80c6762a37
Tweak ChannelReceive interface, to make it closer to ChannelReceiveProxy
...
This is a preparation for deleting ChannelReceiveProxy, Changes
signature of some methods, and demotes methods OnData and
OnReceivedPayloadData to private.
Bug: webrtc:9801
Change-Id: Ib00a80c6482ed5238f3cc8233860c70f11484df9
Reviewed-on: https://webrtc-review.googlesource.com/c/110606
Commit-Queue: Niels Moller <nisse@webrtc.org >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#25599}
2018-11-12 13:25:32 +00:00
140b1d94dc
Eliminate use of EventWrapper from android audio device tests
...
Bug: webrtc:3380
Change-Id: I746d2245966afe89065472d4a6a7447f8c63f9f9
Reviewed-on: https://webrtc-review.googlesource.com/c/110163
Reviewed-by: Henrik Andreassson <henrika@webrtc.org >
Reviewed-by: Magnus Jedvert <magjed@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#25598}
2018-11-12 13:22:46 +00:00
f4a3f9cc25
Add RtcEvent::timestamp_ms()
...
Bug: webrtc:8111
Change-Id: I0ec7eda2b2afcd945625fb9f5d592e73a97992e3
Reviewed-on: https://webrtc-review.googlesource.com/c/109861
Commit-Queue: Elad Alon <eladalon@webrtc.org >
Reviewed-by: Björn Terelius <terelius@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#25597}
2018-11-12 13:18:07 +00:00
89f874eb39
Add offer_extmap_allow_mixed to RTCConfiguration
...
Bug: webrtc:9986
Change-Id: I346e03a46f35c7d59d3ae769842e3aeec9d2d50d
Reviewed-on: https://webrtc-review.googlesource.com/c/110501
Commit-Queue: Johannes Kron <kron@webrtc.org >
Reviewed-by: Steve Anton <steveanton@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#25596}
2018-11-12 12:35:45 +00:00
5ae3a028c8
Revert "Run robolectric tests for Android on several Android API versions"
...
This reverts commit e598e6bff9528f77dc9f4fb3a5954ec5fb6790b0.
Reason for revert: Main suspect of increased Android tests flakiness
Original change's description:
> Run robolectric tests for Android on several Android API versions
>
> Depends on https://bugs.chromium.org/p/chromium/issues/detail?id=901324
>
> Bug: webrtc:9955
> Change-Id: I5e3f4c05b8258b90728644846f425ee131fda8d4
> Reviewed-on: https://webrtc-review.googlesource.com/c/109160
> Reviewed-by: Artem Titarenko <artit@webrtc.org >
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org >
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org >
> Commit-Queue: Artem Titarenko <artit@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#25582}
TBR=phoglund@webrtc.org ,magjed@webrtc.org ,sakal@webrtc.org ,artit@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:9955
Change-Id: I62c4c9c3238f777b6017701bc1332d8661308f9c
Reviewed-on: https://webrtc-review.googlesource.com/c/110609
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#25595}
2018-11-12 12:30:06 +00:00
20f60f0dc6
Fuzzer crash in AGC2.
...
Gain specified by fuzzer in APM config was too high.
Bug: chromium:901661
Change-Id: Id3ea8d23a4284a35c827bb16125902d84e37ca1e
Reviewed-on: https://webrtc-review.googlesource.com/c/110604
Commit-Queue: Alex Loiko <aleloi@webrtc.org >
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#25594}
2018-11-12 12:16:47 +00:00
cfe3b6afd9
Remove most of api/ortc/.
...
It's not currently used or maintained, so it shouldn't be a part of out API.
Bug: webrtc:9824
Change-Id: Ic44c5ea3a9eab8fb75e87a5005cbf6cdd4b1d4ad
Reviewed-on: https://webrtc-review.googlesource.com/c/107645
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Reviewed-by: Minyue Li <minyue@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#25593}
2018-11-12 11:24:07 +00:00
8584667583
Fix overflow for high bitrates in BitrateProber
...
Bug: webrtc:9395
Change-Id: Ic63d9a5ca40673eb87419d0d9e2e3b67fb1a81e4
Reviewed-on: https://webrtc-review.googlesource.com/c/110460
Reviewed-by: Sebastian Jansson <srte@webrtc.org >
Commit-Queue: Johannes Kron <kron@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#25592}
2018-11-12 10:50:14 +00:00
09102a02cf
Revert "Roll "Enable SSE, SSE2, and run-time detected SSE4.1 for libopus.""
...
This reverts commit 466620b326c5743d9e3ce0d5af967fd977c5cf38.
Reason for revert: Break downstream clients which are still expecting the previous references for NetEqDecodingTest.TestOpusBitExactness.
Original change's description:
> Roll "Enable SSE, SSE2, and run-time detected SSE4.1 for libopus."
>
> We manually roll third_party since we need to update impacted tests,
> namely bit-exact comparison of expected neteq_opus results.
> They have changed due to SSE optimization enabled here:
> 85d355e530db720b4ab9..ae94013397db720b4ab9..ae94013397fee916f36b..f42826395602b0a894b0..3f6180968495aadfb43f..fb48cd850c03b25bebb5..f6a2832441360db5b8aa..8209b47661https://boringssl.googlesource.com/boringssl.git/+log/dd412c428a..384d0eaf19
> * src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/2e722b007d..f04a3a61ad
> * src/third_party/depot_tools: 4d2d5b4bbe..edcefdcf7df56830ed40..fb0d66d04ca8e76f0ca5..f8ace9b670db720b4ab9..ae94013397db720b4ab9..ae94013397https://webrtc-review.googlesource.com/c/110040
> Commit-Queue: Yves Gerey <yvesg@webrtc.org >
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#25588}
TBR=phoglund@webrtc.org ,ivoc@webrtc.org ,yvesg@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:9530
Change-Id: I01549abdcfbcad70881ff9aeee913df68d0f0052
Reviewed-on: https://webrtc-review.googlesource.com/c/110602
Reviewed-by: Yves Gerey <yvesg@google.com >
Commit-Queue: Yves Gerey <yvesg@google.com >
Cr-Commit-Position: refs/heads/master@{#25591}
2018-11-12 09:55:10 +00:00
0b1b5c1b2a
Hide RtcEvent members behind accessors
...
Bug: webrtc:8111
Change-Id: I3d350a6e159330aed7362162006860ac86ed7c32
Reviewed-on: https://webrtc-review.googlesource.com/c/109881
Commit-Queue: Elad Alon <eladalon@webrtc.org >
Reviewed-by: Björn Terelius <terelius@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#25590}
2018-11-10 23:34:07 +00:00
eb809f30d1
Event logs - separate audio_level and voice_activity
...
Bug: webrtc:8111
Change-Id: I44d81c5b4f5b854e8accd84521fbbd7b50228903
Reviewed-on: https://webrtc-review.googlesource.com/c/109571
Commit-Queue: Elad Alon <eladalon@webrtc.org >
Reviewed-by: Björn Terelius <terelius@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#25589}
2018-11-10 01:41:28 +00:00
466620b326
Roll "Enable SSE, SSE2, and run-time detected SSE4.1 for libopus."
...
We manually roll third_party since we need to update impacted tests,
namely bit-exact comparison of expected neteq_opus results.
They have changed due to SSE optimization enabled here:
85d355e530db720b4ab9..ae94013397db720b4ab9..ae94013397fee916f36b..f42826395602b0a894b0..3f6180968495aadfb43f..fb48cd850c03b25bebb5..f6a2832441360db5b8aa..8209b47661https://boringssl.googlesource.com/boringssl.git/+log/dd412c428a..384d0eaf19
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/2e722b007d..f04a3a61ad
* src/third_party/depot_tools: 4d2d5b4bbe..edcefdcf7df56830ed40..fb0d66d04ca8e76f0ca5..f8ace9b670db720b4ab9..ae94013397db720b4ab9..ae94013397https://webrtc-review.googlesource.com/c/110040
Commit-Queue: Yves Gerey <yvesg@webrtc.org >
Reviewed-by: Ivo Creusen <ivoc@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#25588}
2018-11-09 22:30:47 +00:00
56a4b32398
Rename fields in rtc_event_log2.proto
...
1. s/deltas_ms/ms-deltas
2. s/deltas_bps/bps_deltas
3. s/raw_packet_deltas/raw_packet_blobs
Bug: webrtc:8111
Change-Id: Ib2f7457275e0b930a6aa73d628a707676c74a2a4
Reviewed-on: https://webrtc-review.googlesource.com/c/109142
Commit-Queue: Elad Alon <eladalon@webrtc.org >
Reviewed-by: Björn Terelius <terelius@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#25587}
2018-11-09 22:29:43 +00:00
a2eb0a7841
Fix up an outdated comment in peerconnection_integrationtest.cc.
...
Bug: webrtc:9719
Change-Id: Ied844fdb941b80ab84d43775cc315c075677dac0
Reviewed-on: https://webrtc-review.googlesource.com/c/110562
Reviewed-by: Steve Anton <steveanton@webrtc.org >
Commit-Queue: Bjorn Mellem <mellem@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#25586}
2018-11-09 18:53:05 +00:00
7127f342a9
Signal Network route change in fake ice.
...
Fake Ice currently does not signal the network route change. Also, it is not aware of the network thread, so added a setter for a network thread.
Bug: None
Change-Id: I25326282f32d36229422eca7368b53ee7b52ec72
Reviewed-on: https://webrtc-review.googlesource.com/c/110363
Commit-Queue: Peter Slatala <psla@webrtc.org >
Reviewed-by: Qingsi Wang <qingsi@webrtc.org >
Reviewed-by: Bjorn Mellem <mellem@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#25585}
2018-11-09 17:15:21 +00:00
d95b0a2fbd
Use delta-encoding in new WebRTC event logs
...
The new event log format makes use of delta encoding to compress
parts of the log.
Bug: webrtc:8111
Change-Id: I7bec839555323a7537dcec831d4ac1d5eb109932
Reviewed-on: https://webrtc-review.googlesource.com/c/109161
Commit-Queue: Elad Alon <eladalon@webrtc.org >
Reviewed-by: Björn Terelius <terelius@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#25584}
2018-11-09 16:39:16 +00:00
72467209a8
Clean up root OWNERS.
...
Remove some redundant lines and Tina who has left the project :(
Bug: None
Change-Id: I8a8cba3c2b13d93e668754fbe6b06daa09095534
Notry: true
Reviewed-on: https://webrtc-review.googlesource.com/c/110503
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#25583}
2018-11-09 14:23:59 +00:00
e598e6bff9
Run robolectric tests for Android on several Android API versions
...
Depends on https://bugs.chromium.org/p/chromium/issues/detail?id=901324
Bug: webrtc:9955
Change-Id: I5e3f4c05b8258b90728644846f425ee131fda8d4
Reviewed-on: https://webrtc-review.googlesource.com/c/109160
Reviewed-by: Artem Titarenko <artit@webrtc.org >
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org >
Reviewed-by: Patrik Höglund <phoglund@webrtc.org >
Commit-Queue: Artem Titarenko <artit@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#25582}
2018-11-09 13:41:10 +00:00
9973fa88ae
Pass HdrMetadata between VideoFrame and EncodedImage for VP9
...
Bug: webrtc:8651
Change-Id: Ie4d7ee19bead84eda7788076662c4066edc3f024
Reviewed-on: https://webrtc-review.googlesource.com/c/109583
Reviewed-by: Erik Språng <sprang@webrtc.org >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Reviewed-by: Niels Moller <nisse@webrtc.org >
Commit-Queue: Johannes Kron <kron@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#25581}
2018-11-09 13:33:37 +00:00
6c373cccbb
Add support for audio in latency visualization.
...
The RTC event log analyzer would previously only plot network latency
for incoming video streams. (The latency is computed from the capture
time in the RTP header, and the packet receive time.) This CL adds
support for audio packets, which requires estimating the RTP clock
frequency for the incoming packets.
Bug: None
Change-Id: Idf1ff9febfdd4097976b22a61f1c5679deb6068c
Reviewed-on: https://webrtc-review.googlesource.com/c/108784
Reviewed-by: Sebastian Jansson <srte@webrtc.org >
Commit-Queue: Björn Terelius <terelius@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#25580}
2018-11-09 13:10:57 +00:00
d8aa9f93e8
Fix flaky JsepTransportControllerTests.
...
In a handful of places we wait for the old IceConnectionState to reach some value and then we assume that the new connection states have also been updated. However those are updated in response to different events that might not have fired yet, so sometimes these tests will fail.
This change makes us wait explicitly for those states to update.
Bug: webrtc:9983
Change-Id: I5cb6652ee29c0b86c0834174442140a3863e08e4
Reviewed-on: https://webrtc-review.googlesource.com/c/110441
Reviewed-by: Per Kjellander <perkj@webrtc.org >
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#25579}
2018-11-09 11:53:15 +00:00
ad1d9f0d78
Add RTP header extension for HDR metadata
...
Bug: webrtc:8651
Change-Id: I1c956eaac1532ac0d3820084edb4054a4cc9c68d
Reviewed-on: https://webrtc-review.googlesource.com/c/109924
Commit-Queue: Johannes Kron <kron@webrtc.org >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Reviewed-by: Åsa Persson <asapersson@webrtc.org >
Reviewed-by: Alex Loiko <aleloi@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#25578}
2018-11-09 11:10:12 +00:00
ee45f900c4
In RTP to NTP estimator do not allow huge jumps in NTP timestamps
...
Bug: webrtc:9698
Change-Id: I64b5ec4d611fd2981bbc11ef2652e97cfd1e72c7
Reviewed-on: https://webrtc-review.googlesource.com/c/110247
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org >
Reviewed-by: Niels Moller <nisse@webrtc.org >
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#25577}
2018-11-09 10:23:23 +00:00
06f6bc9ec4
Reintroduce missing dependencies in libwebrtc.a library.
...
Some targets used to be included transitively via ortc.
Since ortc module has been removed (Bug: webrtc:9824),
this CL explicitly add them in main //:webrtc target.
As a result, the following functions are exposed again:
CreateBuiltinVideoDecoderFactory()
CreateBuiltinVideoEncoderFactory()
CreatePeerConnectionFactory()
[...]
Bug: webrtc:9824
Bug: webrtc:9973
Change-Id: Iebfae582f8887bf76338c73fc85c4608e96c3f0d
Reviewed-on: https://webrtc-review.googlesource.com/c/110248
Commit-Queue: Yves Gerey <yvesg@google.com >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#25576}
2018-11-09 09:44:55 +00:00
175aa2e95c
Implement data channels over media transport.
...
This changes PeerConnection to allow sending and receiving data channel
messages over the media transport. If |use_media_transport_for_data_channels|
is set, PeerConnection will use a DCT_MEDIA_TRANSPORT mode for data
channels.
DCT_MEDIA_TRANSPORT acts exactly like DCT_SCTP within the data channel
and peer connection layers. On the transport layer, it uses the media
transport instead of SCTP. It appears as an RTP data channel in SDP
(just as media over media-transport appears as RTP in SDP).
Bug: webrtc:9719
Change-Id: I6a90142bd3f43668479c825ed02689dcd0d58b78
Reviewed-on: https://webrtc-review.googlesource.com/c/109740
Commit-Queue: Bjorn Mellem <mellem@webrtc.org >
Reviewed-by: Steve Anton <steveanton@webrtc.org >
Reviewed-by: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#25575}
2018-11-09 00:40:32 +00:00
c2ebe21ba9
Reland "Use the factory instead of using the builtin code path in VideoCodecInitializer"
...
Compared the original CL: https://webrtc-review.googlesource.com/c/src/+/94782
This new CL added backward compatible functions to WebRtcMediaEngineFactory so that internal projects will not be broken.
Because of that, now we can revert all the changes to SDK and PeerConnection and do it in following CLs. This makes this CL cleaner.
One temporary disadvantage of this is the media engine now need to take a dependency onto builtin video bitrate factory, but practically it just moved code around and should not result in a large binary size change. We can remove this dependency later if needed.
Bug: webrtc:9513
Change-Id: I38708762ff365e4ca05974b99fac71edc739a756
Reviewed-on: https://webrtc-review.googlesource.com/c/109040
Commit-Queue: Jiawei Ou <ouj@fb.com >
Reviewed-by: Kári Helgason <kthelgason@webrtc.org >
Reviewed-by: Niels Moller <nisse@webrtc.org >
Reviewed-by: Erik Språng <sprang@webrtc.org >
Reviewed-by: Seth Hampson <shampson@webrtc.org >
Reviewed-by: Sebastian Jansson <srte@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#25574}
2018-11-08 19:10:47 +00:00
0393c64af1
[Win/boringSSL] Add nasm as part of required dependencies.
...
In order to fix the roll https://webrtc-review.googlesource.com/c/src/+/110080 ,
this CL updates WebRTC DEPS to be on a par with Chromium's CL:
"Add nasm support to Chromium, use it for boringssl."
7d284aff8chttps://webrtc-review.googlesource.com/c/110340
Reviewed-by: Oleh Prypin <oprypin@webrtc.org >
Commit-Queue: Yves Gerey <yvesg@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#25573}
2018-11-08 18:02:49 +00:00
ada077f447
Callback changes to media transport interface:
...
1) allow multiple target rate observers
2) add getter for overhead
3) add getter for target rate
4) add callback for network route changed.
Bug: webrtc:9719
Change-Id: I06518cd9aed0ebabd204a7f6af3b86f51fd694e0
Reviewed-on: https://webrtc-review.googlesource.com/c/109940
Reviewed-by: Steve Anton <steveanton@webrtc.org >
Reviewed-by: Qingsi Wang <qingsi@webrtc.org >
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org >
Reviewed-by: Bjorn Mellem <mellem@webrtc.org >
Commit-Queue: Peter Slatala <psla@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#25572}
2018-11-08 17:47:09 +00:00
87e1619fb9
Add owners for media_transport_interface
...
Bug: webrtc:9982
No-Try: True
Change-Id: Ib9e468146e11152cbdca8df9a8d1f26d85a5e287
Reviewed-on: https://webrtc-review.googlesource.com/c/110280
Reviewed-by: Steve Anton <steveanton@webrtc.org >
Commit-Queue: Peter Slatala <psla@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#25571}
2018-11-08 17:45:39 +00:00
d3438aa1ed
Add ability to specify if rate controller of video encoder is trusted.
...
If rate controller is trusted, we disable the frame dropper in the
media optimization module.
This is a re-land of
https://webrtc-review.googlesource.com/c/src/+/105020
Bug: webrtc:9890
Change-Id: I418e47a43a1a98cb2fd5295c03360b954f2288f2
Reviewed-on: https://webrtc-review.googlesource.com/c/109141
Commit-Queue: Erik Språng <sprang@webrtc.org >
Reviewed-by: Niels Moller <nisse@webrtc.org >
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#25570}
2018-11-08 16:41:12 +00:00
6528d8a954
In Android encoders, cache EncoderInfo in InitEncode.
...
GetEncoderInfo() is now called every frame, so we should not do
expensive parsing or logging in there. Instead, prepare an EncoderInfo
instance in InitEncode() and just return that in GetEncoderInfo().
Bug: webrtc:9890
Change-Id: Idc9e79e681c6f7ff4f9b446aa298c156f25bc6f6
Reviewed-on: https://webrtc-review.googlesource.com/c/110161
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org >
Commit-Queue: Erik Språng <sprang@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#25569}
2018-11-08 16:40:01 +00:00
260770c28c
Delete rtc::Filesystem. Move needed functions to filerotatingstream.cc.
...
Bug: webrtc:6424
Change-Id: I2a1f215cb4521f21c2f8defd03f0f28c1deae24a
Reviewed-on: https://webrtc-review.googlesource.com/c/109003
Commit-Queue: Niels Moller <nisse@webrtc.org >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#25568}
2018-11-08 16:23:36 +00:00
b0550bdf96
Eliminate use of EventWrapper from mac audio device
...
Bug: webrtc:3380
Change-Id: I9b34588a6a2b035f1787782421e4fc3e6650ef1a
Reviewed-on: https://webrtc-review.googlesource.com/c/110244
Reviewed-by: Henrik Andreassson <henrika@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#25567}
2018-11-08 15:52:53 +00:00
c94b22e2e6
Add magjed/nisse/sprang/brandtr as api/video_codecs owners
...
Bug: None
Change-Id: If6efd711b38befa106e987562a455a5d5feb17d5
Reviewed-on: https://webrtc-review.googlesource.com/c/110245
Reviewed-by: Stefan Holmer <stefan@webrtc.org >
Commit-Queue: Erik Språng <sprang@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#25566}
2018-11-08 15:43:03 +00:00
c5dd3009b4
Introduce RtpPacket::GetExtension accessor that return result
...
instead of using output parameter.
Bug: None
Change-Id: I1d5c150b7cb6302aa29e040e8c9fe68bddfd8c0e
Reviewed-on: https://webrtc-review.googlesource.com/c/110240
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#25565}
2018-11-08 15:34:40 +00:00
357f596558
Split a separate codecs target off of :video_jni
...
This will allow clients to include only the software codecs they need
rather than being forced to bundle them all.
- libjingle_peerconnection_jni keeps its allow_poison for now, until
dependent targets bundle their own codecs explicitly.
- native_api_codecs and native_api_video lose their allow_poison
because dependent targets are already bundling codecs explicitly.
- libjingle_peerconnection_metrics_default_jni and
native_api_peerconnection lose their allow_poison because they
were not actually poisoned.
legacy_hwcodecs_jni and default_video_codec_factory_jni exist for
clients that want to continue bundling the same codecs they get by
default today.
Bug: webrtc:7925
Change-Id: Idf853a6bc77f43decd35ad2a0f467937fec8f8b5
Reviewed-on: https://webrtc-review.googlesource.com/c/108221
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org >
Commit-Queue: Jonathan Yu <yujo@chromium.org >
Cr-Commit-Position: refs/heads/master@{#25564}
2018-11-08 15:27:37 +00:00
5bb1ed6144
Eliminate use of EventWrapper from ios audio device tests
...
Bug: webrtc:3380
Change-Id: I2d2f8a7152212e80600449d49e7f7316dd89bfc2
Reviewed-on: https://webrtc-review.googlesource.com/c/110200
Reviewed-by: Henrik Andreassson <henrika@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#25563}
2018-11-08 15:12:50 +00:00
a33c7af42b
Tolerate optional chunks in WAV files
...
Wave files may contain optional chunks, such as a metadata one.
This CL makes WavReader tolerant to such chunks - it just ignores them.
For more details on the Wave format, please refer to
https://sites.google.com/site/musicgapi/technical-documents/wav-file-format .
Bug: webrtc:8762
Change-Id: Ie0e19dea75661808e7781f51faa1d0f0affeb3e1
Reviewed-on: https://webrtc-review.googlesource.com/c/40300
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#25562}
2018-11-08 14:34:20 +00:00
c496d58882
Add flag for fast jitter buffer playout in neteq simulation
...
It is currently not possible to run e.g. neteq_rtpplay in the fast
accelerate mode.
Bug: None
Change-Id: I5e0ce3fae2ad5585fe9fb545109bb0c9a87fd201
Reviewed-on: https://webrtc-review.googlesource.com/c/110162
Commit-Queue: Sam Zackrisson <saza@webrtc.org >
Reviewed-by: Minyue Li <minyue@webrtc.org >
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#25561}
2018-11-08 14:32:48 +00:00
e6c2c0853f
MsanUninitialized: restric type check to msan case.
...
This change is needed to avoid undesired failures caused by
IsTriviallyCopyable behavior differences across different compilers and
STL implementations.
It will allow https://webrtc-review.googlesource.com/c/src/+/40300 to
land.
Bug: webrtc:8762
Change-Id: Ide32062605320c706b8a2f3f149d73b967b1fe30
Reviewed-on: https://webrtc-review.googlesource.com/c/110143
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#25560}
2018-11-08 13:28:34 +00:00
c4e9825c04
Delete classes EventFactory and EventFactoryImpl.
...
Followup to cl https://webrtc-review.googlesource.com/c/src/+/107890
Bug: webrtc:3380
Change-Id: Iac4389186be3ffbc55e53e18aa302465cd771da4
Reviewed-on: https://webrtc-review.googlesource.com/c/110140
Reviewed-by: Philip Eliasson <philipel@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#25559}
2018-11-08 13:15:39 +00:00
2a74263e3f
Make the bitrate_allocator param optional to prepare for its removal
...
in https://webrtc-review.googlesource.com/109040
Bug: webrtc:9513
Change-Id: I676e5e0242f068b12764a52bf8b6a6865ea7f120
Reviewed-on: https://webrtc-review.googlesource.com/c/110142
Reviewed-by: Erik Språng <sprang@webrtc.org >
Commit-Queue: Oleh Prypin <oprypin@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#25558}
2018-11-08 12:59:46 +00:00
cd2e105128
Reenable test RampUpTest.AudioTransportSequenceNumber
...
Flakiness should be fixed with cl
https://webrtc-review.googlesource.com/96900
Bug: webrtc:8878
Change-Id: I536d670fdf3b9e52091931e2f37ff9b8d02c2f77
Reviewed-on: https://webrtc-review.googlesource.com/c/110160
Reviewed-by: Sebastian Jansson <srte@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#25557}
2018-11-08 12:28:19 +00:00
694ed1793c
Add a style rule about not using const optional<T>& arguments
...
Motivated by discussions here:
https://webrtc-review.googlesource.com/c/src/+/109583
Bug: none
Change-Id: Ia0723adf9fa7c970137ffc9cb5612cb3360d7f5f
Notry: true
Reviewed-on: https://webrtc-review.googlesource.com/c/109568
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
Reviewed-by: Niels Moller <nisse@webrtc.org >
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#25556}
2018-11-08 11:57:35 +00:00
f0e7440a35
Add missing conditional defines to neteq test and tools targets
...
The .cc source files listed below #ifdef for WEBRTC_CODEC_OPUS and
WEBRTC_CODEC_ILBC but the build files don't include the defines.
modules/audio_coding/neteq/tools/neteq_test.cc
modules/audio_coding/neteq/tools/neteq_test_factory.cc
Bug: None
Change-Id: I6065021f68e58d0e5663acd006a9865bf265adc0
Reviewed-on: https://webrtc-review.googlesource.com/c/109925
Reviewed-by: Minyue Li <minyue@webrtc.org >
Commit-Queue: Sam Zackrisson <saza@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#25555}
2018-11-08 11:25:10 +00:00
689983f6bc
Deprecate EventFactory and delete all usage.
...
Will be deleted as soon as downstream calls of
VideoCodingModule::Create are updated.
Tbr: sprang@webrtc.org # Trivial change in video/
Bug: webrtc:3380
Change-Id: Iaeb6da2fb68991225fe9086ddddd4a553e1620b4
Reviewed-on: https://webrtc-review.googlesource.com/c/107890
Reviewed-by: Niels Moller <nisse@webrtc.org >
Reviewed-by: Philip Eliasson <philipel@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#25554}
2018-11-08 11:00:37 +00:00
54b4924349
Update H264 encoder to use GetEncoderInfo
...
Bug: webrtc:9890
Change-Id: I952b979346d97c42a4f60e9e2b091da563dfffab
Reviewed-on: https://webrtc-review.googlesource.com/c/109921
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Commit-Queue: Erik Språng <sprang@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#25553}
2018-11-07 17:01:50 +00:00
10608708eb
Update LibVpxVp8Encoder to use GetEncoderInfo
...
Bug: webrtc:9890
Change-Id: I76566bc38137c81b029fa848da89c96454260895
Reviewed-on: https://webrtc-review.googlesource.com/c/109920
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Commit-Queue: Erik Språng <sprang@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#25552}
2018-11-07 17:00:10 +00:00
727d1649c6
Update VP9 encoder to use GetEncoderInfo
...
Bug: webrtc:9890
Change-Id: I74c1e098c800a44e2e038cd8a01be6c61bec97f5
Reviewed-on: https://webrtc-review.googlesource.com/c/109922
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Commit-Queue: Erik Språng <sprang@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#25551}
2018-11-07 16:51:40 +00:00
5473a45688
Remove multiple RTX codec entries in GetRtpReceiver/SenderCapabilities
...
The spec says there should only be a single entry with no parameters.
Bug: webrtc:9970
Change-Id: I8b55f10b8cb795021269827c6e0e9f12ab86a3c9
Reviewed-on: https://webrtc-review.googlesource.com/c/109588
Commit-Queue: Steve Anton <steveanton@webrtc.org >
Reviewed-by: Steve Anton <steveanton@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#25550}
2018-11-07 15:35:56 +00:00
