Commit Graph

4868 Commits

Author SHA1 Message Date
f5e5d250bc BalancedDegradationSettings: add option to configure a min framerate diff.
If a framerate reduction (input fps - restricted fps) is less than the
configured diff, shorten interval to next qp check.

Bug: none
Change-Id: Ia0b9e0638e5ba75cdc20a1bb45bfcb7d858c5f89
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149040
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28880}
2019-08-16 16:13:46 +00:00
bbeb10925e Reporting audio device underrun counter
Bug: webrtc:10884
Change-Id: I35636fcbc1e2a19a89242379cdff6ec5c12fd21a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149200
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Alex Narest <alexnarest@google.com>
Cr-Commit-Position: refs/heads/master@{#28874}
2019-08-16 11:49:55 +00:00
9b29d69650 Make ANA frame length controller more robust to encoder frame lengths.
Bug: webrtc:10820
Change-Id: Ic3a30976d0181de9cdd35e44d4c5439cadad4812
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149163
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28873}
2019-08-16 10:55:39 +00:00
533c225c93 Revert "Correct conversion between float and fixed formats"
This reverts commit 67e43c8b95057a889ba9946e47d50a265e1e9ac9.

Reason for revert: speculative revert since we see failing bots on Android after this change

https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Android%20Tests%20%28dbg%29%20%28K%20Nexus5%29/4124

Original change's description:
> Correct conversion between float and fixed formats
> 
> This CL changes the way that values are converted
> between fixed and floating point to
> -Avoid the former asymmetric conversion causing
> nonlinear distortions.
> -Reduce the complexity.
> 
> Bug: webrtc:6594
> Change-Id: I64d0cc31c5d16f397686a59a062cfbc4b336d94d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132783
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28867}

TBR=henrik.lundin@webrtc.org,gustaf@webrtc.org,peah@webrtc.org

Change-Id: Id828a09de7075e48556fe2d0beba7a0c6ec227f6
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:6594
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149165
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28872}
2019-08-16 10:40:11 +00:00
67e43c8b95 Correct conversion between float and fixed formats
This CL changes the way that values are converted
between fixed and floating point to
-Avoid the former asymmetric conversion causing
nonlinear distortions.
-Reduce the complexity.

Bug: webrtc:6594
Change-Id: I64d0cc31c5d16f397686a59a062cfbc4b336d94d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132783
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28867}
2019-08-15 13:51:39 +00:00
a1351271e6 Remove all AudioBuffer code that is not related to storing audio data
This CL moves/removes all code from the AudioBuffer that:
-Is not directly handling audio data (e.g., keytaps, VAD descisions).
-Is caching aggregated versions of the rest of the audio data.
-Is not used (or only used in testing)

Bug: webrtc:10882
Change-Id: I737deb3f692748eff30f46ad806b2c6f6292802c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149072
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28866}
2019-08-15 13:49:29 +00:00
3aa0d76cb0 Use struct parser for AlrDetector config.
Bug: webrtc:9883
Change-Id: Ib58fa5ba87607a268f4960898625b1a5adcab69a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148596
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28862}
2019-08-14 18:23:05 +00:00
c759f832e9 Avoid copying of vectors in RtpPacketInfos.
Bug: chromium:982260
Change-Id: Ia4dab497b662e825f80c16530cdf615b62f0a5c9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148523
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28859}
2019-08-14 15:46:02 +00:00
704c8c4446 Re-enable AudioDeviceTest in combination with sanitizers.
Reactivate all tests which aren't flaky anymore.

Bug: webrtc:9751, webrtc:10867
Change-Id: I1d76e0f3e6cc82e78fc46214202f40a9666d41fd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149060
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#28853}
2019-08-14 08:45:18 +00:00
78c56cba00 Delete deprecated version of ReceiveStatistics::Create
Bug: webrtc:10679
Change-Id: I885f38a80c0fe10f1596f33fa95e40a91b23001c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148445
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28852}
2019-08-14 08:40:09 +00:00
608e6ba394 Add AudioDecoderIsacT::Config to include sampling rate and BWInfo object
This CL will make AudioDecoderIsacT symmetrical to AudioEncoderIsacT.

Bug: webrtc:10826
Change-Id: I78d1cf7bc2245bf4a282aabd81c8ece6ca23f285
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146683
Commit-Queue: Jiawei Ou <ouj@fb.com>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28847}
2019-08-14 00:40:19 +00:00
d78196576d Delete StreamDataCountersCallback from ReceiveStatistics
Bug: webrtc:10679
Change-Id: Ife6a4f598c5b70478244b15fc884f6a424d1505b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148521
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28841}
2019-08-13 14:47:08 +00:00
01525f9e03 Delete method StreamStatistician::GetDataCounters
Usage replaced with GetReceiveStreamDataCounters.

Bug: None
Change-Id: Ic5f62ff8a8d33b9eec21657512ba6a0a44635e6d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148801
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28840}
2019-08-13 13:46:45 +00:00
82d75a6214 Use unit types in RoundRobingPacketQueue and PacedSender
This CL replaces various int types with DataRata, DataSize, Timestamp
and TimeDelta classes.

This is part of larger refactoring work where most of PacedSender will
be broken out into a class handling the logic and another responsible
for thread handling. Splitting that up for easier reviewing.

Bug: webrtc:10809
Change-Id: If57a238e5090c47bf3a99c2042783ae584b425f1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148591
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28835}
2019-08-12 17:10:21 +00:00
110a4de4e2 Roll chromium_revision 8f0166a59b..f0fd984a31 (685582:685691)
!! **Manual change** Less strict audio codec tests to accommodate opus fix [1].
!! This is meant to be a temporary mitigation.
[1] https://chromium-review.googlesource.com/c/chromium/src/+/1746617

Change log: 8f0166a59b..f0fd984a31
Full diff: 8f0166a59b..f0fd984a31

Changed dependencies
* src/base: 17d8ac209c..f6cc884505
* src/build: d6837de8f1..956965a6ea
* src/ios: 76e0b0bc60..6780db9c3e
* src/testing: 5d328647a1..48823ed18a
* src/third_party: d70201c684..82063e79f0
* src/third_party/depot_tools: 1b4c7e9f38..6d98232fde
* src/tools: b8953a5bf5..2aa12eadc5
DEPS diff: 8f0166a59b..f0fd984a31/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: I9869cc3f493bc82361e4f93ad846b32390edb340
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148700
Reviewed-by: Yves Gerey <yvesg@webrtc.org>
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28833}
2019-08-12 15:53:01 +00:00
58b496b4d8 Let StreamStatistician::GetReceiveStreamDataCounters return counters by value
Tbr: ossu@webrtc.org # Trivial update of audio/ call site
Bug: None
Change-Id: I3763e83f6c0e18be1b696dd7b2ba5841045c9159
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148820
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28830}
2019-08-12 12:36:00 +00:00
412282acf9 [tsan] Guard audio_device_pulse_linux members from concurrent access.
This CL also fixes data races caused by tests themselves.

TBR= henrika@webrtc.org

Bug: webrtc:9751
Change-Id: Ie7c785b27142fd465f5b4dc9fb0628bd7274f1d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146600
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#28829}
2019-08-12 11:55:52 +00:00
1691e88584 Remove unused fallback method in PacedSender
Bug: webrtc:10809
Change-Id: I30279082c9fa616a686259eb1efc0ebcc1819f61
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148592
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28828}
2019-08-12 10:33:48 +00:00
dc5ed5c023 Delete NACK-related methods from AudioCodingModule
Unused since cl https://webrtc-review.googlesource.com/c/src/+/111504

Bug: None
Change-Id: I210f9c286961a2aec73c7e5c4cf8d04160f5a190
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148076
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28827}
2019-08-12 09:41:10 +00:00
b75d14c802 audioproc_f: input AEC dump as string, output audio to vector
This CL adds the following options:

pass an input AEC dump as a string (currently, the tool can only accept a path to an AEC dump file)
write the processed capture samples to a given vector

Bug: webrtc:10808
Change-Id: I02863c97ec3cd8c03ade2ea8521836f2e7417050
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145208
Commit-Queue: Sonia-Florina Horchidan <soniahorchidan@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28826}
2019-08-12 09:17:36 +00:00
81df62b456 Add field trial to introduce extra delay after target level calculation.
Bug: webrtc:10817
Change-Id: Id9eced821df2859b2cb7174062b6f5e29e145f62
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145902
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28825}
2019-08-12 08:55:23 +00:00
1544915bb4 Avoid capturing extraneous windows in CroppingWindowCapturerWin
This change reduces cases where capturing a window with the cropping
capturer captures unrelated windows from the same process. For instance:
- Capturing an Explorer window could include portions of taskbar UI,
  e.g. when an auto-hide taskbar or window preview thumbnails are shown
  overtop.
- Capturing a window from a process with multiple windows could include
  menus/tooltips from another window.

Instead of capturing any window with an empty/matching title created by
the same process, the cropping capturer will capture any window created
by the same thread. While not foolproof, this heuristic seems to capture
menus/tooltips from the window of interest while excluding those from
other top-level windows in practice (assuming those were created by a
separate thread / independent message pump).

Bug: webrtc:10856
Change-Id: I2072c79da9e0158475b442a43b5b96d6ad307bc2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148641
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Jamie Walch <jamiewalch@chromium.org>
Cr-Commit-Position: refs/heads/master@{#28824}
2019-08-10 03:40:17 +00:00
b90d38a978 Delete unused Opus-specific methods of AudioCodingModule
Bug: None
Change-Id: Ib191e4beadf85cd57e765bc52d305e274e50a473
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148400
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28815}
2019-08-09 07:06:36 +00:00
5297cf368d Delete unused class MockTargetTransferRateObserver
Bug: None
Change-Id: I60e9dc05450207dfd572ae17a42cf1adaed4c1b6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148525
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28813}
2019-08-09 06:15:06 +00:00
940c2b5005 AEC3: Reduce level of log messages
This change reduces the level of several non-critical log messages in
order to reduce log spamming.

Bug: webrtc:8671
Change-Id: I6faae7a2ae4eeafd18c2770208485a75ad946e20
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148528
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28809}
2019-08-08 14:08:05 +00:00
0d210ee2f6 Change return type of of ReceiveStatistics::Create to unique_ptr.
There are currently three overloads with different number of arguments,
and one of those return a raw pointer. This cl changes that to unique_ptr.
The transition plan is to update those downstream call sites that
currently require a raw pointer to use one of the other overloads.

Bug: webrtc:10679
Change-Id: I234605e99c04a59fbe6f478581ed8edd96a9b05a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148447
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28804}
2019-08-08 08:19:43 +00:00
c2fe547eba Remove unused fallbacks in PacedSender
Bug: webrtc:10809
Change-Id: I322e5f0dbfb8648aee4f88d37b8a0938a48c0f3e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148440
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28803}
2019-08-08 08:18:38 +00:00
eac47f7fae Removing unused fallback variant for the reverb computation
This CL removes a long unused fallback behavior for the reverb
computation.

Bug: webrtc:8671
Change-Id: I4b57795a9bb33769237858f40392881ee235653e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148520
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28802}
2019-08-08 08:00:38 +00:00
5b5d97c938 Reland of "Reporting of decoding_codec_plc events""
This is a reland of 0a88ea050cda58de81d624cf2764d46929447ed5.

The new stat will not be reported unless it is GT 0.

Reporting of decoding_codec_plc events

Bug: webrtc:10838
Change-Id: Ic8585b4eeae9a2643374f15bc2578d1141e59683
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148448
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Alex Narest <alexnarest@google.com>
Cr-Commit-Position: refs/heads/master@{#28797}
2019-08-07 18:41:46 +00:00
b1686786e8 Add RTC_ prefix to non-standard format specifier macro "PRIdNS"
Some of the macros in format_macros.h follow the C standard and try to fill holes in it (on Windows). But this one has no direct equivalent in the standard and is just mimicking the naming convention. That's not nice.

References:
https://devblogs.microsoft.com/cppblog/c99-library-support-in-visual-studio-2013/
https://stackoverflow.com/a/2524673

Change-Id: I53f3faca2976a5b5d4b04a67ffb56ae0f4e930b2
Bug: webrtc:10852
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147862
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28794}
2019-08-07 13:36:05 +00:00
12ebfa69ba Delete RtcpStatisticsCallback from ReceiveStatistics
Update VideoReceiveStream::GetStats to use
StreamStatistician::GetStatistics instead, similarly to the audio
receiver.

Bug: webrtc:10679
Change-Id: I8a701e8a8e921c87895424362dc83500737c916d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142233
Reviewed-by: Erik Språng <sprang@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@{#28793}
2019-08-07 13:33:55 +00:00
83bbe91398 Delete deprecated rtc_event_log header
Bug: webrtc:10206
Change-Id: I9ed3148843c647372993729b87c0e74741ab540b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147870
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28791}
2019-08-07 10:58:17 +00:00
e08648dc70 Add AbsoluteCaptureTime to RtpPacketInfo.
This change stores the optional `AbsoluteCaptureTime` header extension in `RtpPacketInfo` so that we later can consume it in `SourceTracker`.

Bug: webrtc:10739
Change-Id: I975e8863117fcda134535cd49ad71079a7ff38ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148068
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28790}
2019-08-07 10:12:56 +00:00
75caef7a4b Delete unused ACM members isac_decoder_16k_ and isac_decoder_32k_
Unused since https://webrtc-review.googlesource.com/c/103821.

Bug: webrtc:8396
Change-Id: Ia83f02f16d6ea8c260ea765b41506f2691e035bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148072
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28788}
2019-08-07 09:53:22 +00:00
c653172e74 Delete obsolete method AudioCodingModule::SetBitRate
Bug: None
Change-Id: I2291f7b4b46d269592eacad67a126010b750fac2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148079
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28784}
2019-08-07 08:37:25 +00:00
4d7c405599 Split out RtcpCnameCallback from RtcpStatisticsCallback
Cname callback is used only on receive side, and statistics (soon)
only on the send side.

Bug: webrtc:10679
Change-Id: I122e9cafaea93cd0ba75dc955a652d9d4bddc379
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147867
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28767}
2019-08-06 08:29:57 +00:00
ed44f5464a In ChannelReceive, use AcmReceiver directly, not AudioCodingModule
Bug: webrtc:9801
Change-Id: I02d76bc89c363247c8dc782db316a9f87a2b93ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/111504
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28766}
2019-08-06 08:23:26 +00:00
45231be79c AEC3: Removing unused code in the echo subtractor
Bug: webrtc:8671
Change-Id: I77e9c55fe2e1030e5b74c02d4bc9222de422f6f4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137045
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28760}
2019-08-05 14:53:59 +00:00
cdbaeeb737 Aec3:Remove unused legacy code
Bug: webrtc:8671
Change-Id: I54f14cc304dbe1639fbb356dbdf9beeb1781ede4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137801
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28759}
2019-08-05 14:13:54 +00:00
d7ee76cadd Wire up field trials for some experimental screenshare settings
Bug: b/132074409
Change-Id: I83d5334255bad4fcf585b9850506bbfe1914ba57
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147868
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28758}
2019-08-05 13:47:01 +00:00
b56cca3282 Remove the old ContributingSources class.
This change removes the old `ContributingSources` class. It has been replaced by the new `SourceTracker`.

Bug: webrtc:10793
Change-Id: Ibd481cf6584837c46b229b9fc2a071362f07d361
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147878
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28756}
2019-08-05 12:30:22 +00:00
8bbdb5b9bd Update VideoBitrateAllocator allocate to take a struct with more fields
We want to evaluate more data in order to make better choices in the
bitrate allocators.
In order to freely update the parameter list without
breaking the API many times for projects customizing them, we'll use a
struct instead.

Bug: webrtc:10126
Change-Id: I443f86781c5134950294cdd1e3197a47447cf973
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141418
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28748}
2019-08-02 13:52:54 +00:00
9a9f18a736 Get WebRTC.Video.ReceivedPacketsLostInPercent from ReceiveStatistics
Old way to produce this histogram was based on RtcpStatisticsCallback
reporting sent RTCP messages, with some additional processing by the
ReportBlockStats class. After this cl, to grand average fraction loss
is computed by StreamStatistician, queried by VideoReceiveStream when
the stream is closed down, and passed to ReceiveStatisticsProxy which
produces histograms.

This is a preparation for deleting the RtcpStatisticsCallback from
ReceiveStatistics.

Bug: webrtc:10679
Change-Id: Ie37062c1ae590fd92d3bd0f94c510e135ab93e8d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147722
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@{#28747}
2019-08-02 12:38:34 +00:00
59bbd65561 Add ToString method for AudioProcessing::Config
Bug: webrtc:10138
Change-Id: Icc0b40596867fd6fb8c1de5ca209dbafb504e50d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147726
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28745}
2019-08-02 10:01:56 +00:00
bd3f30535c Request a new key frame if packet buffer is cleared
Bug: webrtc:10843
Change-Id: I1eab0891f3e68b7d504dc637790604a25c243856
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147721
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28735}
2019-08-01 14:35:54 +00:00
84de3d95cf Factor framework dependencies out of audio_device_impl
Bug: None
Change-Id: I7d8d737134bb1a9dcf376cd39e74e73a5a6a0e97
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147723
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28733}
2019-08-01 12:06:14 +00:00
fd643a4782 Build core audio for older windows versions
Some of the constants and structure definitions used are only available with
specific and recent versions of the windows SDK. This change allows this
to build with a toolchain targeting WINVER 0x0601 (Windows 7)

Bug: None
Change-Id: I3339f7c44c375fb7d583b78aa137f748c9776a07
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147440
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Paul Roberts <pacaro@google.com>
Cr-Commit-Position: refs/heads/master@{#28730}
2019-07-31 22:58:00 +00:00
60ffc31ae1 Fix potential crash if nack is being processed when media gets disabled
This is a race that can happen if a nack arrives before media is
disabled, but the packet is not processed until after the disabling
is complete.

Bug: webrtc:10633, b/138636698
Change-Id: Ic90462b815163ab58c324e5cdb95c8d199c0b772
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147277
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28718}
2019-07-31 12:28:04 +00:00
7702c8ac04 Fix SendSideBweWithOverhead using new pacer code path.
This field trial was read in RTPSender, and the altered packet size
passed along to the pacer. Now, the pacer packet queue looks directly
at the packet instance, so it needs to be aware of the experiment flag
in order to make the right decision.

Bug: webrtc:10633, b/138582168
Change-Id: If1148f39c463e11ad49a659913465f131cf9b526
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147270
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28714}
2019-07-31 09:33:36 +00:00
e08ca23ec9 Include overlapping dialogs when capturing a window via screen cropping
This change includes windows owned by the primary captured window in the
captured frames if these conditions are met:
1) The owned window (e.g. dialog) overlaps the primary window (in whole
or part)
2) The primary window is otherwise eligible for the crop-from-screen
path (CroppingWindowCapturer is being used, and other conditions in
ShouldUseScreenCapturer are met)

In practice, this means that dialog windows / message boxes are captured
in many cases where they aren't today. This seems beneficial to some
scenarios (e.g. demonstrating / recording how to do something, or
requesting help with something, that involves dialogs).

This is a logical revert of a change for https://crbug.com/webrtc/8062 .
There's some commentary in the newer bug that attempts to make a case
for revisiting that change. (In summary: cases where a dialog would be
substantialy clipped / partial seem relatively uncommon and have
workarounds. Clipping may already occur for menus & tooltips. Clipping
seems less surprising than complete absence.)

Changing the GA_ROOT flag back to GA_ROOTOWNER is sufficient to restore
the older behavior. The removal of the EnumChildWindows call is just a
minor optimization (it was unnecessary/superfluous, since every child
window would match the GA_ROOT check; dialogs are owned root windows,
not child windows).

Removing condition (2) above (capturing dialogs & other related
overlapping windows when not using the crop-from-screen path) is tracked
by https://crbug.com/980864 .

Bug: webrtc:10767
Change-Id: If7b418365685a7b96dc93901ef9367844f9ee99e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147421
Commit-Queue: Jamie Walch <jamiewalch@chromium.org>
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Cr-Commit-Position: refs/heads/master@{#28711}
2019-07-30 23:54:50 +00:00