629de6f7ed
Merge RtpPacket HasExtension and IsExtensionReserved functions
...
RtpPacket doesn't keep difference between reserved and set extension.
Bug: None
Change-Id: I1c79f4ebd7ba20ae5da0194c3faa418050db7d8e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166340
Reviewed-by: Erik Språng <sprang@webrtc.org >
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30316}
2020-01-20 11:37:25 +00:00
df2c601616
Move Offset constants from VideoSendTiming value to VideoTimingExtension class
...
These constants describes how value should be put on the wire and thus
belong to the extension builder/writer class rather than extension value class
Bug: None
Change-Id: I65ca3923eddcc2e48563ad69b98356c159ad86be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166461
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30305}
2020-01-17 15:57:38 +00:00
db6ca7f2d7
Add safety checks in RtpPacket::ZeroMutableExtensions and fuzz it
...
Bug: chromium:1042535
Change-Id: I0f7ef1086631b5beb2e0c89d57534d2551289117
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166441
Reviewed-by: Niels Moller <nisse@webrtc.org >
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30303}
2020-01-17 14:22:04 +00:00
64f1f3f04e
Replace RTC_FALLTHROUGH with ABSL_FALLTHROUGH_INTENTED
...
Bug: None
Change-Id: I7287403f3fb13b8e30f92ca3cf1882b03bb53a6e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166176
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30283}
2020-01-16 15:20:35 +00:00
c6f81a71e5
Remove higher_spatial_layers from RTPVideoHeader structure as unused.
...
The idea to communicate spatial dependencies with spatial layers bitmask
wasn't fully implemented and was dropped in later version of the descriptor.
Bug: webrtc:10342
Change-Id: I1ed191c3a2a9d2e1e9ddf313f781ca8257c34dfa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166165
Reviewed-by: Markus Handell <handellm@webrtc.org >
Reviewed-by: Niels Moller <nisse@webrtc.org >
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30278}
2020-01-16 11:11:39 +00:00
edb80cff01
Delete RtpDepacketizer interface as no longer used
...
Bug: webrtc:11152
Change-Id: I0c5f2167ba39c22f4491d2e34f3462b9ecb9bf2f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166160
Reviewed-by: Markus Handell <handellm@webrtc.org >
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30276}
2020-01-16 09:00:16 +00:00
61d6471912
Change H264 depacketizer to implement VideoRtpDepacketizer interface
...
Bug: webrtc:11152
Change-Id: If5169f47d85918356fa66e2bf3422d722044aa1f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165581
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Reviewed-by: Markus Handell <handellm@webrtc.org >
Reviewed-by: Sam Zackrisson <saza@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30264}
2020-01-15 12:26:55 +00:00
d06588a758
Change Av1 depacketizer to implement VideoRtpDepacketizer interface
...
Bug: webrtc:11152
Change-Id: I322115263f60439bee36277157a0acef9bd28e3e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165343
Reviewed-by: Sam Zackrisson <saza@webrtc.org >
Reviewed-by: Philip Eliasson <philipel@webrtc.org >
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30260}
2020-01-15 10:16:03 +00:00
b2b2031457
Concatenate string literals at compile time.
...
This CL was generated by running:
git ls-files | grep ".cc" | xargs perl -i -ne 'BEGIN {undef $/}; s/("[\s\n]*<<[\s\n]*")/" "/g; print;'; git cl format
After that I manually edited modules/audio_processing/gain_controller2.cc to preserve its original
formatting.
This primary benefit of this change is a small reduction in binary size.
Bug: None
Change-Id: I689fa7ba9c717c314bb167e5d592c3c4e0871e29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165961
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30251}
2020-01-14 14:47:48 +00:00
7d43801a07
Delete RtpGenericDepacketizer as no longer used
...
Bug: webrtc:11152
Change-Id: I275765e1aa013d8188d43e2911e8ab022563d1d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165394
Reviewed-by: Markus Handell <handellm@webrtc.org >
Reviewed-by: Sam Zackrisson <saza@webrtc.org >
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30234}
2020-01-13 13:45:37 +00:00
b42aeaa3fb
Move RtpDepacketizerH264 into own files
...
Bug: webrtc:11152
Change-Id: Iab4975e9f378b177a2abf34559f9b74752e69843
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165582
Reviewed-by: Markus Handell <handellm@webrtc.org >
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30212}
2020-01-10 15:33:54 +00:00
5c35f2fb1b
Delete RtpDepacketizerVp9 in favor of VideoRtpDepacketizerVp9
...
Bug: webrtc:11152
Change-Id: Ic50f2dc49ca420b3406d4dea11ed20328aa59136
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165382
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Reviewed-by: Sam Zackrisson <saza@webrtc.org >
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30195}
2020-01-09 13:07:44 +00:00
26e1b7ac01
Delete RtpDepacketizerVp8 in favor of VideoRtpDepacketizerVp8
...
Bug: webrtc:11152
Change-Id: I1a6225701ecd6f7a34c946d7296f0ab0cbb5eaef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165342
Reviewed-by: Markus Handell <handellm@webrtc.org >
Reviewed-by: Sam Zackrisson <saza@webrtc.org >
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30190}
2020-01-09 12:10:19 +00:00
57218b4e22
Delete RtpDepacketizer::Create factory
...
Bug: webrtc:11152
Change-Id: I09824b97506a11f917cd71f2f0d30306538eee13
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163023
Reviewed-by: Markus Handell <handellm@webrtc.org >
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30178}
2020-01-08 11:41:06 +00:00
27f4d325ad
Add VideoRtpDepacketizerGeneric
...
Bug: webrtc:11152
Change-Id: I27d6a62093d36a4e77dd35d4c115af8cdcc0178a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162202
Reviewed-by: Markus Handell <handellm@webrtc.org >
Reviewed-by: Philip Eliasson <philipel@webrtc.org >
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30160}
2020-01-07 09:27:34 +00:00
00a1bcb441
Ensure that unset capture timestamp wouldn't cause incorrect SR rtp timestamps
...
If for some reason capture timestamp is unset, the default value of 0 would be
passed to RtcpSender. This will cause rtp timestamps to grow at double the rate
in Sender Reports because it has time since the last frame capture as a term.
Bug: none
Change-Id: I2fe09dabef6b0957fb504deaa06393dedc4a9e70
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162481
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30105}
2019-12-17 12:03:24 +00:00
32fe4ef967
Move vp9 rtp depacketization to VideoRtpDepacketizerVp9
...
Bug: webrtc:11152
Change-Id: I560d4cd62fabae093e3df592f0e7cc4001c10657
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162420
Reviewed-by: Markus Handell <handellm@webrtc.org >
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30102}
2019-12-16 17:11:13 +00:00
eae6896f76
Move vp8 rtp depacketization to VideoRtpDepacketizerVp8
...
Bug: webrtc:11152
Change-Id: Ic2b7fd091cb4d095ce29fbe06196f6424c08fce1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161451
Reviewed-by: Markus Handell <handellm@webrtc.org >
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30088}
2019-12-13 15:10:46 +00:00
5e9cac984f
Don't try to resend packets that were removed out of order.
...
Bug: webrtc:11206
Change-Id: Iae05e1db80afd871d37aca203e17bad40dbc9522
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162041
Reviewed-by: Erik Språng <sprang@webrtc.org >
Commit-Queue: Sebastian Jansson <srte@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30083}
2019-12-13 10:29:49 +00:00
1e51a388bc
Makes padding prefer video SSRCs instead of audio.
...
Some clients will not count audio packets into the bandwidth estimate
despite negotiating e.g. abs-send-time for that SSRC.
If padding is sent on such an RTP module, we might get stuck in a low
resolution.
This CL works around that by preferring to send padding on video SSRCs.
Bug: webrtc:11196
Change-Id: I1ff503a31a85bc32315006a4f15f8b08e5d4e883
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161941
Commit-Queue: Erik Språng <sprang@webrtc.org >
Reviewed-by: Sebastian Jansson <srte@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30066}
2019-12-11 16:32:14 +00:00
a3ecb7a656
Migrate tests from RtpDepacketizer to VideoRtpDepacketizer interface
...
Bug: webrtc:11152
Change-Id: I1b1c5183d35b791c09c14c9d1f0ca240c1749d9a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161455
Reviewed-by: Niels Moller <nisse@webrtc.org >
Reviewed-by: Philip Eliasson <philipel@webrtc.org >
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30055}
2019-12-10 17:37:46 +00:00
951e289853
Add VideoTimingExtension to kFecOrPaddingExtensionSizes.
...
As of https://webrtc-review.googlesource.com/c/src/+/158899 , FEC may be
used on packets with VideoTimingExtension. This may result in creation
of FEC packets that exceed the maximum configured RTP packet size.
This problem occurs most frequently with datagram transports that define a
smaller maximum packet size.
Bug: webrtc:9719
Change-Id: I842216a6696a695f0a3f01a221e538605fc5b9bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161557
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
Commit-Queue: Bjorn Mellem <mellem@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30045}
2019-12-09 18:46:53 +00:00
80bc1acb9c
Add implementations of the VideoRtpDepacketizer interface
...
while suboptimal, these implementions are complete and allow to
swap code from using RtpDepacketizer interface to VideoRtpDepacketizer
Bug: webrtc:11152
Change-Id: Ie7823feeb5b0563b74754255aaedfada9d446ac5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161380
Reviewed-by: Philip Eliasson <philipel@webrtc.org >
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30031}
2019-12-06 15:20:29 +00:00
fc50e44a03
Introduce VideoRtpDepacketizer interface to replace RtpDepacketizer
...
Bug: webrtc:11152
Change-Id: I20fd81233080d45d8978e5e57d0be6b592f44f43
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161322
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Reviewed-by: Erik Språng <sprang@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30018}
2019-12-05 15:05:30 +00:00
cae277959b
Introduce InbandComfortNoise RTP header extension.
...
BUG: webrtc:11085
Change-Id: I9b556a0d67d3c239abc247787103af9e50af4e65
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159710
Commit-Queue: Minyue Li <minyue@webrtc.org >
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org >
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30014}
2019-12-05 13:35:01 +00:00
577b88dae7
Add new_request flag to SendFullIntraRequest
...
This allows one to request the same sequence number again
in the case of resending an FIR to the a sender before the
sender has time to send a key-frame.
Bug: webrtc:11171
Change-Id: Idd8e8120ccbcc194cefb8d0cf3f7cc64e7f76aa5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161236
Commit-Queue: Evan Shrubsole <eshr@google.com >
Reviewed-by: Erik Språng <sprang@webrtc.org >
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30006}
2019-12-04 13:45:02 +00:00
499b3b6c7e
In RtpDepacketizerAV1 use aggregation header to detect key frames
...
instead of guessing based on presence of the sequence header OBU.
Bug: webrtc:11042
Change-Id: I9a0674249feceebb07299ea965c62e87499f6baf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161013
Reviewed-by: Philip Eliasson <philipel@webrtc.org >
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29958}
2019-11-29 10:14:20 +00:00
0682ca9a83
Use AV1 packetizer/depacketizer for AV1 bitstreams
...
Bug: webrtc:11042
Change-Id: Ibf45a99d8016dccbe109d946ac967efa927312e4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161011
Reviewed-by: Philip Eliasson <philipel@webrtc.org >
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29953}
2019-11-28 18:01:10 +00:00
2dec496f80
Add directive to make TRACE_EVENT macros optional.
...
Bug: webrtc:11132
Change-Id: I801994ad262e1acff73e4c20afd7a7343b56268c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160654
Commit-Queue: Doudou Kisabaka <doudouk@google.com >
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29949}
2019-11-28 15:58:24 +00:00
096a46f38f
Implement AV1 RtpPacketizer class
...
Bug: webrtc:11042
Change-Id: Id1fc0acfa87a4520344f2636f50cb4d4e7284829
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160416
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Reviewed-by: Philip Eliasson <philipel@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29947}
2019-11-28 14:39:02 +00:00
5314b13a8d
Fix undefined-shift in RtpDepacketizerAv1::AssembleFrame
...
Bug: chromium:1028348
Change-Id: I824e84138acbf4e73fc21ee8248e29e5cc7a0ba0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160643
Reviewed-by: Sam Zackrisson <saza@webrtc.org >
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29945}
2019-11-28 11:27:33 +00:00
dc36829db0
Add VideoCodecType::kVideoCodecAV1 value
...
Bug: webrtc:11042
Change-Id: I3c5151c9e47679760f8f7d79270488fa8f4c7db5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159282
Reviewed-by: Åsa Persson <asapersson@webrtc.org >
Reviewed-by: Philip Eliasson <philipel@webrtc.org >
Reviewed-by: Niels Moller <nisse@webrtc.org >
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29927}
2019-11-27 10:18:45 +00:00
038fd99780
Add RtpDepacketizerAv1::AssembleFrame function
...
Bug: webrtc:11042
Change-Id: I677fc6a9affacf3b7c80adc2c3493c16806db1f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160003
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Reviewed-by: Philip Eliasson <philipel@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29862}
2019-11-21 14:50:41 +00:00
d51cc7bd71
Add absolute capture time property to rtp sources.
...
This part of the effort to implement A/V sync metric.
Bug: webrtc:10739
Change-Id: I4adba1b99b37b31868168e37d9aa8e03f8ea6d4e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159886
Commit-Queue: Ruslan Burakov <kuddai@webrtc.org >
Reviewed-by: Björn Terelius <terelius@webrtc.org >
Reviewed-by: Minyue Li <minyue@webrtc.org >
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
Reviewed-by: Ruslan Burakov <kuddai@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29849}
2019-11-20 18:50:45 +00:00
063c7d18c0
In dependency descriptor remove extended fields indicator
...
to follow PR64 spec change
https://github.com/AOMediaCodec/av1-rtp-spec/pull/64
Bug: webrtc:10342
Change-Id: Ic082d5e551b5f38427d5a43be987b0d35f6ea155
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160001
Reviewed-by: Philip Eliasson <philipel@webrtc.org >
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29832}
2019-11-19 13:12:10 +00:00
ccf12c6e97
Reland "Add AV1 RtpDepacketizer class"
...
This is a reland of 49470c2ac460ed8cce250942e8525c5f14e32778
Tentative reland to rule-out bot flakiness.
Original change's description:
> Add AV1 RtpDepacketizer class
>
> Implement Parse function that extracts is_first_packet_in_frame,
> is_last_packet_in_frame, and frame_type fields.
>
> Bug: webrtc:11042
> Change-Id: I9360ea52ef274281b5c5e4c31955100b92155bfe
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159180
> Reviewed-by: Philip Eliasson <philipel@webrtc.org >
> Reviewed-by: Sam Zackrisson <saza@webrtc.org >
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#29814}
TBR=saza@webrtc.org ,philipel@webrtc.org
Bug: webrtc:11042
Change-Id: Ibd672ce685bcab86960500740465539ed70fcdf4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159941
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29819}
2019-11-18 15:23:08 +00:00
9f99175710
Revert "Add AV1 RtpDepacketizer class"
...
This reverts commit 49470c2ac460ed8cce250942e8525c5f14e32778.
Reason for revert: Seems to trigger linker error on iOS64. See:
https://ci.chromium.org/p/webrtc/builders/ci/iOS64%20Debug/17733
Original change's description:
> Add AV1 RtpDepacketizer class
>
> Implement Parse function that extracts is_first_packet_in_frame,
> is_last_packet_in_frame, and frame_type fields.
>
> Bug: webrtc:11042
> Change-Id: I9360ea52ef274281b5c5e4c31955100b92155bfe
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159180
> Reviewed-by: Philip Eliasson <philipel@webrtc.org >
> Reviewed-by: Sam Zackrisson <saza@webrtc.org >
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#29814}
TBR=danilchap@webrtc.org ,saza@webrtc.org ,philipel@webrtc.org
Change-Id: I2eb5994d8e31e12d6cb6e9f792b691ed10d9df81
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11042
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159940
Reviewed-by: Yves Gerey <yvesg@google.com >
Commit-Queue: Yves Gerey <yvesg@google.com >
Cr-Commit-Position: refs/heads/master@{#29815}
2019-11-18 12:14:56 +00:00
49470c2ac4
Add AV1 RtpDepacketizer class
...
Implement Parse function that extracts is_first_packet_in_frame,
is_last_packet_in_frame, and frame_type fields.
Bug: webrtc:11042
Change-Id: I9360ea52ef274281b5c5e4c31955100b92155bfe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159180
Reviewed-by: Philip Eliasson <philipel@webrtc.org >
Reviewed-by: Sam Zackrisson <saza@webrtc.org >
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29814}
2019-11-18 09:39:34 +00:00
3c78ea4794
Enable FEC protection of packets with VideoTimingExtension
...
Bug: webrtc:10750
Change-Id: I532283ea51eb40cdeca5ff11be2f71da97058e41
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158899
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29727}
2019-11-07 13:46:19 +00:00
e9f663c8cb
In dependency descritpor add active decode targets bitmask field
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to follow spec draft change.
Bug: webrtc:10342
Change-Id: I8cd9f26a2061ecd62a3a7826c4086141203ee5cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159022
Reviewed-by: Sam Zackrisson <saza@webrtc.org >
Reviewed-by: Philip Eliasson <philipel@webrtc.org >
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29726}
2019-11-07 13:41:49 +00:00
bae12756da
Using unit types in TransportFeedbackAdapter.
...
Bug: webrtc:9883
Change-Id: I6d7d653079bb969fa3bc6f62fd35f2aa870edab6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158792
Reviewed-by: Erik Språng <sprang@webrtc.org >
Commit-Queue: Sebastian Jansson <srte@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29705}
2019-11-06 12:25:00 +00:00
26452ff7db
Cleanup of TransportFeedbackAdapter.
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* Removes legacy defines from rtp_rtcp_defines.
* Simplifies the feedback adaptation logic, this is achieved
by using the ability to preserve lost packets information
from the RTCP message.
* Extracts in flight data tracking to a separate helper class.
* Removes legacy fields and constructors from the PacketFeedback
structure.
* Removes the legacy GetTransportFeedbackVector method.
Apart from reducing total LOC, this prepares for moving the adaptation
to run on a TaskQueue.
Bug: webrtc:9883
Change-Id: I5ef4eace0948f119f283cd71dc2b8d0954a1449b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158781
Commit-Queue: Sebastian Jansson <srte@webrtc.org >
Reviewed-by: Erik Språng <sprang@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29674}
2019-11-01 11:55:16 +00:00
d1ea4c93d3
Update comments on Audio Level RTP header extension.
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Bug: None
Change-Id: Id9f10ea2236ba4a154cd82f2e2b05e3fa03442f3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158745
Commit-Queue: Minyue Li <minyue@webrtc.org >
Reviewed-by: Johannes Kron <kron@webrtc.org >
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29666}
2019-10-31 13:11:41 +00:00
9cdc9cc1c4
Cleanup of deprecated RTPSender code
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Also reformats RtpRtcpImpl::RtpSender by removing _ suffixes from
struct members.
Bug: webrtc:11036
Change-Id: I52cdcdff0727b62673323f64a6dc37d56ba4efbc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158532
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Commit-Queue: Erik Språng <sprang@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29642}
2019-10-29 10:08:12 +00:00
77b7529515
Reland "Use RtpSenderEgress directly instead of via RTPSender"
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This is a reland of b533010bc66a0628c8efa4d6c3dd60b8336f2736
Patchset 1 is identical to previously landed CL.
Patchset 2 contains a workaround to migrate downstream tests.
Original change's description:
> Use RtpSenderEgress directly instead of via RTPSender
>
> Bug: webrtc:11036
> Change-Id: Ida4e8bc705ae43ceb1b131114707b30d10ba8642
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158521
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
> Commit-Queue: Erik Språng <sprang@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#29626}
Bug: webrtc:11036
Change-Id: I8054169036a7f9f262308cac59f12ac8f9c73c17
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158531
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Commit-Queue: Erik Språng <sprang@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29635}
2019-10-28 17:13:30 +00:00
cff20c2615
Adds protected bitrate helper methods to RtpRtcpImpl
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Bug: webrtc:11036
Change-Id: Iac7f79b60b9f4150868e4e2c59c04c6f866011de
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158527
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Commit-Queue: Erik Språng <sprang@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29631}
2019-10-28 12:52:37 +00:00
a81e2b4510
Revert "Use RtpSenderEgress directly instead of via RTPSender"
...
This reverts commit b533010bc66a0628c8efa4d6c3dd60b8336f2736.
Reason for revert: Breaks downstream tests.
Original change's description:
> Use RtpSenderEgress directly instead of via RTPSender
>
> Bug: webrtc:11036
> Change-Id: Ida4e8bc705ae43ceb1b131114707b30d10ba8642
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158521
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
> Commit-Queue: Erik Språng <sprang@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#29626}
TBR=ilnik@webrtc.org ,sprang@webrtc.org
Change-Id: Ib3354f6907d21462a8ad0c37eb8f6e94c48af217
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11036
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158526
Reviewed-by: Erik Språng <sprang@webrtc.org >
Commit-Queue: Erik Språng <sprang@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29627}
2019-10-28 11:17:18 +00:00
b533010bc6
Use RtpSenderEgress directly instead of via RTPSender
...
Bug: webrtc:11036
Change-Id: Ida4e8bc705ae43ceb1b131114707b30d10ba8642
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158521
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Commit-Queue: Erik Språng <sprang@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29626}
2019-10-28 10:38:14 +00:00
67ac9e8ecb
Prepares RTPSender for extracting RtpSenderEgress
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The post-pacing part of the RTP sender has been moved from RTPSender
into the new RtpSenderEgress class. However, that class is not directly
used and instead a subset of method calls are passed through RTPSender.
This CL prepares for removing dependencies between RTPSender and
RtpSenderEgress. All current behavior is preserved, and unit tests are
unchanged to verify this.
For more context, see patch set 2.
Change-Id: If795f2603aeb6302ac1565d9efaea514af240dc7
Bug: webrtc:11036
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158020
Commit-Queue: Erik Språng <sprang@webrtc.org >
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29616}
2019-10-25 14:11:51 +00:00
a9229043e3
Calls OnPacketsAcknowledged on RtpRtcp instead of RTPSender directly.
...
This prepares for splitting RtpSenderEgress out of RTPSender.
For context, see:
https://webrtc-review.googlesource.com/c/src/+/158020
Bug: webrtc:11036
Change-Id: I6d385ba255ce23f4c6685a3737eeb243ce2ec6ff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158201
Reviewed-by: Sebastian Jansson <srte@webrtc.org >
Commit-Queue: Erik Språng <sprang@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29601}
2019-10-24 12:13:56 +00:00
