScreenshareSlidesVP8_2TL_VeryLossyNet'
dies with an assert related to rtc::Optional.
Original issue's description:
> Delete VideoFrame default constructor, and IsZeroSize method.
>
> This ensures that the video_frame_buffer method never can return a
> null pointer.
>
> BUG=webrtc:6591
>
> Committed: https://crrev.com/bfcf561923a42005e4c7d66d8e72e5932155f997
> Cr-Commit-Position: refs/heads/master@{#15574}
TBR=magjed@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6591
Review-Url: https://codereview.webrtc.org/2574123002
Cr-Commit-Position: refs/heads/master@{#15597}
An external audio mixer will be passed from PeerConnectionFactory to
AudioTransportProxy.
This CL has rewritten based on reverted CL
https://codereview.chromium.org/2539213003/
The only difference is that
static MediaEngineInterface* Create(
webrtc::AudioDeviceModule* adm,
const rtc::scoped_refptr<webrtc::AudioDecoderFactory>&
audio_decoder_factory,
WebRtcVideoEncoderFactory* video_encoder_factory,
WebRtcVideoDecoderFactory* video_decoder_factory);
in media/engine/webrtcmediaengine.h is kept in this CL instead of
replaced for backward compatibility.
BUG=webrtc:6457
Review-Url: https://codereview.webrtc.org/2570993002
Cr-Commit-Position: refs/heads/master@{#15580}
This ensures that the video_frame_buffer method never can return a
null pointer.
BUG=webrtc:6591
Review-Url: https://codereview.webrtc.org/2541863002
Cr-Commit-Position: refs/heads/master@{#15574}
Reason for revert:
A interface change broke some downstream code in google3.
Original issue's description:
> Support external audio mixer in webrtc.
>
> An external audio mixer will be passed from PeerConnectionFactory to
> AudioTransportProxy.
>
> BUG=webrtc:6457
>
> Committed: https://crrev.com/f6bcac59e88c3be5ffd73bbb1098f2d82ee316a1
> Cr-Commit-Position: refs/heads/master@{#15556}
TBR=solenberg@webrtc.org,aleloi@webrtc.org,deadbeef@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6457
Review-Url: https://codereview.webrtc.org/2562333003
Cr-Commit-Position: refs/heads/master@{#15557}
An external audio mixer will be passed from PeerConnectionFactory to
AudioTransportProxy.
BUG=webrtc:6457
Review-Url: https://codereview.webrtc.org/2539213003
Cr-Commit-Position: refs/heads/master@{#15556}
Was added for video initially, but not for audio.
BUG=webrtc:6862
Review-Url: https://codereview.webrtc.org/2568553002
Cr-Commit-Position: refs/heads/master@{#15552}
This also deletes unused features of the video_capturer interface, the classes
VideoCaptureFeedBack, VideoCaptureEncodeInterface and related methods,
and the module id which used to be passed as an argument to the
VideoCaptureDataCallback.
In theory the module id could have been used to let a single
VideoCaptureDataCallback serve several capturers, and demultiplex
on the id, but in practice, it was unused. With this change, it is
required to use a separate VideoSinkInterface for each capturer.
BUG=webrtc:6789
Review-Url: https://codereview.webrtc.org/2534553002
Cr-Commit-Position: refs/heads/master@{#15540}
