The H264SpsPpsTracker class:
- Keeps track of all received SPS/PPS.
- Decides whether a packet should be inserted into the PacketBuffer or not.
- Don't insert if this packet only contains SPS and/or PPS.
- Don't insert if this is the first packet of and IDR and we have not
received the required SPS/PPS.
- Insert start codes, and in the case of the first packet of an IDR prepend
the bitstream with the given SPS/PPS for this IDR.
BUG=webrtc:5514
Review-Url: https://codereview.webrtc.org/2466993003
Cr-Commit-Position: refs/heads/master@{#14906}
Reason for revert:
Reverting because of the reasons given in comment #16:
"This change breaks a scenario that is unfortunately not covered by unit tests,
but can easily happen in a real call.
The scenario that is broken by the change is this:
1. A sends an offer to B, with a set of codecs C_a (which is a subset of C_b,
the codecs supported by B)
2. B responds with an answer, and sets the receive codecs to C_a.
3. At a later time, B generates a new offer which by default includes all codecs
in C_b.
4. B calls SetLocalDescription() with this offer, that adds new receive codecs.
5. Adding the new codecs fails, because of the check at
https://cs.chromium.org/chromium/src/third_party/webrtc/voice_engine/channel.....
This causes SetLocalDescription() itself to fail. The net effect is that B
cannot set a local description it just generated.
Before the CL mentioned above, we'd stop playout before changing the codecs, and
the operation would succeed."
Original issue's description:
> Removed the legacy behavior of stopping playout when setting new receive codecs.
>
> BUG=webrtc:4690
>
> Committed: https://crrev.com/917d4e1e7131f35764cff932a8793151585e8179
> Cr-Commit-Position: refs/heads/master@{#14610}
TBR=solenberg@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:4690
Review-Url: https://codereview.webrtc.org/2478433003
Cr-Commit-Position: refs/heads/master@{#14905}
This CL sets the data channel type of the session options before setting the bundle-enabled flag of the session options, so that bundle-enabled will be correctly set and the bundle group will be created.
BUG=webrtc:6218
Review-Url: https://codereview.webrtc.org/2473603002
Cr-Commit-Position: refs/heads/master@{#14904}
This change copies ScreenCapturerDifferWrapper to a new
DesktopCapturerDifferWrapper, and adds DesktopCapturer::CreateWindowCapturer and
DesktopCapturer::CreateScreenCapturer functions to replace
WindowCapturer::Create and ScreenCapturer::Create.
BUG=webrtc:6513
Committed: https://crrev.com/b763e39beba92b45baa09542f949daabbe6258a3
Review-Url: https://codereview.webrtc.org/2468753002
Cr-Original-Commit-Position: refs/heads/master@{#14880}
Cr-Commit-Position: refs/heads/master@{#14903}
Removed EncodedImageCallback::Encoded() and RtpRtcp::SendOutgoingData().
These methods should no longer be used anywhere and it's safe to remove
them.
BUG=chromium:621691
Review-Url: https://codereview.webrtc.org/2405173006
Cr-Commit-Position: refs/heads/master@{#14902}
Suppress WebRtcVideoEncoderFactory overloaded virtual function warning
in WebRtcSimulcastEncoderFactory and FakeWebRtcVideoEncoderFactory.
This warning is triggered by the change in this CL:
https://codereview.webrtc.org/2449993003/.
BUG=webrtc:6402, webrtc:6337
Review-Url: https://codereview.webrtc.org/2468253002
Cr-Commit-Position: refs/heads/master@{#14901}
This is intended to make SequencedTaskChecker work for native dispatch queues
on iOS and macOS. These labels can be compared by their pointers to determine
if a task is running on the same queue.
BUG=webrtc:6643
Review-Url: https://codereview.webrtc.org/2464383002
Cr-Commit-Position: refs/heads/master@{#14900}
Before the removal and copy of script of video file on the android
device was done asynchronously, which was a bug.
BUG=webrtc:6545
NOTRY=True
Review-Url: https://codereview.webrtc.org/2470663004
Cr-Commit-Position: refs/heads/master@{#14898}
Before only C420 as format name was accepted, now C420mpeg2 is also
accepted. Both means the same thing.
BUG=webrtc:6545
NOTRY=True
Review-Url: https://codereview.webrtc.org/2468943002
Cr-Commit-Position: refs/heads/master@{#14897}
In the new APM statistics interface, the default values did not match those previously used in AudioSendStream::Stats.
BUG=webrtc:6525
Review-Url: https://codereview.webrtc.org/2469783002
Cr-Commit-Position: refs/heads/master@{#14896}
This fix is made to remove the discrepancy between GYP and GN audio_decoder_factory_interface target.
BUG=webrtc:6412
NOTRY=True
Review-Url: https://codereview.webrtc.org/2472643003
Cr-Commit-Position: refs/heads/master@{#14894}
This CL replaces the use of webrtc::Video as an internal
variable in the H.264 encoder with the specific fields
that are used by this encoder.
In support of refactorings discussed around:
BUG=600254
Committed: https://crrev.com/2549437b5ccf5aae2e6f1a1491c5f505d1859f9c
Review-Url: https://codereview.webrtc.org/2468903003
Cr-Original-Commit-Position: refs/heads/master@{#14887}
Cr-Commit-Position: refs/heads/master@{#14892}
- Add histogram: "WebRTC.Video.RtpToNtpFreqOffsetInKhz"
The absolute value of the difference between the estimated frequency during RTP timestamp to NTP time conversion and the actual value (i.e. 90 kHz) is measured per received video frame. The max offset during 40 second intervals is stored. The average of these stored offsets per received video stream is recorded when a stream is removed.
Updated rtp_to_ntp.cc:
- Add validation for only inserting newer RTCP sender reports to the rtcp list.
- Move calculation of frequency/offset (from RTP/NTP timestamp pairs) to UpdateRtcpList. Calculated when a new RTCP SR in inserted (and not in RtpToNtpMs per packet).
BUG=webrtc:6579
Review-Url: https://codereview.webrtc.org/2385763002
Cr-Commit-Position: refs/heads/master@{#14891}
Reason for revert:
Landed the wrong patchset. Nothing broken.
Original issue's description:
> Remove webrtc::Video from H264 encoder internals
>
> This CL replaces the use of webrtc::Video as an internal
> variable in the H.264 encoder with the specific fields
> that are used by this encoder.
>
> In support of refactorings discussed around:
>
> BUG=600254
>
> Committed: https://crrev.com/2549437b5ccf5aae2e6f1a1491c5f505d1859f9c
> Cr-Commit-Position: refs/heads/master@{#14887}
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=600254
Review-Url: https://codereview.webrtc.org/2472673002
Cr-Commit-Position: refs/heads/master@{#14888}
This CL replaces the use of webrtc::Video as an internal
variable in the H.264 encoder with the specific fields
that are used by this encoder.
In support of refactorings discussed around:
BUG=600254
Review-Url: https://codereview.webrtc.org/2468903003
Cr-Commit-Position: refs/heads/master@{#14887}
Reason for revert:
Prevents WebRTC rolls into Chrome.
https://build.chromium.org/p/chromium.linux/builders/Blimp%20Linux%20%28dbg%29/builds/14848/steps/compile/logs/stdio
The reason for reverting is: Breaks
https://build.chromium.org/p/chromium.linux/builders/Blimp%20Linux%20%28dbg%2...
[881/894] SOLINK ./libcontent.so
FAILED: libcontent.so libcontent.so.TOC
../../third_party/webrtc/modules/desktop_capture/desktop_capturer.cc:45: error:
undefined reference to
'webrtc::DesktopCapturer::CreateRawWindowCapturer(webrtc::DesktopCaptureOptions
const&)'
../../third_party/webrtc/modules/desktop_capture/desktop_capturer.cc:56: error:
undefined reference to
'webrtc::DesktopCapturer::CreateRawScreenCapturer(webrtc::DesktopCaptureOptions
const&)'
clang: error: linker command failed with exit code 1 (use -v to see invocation)
ninja: build stopped: subcommand failed.
Original issue's description:
> Add CreateWindowCapturer() and CreateScreenCapturer() in DesktopCapturer
>
> This change copies ScreenCapturerDifferWrapper to a new
> DesktopCapturerDifferWrapper, and adds DesktopCapturer::CreateWindowCapturer and
> DesktopCapturer::CreateScreenCapturer functions to replace
> WindowCapturer::Create and ScreenCapturer::Create.
>
> BUG=webrtc:6513
>
> Committed: https://crrev.com/b763e39beba92b45baa09542f949daabbe6258a3
> Cr-Commit-Position: refs/heads/master@{#14880}
TBR=sergeyu@chromium.org,zijiehe@chromium.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6513
Review-Url: https://codereview.webrtc.org/2471773002
Cr-Commit-Position: refs/heads/master@{#14884}
They can be removed and we can use the default system controls.
It's less code and also has more native look.
BUG=webrtc:6617
Review-Url: https://codereview.webrtc.org/2455413002
Cr-Commit-Position: refs/heads/master@{#14882}
To achieve this, several changes needed to be made on both UI and
app logic level.
* Settings view controller is added (modally shown when the settings
button is pressed).
- From there the user can see the current capture resolution
and select another capture resolution.
* Model class for the capture resolution added.
- Improves readability and makes separation of concerns cleaner
- Handles persisting
- Provides defaults
- Maps video resolution setting to RTCMediaConstraints dictionary
* Test for the model class
In future it would be possible to extend this CL and add further settings (i.e
bit rate).
Also it would be easy to remove the hardcoded resolutions and use dynamic values
depending on device capability.
BUG=webrtc:6473
Review-Url: https://codereview.webrtc.org/2462623002
Cr-Commit-Position: refs/heads/master@{#14881}
This change copies ScreenCapturerDifferWrapper to a new
DesktopCapturerDifferWrapper, and adds DesktopCapturer::CreateWindowCapturer and
DesktopCapturer::CreateScreenCapturer functions to replace
WindowCapturer::Create and ScreenCapturer::Create.
BUG=webrtc:6513
Review-Url: https://codereview.webrtc.org/2468753002
Cr-Commit-Position: refs/heads/master@{#14880}
With this change, instead of
RTC_DCHECK_GE(unsigned_var, 17u);
we can simply write
RTC_DCHECK_GE(unsigned_var, 17);
or even
RTC_DCHECK_GE(unsigned_var, -17); // Always true.
and the mathematically sensible thing will happen.
Perhaps more importantly, we can replace checks like
// index is size_t, num_channels is int.
RTC_DCHECK(num_channels >= 0
&& index < static_cast<size_t>(num_channels));
or, even worse, just
// Surely num_channels isn't negative. That would be absurd!
RTC_DCHECK_LT(index, static_cast<size_t>(num_channels));
with simply
RTC_DCHECK_LT(index, num_channels);
In short, you no longer have to keep track of the signedness of the arguments, because the sensible thing will happen.
BUG=webrtc:6645
Review-Url: https://codereview.webrtc.org/2459793002
Cr-Commit-Position: refs/heads/master@{#14878}
This removes the VideoSendStream::LoadObserver interface and the implementation in WebrtcVideoSendStream and replace it with VideoSinkWants through the VideoSourceInterface.
To do that that, some stats for CPU adaptation is moved into VideoSendStream. Also handling of the CVO rtp header extension is moved to VideoSendStreamImpl.
BUG=webrtc:5687
TBR=mflodman@webrtc.org
Review-Url: https://codereview.webrtc.org/2304363002
Cr-Commit-Position: refs/heads/master@{#14877}
In this CL:
- Don't insert a packet if we have explicitly cleared past it.
- Added some logging to ExpandBufferSize.
- Renamed IsContinuous to PotentialNewFrame.
- Unittests updated/added for this new behavior.
- Refactored TestPacketBuffer unittests.
BUG=webrtc:5514
R=danilchap@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2399373002 .
Cr-Commit-Position: refs/heads/master@{#14871}
This is a proposal for a new RTCP message. Feel free to comment on the
message structure, selected type ids etc, as well as code for
serialization/deserialization. Once we agree on this, I'll continue
with wiring it up in the actual rtcp sender and receiver.
BUG=webrtc:6301
Review-Url: https://codereview.webrtc.org/2306873003
Cr-Commit-Position: refs/heads/master@{#14867}
Writable() and the related signal are already part of rtc::PacketTransportInterface. Sense of code symmetry aesthetics dictates that receiving() and the related signal should be declared in the same place.
BUG=webrtc:6531
Review-Url: https://codereview.webrtc.org/2444793003
Cr-Commit-Position: refs/heads/master@{#14865}
Reason for revert:
Dependend cl Breaks Chrome FYI.
peerconnection_unittest anropar RTCStatsReport::Create without parameters.
Original issue's description:
> RTCInboundRTPStreamStats[1] added.
>
> Not all stats are collected in this CL, this must be addressed before
> closing the issue.
>
> [1] https://w3c.github.io/webrtc-stats/#inboundrtpstats-dict*
>
> BUG=chromium:627816, chromium:657855, chromium:657854
>
> Committed: https://crrev.com/0d7bf169402ea9345d163998f4f7df89229ac470
> Cr-Commit-Position: refs/heads/master@{#14861}
TBR=hta@webrtc.org,deadbeef@webrtc.org,hbos@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:627816, chromium:657855, chromium:657854
Review-Url: https://codereview.webrtc.org/2470683002
Cr-Commit-Position: refs/heads/master@{#14862}
Usually .sha1 files are downlaoded using DEPS hooks but since this
bucket is internal we can't run it everywhere since it would fail
non-Googler checkouts. Instead we download the binaries by calling
a Python script, which will be added as a separate build step on the
buildbots.
The .sha1 files are copied from
https://cs.chromium.org/chromium/src/chrome/test/data/webrtc/resources/tools/
leaving out pesq and sox.
BUG=webrtc:6633
TESTED=Ran the download.py script on Mac and verified the files were downloaded.
R=mandermo@google.com, phoglund@webrtc.org
Review URL: https://codereview.webrtc.org/2462023002 .
Cr-Commit-Position: refs/heads/master@{#14859}
- The legacy API is not used in WVoE/MC.
- Removed use of the API (along with StartReceive()) from unit tests.
BUG=webrtc:4690
Review-Url: https://codereview.webrtc.org/2453243003
Cr-Commit-Position: refs/heads/master@{#14858}
