This CL is the step 1 for adding alpha channel support over the wire in webrtc.
- Add the footprint for adapter classes that wraps actual codecs.
- This CL does not add a webrtc::VideoFrame container that can carry alpha to
make the CL shorter for an easier review. Therefore, it exercises a code path
for when we receive no alpha input, just regular I420 frames.
- Unittest sends a video frame for encode/decode through these adapters and
checks the output PSNR.
- See https://webrtc-review.googlesource.com/c/src/+/7800 for the experimental
CL that gives an idea about how it will come together.
Design Doc: https://goo.gl/sFeSUT
Bug: webrtc:7671
Change-Id: I9d3be13647a0a958feceb8d7a9aa93852fc6a1fa
Reviewed-on: https://webrtc-review.googlesource.com/11841
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20490}
This reverts commit 1c9623c70db42550d152c127e01434004087f743.
Reason for revert: Breaks brace initialization:
cricket::VideoDecoderParams params = { "deadbeef" };
I suggest adding an initializer list constructor.
Original change's description:
> Add helper functions for migrating to new video codec factories
>
> This CL adds helper functions in media/engine/convert_legacy_video_factory.h to
> convert from the old WebRtcVideoEncoder and WebRtcVideoDecoder to the new
> webrtc::VideoEncoder and webrtc::VideoDecoder.
>
> The purpose is to make it as easy as possible for clients to migrate to the new
> API and allow us to stop depending on the internal SW codecs as soon as possible.
>
> There still exists an ugly decoder adapter class in the video engine. The reason
> is that we need to continue to pass in the |receive_stream_id| decoder params to
> some legacy clients.
>
> Bug: webrtc:7925
> Change-Id: I43ff03e036411a85d4940fe517a34489f171d698
> Reviewed-on: https://webrtc-review.googlesource.com/15181
> Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20475}
TBR=magjed@webrtc.org,andersc@webrtc.org
Change-Id: I0d1084dc86979fbca748d9ba287d1db3dbe52b44
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7925
Reviewed-on: https://webrtc-review.googlesource.com/17160
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20486}
The method prototype is being changed to returning a const char*,
we don't rely on it in webrtc and the point of the DCHECK that
previously referenced it, was to avoid usage of std::string for
histogram names.
Bug: webrtc:8472
Change-Id: I69b588d4a8f339911a051fd232d63ea5bb1f9a45
Reviewed-on: https://webrtc-review.googlesource.com/16940
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20484}
Enable cpplint check in the PRESUBMIT for pc/ and fix all existing
warnings.
Bug: webrtc:5583
Change-Id: If39994692ab6f6f3c83c74f23850f02fdfe810e8
Reviewed-on: https://webrtc-review.googlesource.com/16540
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20482}
This affects the new injectable codecs.
Bug: webrtc:8459
Change-Id: I484a3ae4c29fd8bae8b13308315758b3689bdd4d
Reviewed-on: https://webrtc-review.googlesource.com/16861
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20478}
This is a reland of 30915a742d86df55ac5c04501c0e8104675a612e
Original change's description:
> Simple Default ObjC video codec factories.
>
> Move the simple video encoder/decoder factory from AppRTCMobile into the
> public API so users who don't have special requirements for video codecs
> can easily get started.
>
> Also clean up the API a little.
>
> This CL replaces the more flexible default factories in
> https://webrtc-review.googlesource.com/c/src/+/7741 and clients that
> want to implement their own codecs will have to supply their own
> encoder/decoder factories as well. The benefits of the approach in
> this CL are a simpler API and less effects on the rest of the code.
>
> Bug: None
> Change-Id: I4ed94090d778b4fc38b49864de1d4de4ff125d6a
> Reviewed-on: https://webrtc-review.googlesource.com/15141
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20441}
Bug: None
Change-Id: If0910cc540dc835dfec4eeb5bea527d88482d110
Reviewed-on: https://webrtc-review.googlesource.com/16780
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20476}
This CL adds helper functions in media/engine/convert_legacy_video_factory.h to
convert from the old WebRtcVideoEncoder and WebRtcVideoDecoder to the new
webrtc::VideoEncoder and webrtc::VideoDecoder.
The purpose is to make it as easy as possible for clients to migrate to the new
API and allow us to stop depending on the internal SW codecs as soon as possible.
There still exists an ugly decoder adapter class in the video engine. The reason
is that we need to continue to pass in the |receive_stream_id| decoder params to
some legacy clients.
Bug: webrtc:7925
Change-Id: I43ff03e036411a85d4940fe517a34489f171d698
Reviewed-on: https://webrtc-review.googlesource.com/15181
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20475}
Adds a unittest to test this.
A Reset() with unsupported frequencies will fail, but currently leaves the resampler in an illegal state.
Subsequent calls to ResetIfNeeded(), which depends on the internal state, will then have unreliable behavior.
The following sequence of calls demonstrate this: It appears as though the resampler is successfully reinitialized to upsample from 44 kHz to 48 kHz, but will in fact crash on Push().
Resampler::Reset() with in=44000, out=32000 // Returns 0
Resampler::ResetIfNeeded() with in=44000, out=48000 // Returns -1
Resampler::ResetIfNeeded() with in=44000, out=48000 // Returns 0
Resampler::Push() with some data
Bug: webrtc:8426
Change-Id: Id1e0528ffcb7a86702d4c2f4c5103a1db419c07d
Reviewed-on: https://webrtc-review.googlesource.com/16424
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20474}
The RTPFragmentationHeader was used when sending audio using RED
for loss protection. This feature has been deprecated and
gradually removed. This cl removes remnants of support from
the RTP send path.
Bug: webrtc:6471
Change-Id: Ia1249047b09c16f79498827f74c2ce07aa38b8f7
Reviewed-on: https://webrtc-review.googlesource.com/16427
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20473}
This CL replaces one 'int32_t' with 'uint32_t'. The value is a
non-negative energy, and the number of leading zeros is
computed. During computation, a shift can cause it to overflow.
Issue was found by the Audio Processing fuzzer.
Bug: chromium:778939, chromium:778921, chromium:778919
Change-Id: I3d7e0b547e6b0edcd9995903517ea851142a08c1
Reviewed-on: https://webrtc-review.googlesource.com/16433
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20470}
The ObjC API (the files in sdk/objc/Framework/Headers/WebRTC/) needs to
be pure ObjC. The changes that are reverted here introduced C++ which
turns it into ObjC++.
We don't have a test protectcing this right now, but it's probably
something we should add to catch changes like this in the future.
TBR=alexnarest@webrtc.org,deadbeef@webrtc.org
Bug: webrtc:8243
Change-Id: Idea688f4014cd44c27cf2cb2a5ec8a9ea7da3c00
Reviewed-on: https://webrtc-review.googlesource.com/16429
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20463}
Second attempt to land https://webrtc-review.googlesource.com/c/src/+/15481.
This time with an extra (dummy) interface to ensure that we don't
break downstream clients.
Improves native Android audio implementations.
Bug: webrtc:8453
Change-Id: I659a3013ae523a2588e4c41ca44b7d0d2d65efb7
Reviewed-on: https://webrtc-review.googlesource.com/16425
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20462}
Always enabling verbose mode means about 100% more text is printed,
but this should not be a problem as the only time that we explicitly
look at the logs is when the bots are failing, or when we want to save
all output for plotting.
BUG=webrtc:8448
Change-Id: Ia5feab5220d047440d15cddb7d3fbca1c5a4aaf5
Reviewed-on: https://webrtc-review.googlesource.com/16140
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20461}
DEPS roll is blocked on it because it pins a broken clang version,
but Chromium has no commitment to fixing it.
Bug: chromium:774973
Change-Id: Id04fadde599293bca7b6c25faa2e9926c1265dc7
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/16423
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20460}
Always use the packetization formely known as kEqualSize.
The RTPFragementation header is ignored, which is no change
in behaviour, since the caller previously always passed null.
Bug: webrtc:6471
Change-Id: Id9e2f985280c2ee8cc33fcf0e5c1fc3ee61c1aff
Reviewed-on: https://webrtc-review.googlesource.com/15222
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20459}
In order to avoid conflicts with downstream projects WebRTC is going
to prefix its LOG macros with RTC_.
This CL renames all the LOG macros to macros with the RTC_ prefix and
it also defines backward compatibility LOG macros in order to let
downstream projects to switch to RTC_ prefixed macros without breaking
them.
A follow-up CL will remove the usage of LOG macros in WebRTC.
Bug: webrtc:8452
Change-Id: Ic3e495cba6c772f65259dc65ee278560a59d02d7
Reviewed-on: https://webrtc-review.googlesource.com/15442
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20456}
CGWindowID is 32-bit, WindowId is 64-bit, using WindowId to receive int value
from CFNumberGetValue() causes the top 32 bits to be random. WindowFinderMac is
impacted by this issue and returns a random number. WindowCapturerMac cannot
match the window_id_ with the the random number.
Meanwhile MouseCursorMonitorMac uses window title to match "Dock" window. See,
https://cs.chromium.org/chromium/src/third_party/webrtc/modules/desktop_capture/mouse_cursor_monitor_mac.mm?rcl=a194e58e799ccab6c999998e5d0f75725aa3f748&l=174
This logic should not be necessary on 10.12 or upper, the name of dock window
is not "Dock" anymore. But to ensure the consistency on old platforms, I have
also added this logic back into GetWindowList() function.
Bug: chromium:778049
Change-Id: Ie827bcd5d31f2ca69ff24c24cf640cb7cc50d419
Reviewed-on: https://webrtc-review.googlesource.com/15782
Commit-Queue: Zijie He <zijiehe@chromium.org>
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Cr-Commit-Position: refs/heads/master@{#20451}
This is done in preparation to make all javac warnings into errors for
WebRTC targets.
Bug: webrtc:6597
Change-Id: I402043157bd75943adf0de52111e5a1bb179c6d1
Reviewed-on: https://webrtc-review.googlesource.com/15104
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20450}
Apparently WebSocketObserver gets garbage collected if it is not stored
by us. This caused some external tests to break.
Bug: None
Change-Id: If62786e84f84a5a63172d67962bb4de8ae3e8479
Reviewed-on: https://webrtc-review.googlesource.com/16100
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20449}
Summary:
Adds AudioTrackStartErrorCode to separate different types of error
codes in combination with StartPlayout.
Harmonizes WebRtcAudioRecord and WebRtcAudioTrack implementations
to ensure that init/start/stop is performed identically.
Adds thread checking in WebRtcAudio track.
Bug: webrtc:8453
Change-Id: Ic913e888ff9493c9cc748a7b4dae43eb6b37fa85
Reviewed-on: https://webrtc-review.googlesource.com/15481
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20448}
This reverts commit 30915a742d86df55ac5c04501c0e8104675a612e.
Reason for revert: Breaks downstream.
Original change's description:
> Simple Default ObjC video codec factories.
>
> Move the simple video encoder/decoder factory from AppRTCMobile into the
> public API so users who don't have special requirements for video codecs
> can easily get started.
>
> Also clean up the API a little.
>
> This CL replaces the more flexible default factories in
> https://webrtc-review.googlesource.com/c/src/+/7741 and clients that
> want to implement their own codecs will have to supply their own
> encoder/decoder factories as well. The benefits of the approach in
> this CL are a simpler API and less effects on the rest of the code.
>
> Bug: None
> Change-Id: I4ed94090d778b4fc38b49864de1d4de4ff125d6a
> Reviewed-on: https://webrtc-review.googlesource.com/15141
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20441}
TBR=magjed@webrtc.org,andersc@webrtc.org,kthelgason@webrtc.org
Change-Id: I3d4395cc9667e6c6cdb33a3b0f5c5fb5bfde9028
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/15182
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20446}
I want to move away from the old encoder factory interface
cricket::WebRtcEncoderFactory to the new webrtc::VideoEncoderFactory. I
created a new webrtc::SdpVideoFormat that essentially is a subset of the
cricket::VideoCodec variables. E.g. the encoder factories shouldn't have
to assign payload types to the codecs, so the payload is not part of
webrtc::SdpVideoFormat. I also didn't add the "feedback_params" that is
used in cricket::VideoCodec to webrtc::SdpVideoFormat. This is causing
problems now, because the internal encoder factory is adding flexfec
feedback params. To avoid this problem, I add these feedback params in
WebRtcVideoEngine instead, like we do for the other codecs.
Bug: webrtc:7925
Change-Id: I7c6ae8d1e1f47f3631c4804c223ec21da8d73685
Reviewed-on: https://webrtc-review.googlesource.com/15223
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20444}
Adds a constructor for DirectTransport that takes a pointer to an instance
of a class derived from FakeNetworkPipe. Said class can override Process()
and SendPacket(...) members thereby emulating any desired network behavior.
Bug: b/67487983
Change-Id: I829fd3506124db61587af19192a14fdf62b06ca5
Reviewed-on: https://webrtc-review.googlesource.com/14620
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20443}
