To make testing easier all of PacketQueues functions have been made virtual,
and PacketQueue2 now inherits PacketQueue. This change was made to minimize
changes in PacedSender.
Bug: webrtc:8287, webrtc:8288
Change-Id: I2593340e7cc7da617370b0a33e7b9deeb46d9487
Reviewed-on: https://webrtc-review.googlesource.com/9380
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20385}
when first key frame is encoded) to avoid a too high initial estimate.
Bug: webrtc:8375
Change-Id: I404e394d8f2ac648170dd3828065435a4d2c6147
Reviewed-on: https://webrtc-review.googlesource.com/14061
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20370}
4 new counters added to SendStatisticsProxy and reported to UMA and logs.
Bug: webrtc:8355
Change-Id: Idf9b8dfc295c92821e058a97cb3894dc6a446082
Reviewed-on: https://webrtc-review.googlesource.com/12260
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20347}
Now some metrics would be aggregated over 1s window instead of 10s.
Bug: webrtc:8402
Change-Id: If0b5a70257c767a1741b451585f3da501b903374
Reviewed-on: https://webrtc-review.googlesource.com/11400
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20316}
TBR=sprang@webrtc.org
This is a reland of af721b72cc1bdc5d945629ad78fbea701b6f82b9
Original change's description:
> Remove sent framerate and bitrate calculations from MediaOptimization.
>
> Add RateTracker for sent framerate and bitrate in SendStatisticsProxy.
>
> Store sent frame info in map to solve potential issue where sent framerate statistics could be
> incorrect.
>
> Bug: webrtc:8375
> Change-Id: I4a6e3956013438a711b8c2e73a8cd90c52dd1210
> Reviewed-on: https://webrtc-review.googlesource.com/7880
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Åsa Persson <asapersson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20225}
Bug: webrtc:8375
Change-Id: I06ea90ae8646ba11ddd8ddceb82ea82d75ae2109
Reviewed-on: https://webrtc-review.googlesource.com/11320
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20308}
Track IDs are assigned by application during track creation.
Track IDs are used by custom bitrate allocation strategies to identify tracks.
Track ID can be empty, in that case bitrate allocation strategies will not be able to handle
these tracks specifically and will handle them as a default.
Bug: webrtc:8243
Change-Id: I89987e33328320bfd0539ad532342df6da144c98
Reviewed-on: https://webrtc-review.googlesource.com/4820
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20285}
This should be a separate utility class, rather than internal thing in
ReceiveStatisticsProxy.
Bug: webrtc:8347
Change-Id: I9c8238e625999dba5776d6038c819732d07e9656
Reviewed-on: https://webrtc-review.googlesource.com/7609
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20267}
This reverts commit af721b72cc1bdc5d945629ad78fbea701b6f82b9.
Reason for revert: <INSERT REASONING HERE>
Original change's description:
> Remove sent framerate and bitrate calculations from MediaOptimization.
>
> Add RateTracker for sent framerate and bitrate in SendStatisticsProxy.
>
> Store sent frame info in map to solve potential issue where sent framerate statistics could be
> incorrect.
>
> Bug: webrtc:8375
> Change-Id: I4a6e3956013438a711b8c2e73a8cd90c52dd1210
> Reviewed-on: https://webrtc-review.googlesource.com/7880
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Åsa Persson <asapersson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20225}
TBR=asapersson@webrtc.org,sprang@webrtc.org
Change-Id: Ic914f03ff7065ac410ae06b6f82b56a935399b66
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8375
Reviewed-on: https://webrtc-review.googlesource.com/8480
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20248}
Remove REMB accessor as used for debug checks only.
Merge SetRembData and SetRembStatus(true) eliminating
state 'remb can be send, but no data available yet'
Bug: None
Change-Id: I4c1c19435657e5cde02a17de90ec6de9f00b7daf
Reviewed-on: https://webrtc-review.googlesource.com/7983
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20240}
Add RateTracker for sent framerate and bitrate in SendStatisticsProxy.
Store sent frame info in map to solve potential issue where sent framerate statistics could be
incorrect.
Bug: webrtc:8375
Change-Id: I4a6e3956013438a711b8c2e73a8cd90c52dd1210
Reviewed-on: https://webrtc-review.googlesource.com/7880
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20225}
This CL is the same CL we had at https://codereview.webrtc.org/3014543002/.
Since we cannot land it with Rietveld anymore let's move the discussion
to Gerrit.
BUG=webrtc:7641
CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal
Change-Id: I5662bec318544b07f476c12ecada997d726e7361
Reviewed-on: https://webrtc-review.googlesource.com/7981
Reviewed-by: Henrik Kjellander <kjellander@google.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20224}
The call to |interframe_delay_max_moving_.Add()| below depends on |now|
non decreasing in consequtive calls. However, if two threads are
competing for the lock it may happen that current thread calculates |now|
before the other thread, yet it will get the lock later. This will result
in decreasing local time in consecutive calls and trigger a DCHECK.
The same also applies to |timing_frame_info_counter_|.
Bug: none
Change-Id: I3376d88d4448c2c105e9227a445b11cd6ba8d341
Reviewed-on: https://webrtc-review.googlesource.com/7861
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20217}
Only allow gaps in picture id for key frames.
When a VideoSendStream is destroyed, frames in the queue not yet sent are lost. The recreated stream
should start with a key frame.
Also enable PictureIdIncreasingAfterStreamCountChangeSimulcastEncoderAdapter if forced fallback is
enabled. In this case, the picture id is set in the PayloadRouter and the sequence should be
continuous.
Bug: none
Change-Id: If7987166c86d6a8edbe5e479701f7f04c49cd89c
Reviewed-on: https://webrtc-review.googlesource.com/7363
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20216}
The SW and HW encoder have separate picture id sequences.
Set picture id to not cause sequence discontinuties at encoder changes.
Bug: webrtc:6634
Change-Id: Ie47168791399303d88cbec3ef6ae8ef8c16ced30
Reviewed-on: https://webrtc-review.googlesource.com/5481
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20188}
Histogram based percentile counter is added in ReceiveStatisticsProxy.
New 95th percentile metric is reported in the same way as interframe
delay.
Bug: webrtc:8347
Change-Id: I5e476cbb6361dd341cdb97c37d883c3923e5f611
Reviewed-on: https://webrtc-review.googlesource.com/6880
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20184}
This is needed in the general case, now that we aim to support codecs
other than those built-in to WebRTC.
BUG=webrtc:8159
Change-Id: I40a41252bf69ad5d4d0208e3c1e8918da7394706
Reviewed-on: https://webrtc-review.googlesource.com/5380
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20136}
We need to support two modes of writing to the output:
1. Current way - the application lets lets WebRTC know which file to write to, and WebRTC is then in charge of the writing.
2. New way - the application would receive indications from WebRTC about (encoded) RTC events, and would itself be in charge of processing them (be it writing it to a file, uploading it somewhere, etc.).
We achieve this by creating an interface for output - RtcEventLogOutput. By providing an instance of the subclass, RtcEventLogOutputFile, the old behavior is achieved. The subclass of the new behavior is to be added by a later CL.
TBR=stefan@webrtc.org
Bug: webrtc:8111
Change-Id: I9c50521a7f7144d86d8353a65995795862e19c44
Reviewed-on: https://webrtc-review.googlesource.com/2686
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20135}
The two timestamps, rtp time and corresponding system time, are always
used together, for audio/video sync. The new method reads both
timestamps, without releasing a lock in between. Ensures that the
caller gets values corresponding to the same packet.
Bug: webrtc:7135
Change-Id: I25bdcbe9ad620016bfad39841b339c266efade14
Reviewed-on: https://webrtc-review.googlesource.com/4062
Commit-Queue: Niels Moller <nisse@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20120}
RtcEventLogImpl no longer hard-codes the way encoding is done. It now relies on RtcEventEncoder for it. This gives two benefits:
1. We can decide between the current encoding and the new encoding (which is still WIP) without code duplication (no need for RtcEventLogImplNew).
2. Encoding is done only when the event needs to be written to a file. This both avoids unnecessary encoding of events which don't end up getting written to a file, as well as is useful for the new, delta-based encoding, which is stateful.
BUG=webrtc:8111
Change-Id: I9517132e5f96b8059002a66fde8d42d3a678c3bb
Reviewed-on: https://webrtc-review.googlesource.com/1365
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20118}
Because RtcEventLog is created and stopped in different threads,
SequencedTaskChecker causes failure at the end of a test.
Bug: none
Change-Id: Ibaec3162eedebd180b101ec46a171efee5fe667e
Reviewed-on: https://webrtc-review.googlesource.com/5401
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20078}
Stop using PayloadUnion's public member variables, since a future CL
will make them private.
BUG=webrtc:8159
Change-Id: Ia3dada56be7ef00ed80f3733209b18c178a36561
Reviewed-on: https://webrtc-review.googlesource.com/4380
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20027}
On android, the flag to store the frame with the worst PSNR was called
'--test_artifacts_dir'.
I think test artifacts is a better name.
TBR=sprang@webrtc.org
Bug: chromium:745469
Change-Id: I358ea2985a1df2da12b81df173d74ac193556a49
Reviewed-on: https://webrtc-review.googlesource.com/4080
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20000}
This is a workaround until downstream projects have been fixed.
BUG=webrtc:8220
Review-Url: https://codereview.webrtc.org/3017613002
Cr-Commit-Position: refs/heads/master@{#19966}
Replaced with scalars ulpfec_payload_type and red_payload_type.
In particular, ulpfec.red_rtx_payload_type, which duplicated info in
rtx_associated_payload_types, is deleted. This is a followup to cl
https://codereview.webrtc.org/3012963002.
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/3019453002
Cr-Commit-Position: refs/heads/master@{#19965}
Reason for revert:
Fixes has landed.
Original issue's description:
> Revert of Fix the video buffer size should take rtt into consideration (patchset #3 id:40001 of https://codereview.chromium.org/2980413002/ )
>
> Reason for revert:
> We are not certain this is the behavior we want.
>
> Original issue's description:
> > Fix the video buffer size should take rtt into consideration
> >
> > BUG=webrtc:8010
> >
> > Review-Url: https://codereview.webrtc.org/2980413002
> > Cr-Commit-Position: refs/heads/master@{#19285}
> > Committed: f1e08d0b58
>
> TBR=sprang@webrtc.org,gustavogb@gmail.com
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:8010
>
> Review-Url: https://codereview.webrtc.org/3002033002
> Cr-Commit-Position: refs/heads/master@{#19442}
> Committed: bdbc8895f3TBR=sprang@webrtc.org,gustavogb@gmail.com
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:8010
Review-Url: https://codereview.webrtc.org/3016633002
Cr-Commit-Position: refs/heads/master@{#19944}
* Add audio_ prefix to CallTest::{en,de}coder_factory_.
* Let VideoQualityTest only instantiate encoders using encoder factories.
* Add HW encoder factories to VideoQualityTest.
* Add full stack tests:
- sqcif7 at 30 kbps: MediaCodec and libvpx.
- 240p10 at 100 kbps: MediaCodec, libvpx, and MediaCodec+libvpx.
BUG=webrtc:8219
Review-Url: https://codereview.webrtc.org/3005253002
Cr-Commit-Position: refs/heads/master@{#19923}
It's in the way of a refactoring.
Also change PayloadTypeToPayload---the method all callers can use instead---to return Optional<Payload> instead of const Payload* (for thread safety reasons: an object that protects itself with a mutex shouldn't be handing out pointers to parts of itself).
BUG=webrtc:8159
Change-Id: I7ef0d545077ffdea016b309f2165e3c4955a2928
Reviewed-on: https://webrtc-review.googlesource.com/2360
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19917}
Stats added for number of forced SW fallback changes per minute and percentage of time fallback is enabled for sent video streams:
- "WebRTC.Video.Encoder.ForcedSwFallbackChangesPerMinute.Vp8"
- "WebRTC.Video.Encoder.ForcedSwFallbackTimeInPercent.Vp8"
BUG=webrtc:6634
Review-Url: https://codereview.webrtc.org/3012863002
Cr-Commit-Position: refs/heads/master@{#19862}
Now that we have moved WebRTC from src/webrtc to src/, common_types.h
and typedefs.h are triggering a cpplint error.
The cpplint complaint is:
Include the directory when naming .h files [build/include] [4]
This CL disables the error but we have to remove these two headers
from the root directory.
NOPRESUBMIT=true
Bug: webrtc:5876
Change-Id: I08e1b69aadcc4b28ab83bf25e3819d135d41d333
Reviewed-on: https://webrtc-review.googlesource.com/1577
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19859}
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
In order to eliminate the WebRTC Subtree mirror in Chromium,
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
