Use AdaptDown/AdaptUp instead of ScaleDown/ScaleUp, since we may want to
adapt using other means than resolution.
Also, extend vie_encoder with unit test that actually uses frames scaled
to resolution as determined by VideoAdapter, since that seems to be the
default implementation.
BUG=webrtc:4172
Review-Url: https://codereview.webrtc.org/2652893015
Cr-Commit-Position: refs/heads/master@{#16402}
Note:
* PLR is calculated over all of the known packets.
* RPLR is calculated over all of the known packet *pairs*. That is, only over sets of subsequent packets where the reception status is known for both.
BUG=webrtc:7058
Review-Url: https://codereview.webrtc.org/2629883003
Cr-Commit-Position: refs/heads/master@{#16401}
(Or, in less flattering terms, fixing a performance issue introduced
a few months ago by me).
In GN release mode (is_debug = false), the version of the mixer code
before this CL generated code that multiplied each sample (tens of
thousands/second for each input stream) with a floating point number.
This number is almost always exactly 1.0f. The only situation when it's
not 1 is when an audio steam is added or removed.
For one input stream early return leads to a 30% improvement of audio
mixing time profiled on x86-64 under a release build (is_debug = false,
enable_profiling, enable_full_stack_frames_for_profiling) with 16kHz and no
APM limiter. There can be up to 3 streams.
BUG=chromium:687502
Review-Url: https://codereview.webrtc.org/2659423002
Cr-Commit-Position: refs/heads/master@{#16396}
Reason for revert:
due to failures on perf tests (not on perf stats, but fails running due to dcheck failures), see e.g., https://build.chromium.org/p/client.webrtc.perf/builders/Android32%20Tests%20(K%20Nexus5)
Original issue's description:
> Drop frames until specified bitrate is achieved.
>
> This CL fixes a regression introduced with the new quality scaler
> where the video would no longer start in a scaled mode. This CL adds
> code that compares incoming captured frames to the target bitrate,
> and if they are found to be too large, they are dropped and sinkWants
> set to a lower resolution. The number of dropped frames should be low
> (0-4 in most cases) and should not introduce a noticeable delay, or
> at least should be preferrable to having the first 2-4 seconds of video
> have very low quality.
>
> BUG=webrtc:6953
>
> Review-Url: https://codereview.webrtc.org/2630333002
> Cr-Commit-Position: refs/heads/master@{#16391}
> Committed: 83399caec5TBR=perkj@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,kthelgason@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6953
Review-Url: https://codereview.webrtc.org/2666303002
Cr-Commit-Position: refs/heads/master@{#16395}
Delete the calls from RtpStreamReceiver (for video) and
AudioReceiveStream.
BUG=webrtc:6847
Review-Url: https://codereview.webrtc.org/2659563002
Cr-Commit-Position: refs/heads/master@{#16393}
This CL fixes a regression introduced with the new quality scaler
where the video would no longer start in a scaled mode. This CL adds
code that compares incoming captured frames to the target bitrate,
and if they are found to be too large, they are dropped and sinkWants
set to a lower resolution. The number of dropped frames should be low
(0-4 in most cases) and should not introduce a noticeable delay, or
at least should be preferrable to having the first 2-4 seconds of video
have very low quality.
BUG=webrtc:6953
Review-Url: https://codereview.webrtc.org/2630333002
Cr-Commit-Position: refs/heads/master@{#16391}
It appears that thread.cc was the only thing in the webrtc codebase that was
doing this incorrectly (platform_thread.cc, for instance, is ok).
BUG=chromium:687251
Review-Url: https://codereview.webrtc.org/2668693005
Cr-Commit-Position: refs/heads/master@{#16387}
If the frame buffer is cleared while the decoding thread is waiting to acquire
the lock in order to return the |next_frame_it| will be invalidated.
BUG=chromium:679306
Review-Url: https://codereview.webrtc.org/2668743002
Cr-Commit-Position: refs/heads/master@{#16384}
Reason for revert:
Relanding because breakage was not related to this change.
Original issue's description:
> Revert of Remove usage of deprecated g_type_init API (patchset #1 id:1 of https://codereview.webrtc.org/2660823003/ )
>
> Reason for revert:
> Reverting due to internal breakage. Will investigate and re-land
>
> Original issue's description:
> > Remove usage of deprecated g_type_init API
> >
> > BUG=webrtc:7024
> >
> > Review-Url: https://codereview.webrtc.org/2660823003
> > Cr-Commit-Position: refs/heads/master@{#16376}
> > Committed: b2caab7f60
>
> TBR=kjellander@webrtc.org,perkj@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7024
>
> Review-Url: https://codereview.webrtc.org/2666103002
> Cr-Commit-Position: refs/heads/master@{#16379}
> Committed: d1685ab547TBR=kjellander@webrtc.org,perkj@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7024
Review-Url: https://codereview.webrtc.org/2666923002
Cr-Commit-Position: refs/heads/master@{#16380}
Reason for revert:
Reverting due to internal breakage. Will investigate and re-land
Original issue's description:
> Remove usage of deprecated g_type_init API
>
> BUG=webrtc:7024
>
> Review-Url: https://codereview.webrtc.org/2660823003
> Cr-Commit-Position: refs/heads/master@{#16376}
> Committed: b2caab7f60TBR=kjellander@webrtc.org,perkj@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7024
Review-Url: https://codereview.webrtc.org/2666103002
Cr-Commit-Position: refs/heads/master@{#16379}
This is the last CL in webrtc:7030, we have moved all the content
of the folder and we can get rid of it.
BUG=webrtc:7030
TBR=stefan@webrtc.org
NOTRY=True
Review-Url: https://codereview.webrtc.org/2661503004
Cr-Commit-Position: refs/heads/master@{#16369}
We read past the end of the initialized part of the buffer, seemingly
on purpose (no one knows the details of this code anymore). The right
thing to do is probably to zero that part of the buffer.
(The *right* right thing to would be to rewrite this so that it was
easier to see what data was supposed to be where when, but
priorities...)
BUG=chromium:683040
Review-Url: https://codereview.webrtc.org/2659383002
Cr-Commit-Position: refs/heads/master@{#16365}
This allows controllers to do internal logic upon the network metric changes, e.g., filtering.
BUG=webrtc:6303
Review-Url: https://codereview.webrtc.org/2643133003
Cr-Commit-Position: refs/heads/master@{#16363}
The main difference to the old computation is that the expected number of packets during an interval is now computed as the change in highest sequence number encountered, rather than the sequence number difference between the first and last packet in the interval.
BUG=webrtc:7046
Review-Url: https://codereview.webrtc.org/2656333002
Cr-Commit-Position: refs/heads/master@{#16361}
This should help pave the way for injectable audio codecs, since
external implementations need to be able to signal arbitrary fmtp
parameters.
BUG=webrtc:5806
Review-Url: https://codereview.webrtc.org/2661453003
Cr-Commit-Position: refs/heads/master@{#16360}
Also mark the render_time_ms getter function and the ntp timestamp
as deprecated.
BUG=webrtc:6977
Review-Url: https://codereview.webrtc.org/2633493002
Cr-Commit-Position: refs/heads/master@{#16354}
WebRTC's NO_RETURN definition conflicts with WebKit's. This can be avoided by
only defining it if not already defined.
BUG=webrtc:7054
Review-Url: https://codereview.webrtc.org/2657823004
Cr-Commit-Position: refs/heads/master@{#16353}
Since we can now have multiple H264 payload type, we need to move all of them to the beginning of the codec list, instead of greedily taking the first payload type that matches the preferred codec name.
BUG=webrtc:6738
Review-Url: https://codereview.webrtc.org/2658573009
Cr-Commit-Position: refs/heads/master@{#16349}
Video modules are added in reverse order to ensure that the padding order is the same as before, prioritizing high resolution streams.
BUG=webrtc:7043
Review-Url: https://codereview.webrtc.org/2655033002
Cr-Commit-Position: refs/heads/master@{#16329}
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}
This avoids issues where the bitrate produced by the codec is far lower than the target bitrate in the beginning, which causes the delay-based BWE to be initialized accordingly.
BUG=webrtc:5079
Review-Url: https://codereview.webrtc.org/2653883002
Cr-Commit-Position: refs/heads/master@{#16327}
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}
Test disabled on Windows due to failures on Win Msan, Win64 Debug, Win
SyzyAsan, Win32 Debug and others.
TBR=sprang@webrtc.org
BUG=webrtc:6886
NOTRY=True
Review-Url: https://codereview.webrtc.org/2657233002
Cr-Commit-Position: refs/heads/master@{#16320}
This CL adds code to the BlockProcessor, which basically constitutes
the second layer in echo canceller 3. The CL includes two incomplete
classes (EchoRemover and EchoPathDelayEstimator) which will be completed
in upcoming CLs. Because of this, some of the unittests are disabled
until those are added.
BUG=webrtc:6018
Review-Url: https://codereview.webrtc.org/2611223003
Cr-Commit-Position: refs/heads/master@{#16319}
This means that smaller probe packets will be allowed at lower bitrates.
BUG=webrtc:7043
Review-Url: https://codereview.webrtc.org/2650393002
Cr-Commit-Position: refs/heads/master@{#16317}
