1778 Commits

Author SHA1 Message Date
Danil Chapovalov
527ff1eec2 Remove raw extensions accessors from rtp packet
These accessors were introduced in https://codereview.webrtc.org/2789773004
for dynamic size extensions.
They are now implemented without need of these raw functions

Bug: None
Change-Id: Id43f0bcbf951d60ebeece49556b093956c5ad2bf
Reviewed-on: https://webrtc-review.googlesource.com/92626
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24242}
2018-08-09 10:43:37 +00:00
Niels Möller
98d7c52a7c Delete unused constants from rtp_rtcp_config.h
Bug: None
Change-Id: Iced341f0574e26ac3be3292870fb7d7522b75ce1
Reviewed-on: https://webrtc-review.googlesource.com/93280
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24238}
2018-08-09 08:38:51 +00:00
Niels Moller
1788dcb506 Revert "Refactor RtpVideoStreamReceiver without RtpReceiver."
This reverts commit 0b9e01d605a174a52635626c885738a222abff46.

Reason for revert: Appears to breaks AV sync, failing perftests: 
CallPerfTest.PlaysOutAudioAndVideoInSyncWithVideoNtpDrift
CallPerfTest.PlaysOutAudioAndVideoInSyncWithAudioFasterThanVideoDrift
CallPerfTest.PlaysOutAudioAndVideoInSyncWithVideoFasterThanAudioDrift



Original change's description:
> Refactor RtpVideoStreamReceiver without RtpReceiver.
> 
> Bug: webrtc:7135
> Change-Id: Iabf3330e579b892efc160683f9f90efbf6ff9a40
> Reviewed-on: https://webrtc-review.googlesource.com/92398
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24232}

TBR=danilchap@webrtc.org,magjed@webrtc.org,nisse@webrtc.org,sprang@webrtc.org

Change-Id: I70a1dcb519c51937e35e04ac82b2ab495bec0a3d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7135
Reviewed-on: https://webrtc-review.googlesource.com/93260
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24235}
2018-08-09 06:19:14 +00:00
Qingsi Wang
b2db53998d Parse the number of packets lost in RTCP SR as a signed integer.
The cumulative number of packets lost in a RTCP sender report can be
negative if there are duplicates. This CL fixes a bug that the parser of
RTCP reports treats the field as an unsigned integer, and incorrectly
reports large packet losses when a negative loss is reported.

Bug: webrtc:9601
Change-Id: I1109ac0741614d61bda743e13a390b7d3e147a9c
Reviewed-on: https://webrtc-review.googlesource.com/92942
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@google.com>
Cr-Commit-Position: refs/heads/master@{#24234}
2018-08-08 16:44:11 +00:00
Niels Möller
0b9e01d605 Refactor RtpVideoStreamReceiver without RtpReceiver.
Bug: webrtc:7135
Change-Id: Iabf3330e579b892efc160683f9f90efbf6ff9a40
Reviewed-on: https://webrtc-review.googlesource.com/92398
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24232}
2018-08-08 15:21:55 +00:00
Alex Loiko
a837dd790d Reset Agc2 on analog gain changes.
Agc2 applies a digital gain to the nearend signal.
When the analog level changes, the digital gain calculation is no
longer valid. Therefore Agc2 should be notified to analog gain
changes.

This CL also allow audioproc_f to chain AGC1 and AGC2. In a dependent
CL we will allow using AGC1 for analog gain and AGC2 for digital
gain.

Bug: webrtc:7494
Change-Id: Id75b3728fbf2de1d84b7fba005e4670c7a2985d9
Reviewed-on: https://webrtc-review.googlesource.com/89387
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24231}
2018-08-08 14:36:37 +00:00
Sebastian Jansson
436d036b62 Limits reported cumulative packets lost to 0.
This ensures that we don't break clients that can't handle
negative values.

Bug: webrtc:9598
Change-Id: I33c3933982577752eceb738d7e0bd2a6825d2249
Reviewed-on: https://webrtc-review.googlesource.com/93020
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24230}
2018-08-08 13:27:36 +00:00
Niels Möller
3ed46bd83b Delete RTPReceiverStrategy::OnNewPayloadTypeCreated and related code.
Bug: webrtc:7135
Change-Id: Ic20d98cbfb8154f5abbc2501cbcccb950148e732
Reviewed-on: https://webrtc-review.googlesource.com/92396
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24219}
2018-08-08 08:01:32 +00:00
Karl Wiberg
133cff009b AudioCodingModuleTest.TestAllCodecs: Create audio encoders the new way
Specifically, don't expect the ACM to be able to create encoders; we
have to give it an encoder that we make ourselves.

To make it work, I had to add support for the "ptime" parameter to the
L16 codec.

Bug: webrtc:8396
Change-Id: I3869422882611d2eed65d6c849ea7cd3ad6bd126
Reviewed-on: https://webrtc-review.googlesource.com/87423
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24217}
2018-08-08 01:38:05 +00:00
Yves Gerey
435c8de312 Clean up LinkedSet (LRU) code.
* More canonical and efficient 'move to front'.
 * Don't use 'new' when value semantic is fine.
 * Simplify flow (remove One-off private method).
 * Remove dead code.

Bug: webrtc:9575
Change-Id: Ie6a3c4e3d5e2342e77e54fd59fffa05f6e5f9ebe
Reviewed-on: https://webrtc-review.googlesource.com/92802
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24215}
2018-08-07 17:32:07 +00:00
Emircan Uysaler
704a7bd55a Use rtc::saturated_cast instead of static_cast in VCMFecMethod
Bug: webrtc:9439
Change-Id: Ia76a37ab5ae4871c7437b1b4c242556cd33bee40
Reviewed-on: https://webrtc-review.googlesource.com/92701
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24214}
2018-08-07 17:08:42 +00:00
Jiawei Ou
5f7d00eb3d Release audio unit when ios audio device failed to initialize playout and recording.
TBR=henrika@webrtc.org

Bug: webrtc:9552
Change-Id: I7c3e0c1c2126603e7b1cc412cb37cac57eb3cdbf
Reviewed-on: https://webrtc-review.googlesource.com/90085
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24209}
2018-08-07 14:34:12 +00:00
Artem Titov
10d70caa13 Fix guards for headers in third party
Bug: webrtc:8366
Change-Id: I86309265c822dd4430c5578d813bdddc77102d05
Reviewed-on: https://webrtc-review.googlesource.com/90416
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24203}
2018-08-07 09:39:06 +00:00
Yves Gerey
9a29c03355 Fix random crashes - invariant broken in LinkedSet (LRU) implementation.
Root cause: IsNewSequenceNumber didn't respect strict weak ordering requirements.
            (e.g. 0, 0x1000, 0x2000, ... 0x9000 are increasing, but 0x9000 < 0)
Solution: Unwrap the sequence numbers into int64_t for proper sorting.

This CL also introduce a simpler interface,
which does a better job at hiding implementation details.

Bug: webrtc:9575
Change-Id: Ic9922426de32278e8b51c6ecef8e2efeb0997512
Reviewed-on: https://webrtc-review.googlesource.com/91165
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24202}
2018-08-07 09:18:41 +00:00
Sebastian Jansson
eb73a7bd16 Removes unnecessary webrtc_cc namespaces.
Bug: webrtc:9586
Change-Id: I6407ee465d725d7469c409e5bea1c55354ab7f95
Reviewed-on: https://webrtc-review.googlesource.com/92385
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24199}
2018-08-06 17:18:45 +00:00
Sebastian Jansson
13ef7d25f6 Adds feedback only mode to GoogCC.
This CL adds a factory for creating a GoogCC network controller that
can be used without RTCP specific messages. This prepares for enabling
use of other underlying protocols as long as they can provide per
packet feedback.

Bug: None
Change-Id: I6671181949d97abd18843d0f4edf75040cc3f007
Reviewed-on: https://webrtc-review.googlesource.com/84583
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24198}
2018-08-06 15:43:37 +00:00
Sebastian Jansson
f70bc5eeff Removes pause check from RoundRobinPacketQueue.
This CL removes a check in RoundRobinPacketQueue::FinalizePop. This
check will trigger if a the pause state is changed in PacedSender while
a packet is sent. This is a rare occurrence but would yield flaky
behavior. The check should not be required for the code to function
since the paused state is not read in FinalizePop other than for this
check.

Bug: webrtc:9586
Change-Id: Ib9476168eb637dc2f9710d0592bed92c4b03dacb
Reviewed-on: https://webrtc-review.googlesource.com/92090
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24197}
2018-08-06 15:22:41 +00:00
Niels Möller
fd77b78821 Delete RtpReceiverImpl::CheckPayloadChanged.
Also delete related code in RtpReceiverAudio, RtpReceiverVideo and
RtpPayloadRegistry.

Only intended change in behavior is that packets with unknown payload
types are not discarded at this level of the stack. They are discarded
higher up, in Channel::ReceivePacket (audio) and
RtpVideoStreamReceiver::ReceivePacket (video).

Bug: webrtc:8995
Change-Id: I807997120bb40a95b0575c55db6e20a0cac651bf
Reviewed-on: https://webrtc-review.googlesource.com/92087
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24196}
2018-08-06 15:08:12 +00:00
Emircan Uysaler
ff52e88a74 Revert "Extract color space from Vp8 decoder"
This reverts commit fad2aa23b406ca5d85b8aa9ab891f2067e51c782.

Reason for revert: There seems to be a mismatch with Chrome's default for VP8.

Original change's description:
> Extract color space from Vp8 decoder
> 
> Makes use of ColorSpace class to extract info from Vp8 stream.
> 
> Bug: webrtc:9522
> Change-Id: Id9d46eeea5497c4da31db27bfcf2743612ae4157
> Reviewed-on: https://webrtc-review.googlesource.com/90183
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24086}

TBR=sprang@webrtc.org,emircan@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:9522
Change-Id: Ie589963159c9e7ccbc52bf3fdfcbc383656a4ca9
Reviewed-on: https://webrtc-review.googlesource.com/92500
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24191}
2018-08-04 00:26:21 +00:00
philipel
7d745e5a89 Reland "Remove RTPVideoHeader::h264() accessors."
Downstream projects have been updated, so this can now be relanded.
This is a revert (and rebase) of: https://webrtc-review.googlesource.com/c/src/+/88820

Bug: none
Change-Id: I424664ddef7aeebd3c6c94ae67c7f70a342dc9a4
Reviewed-on: https://webrtc-review.googlesource.com/92082
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24181}
2018-08-03 09:16:50 +00:00
Sebastian Jansson
da2ec40590 Always sends probes when they are generated.
This changes makes the usage of the new probe controller reflect how the
old probe controller was used. That is probes are now sent as soon as
they are generated. This is to avoid regressions in performance doe to
the timing of the sent probes.

Bug: chromium:868776
Change-Id: I722585689258c9b01e8f1dc47249b284a05a2793
Reviewed-on: https://webrtc-review.googlesource.com/91441
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24175}
2018-08-02 15:36:12 +00:00
Niels Möller
dc6e68b4a7 Delete class TelephoneEventHandler and related code.
Followup to https://webrtc-review.googlesource.com/91125.

Bug: webrtc:7135
Change-Id: I7011cc65ac756931d8134763da57ec1bc9c584d6
Reviewed-on: https://webrtc-review.googlesource.com/91163
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24174}
2018-08-02 15:02:23 +00:00
Niels Möller
ab4a530b87 Delete telephone-event handling from RTPReceiverAudio.
Bug: webrtc:7135
Change-Id: Ic8b96f44ba25ff9265570dd43d3c76ed0177abfb
Reviewed-on: https://webrtc-review.googlesource.com/91125
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24172}
2018-08-02 12:55:40 +00:00
philipel
f8d81d33ed Add members for the codec agnostic descriptor to RTPVideoHeader.
TBR=danilchap@webrtc.org

Bug: webrtc:9361, webrtc:9582
Change-Id: I0303fc89bafab59e68ec81979e0e4372e79a4f51
Reviewed-on: https://webrtc-review.googlesource.com/91866
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24170}
2018-08-02 09:12:31 +00:00
Joachim Bauch
d3b7ec2e91 Allow all "token" chars from RFC 4566 when checking for legal mid names.
Previously only alphanumeric characters were allowed.

Bug: webrtc:9537
Change-Id: I3fd793ad88520b25ecd884efe3a698f2f0af4639
Reviewed-on: https://webrtc-review.googlesource.com/89388
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24167}
2018-08-01 18:20:42 +00:00
Per Åhgren
78026754a7 AEC3: Utilize shadow filter output to respond to audio path changes
This CL adds functionality to use the shadow filter output instead
of the main filter output for cases when the former is better than
the latter. One case when that happens is when there have been an
echo path change, either in the acoustic path, in the audio buffers
or due to some active audio processing effects being applied on
the device.

The CL causes less echo leaks, in particular on devices with
active render processing.

Bug: webrtc:9581,chromium:869821
Change-Id: Icb8df1b94141598da82dc188051ac59e43338938
Reviewed-on: https://webrtc-review.googlesource.com/91820
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24166}
2018-08-01 15:20:33 +00:00
Oleh Prypin
d2f4e8bd90 Explicitly add -mfpu=neon to all targets that use NEON
Remove obsolete comment about Chromium not defining NEON for Android.

Semi-related fix: don't use `rtc_remove_configs` directly, `suppressed_configs` is the "public interface".

Bug: webrtc:9579
Change-Id: I512628feb462a29432f1356cfef00efe1ddaf84f
Reviewed-on: https://webrtc-review.googlesource.com/91761
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24165}
2018-08-01 13:15:42 +00:00
Alessio Bazzica
55bf92adf4 RNN VAD: more specific build target names.
Bug: webrtc:9076
Change-Id: Ie35ce0f864318a1ddc552285a5535fe411168202
Reviewed-on: https://webrtc-review.googlesource.com/91760
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24162}
2018-08-01 09:07:26 +00:00
Alessio Bazzica
2a99c0bf67 Fix MovingMoments::CalculateMoments.
Protect from negative second moments, which are unexpected in TransientDetector::Detect
and may lead to invalid results.

Bug: chromium:866925
Change-Id: Id1d5b2ebb51e54d9d332b869c6f63dcd03cc461c
Reviewed-on: https://webrtc-review.googlesource.com/91164
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24153}
2018-07-31 15:08:12 +00:00
Sergey Silkin
52233a3f28 Increase RtpFrameReferenceFinder's frame buffer length to 100 frames.
This mitigates the long freeze issue caused by overflow of frame
buffer in RtpFrameReferenceFinder and subsequent removal of old, but
not yet decoded frames, from the buffer.

Bug: webrtc:9550
Change-Id: I03390bb58847688c6cb3f4868bf21269ad07073a
Reviewed-on: https://webrtc-review.googlesource.com/91124
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24152}
2018-07-31 13:41:25 +00:00
Per Åhgren
ef5d5af3a0 AEC3: Increasing the accuracy of the detection for early reverb
This CL introduces an adaptive estimation of the early reverb
in the estimation for the room reverberation. The benefits of
this is that for room with long early reflections there is
a lower risk of underestimating the reverberation.

This CL is for a landing the code in
https://webrtc-review.googlesource.com/c/src/+/87420,
and the review of the code was done in that CL. The author of
code is devicentepena@webrtc.org

Bug: webrtc:9479, chromium:865397
Change-Id: Id6f57e2a684664aef96e8c502e66775f37da59da
Reviewed-on: https://webrtc-review.googlesource.com/91162
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24146}
2018-07-30 22:34:19 +00:00
Niels Möller
1bd66642c3 Set RtpReceiverAudio::telephone_event_forward_to_decoder_ true on construction.
All users call SetTelephoneEventForwardToDecoder(true). Setting the
flag to true on construction, enables deletion of those calls,
followed by deletion of the flag itself.

The unused getter method TelephoneEventForwardToDecoder() is deleted
right away.

Bug: webrtc:7135
Change-Id: I8c52c957b3f074be7ffc425b3588402d1e42b844
Reviewed-on: https://webrtc-review.googlesource.com/90402
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24141}
2018-07-30 12:24:49 +00:00
Sam Zackrisson
0b0f3596bd Remove old temporary webrtc::PostProcessing typedef
Related bug closed since half a year back.

Bug: webrtc:8665
Change-Id: I77007caaa97b5db04f5cf144323cac7a576a7fde
Reviewed-on: https://webrtc-review.googlesource.com/90872
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24135}
2018-07-27 15:43:57 +00:00
Per Åhgren
f954ba5c11 AEC3: Increasing the transparency during call startup
This CL increases the AEC3 transparency during call
startup and after echo path delay changes in 3 ways:
1. The exit requirements for the initial mode is
made less strict.
2. The requirements for using the linear echo model
are made less strict.
3. The duplicated reverb modelling in the linear mode
removed.


Bug: webrtc:9572,chromium:868329
Change-Id: I79ea0796ed26408e35576bb39eaae4e4848b4f83
Reviewed-on: https://webrtc-review.googlesource.com/90868
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24132}
2018-07-27 14:18:42 +00:00
Sam Zackrisson
8b5d2cc93e Add unused AEC toggling config to API
This will be the one way of toggling AEC. The EchoControlMobile and
EchoCancellation interfaces will be removed.

The settings introduced here are not used yet, to allow for smooth
downstream fixes.

Bug: webrtc:9535
Change-Id: I3b1a524a0ab7daf63419d7e5ed47417b9282dbf6
Reviewed-on: https://webrtc-review.googlesource.com/90864
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24129}
2018-07-27 12:57:45 +00:00
Artem Titov
421f868d8a Fix refered LICENSE file path in webrtc license notice
TBR=phoglund

Bug: webrtc:8366
Change-Id: I044cc2153d30b6a88a96b96717ee5680634f6a07
Reviewed-on: https://webrtc-review.googlesource.com/90417
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24122}
2018-07-27 07:50:55 +00:00
Per Åhgren
e4db6a1518 AEC3: Improved the accuracy of the adaptive filter
This CL adds a functionality that jump-starts the
AEC3 shadow filter whenever it performs consistently
worse than the main filter.
The jump-start is done such that the shadow filter
is re-initialized using the main filter coefficients.

The effects of this is a significantly more accurate
main linear filter which leads to less echo leakage
and better transparency

Bug: webrtc:9565, chromium:867873
Change-Id: Ie0b23cd536adc7ce96fc3ed2a7db112aec7437f1
Reviewed-on: https://webrtc-review.googlesource.com/90413
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24117}
2018-07-26 14:51:32 +00:00
Artem Titov
75caa597a3 Untangle fft third party lib from dependon WebRTC
TBR=phoglund

Bug: webrtc:9558
Change-Id: I6cc1936549f008694c3617c1d990524c34da16e3
Reviewed-on: https://webrtc-review.googlesource.com/90411
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24115}
2018-07-26 13:44:30 +00:00
Mirko Bonadei
3e5281f2b3 Enable clang::find_bad_constructs for congestion_controller/goog_cc.
This CL removes //build/config/clang:find_bad_constructs from the
suppressed_configs list, which means that clang:find_bad_constructs
is now enabled on these translation units.

Bug: webrtc:9251, webrtc:163
Change-Id: I289795c92958fd43fed6165894510ad63ca9d24d
Reviewed-on: https://webrtc-review.googlesource.com/90415
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24114}
2018-07-26 13:32:29 +00:00
Niels Möller
a15fd0dee6 Add missing include of stdint.h in MIPS code.
Needed after cl https://webrtc-review.googlesource.com/c/src/+/90249,
which deleted the include of typedefs.h.

Bug: webrtc:6854
Change-Id: I4ab86fae40843613a76da378658343198a800d0c
Reviewed-on: https://webrtc-review.googlesource.com/90414
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24112}
2018-07-26 11:30:19 +00:00
Artem Titov
333a50562c Move fft4g to proper third_party directory
Bug: webrtc:8366
Change-Id: I98d3ae56a1d14b3ecacd85a4b3d234e215c8bc58
Reviewed-on: https://webrtc-review.googlesource.com/85642
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24103}
2018-07-25 15:44:53 +00:00
Per Åhgren
7f5175a455 AEC3: Corrected the filter adjustment during analog gain changes
This CL corrects the way that the echo subtractor output is
adjusted during the adjustment of the adaptive filter when the
analog AGC gain changes.

The CL also ensures that the main adaptive filter is not updated
when this occurs.

Bug: webrtc:9561,chromium:867373
Change-Id: I636f936128f7d9f0d82ca4140b59f148eb35d6a4
Reviewed-on: https://webrtc-review.googlesource.com/90401
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24101}
2018-07-25 15:00:33 +00:00
Niels Möller
a12c42a6b2 Delete root header file typedef.h.
Usage replaced with stdint.h, rtc_base/system/arch.h and
rtc_base/system/unused.h, as appropriate.

Bug: webrtc:6854
Change-Id: I97225465d14b969903d92979e2df3c3c05d35f18
Reviewed-on: https://webrtc-review.googlesource.com/90249
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24100}
2018-07-25 14:59:26 +00:00
Artem Titov
52b9000380 Move g722 to proper third_party directory
Bug: webrtc:8366
Change-Id: I81b051dd25da2d7eaa2902af284d8b669ad8e3c9
Reviewed-on: https://webrtc-review.googlesource.com/85620
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24096}
2018-07-25 11:56:59 +00:00
Artem Titov
e095b81940 Move g711 to proper third_party directory
Bug: webrtc:8366
Change-Id: Ic57bd5c5c01871aee2956b2a098a79b106f54c9e
Reviewed-on: https://webrtc-review.googlesource.com/85375
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24095}
2018-07-25 10:27:08 +00:00
Niels Möller
5c302c0ec5 Delete obsolete file call/video_config.h.
It was moved to api/video_codecs/video_encoder_config.h in cl
https://webrtc-review.googlesource.com/77683.

Bug: webrtc:8830
Change-Id: I197fd3270d3dea0a5ec98b22cc675c407c388e93
Reviewed-on: https://webrtc-review.googlesource.com/90243
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24092}
2018-07-25 09:11:39 +00:00
Artem Titov
8a838fd207 Move fft to proper third_party directory
Bug: webrtc:8366
Change-Id: I741a381fe1cf18909baefd89743b2ff4fe0a6bae
Reviewed-on: https://webrtc-review.googlesource.com/86822
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24091}
2018-07-25 08:39:28 +00:00
Emircan Uysaler
715cc238d2 Refactor VP9 decoder color space code
Bug: webrtc:9522
Change-Id: I4106fd1d1386156e0c5b80f77763643694ead284
Reviewed-on: https://webrtc-review.googlesource.com/90182
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24087}
2018-07-24 18:17:06 +00:00
Emircan Uysaler
fad2aa23b4 Extract color space from Vp8 decoder
Makes use of ColorSpace class to extract info from Vp8 stream.

Bug: webrtc:9522
Change-Id: Id9d46eeea5497c4da31db27bfcf2743612ae4157
Reviewed-on: https://webrtc-review.googlesource.com/90183
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24086}
2018-07-24 17:21:06 +00:00
Emircan Uysaler
d4c16b131f Extract color space from H264 decoder
Makes use of ColorSpace class to extract info from H264 stream.

Bug: webrtc:9522
Change-Id: I651d16707260bb2867b1eda95dd4956d62c47279
Reviewed-on: https://webrtc-review.googlesource.com/90180
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24085}
2018-07-24 17:07:16 +00:00