The same information can be found in `AudioFrame.packet_infos_`.
Bug: none
Change-Id: Ib63bc41ffb896677a445d875afce0a98acea6999
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265161
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37153}
When target_os is set to "fuchsia":
BUILD: suppress Wundef flag
DEPS: download the Fuchsia SDK
audio_encoding: add header include
video_capture: video_capture_factory is not yet implemented for Fuchsia
so we add a null capture factory when building for Fuchsia.
Bug: webrtc:14061
Change-Id: Id6ca7418859c85293a0a5e2a8427807ee039db2c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262200
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37030}
It replaces the relative arrival delay tracker which is equivalent.
This results in a slight bit-exactness change but nothing that should affect quality.
Bug: webrtc:13322
Change-Id: I6ed5d6fdfa724859122928a8838acce27ac2e5d0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263380
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37004}
This is to more accurately simulate Opus CNG.
Bug: None
Change-Id: I3244d88e1f7410190551b6fa24cdd08599b5771e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262661
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36913}
A packet arrival history is used to store the timing of incoming packets and tracks the earliest and latest packets by taking the difference between rtp timestamp and arrival time. The history is windowed to 2 seconds by default. The packet arrival history will replace the relative arrival delay tracker in a follow up cl.
The playout delay is estimated by taking the difference between the current playout timestamp and the earliest packet arrival in the history. This method works better when DTX is used compared to the buffer level filter that it replaces.
The threshold for acceleration is changed to be the maximum of the target delay and the maximum packet arrival delay in the history. This prevents any acceleration immediately after an underrun and gives some time to adapt the target delay to new network conditions.
The logic when to decode the next packet after a packet loss is also changed to do concealment for the full loss duration unless the delay is too high.
The new mode is default disabled and can be enabled using a field trial.
Bug: webrtc:13322,webrtc:13966
Change-Id: Idfa0020584591261475b9ca350cc7c6531de9911
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259820
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36899}
Currently only implemented for codec internal CNG (Opus).
Bug: webrtc:13322
Change-Id: I00622f2967f066dba64a792e26081038ae0cb0d9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259200
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36590}
This is a slight change in behavior that fixes a bug where all expansions are not counted due to more than 10ms available in the sync buffer, which can happen after repeated expansions.
The counter should also be updated when in muted mode.
Bug: webrtc:13322
Change-Id: I067689ee251d3d1ae990a27cdd271f718b0d6f2f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257360
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36483}
This is used until the first RTT measurement becomes available.
100ms is a reasonable default and used in other places.
Bug: webrtc:10178
Change-Id: I14f530504a4866fbe75f025dfe184fd6e296b75e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256861
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36384}
Recent Clang versions have enhanced -Wunused-but-set-variable which now
warns about this.
Bug: chromium:1309955
Change-Id: Ie70df85f5a6d2cbabb4e10960bfd926ff7bf32cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257162
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Hans Wennborg <hans@chromium.org>
Cr-Commit-Position: refs/heads/main@{#36381}
This reverts commit a37384899b.
Reason for revert: It breaks some downstream tests, let's reland on Monday adding a fix for them as well (Mac M1 is still broken).
Original change's description:
> Update NetEq bitexactness tests to only run on Linux.
>
> Running bitexactness tests only on Linux makes it significantly easier to
> update them, while still giving many of the same benefits.
>
> Bug: webrtc:12518, b/216736217
> Change-Id: I7f3c9a27c0fc14b7ee0e83aede2e7702cfa79141
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249787
> Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
> Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
> Reviewed-by: Artem Titov <titovartem@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35829}
TBR=mbonadei@webrtc.org,ivoc@webrtc.org,titovartem@webrtc.org,jakobi@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
Change-Id: I53e3d18d53949eb9dded9ce29de99e091a480705
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:12518, b/216736217
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249980
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35836}
Running bitexactness tests only on Linux makes it significantly easier to
update them, while still giving many of the same benefits.
Bug: webrtc:12518, b/216736217
Change-Id: I7f3c9a27c0fc14b7ee0e83aede2e7702cfa79141
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249787
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35829}
This is a safe cleanup change since top-level const applied to
parameters in function declarations (that are not also
definitions) are ignored by the compiler. Hence, such changes do
not change the type of the declared functions and are simply
no-ops.
Bug: webrtc:13610
Change-Id: Ibafb92c45119a6d8bdb6f9109aa8dad6385163a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249086
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35802}
If the number of samples does not fit in an AudioFrame, we should return
kSampleUnderrun to avoid crashes further downstream.
Bug: chromium:1265806
Change-Id: Ie94e1de53810167fd9b52ade72b3cb669a2a4f06
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238666
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35459}
This reverts commit 4cbfe4192c.
Reason for revert: The fix in this CL is ineffective. A better one has been created here: https://webrtc-review.googlesource.com/c/src/+/238666
Original change's description:
> Fix out-of-bounds memory access due to large number of audio channels.
>
> 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}
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: chromium:1265806
Change-Id: If695ed92f831c2a9631efdf47f1568f5a15c1841
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238803
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35413}
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}
Follow up for https://webrtc-review.googlesource.com/c/src/+/232061/5. Updated mac M1 tests that was missed because they are not part of CQ
Bug: b/199885455
Change-Id: I77618ac2869ba601f322857f4391b63220d20252
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232220
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35021}
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}
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}
