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}
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}
This is a change from the previous 100Hz frequency.
Also changing the locks slightly in AcmReceiver so that grabbing the
neteq lock right after we've let it go, isn't necessary inside of
AcmReceiver::GetAudio and also to avoid grabbing the neteq lock while
holding the AcmReceiver lock.
Bug: webrtc:12868
Change-Id: If6ee35f3dca20eb5bdbc615123aa099ccecf57c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221371
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34258}
Before this CL, timestamps of received packets were rounded
to the nearest millisecond and stored as int64_t. Due to the
rounding it sometimes happened that timestamps later in the
pipeline that are not rounded seem to occur even before the
video frame was received.
Change-Id: I92d8f3540b23baae2d4a1dc6a7cb3f58bcdaad18
Bug: webrtc:12722
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216398
Reviewed-by: Chen Xing <chxg@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33916}
This is essentially replacing `new rtc::RefCountedObject` with
`rtc::make_ref_counted` in many files. In a couple of places I
made minor tweaks to make things compile such as adding parenthesis
when they were missing.
Bug: webrtc:12701
Change-Id: I3828dbf3ee0eb0232f3a47067474484ac2f4aed2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215973
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33852}
Many things are omitted in this doc and it can definitely be improved,
but I hope it captures the most important parts.
Bug: webrtc:12568
Change-Id: I13097d633ca19cecc9dd43bdb777b0ca48f151dd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215142
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33724}
This is required to be able to update the Opus version and will be
rolled back after.
Bug: webrtc:12518
Change-Id: Icc649039787db44bd55a0dc8e5ba4089df3a9566
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209363
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33375}
Audio interruption metric is not implemented for codecs doing their own PLC.
R=ivoc@webrtc.org, jakobi@webrtc.org
Bug: b/177523033 webrtc:12456
Change-Id: I0aca6fa5c0ff617e76ee1e4ed8d95703c7097223
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206561
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Pablo Barrera González <barrerap@google.com>
Cr-Commit-Position: refs/heads/master@{#33229}
Finding the array element with the largest argmax is a fairly common
operation, so it makes sense to have a Neon optimized version. The
implementation is done by first finding both the min and max value, and
then returning whichever has the largest argmax.
Bug: chromium:12355
Change-Id: I088bd4f7d469b2424a7265de10fffb42764567a1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201622
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33052}
The abs max of a 16 bit integer cannot be represented as a 16 bit integer, because abs(-2^16) is too large. To work around this, we can instead use the index of the max element, convert it to a 32-bit int and then take the absolute value.
Bug: chromium:1158070, chromium:1146835, chromium:1161837
Change-Id: If56177c55ec62b4bd578986a5deae38a91bbc821
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/198123
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32950}
I was not able to reproduce chromium:1146676 locally, so the change in merge.cc is a speculative fix.
Bug: chromium:1146835, chromium:1146676, chromium:1137226
Change-Id: I14472ba5b41e58b2d5f27d9833249c14505af18f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/194264
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32759}
Instead of flushing all packets, it makes sense to flush down to the target level instead. This CL also initiates a flush when the packet buffer is a multiple of the target level, instead of waiting until it is completely full.
Bug: webrtc:12201
Change-Id: I8775147624536824eb88752f6e8ffe57ec6199cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/193941
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32701}
Add a new field trial with more flexible parsing and new options:
- Resample packet delays to only update histogram with maximum observed
delay every X ms.
- Setting the maximum history size (in ms) used for calculating the
relative arrival delay.
Legacy field trial used for configuration is maintained.
Bug: webrtc:10333
Change-Id: I35b004f5d8209c85b33cb49def3816db51650946
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/192789
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32591}
The variables that are used to track the amount of preemptive expansion
and acceleration are not initialized before being passed to their
respective functions. However, these function can fail in certain cases,
and when they do the uninitialized memory will pollute the NetEq statistics.
Bug: chromium:1140376
Change-Id: I004fbaaf8d24de01dd1997fb73bdf93ca88ceaaf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191480
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32544}
