The problem was fixed by implementing the methid PacketDuration() in
AudioDecoderG722StereoImpl, which catches the issue in
AudioDecoder::Decode().
Bug: chromium:1280851
Change-Id: I31f974b9999f3c1c62b0e5dc39bb3e56a9a9388d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251842
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36034}
This CL removes even more top-level const from parameters in function
declarations. This change is safe because top-level const in function
declarations (not function definitions) are ignored by the compiler
and so change is just a no-op cleanup.
Bug: webrtc:13610
Change-Id: Icf6868c27b1fdb9d9915b3a7020eb34bdcf07a09
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249989
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35866}
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}
We should avoid creating codecs with invalid parameters, since this can
expose security issues. For many codecs the IsOk() method to check the
codec config is only called in DCHECKs. This CL ensures IsOk() is always
called, also in non-debug builds.
Bug: chromium:1265806
Change-Id: Ibd3c6c65d3bb547cd2603e11808ac40ac693a8b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238801
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35422}
This reverts commit 4cbfe4192cd5b8289f7896ce14e0bd8c4ae41a97.
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. 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}
