Commit graph

10 commits

Author SHA1 Message Date
Florent Castelli
8037fc6ffa Migrate absl::optional to std::optional
Bug: webrtc:342905193
No-Try: True
Change-Id: Icc968be43b8830038ea9a1f5f604307220457807
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361021
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42911}
2024-09-02 12:16:47 +00:00
Danil Chapovalov
3732b84c17 Pass Clock and RtcEventLog as Environment into AudioReceiveStream
To make Environment available for creating AudioDecoders to use propagated field trials

Bug: webrtc:356878416
Change-Id: I5dc4a3514d1182db6a7a5aa770b87daba529a5c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357442
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42719}
2024-08-02 11:58:23 +00:00
Per K
61fff586b1 Split out time_util to separate target ntp_time_util
Split out time_util.h and cc from target rtp_rtcp to its own target.
This is to avoid possible circular dependencies and not having all targets using them to depend on the full RtpRcp module.


Bug: webrtc:343076000
Change-Id: I7b3c84456b17f1920f71afdd5a644d27e28caed2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352480
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42392}
2024-05-28 13:31:00 +00:00
Florent Castelli
f4673f97ed Move webrtc::AudioDeviceModule include to api/ folder
Bug: webrtc:15874
Change-Id: I5bdb19d5e710838b41e6ca283d406c9f1f21286b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/348060
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42137}
2024-04-22 08:56:31 +00:00
Tony Herre
9c6874607a Consolidate encoded transform mocks into api/test/
Includes removing the duplicate MockTransformableAudioFrame definition
in test/ in favour of the existing one in api/test/

Bug: webrtc:15802
Change-Id: Ib5f86b8b2095dd4e580cd9ff0038134f8a43cd93
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336340
Auto-Submit: Tony Herre <herre@google.com>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41622}
2024-01-26 12:46:34 +00:00
Tony Herre
f921d25320 Remove DCHECK on setting audio rcvr encoded transform twice
Failing a DCHECK on a ChannelReceiver having its encoded transform set
more than once contradicts the comment above - this can happen when
reconfiguring a channel, eg as in the web platform test
webrtc/recvonly-transceiver-can-become-sendrecv.https.html.

It was added after the original code by a different author, and indeed
the video side doesn't have such a check.

Bug: chromium:1502781
Change-Id: Id36e52601da34ebc194ff058e4672046379f576e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328560
Commit-Queue: Tony Herre <herre@google.com>
Auto-Submit: Tony Herre <herre@google.com>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41246}
2023-11-27 10:53:24 +00:00
Harald Alvestrand
d43af9172b Remove internal overrides using old SendRtp and SendRtcp interfaces.
This CL takes away all usages except for Android code.

Low-Coverage-Reason: Refactoring old code
Bug: webrtc:15410
Change-Id: I66bed6a4a2787b4177a82e599b48623ca67cd235
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315940
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40554}
2023-08-15 13:20:21 +00:00
Harald Alvestrand
95d12adf37 Create unit test for the population of capture_start_ntp_time
This verifies that receiving two RTCP SR packets is enough to get
a defined capture start time stat.

Bug: webrtc:13931
Change-Id: Ib5f7c2954eab6500917f25c44f523d3aedae5e94
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291520
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39261}
2023-02-06 14:00:39 +00:00
Harald Alvestrand
ba846ccf24 Add a test that shows when channel_receive fires RR
This seems to happen 2.5 seconds after initialization.
Written as part of debugging a different issue.

Bug: webrtc:13931
Change-Id: I3686cdbc39284505a437ebc0bfd8c74c483624c9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291704
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39245}
2023-02-01 16:38:38 +00:00
Harald Alvestrand
e15b9ff408 Add a basic unittest for webrtc::voe::ChannelReceive
This CL adds an unittest that a ChannelReceive can be constructed
and destroyed without crashing. It is a basis for further testing.

Lack of unit test was discovered while pursuing bug mentioned below.

Bug: webrtc:13931
Change-Id: Iddb110f2df25e3806c74a5d00bbfab6d6d8e267f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291338
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39200}
2023-01-25 20:06:26 +00:00