The new RTCVideoSource interface can be used by custom implementations of RTCVideoCapturer.
BUG=webrtc:7177
TBR=tommi
Review-Url: https://codereview.webrtc.org/2745193002
Cr-Commit-Position: refs/heads/master@{#17409}
Moves settings model and related classes to code common for both iOS
and Mac.
BUG=webrtc:7177,webrtc:6494
Review-Url: https://codereview.webrtc.org/2770113004
Cr-Commit-Position: refs/heads/master@{#17408}
When screen/window capturers fail to use shared memory, they are
supposed to fall back to XGetImage(). It's slower, but should still
allow to capture desktop content. This wasn't implemented correclty in
XScreenPixelBuffer - it was failing completely when shmget() returns an
error.
BUG=705146
Review-Url: https://codereview.webrtc.org/2775793003
Cr-Commit-Position: refs/heads/master@{#17401}
Decouples encode flags and calculates them the same for both default and
screencast temporal layers.
With this change encoders could start using TemporalReferences for
temporal-layers flags, but they can not be used by asynchronous encoders
(hardware encoders) yet.
Also removes 'timestamp' as a dead parameter to FrameEncoded().
BUG=chromium:702017, webrtc:7349
R=marpan@google.com, sprang@webrtc.org, marpan@webrtc.org
Review-Url: https://codereview.webrtc.org/2769263002 .
Cr-Commit-Position: refs/heads/master@{#17397}
JSEP implementations are required to always generate offers with
"actpass", but remote endpoints are not. So we should accept remote
offers with the current negotiated DTLS role.
This was recently clarified in dtls-sdp; it was somewhat ambiguous
before.
Also doing a bit of refactoring of JsepTransport (making a method
private that should have been private, fixing unit tests that were
directly calling said method).
BUG=webrtc:7072
Review-Url: https://codereview.webrtc.org/2770903003
Cr-Commit-Position: refs/heads/master@{#17396}
Implementation owned by call, and passed to VideoSendStream and
AudioSendStream.
BUG=webrtc:6847, webrtc:7135
Review-Url: https://codereview.webrtc.org/2685673003
Cr-Commit-Position: refs/heads/master@{#17389}
This reverts to previous behavior where b=AS only affects the codec bitrate for audio streams, and not the max bandwidth estimate.
BUG=chromium:703903
Review-Url: https://codereview.webrtc.org/2774123002
Cr-Commit-Position: refs/heads/master@{#17386}
This will later allow calling the "PeerConnectionObserver::OnIceCandidate"
method asynchronously while keeping the object alive.
BUG=webrtc:3721
Review-Url: https://codereview.webrtc.org/2748253003
Cr-Commit-Position: refs/heads/master@{#17380}
The test is failing due to libvpx roll which will be fixed in future roll.
It wasn't discovered at trybots since webrtc_perf_tests doesn't run
on Mac because of webrtc:7322.
BUG=webrtc:7401
TBR=marpan@webrtc.org
Review-Url: https://codereview.webrtc.org/2774973002 .
Cr-Commit-Position: refs/heads/master@{#17377}
TWCC-PLR -based FecController doesn’t need smoothing; instead, use "null-smoothing", which just returns the last value (if any).
If we end up using TWCC-PLR, we'll just remove smoothing altogether. Until then, this is the least intrusive way to modify the code while still letting it work correctly for RTCP-PLR.
BUG=webrtc:7058
Review-Url: https://codereview.webrtc.org/2687433004
Cr-Commit-Position: refs/heads/master@{#17375}
Reason for revert:
Makes perf and Chromium FYI bots unhappy.
Original issue's description:
> WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType
>
> This removes one more place where we were unable to handle codecs not
> in the built-in set.
>
> BUG=webrtc:5805
>
> Review-Url: https://codereview.webrtc.org/2686043006
> Cr-Commit-Position: refs/heads/master@{#17370}
> Committed: 1724cfbdbaTBR=ossu@webrtc.org,solenberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2772043002
Cr-Commit-Position: refs/heads/master@{#17374}
1. Rename FecController to FecControllerPlrBased.
2. Introduce FecControllerRplrBased - a version of FecController that makes its decision based on RPLR instead of PLR.
BUG=webrtc:7058
Review-Url: https://codereview.webrtc.org/2672933003
Cr-Commit-Position: refs/heads/master@{#17373}
Only one test case in it, testing an API which is deprecated.
BUG=webrtc:4690
Review-Url: https://codereview.webrtc.org/2757213002
Cr-Commit-Position: refs/heads/master@{#17372}
Otherwise cpplint will trigger during presubmit for unrelated changes
in these files.
BUG=webrtc:5149
NOTRY=True
Review-Url: https://codereview.webrtc.org/2767393003
Cr-Commit-Position: refs/heads/master@{#17371}
This removes one more place where we were unable to handle codecs not
in the built-in set.
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2686043006
Cr-Commit-Position: refs/heads/master@{#17370}
This is part of a series of CLs. Next CLs:
1. CL for RPLR-based FecController
2. CL for allowing experiment-driven configuration of the above (through both field-trials and protobuf)
BUG=webrtc:7058
Review-Url: https://codereview.webrtc.org/2661043003
Cr-Commit-Position: refs/heads/master@{#17368}
This CL is one in a series. To finish the work, the following CLs will be added:
1. CL for connecting RPLR as well
2. CL for RPLR-based FecController
3. CL for allowing experiment-driven configuration of the above (through both field-trials and protobuf)
BUG=webrtc:7058
Review-Url: https://codereview.webrtc.org/2638083002
Cr-Commit-Position: refs/heads/master@{#17365}
Only use of this method was to implement the cross-device case of
MoveFile on unix systems.
In turn, only use of the MoveFile method is in
FileRotatingStream::RotateFiles. Since file rotation should never
involve any copying of log data, the MoveFile fallback to copying can
be deleted.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2747373003
Cr-Commit-Position: refs/heads/master@{#17364}
The ConversationalSpeechTest.* unit tests are now part of modules_unittests.
The rtc_test target has been replaced with an rtc_source_set one.
The latter is included as dependency in audio_processing_unittests.
BUG=webrtc:7218
Review-Url: https://codereview.webrtc.org/2769863005
Cr-Commit-Position: refs/heads/master@{#17360}
The C++ part of the test uses CallTest to set up an audio-only call. It reads an audio file, plays it through a FakeAudioDevice which transfers data through a FakeNetworkPipe for another FakeAudioDevice to receive it and write it to a file. Information about these files is printed to stdout.
The test cases are meant to try different network and audio configs (more are planned in the future).
The Python part of the test runs the C++ part and scans stdout for tests to perform, runs the pairs of files (original and degraded) through the PESQ tool to receive a score and writes that to perf dashboard.
BUG=webrtc:7229
NOTRY=True
Review-Url: https://codereview.webrtc.org/2694203002
Cr-Commit-Position: refs/heads/master@{#17356}
constexpr function should be preferred than a macro. So this change replaces
FOURCC() macro with a constexpr uint32_t FourCC() function.
BUG=679523, 650926
Review-Url: https://codereview.webrtc.org/2771573002
Cr-Commit-Position: refs/heads/master@{#17351}
Moves towards separating which layers may be referenced instead of
referencing libvpx flags directly. This will make strategies easier to
extract and usable from hardware encoders (RTCVideoEncoder, for
instance).
BUG=chromium:702017, webrtc:7349
R=brandtr@webrtc.org, marpan@webrtc.org. sprang@webrtc.org
Review-Url: https://codereview.webrtc.org/2747123005
Cr-Commit-Position: refs/heads/master@{#17349}
This change adds a DesktopCapturerId namespace, and attaches an int to each
DesktopFrame. ScreenCapturerWinGdi and ScreenCapturerWinDirectx now actively set
this field to differentiate themselves.
BUG=679523, 650926
Review-Url: https://codereview.webrtc.org/2759493002
Cr-Original-Commit-Position: refs/heads/master@{#17329}
Committed: 41e3d9ff3b
Review-Url: https://codereview.webrtc.org/2759493002
Cr-Commit-Position: refs/heads/master@{#17347}
The conversational_speech::Timing class models a list of turns.
Each turn, is identified by a speaker, the audiotrack name, and an offset in milliseconds.
The unit test checks that an instance of Timing is correctly populated and that save/reload leads to the same data.
BUG=webrtc:7218
Review-Url: https://codereview.webrtc.org/2750353002
Cr-Commit-Position: refs/heads/master@{#17346}
