Commit Graph

4868 Commits

Author SHA1 Message Date
45a57fda24 Remove unused include from FrameBuffer2.
Bug: None
Change-Id: I766b430beb4f5ba35519931fbff19261a462f2c2
Reviewed-on: https://webrtc-review.googlesource.com/81184
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23517}
2018-06-05 11:33:20 +00:00
3ea3e300dc Fixing some SIGFPEs that are making my tests crash
Bug: none
Change-Id: Ib538e4f131a2c05b9b832bc8235f4f0bb35d04c0
Reviewed-on: https://webrtc-review.googlesource.com/74622
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23515}
2018-06-05 10:03:48 +00:00
27300c3546 Allow 3 encoder threads in libvpx for HD on > 6 core cpus
Bug: webrtc:4172
Change-Id: I50446779403eff0fe2e840afc6cfab9f8a310b1a
Reviewed-on: https://webrtc-review.googlesource.com/77981
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23511}
2018-06-04 16:48:09 +00:00
adb4841173 Remove explicit locking using av_lockmgr_register
av_lockmgr_register is deprecated and no-op since
a04c2c707d

Bug: webrtc:8745
Change-Id: I284c9a6edf88a584c3a5cb5dfae4ccf1be1f8851
Reviewed-on: https://webrtc-review.googlesource.com/39503
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23508}
2018-06-04 12:17:07 +00:00
520ca4e3b8 Delete enum RtpVideoCodecTypes, replaced with VideoCodecType.
Bug: webrtc:8995
Change-Id: I0b44aa26f2f6a81aec7ca1281b8513d8e03228b8
Reviewed-on: https://webrtc-review.googlesource.com/79561
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23507}
2018-06-04 11:53:17 +00:00
b18931d30c Updating ffmpeg deprecated functions TODO.
webrtc:8745 is closed and it was about unblocking the Chromium roll,
webrtc:9352 is the new bug to keep track of the removal of ffmpeg
deprecated functions.

Bug: webrtc:9352
Change-Id: I2818dba804f3d611d4df80559a635e7cf1ee5338
No-Try: True
TBR: phoglund@webrtc.org
Reviewed-on: https://webrtc-review.googlesource.com/80882
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23506}
2018-06-04 11:48:47 +00:00
eed5faefb9 Revert "Disabling VideoCaptureTest on Linux."
This reverts commit 183f4d90bd4f9c3fe5462f78138e657e43954bf5.

Reason for revert: This does not mitigate the bot's flakiness

Original change's description:
> Disabling VideoCaptureTest on Linux.
> 
> Has been really flaky lately, due to NumberOfDevices returning 0.
> 
> TBR=perkj@webrtc.org
> NOTRY=True
> 
> Bug: webrtc:9292
> Change-Id: I5a74236559f13bb6316abced5c12e5d276c398d6
> Reviewed-on: https://webrtc-review.googlesource.com/79680
> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23436}

Bug: webrtc:9292
Change-Id: Id015ec431547f70c335c8e296f8b0a54ff5f4ca1
Reviewed-on: https://webrtc-review.googlesource.com/80381
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23505}
2018-06-04 08:53:11 +00:00
27fe43a1aa Removing warning suppression flags from modules/audio_coding.
Bug: webrtc:9251
Change-Id: I7af3985d337082eea56164357119040383a37074
Reviewed-on: https://webrtc-review.googlesource.com/80483
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23503}
2018-06-04 08:46:01 +00:00
5441398d21 Removing -Wno-write-strings from video_capture_tests.
Bug: webrtc:9251
Change-Id: I6bb182e2ff2676eccdfaca9f608d2134830087f8
Reviewed-on: https://webrtc-review.googlesource.com/80840
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23502}
2018-06-04 08:25:48 +00:00
277a656263 Unstable BWE due to improper bit rate padding for VP9.
Bug: webrtc:9345
Change-Id: I5b1e0b4ed7a8c1d0b942b09433017cac6d53c64b
Reviewed-on: https://webrtc-review.googlesource.com/79000
Commit-Queue: Michael Horowitz <mhoro@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23496}
2018-06-01 20:07:06 +00:00
28deb90728 Reland "Start supporting H264 packetization mode 0."
This is a reland of 3409cfa378e75c0c08d900e0848147929249a62b

Needed to change RtpVideoStreamReceiver to stop deregistering a payload
type if two payload types refer to the same codec (which now happens,
with the packetization mode 0/1 payload types). It's not clear why this
was being done in the first place.

Original change's description:
> Start supporting H264 packetization mode 0.
>
> The work was already done to support it, but it wasn't being negotiated
> in SDP.
>
> This means we'll now see 8 H264 payload types instead of 4; one for each
> combination of BP/CBP profiles, packetization modes 0/1, and RTX/non-RTX.
> This could be problematic in the future, since we're starting to run
> out of dynamic payload types (using 25 of 32).
>
> Bug: chromium:600254
> Change-Id: Ief2340db77c796f12980445b547b87e939170fae
> Reviewed-on: https://webrtc-review.googlesource.com/77264
> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23372}

Bug: chromium:600254
Change-Id: Ice1acc05acd1543d9b46e918de2bba0694d86259
Reviewed-on: https://webrtc-review.googlesource.com/78399
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23494}
2018-06-01 18:03:06 +00:00
e97b5493a5 Fixes leak of AudioDeviceID array due to early return in AudioDeviceMac::GetNumberDevices()
Bug: webrtc:9348
Change-Id: I67a534ec8225180aa67018f7c11f1983262af585
Reviewed-on: https://webrtc-review.googlesource.com/80480
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23490}
2018-06-01 11:53:51 +00:00
c4b7f037b7 AEC3: Adjust active render limits for downsampling factor 8
The signal used for delay estimation at downsampling factor 8 is bandpass
filtered and contains less energy than for other downsampling factors.
This CL adjusts the energy threshold used for determining if there is enough
farend activity to update the matched filters in the delay estimator.
Only downsampling factor 8 is affected.

Bug: webrtc:9288,chromium:846615
Change-Id: I6f38f5609a31e7a08e60571ac75ea75c9962e026
Reviewed-on: https://webrtc-review.googlesource.com/80443
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23486}
2018-06-01 10:07:16 +00:00
f2fae875d5 Add min pushback target bitrate as a parameter that can be set in field trial string.
Bug: None
Change-Id: I9922abadba8164d19e06026fe363efdd337f068e
Reviewed-on: https://webrtc-review.googlesource.com/80122
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23484}
2018-06-01 09:58:36 +00:00
d45b345700 Set max_consec_drop to INT_MAX.
Set recently added max_consec_drop parameter to INT_MAX to keep behavior
of frame dropping logic unchanged.

Bug: none
Change-Id: Ie1d4b428cabc7182ed325c7de4ba8a42cdc826b1
Reviewed-on: https://webrtc-review.googlesource.com/79148
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Marco Paniconi <marpan@google.com>
Cr-Commit-Position: refs/heads/master@{#23482}
2018-06-01 08:30:02 +00:00
7f1583c921 [desktopCapture Windows] ignore Chrome notification window on top
Chrome uses Windows native framework to show the notification of the
ongoing presenting. This notification window is enumerated as a
separated window which is on top most. If this window blocks the target
window, Chrome can't do the cropping and has to switch to GDI methods.
If GDI methods can't capture the target window, then capturing will fail
until the notification is dismissed.

It's hard to identify the notification window in EnumWindows() callback.
So far it works if we ignore window with no title and class name
prefixed with "Chrome_WidgetWin_" and with certain extended styles,
as so does in this CL.

Bug: chromium:847664
Change-Id: Iafabcb1f685adb91bf092475642151e1475cdf61
Reviewed-on: https://webrtc-review.googlesource.com/79742
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Brave Yao <braveyao@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23474}
2018-05-31 17:07:16 +00:00
435187d18d AEC3: CascadedBiQuadFilter can run different filters in cascade
CascadedBiQuadFilter can run identical filters multiple times. This CL
allows the use of different filters in each step. This enables the use
of more elaborate filters. The filters are defined by zeros, poles and
gains.

The 'old' way of initializing CascadedBiQuadFilter with a transfer
function and number of filters is left intact.

Bug: webrtc:9288,chromium:846615
Change-Id: Ie4a5b98eba044415571cdcac087b20870a0b5d33
Reviewed-on: https://webrtc-review.googlesource.com/80060
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23473}
2018-05-31 13:45:15 +00:00
0cedc054a2 Refactor SimulcastTestUtility into SimulcastTestFixture{,Impl}
This will allow exposing the interface to downstream users that
want to test VP8 simulcast. No functional changes to the tests
themselves are expected.

Bug: webrtc:9281
Change-Id: I4128b8f35a4412c5b330cf55c8dc0e173d4570da
Reviewed-on: https://webrtc-review.googlesource.com/77361
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23469}
2018-05-31 11:48:17 +00:00
3d8dbcb686 Adds loss rate filter in BBR controller.
Adds a simple loss rate filter to the BBR network congestion controller.
The loss rate is used to control error correction. Previously the value
was reported as zero which would disable error correction.

Bug: webrtc:8415
Change-Id: Icec8f25fcc9509432ea91eaec30b39a024f92b42
Reviewed-on: https://webrtc-review.googlesource.com/78263
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23467}
2018-05-31 11:10:07 +00:00
750efbe5ce Delete definitions of NULL.
Bug: None
Change-Id: I7cd52ba40c9d1f35a583377c4e729875fbddc068
Reviewed-on: https://webrtc-review.googlesource.com/79941
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23462}
2018-05-31 07:55:15 +00:00
3f1d15b352 Remove deprecated mac capture code.
Bug: webrtc:6898, webrtc:6333, webrtc:7861
Change-Id: Ie33eaa47585012f98b59ccffc0c849c1d9da54da
Reviewed-on: https://webrtc-review.googlesource.com/79920
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23454}
2018-05-30 15:40:01 +00:00
ee20336f6e Drop entire superframe if any layer is overshooting.
Use new frame dropping mode - FULL_SUPERFRAME_DROP - in VP9 encoder and
configure it to drop entire superframe if any layer is overshooting.

Bug: none
Change-Id: Ie22ed5c175e530bcce365d40cba0d10cb608ad4f
Reviewed-on: https://webrtc-review.googlesource.com/79622
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23447}
2018-05-30 11:23:15 +00:00
183f4d90bd Disabling VideoCaptureTest on Linux.
Has been really flaky lately, due to NumberOfDevices returning 0.

TBR=perkj@webrtc.org
NOTRY=True

Bug: webrtc:9292
Change-Id: I5a74236559f13bb6316abced5c12e5d276c398d6
Reviewed-on: https://webrtc-review.googlesource.com/79680
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23436}
2018-05-29 19:16:14 +00:00
79445eadcc Thread checker fails when switching to/from bluetooth headset.
Made some minor changes to resolve the issue. Only affects Debug builds.

NOTRY=TRUE

Bug: webrtc:9310
Change-Id: Ieeeb57d24b559282b2eefd4d8785f7cfe4f44e40
Reviewed-on: https://webrtc-review.googlesource.com/79624
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23434}
2018-05-29 14:50:04 +00:00
7a0bb00422 Split LoggedBweProbeResult into -Success and -Failure.
Also change ParsedEventLog::EventType to enum class.

Bug: webrtc:8111
Change-Id: I4747fb9cbcbdb963fa032770078218e5b416b3da
Reviewed-on: https://webrtc-review.googlesource.com/79280
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23432}
2018-05-29 13:41:04 +00:00
9545e1c9e5 Delete deprecated CreateVideoReceiver and CreateAudioReceiver.
Bug: webrtc:8995
Change-Id: Ic619f3cbc4bd9b5374c00c2e081f2b9811091e12
Reviewed-on: https://webrtc-review.googlesource.com/79400
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23431}
2018-05-29 13:06:54 +00:00
f4d0afbb94 Always apply congestion window in pacer.
A previous change caused a regression in the congestion window behavior.
This CL restores previous behavior.

Bug: webrtc:8415
Change-Id: Id2e42d66bcfb58780c98da2227da39b970f26f0e
Reviewed-on: https://webrtc-review.googlesource.com/79483
Reviewed-by: Ying Wang <yinwa@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23428}
2018-05-29 12:02:04 +00:00
26bc6695cd Pass packet retransmission information in PacketOptions.
bugs.webrtc.org/8439 introduces application data that could e.g. contain
timestamps. We would like to take different actions for this data
depending on whether this is the first time a packet is being sent.

Bug: webrtc:8906
Change-Id: Ib370d76beec2960d961bf44391930faa4b193479
Reviewed-on: https://webrtc-review.googlesource.com/77643
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Petter Strandmark <strandmark@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23426}
2018-05-29 10:12:04 +00:00
2aae2733a7 Remove adapter bools from VideoCodecTestFixture::Config.
It should be the responsibility of the fixture user to provide the exact
codecs that should be tested instead. This reduces the coupling between
the test fixture and the codec instantiation.

Bug: webrtc:9317
Change-Id: I60d8f5c4b516ba33e2293d574ba17602c39f992b
Reviewed-on: https://webrtc-review.googlesource.com/79147
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23425}
2018-05-29 08:02:13 +00:00
e3ca991770 AEC3: Added a mode to properly utilize highly linear setups
Bug: webrtc:9321
Change-Id: I9c1abbd6b1daa1ecff041633318edfb8a011e9c0
Reviewed-on: https://webrtc-review.googlesource.com/79480
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23423}
2018-05-29 07:59:03 +00:00
7c1ccfa881 Move VisualizationParams to VideoCodecTestFixture::Config.
Bug: None
Change-Id: I0a725535c840dda2704dfff33f5e5d3bef3fc0a7
Reviewed-on: https://webrtc-review.googlesource.com/78882
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23422}
2018-05-29 07:18:04 +00:00
7645b18c04 Makes BBR more like the Quic implementation.
Bug: webrtc:8415
Change-Id: I2c21fbe88afec88726cbdd7c6e7626cccb07ce71
Reviewed-on: https://webrtc-review.googlesource.com/77762
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23420}
2018-05-28 17:13:09 +00:00
535bde3752 Adds data in flight information on send packet updates.
This prepares for making the BBR implementation more identical to the
implementation in Quic, this is to ensure that results are comparable.

Bug: webrtc:8415
Change-Id: I7b7e4769772d67cc5112969fefd4e56c6c72432e
Reviewed-on: https://webrtc-review.googlesource.com/76600
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23419}
2018-05-28 15:33:39 +00:00
c5efb0c080 Added an audioproc option to not report the stream delay
Bug: webrtc:9316
Change-Id: If7a20bbac998e9a779579650f3eb9019f974e9a8
Reviewed-on: https://webrtc-review.googlesource.com/79141
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23415}
2018-05-28 13:22:29 +00:00
b563f3db59 Filtering audio playout events with SSRC in NetEq RTP player.
Bug: webrtc:9259
Change-Id: I0b88aa6a7b49bd786637c7ffd9b94c92c608c841
Reviewed-on: https://webrtc-review.googlesource.com/76141
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23414}
2018-05-28 13:16:09 +00:00
92f83cec12 Remove deprecated rtcp SLI/RPSI observers
Bug: webrtc:7338
Change-Id: I39247a3d969637856496b630cadaacac16ef8d09
Reviewed-on: https://webrtc-review.googlesource.com/79260
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23413}
2018-05-28 13:10:54 +00:00
97b4ee5b4c Wire up VAAPI VP8 experimental support in WebRTC.
Experiment flag added to PeerConnectionInterface::RtcConfiguration and
propagated down to VideoStreamEncoder.

Artificial Sdp parameter is added to the sdp format if the flag is set.

Additionally, sdp format is propagated in vp8 simulcast adapters.

Bug: chromium:794608
Change-Id: I2dec54d19ae7bfbd5f2777ec682da5a84194da94
Reviewed-on: https://webrtc-review.googlesource.com/78500
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23412}
2018-05-28 12:30:19 +00:00
66eaed0393 Adding direct congestion window pushback to encoders.
When CongestionWindowPushback experiment is enabled, the pacer is oblivious to the congestion window. The relation between outstanding data and the congestion window affects encoder allocations directly.

Bug: None
Change-Id: Iaacc1d460d44a4ff2d586934c4f9ceb067109337
Reviewed-on: https://webrtc-review.googlesource.com/74922
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23411}
2018-05-28 12:15:59 +00:00
4e6cd5eaeb Get actual list of references from encoder in flexible mode.
In flexible mode, use VP9E_GET_SVC_REF_FRAME_CONFIG to get indices of
reference frame buffers and buffers update by encoded frame.

Set inter_pic_predicted to true only if encoder actually used temporal
prediction.

Bug: webrtc:9244, webrtc:9270
Change-Id: I4e439abeab9e063d50abdcefc59bf58d6596ea6c
Reviewed-on: https://webrtc-review.googlesource.com/74780
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Marco Paniconi <marpan@google.com>
Cr-Commit-Position: refs/heads/master@{#23410}
2018-05-28 11:35:49 +00:00
f782492948 Delete RtpFeedback. The ssrc for a receive stream should be known at
configuration time.

Bug: webrtc:8995
Change-Id: I3d63a76e472a8948c98c98450e96d3301fa2688b
Reviewed-on: https://webrtc-review.googlesource.com/78701
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23409}
2018-05-28 11:05:19 +00:00
0c2e8ce212 Initialize svc_drop_frame in vp9 wrapper.
Thus we don't need to initialize new members added to the structure
in the future.

Bug: None
Change-Id: Id9f5b127c224660f3016973261045b4231a617c1
Reviewed-on: https://webrtc-review.googlesource.com/79080
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23404}
2018-05-28 08:23:19 +00:00
dd09287514 AEC3: Gain limiter: Improving the behavior of the gain limiter.
In this work, we change the behavior of the gain limiter so it also looks at the energy
 on farend around the default delay for deciding the suppression gain
that should be applied at the initial portion of the call.

Bug: webrtc:9311,chromium:846724
Change-Id: I0b777cedbbd7fd689e72070f72237296ce120d3c
Reviewed-on: https://webrtc-review.googlesource.com/78960
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23400}
2018-05-25 15:49:38 +00:00
1388b30661 Adds tracking of outstanding bytes in SendTimeHistory.
This saves having to iterate trough all packets in flight to compute the
number of outstanding bytes.

Bug: webrtc:8415
Change-Id: I35b135f37649a38b44a36d300af42a815f85192d
Reviewed-on: https://webrtc-review.googlesource.com/77727
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23398}
2018-05-25 14:19:48 +00:00
51e23aed9e Remove built-in sw codecs from decoder_database.
All decoders are injectable, no need to create built-in codecs from
there.

Bug: webrtc:7925
Change-Id: Iabf3d59a8e4d721ad29386acbf138b7e5992ce5e
Reviewed-on: https://webrtc-review.googlesource.com/72441
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23397}
2018-05-25 09:54:18 +00:00
78b1c4a487 AEC3: Delay estimator uses bandpass filtered signal with downsampling factor 8
Letting the delay estimator operate at a sampling frequency of 2 kHz
with audio between 0 and 1 kHz makes it sensitive to noisy environments.
This CL bandpass filters the 16 kHz signal before downsampling to 2 kHz
in a way that the downsampled 2 kHz signal contains audio between 1 and
2 kHz. It also sets downsampling factor 8 as default which significantly
reduces computational complexity.

Bug: webrtc:9288,chromium:846615
Change-Id: Iaf67898a1a14326cd61bb7f81c14d3c12a697c8d
Reviewed-on: https://webrtc-review.googlesource.com/78703
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23395}
2018-05-25 09:31:38 +00:00
e058568cc5 iLBC decoding: Ignore a signed overflow
It's always been there, and there's no security risk.

Bug: chromium:843477
Change-Id: I6121943f23b477300cf60ffc4858ef0ab43466dc
Reviewed-on: https://webrtc-review.googlesource.com/78782
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23393}
2018-05-25 08:34:44 +00:00
bc84685497 Remove VideoCodecTestFixtureImpl dependency on Android specifics.
This is needed for downstream users of the impl, as we currently pull
in Chromium specifics in the android_codec_factory_helper. Further,
the downstream users should explicitly supply their own factories
if they do not want to use the internal ones.

Bug: None
Change-Id: Ia7b01a66aadaba3d5accf44e5ca38e1a319e4e34
Reviewed-on: https://webrtc-review.googlesource.com/78420
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23390}
2018-05-24 16:20:11 +00:00
95de63b6fc Rename parsing function in AimdRateControl
Bug: None
Change-Id: I59e54cb4ec87c5d31eb8b14813766f1d1e2a95c4
Reviewed-on: https://webrtc-review.googlesource.com/77240
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23386}
2018-05-24 14:32:11 +00:00
172fd8536e Replaces redundant congestion controller components
This CL replaces components in the congestion controller module
that are identical to equivalent components in the rtp and goog_cc
subfolder. Some redundant components are left as they were not
trivial to replace.

Bug: webrtc:8415
Change-Id: I86a1f164d7b100b8ec8ba7dbc1c9bda2128a4f37
Reviewed-on: https://webrtc-review.googlesource.com/78521
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23384}
2018-05-24 13:35:31 +00:00
d7b9131de4 Move socklen_t definition for windows to win32.h.
Bug: webrtc:6853
Change-Id: Ie73cd959707b32b928acdabd46329830b2bb2c27
Reviewed-on: https://webrtc-review.googlesource.com/78720
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23381}
2018-05-24 11:17:30 +00:00