The number of audio channels can be configured in SDP, and can thus be
set to arbitrary values. However, the audio code has limitations that
prevent a high number of channels from working well in practice.
Bug: chromium:1265806
Change-Id: I6f6c3f68a3791bb189a614eece6bd0ed7874f252
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237807
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35359}
The number of audio channels can be configured in SDP, and can thus be
set to arbitrary values by an attacker. This CL fixes an out-of-bounds
memory access that could occur when the number of channels is set to a
large number.
Bug: chromium:1265806
Change-Id: Ic88ff6d85b978b8eb99bf03cc52457a4552e8c24
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237808
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35354}
Add implementation of RTC_DCHECK_NOTREACHED equal to the RTC_NOTREACHED.
The new macros will replace the old one when old one's usage will be
removed. The idea of the renaming to provide a clear signal that this
is debug build only macros and will be stripped in the production build.
Bug: webrtc:9065
Change-Id: I4c35d8b03e74a4b3fd1ae75dba2f9c05643101db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237802
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35348}
The nack threshold feature is unlikely to provide any value, since
reordered packets are rare. This CL also removes the factory method
from the NackTracker class, since it did not add much value.
Bug: webrtc:10178
Change-Id: Ib6bece4a2d9f95bd4298799aaa15627f5c014b61
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231953
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34993}
This is done by not adding missing packets to the NACK list if the number of samples per packet is too large.
Bug: webrtc:10178
Change-Id: If46398d6d05ea35f30d7028040d3b808559e950b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231841
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34984}
Requesting nacks in more cases allows the delay adaptation to better
predict if it's worth it to increase the jitter buffer delay to wait for
the nacked packets to arrive.
Bug: webrtc:10178
Change-Id: I46adb76c013dbb8df0b99eb3f7a398f8f21c2bf6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231648
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34970}
This is done by adding a reorder optimizer that estimates the probability of receiving reordered packets.
The optimal delay is decided by balancing the cost of increasing the delay against the probability of missing a reordered packet, resulting in a loss. This balance is decided using the `ms_per_loss_percent` parameter.
The usage and parameters can be controlled via field trial.
Bug: webrtc:10178
Change-Id: Ic484df0412af35610e74b3a6070f2bac7a926a63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231541
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34954}
Split out `RelativeArrivalDelayTracker` and `DelayOptimizer` logic.
This is in preparation for adding another `DelayOptimizer` specialized in handling reordered packets.
Bug: webrtc:10178
Change-Id: Id3c1746d91980b171fa524f9b2b71cf11fc75f64
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231224
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34938}
This allows NetEq to adapt to late reordered packets which are common when using retransmissions.
Remaining cleanup of the plumbing from WebRTC API will be done in a follow-up cl.
Bug: webrtc:10178
Change-Id: Ia9911eaafdabd3b69441dc089116d79e24f1b2b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231002
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34898}
This changes behavior slightly but results in a better delay estimate and cleaner code.
Bug: webrtc:10178
Change-Id: If150258bc1ea58149940f17c5660733ff61159c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230740
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34883}
This is a reland of 320c57b7c6bf68d8612a4f135f44b3d29e802113
Original change's description:
> red: remove special-casing of no-redundancy
>
> removes the special-casing of not sending a RED header when there is no redundant payload.
> This avoids switching back and forth between the primary and the red payload format (primarily at the start of the connection).
>
> BUG=webrtc:11640
>
> Change-Id: I8e0044bef1ed7c4168d9527645522392db2ed068
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220932
> Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34703}
Bug: webrtc:11640
Change-Id: I5e5687be575183ee16d74df4a8170e4fedad739f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228422
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34713}
removes the special-casing of not sending a RED header when there is no redundant payload.
This avoids switching back and forth between the primary and the red payload format (primarily at the start of the connection).
BUG=webrtc:11640
Change-Id: I8e0044bef1ed7c4168d9527645522392db2ed068
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220932
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34703}
fill the audio level of the recovery packets from the main packet.
While not exact, this should be close enough. Without this,
the audio level in getStats() will be filled but the audio level
in getSynchronizationSources() will be empty.
In chrome this is easy to test, the audio level graph on
https://webrtc.github.io/samples/src/content/peerconnection/audio/
will be empty all the time prior to this fix.
BUG=webrtc:11640
Change-Id: Ia1e61fd1852445239021a76d08032120a92d83aa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226840
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34635}
As part of go/coil update code search links to not point to the
"master" branch.
Bug: chromium:1226942
Change-Id: I0ae9e84ecc660f789a69fe0b226f93bbc39a8a66
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226081
Commit-Queue: Tony Herre <toprice@chromium.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34531}
This CL completes the removal of assert() and relative headers from
the codebase (excluded
//examples/objc/AppRTCMobile/third_party/SocketRocket which is in a
third_party sub-directory).
Bug: webrtc:6779
Change-Id: I93ed57168d2c0e011626873d66529488c5f484f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225546
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34528}
CL partially auto-generated with:
git grep -l "\bassert(" | grep "\.[c|h]" | \
xargs sed -i 's/\bassert(/RTC_DCHECK(/g'
And with:
git grep -l "RTC_DCHECK(false)" | \
xargs sed -i 's/RTC_DCHECK(false)/RTC_NOTREACHED()/g'
With some manual changes to include "rtc_base/checks.h" where
needed.
A follow-up CL will remove assert() from Obj-C code as well
and remove the #include of <assert.h>.
The choice to replace with RTC_DCHECK is because assert()
is because RTC_DCHECK has similar behavior as assert()
based on NDEBUG.
This CL also contains manual changes to switch from
basic RTC_DCHECK to other (preferred) versions like
RTC_DCHECK_GT (and similar).
Bug: webrtc:6779
Change-Id: I00bed8886e03d685a2f42324e34aef2c9b7a63b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/224846
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34442}
If timestamp_scaler_ is used, then rtp_header.timestamp, passed to UpdateLastDecodedPacket, will advance at a different rate than the scaled timestamp packet->timestamp, passed to UpdateLastDecodedPacket.
NackTracker::EstimateTimestamp uses timestamp_last_received_rtp_, and NackTracker::TimeToPlay uses timestamp_last_decoded_rtp_.
This difference causes TimeToPlay to continuously increase to huge values, so that every missing packet will be returned from GetNackList, even if RTT > real time to play.
Change-Id: Ie6ca347972edea98a202c9cdd26c6ab3f45a73c4
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222841
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34361}
