Commit graph

120 commits

Author SHA1 Message Date
Oskar Sundbom
8e07c134ab Optional: Use nullopt and implicit construction in /video
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.

This CL was uploaded by git cl split.

Bug: None
Change-Id: Ie622c215e06956d8d5629733c76f531b7af45012
Reviewed-on: https://webrtc-review.googlesource.com/23568
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21535}
2018-01-09 15:14:10 +00:00
Åsa Persson
aa329e7cc3 Reland: googBandwidthLimitedResolution stat is not always set depending on configuration.
TBR=brandtr@webrtc.org,stefan@webrtc.org

Currently |bw_resolutions_disabled| is set per VP8EncoderImpl instance and reported via
OnEncodedImage callback.

Instead move logic to SendStatisticsProxy to determine if resolution is bw limited or not based
on info that is reported to SendStatisticsProxy::OnEncodedImage.

Bug: webrtc:8643
Change-Id: I553cea30dcda34b753b5224f15094a1b7b70a750
Reviewed-on: https://webrtc-review.googlesource.com/31460
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Original-Commit-Position: refs/heads/master@{#21249}
Reviewed-on: https://webrtc-review.googlesource.com/33360
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21319}
2017-12-18 11:20:13 +00:00
Guido Urdaneta
62e9ebe589 Revert "googBandwidthLimitedResolution stat is not always set depending on configuration."
This reverts commit 59283e4c66.

Reason for revert: This CL is preventing rolls into Chromium because it fails to compile with MSVC.

Sample error log:

[13258/43857] CXX obj/third_party/webrtc/video/video/send_statistics_proxy.obj
FAILED: obj/third_party/webrtc/video/video/send_statistics_proxy.obj 
ninja -t msvc -e environment.x64 -- E:\b\c\goma_client/gomacc.exe "e:\b\c\win_toolchain\vs_files\a9e1098bba66d2acccc377d5ee81265910f29272\vc\tools\msvc\14.11.25503\bin\hostx64\x64/cl.exe" /nologo /showIncludes  @obj/third_party/webrtc/video/video/send_statistics_proxy.obj.rsp /c ../../third_party/webrtc/video/send_statistics_proxy.cc /Foobj/third_party/webrtc/video/video/send_statistics_proxy.obj /Fd"obj/third_party/webrtc/video/video_cc.pdb"
../../third_party/webrtc/video/send_statistics_proxy.cc(217): error C2220: warning treated as error - no 'object' file generated
../../third_party/webrtc/video/send_statistics_proxy.cc(217): warning C4267: 'initializing': conversion from 'size_t' to 'int', possible loss of data
../../third_party/webrtc/video/send_statistics_proxy.cc(632): warning C4267: '=': conversion from 'size_t' to 'uint32_t', possible loss of data


Original change's description:
> googBandwidthLimitedResolution stat is not always set depending on configuration.
> 
> Currently |bw_resolutions_disabled| is set per VP8EncoderImpl instance and reported via
> OnEncodedImage callback.
> 
> Instead move logic to SendStatisticsProxy to determine if resolution is bw limited or not based
> on info that is reported to SendStatisticsProxy::OnEncodedImage.
> 
> Bug: webrtc:8643
> Change-Id: I6c148e3507a0f04a793775b9f84ce54028b64d0f
> Reviewed-on: https://webrtc-review.googlesource.com/31460
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Åsa Persson <asapersson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21249}

TBR=brandtr@webrtc.org,asapersson@webrtc.org,stefan@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:8643
Change-Id: Ib9ef55b8894ea72236a5dc1e9a839adecd401afb
Reviewed-on: https://webrtc-review.googlesource.com/33100
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21284}
2017-12-14 17:35:53 +00:00
Åsa Persson
59283e4c66 googBandwidthLimitedResolution stat is not always set depending on configuration.
Currently |bw_resolutions_disabled| is set per VP8EncoderImpl instance and reported via
OnEncodedImage callback.

Instead move logic to SendStatisticsProxy to determine if resolution is bw limited or not based
on info that is reported to SendStatisticsProxy::OnEncodedImage.

Bug: webrtc:8643
Change-Id: I6c148e3507a0f04a793775b9f84ce54028b64d0f
Reviewed-on: https://webrtc-review.googlesource.com/31460
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21249}
2017-12-13 14:32:21 +00:00
Åsa Persson
c3ed630560 Add stats googHasEnteredLowResolution.
Indicates if the forced sw fallback has had an effect (or would have had an effect if it had been
enabled).


Bug: webrtc:6634
Change-Id: I574b9001a2fae650fb894a1caa0d0f84257658e3
Reviewed-on: https://webrtc-review.googlesource.com/23300
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20729}
2017-11-17 13:02:07 +00:00
Åsa Persson
45bbc8ac19 Change forced software encoder fallback for VP8 to be only based on resolution and not bitrate.
Switches from VP8 HW to VP8 SW for resolutions <= max_pixels. 

|<- min_pixels  VP8 SW  max_pixels ->|  VP8 HW  |

Bug: webrtc:6634
Change-Id: Ib324df2b8418659c29d999259c0ed47448310696
Reviewed-on: https://webrtc-review.googlesource.com/7362
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20646}
2017-11-13 10:58:42 +00:00
Mirko Bonadei
675513b96a Stop using LOG macros in favor of RTC_ prefixed macros.
This CL has been generated with the following script:

for m in PLOG \
  LOG_TAG \
  LOG_GLEM \
  LOG_GLE_EX \
  LOG_GLE \
  LAST_SYSTEM_ERROR \
  LOG_ERRNO_EX \
  LOG_ERRNO \
  LOG_ERR_EX \
  LOG_ERR \
  LOG_V \
  LOG_F \
  LOG_T_F \
  LOG_E \
  LOG_T \
  LOG_CHECK_LEVEL_V \
  LOG_CHECK_LEVEL \
  LOG
do
  git grep -l $m | xargs sed -i "s,\b$m\b,RTC_$m,g"
done
git checkout rtc_base/logging.h
git cl format

Bug: webrtc:8452
Change-Id: I1a53ef3e0a5ef6e244e62b2e012b864914784600
Reviewed-on: https://webrtc-review.googlesource.com/21325
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20617}
2017-11-09 11:56:32 +00:00
Ilya Nikolaevskiy
d79314f9f9 Reland "Add fine grained dropped video frames counters on sending side"
Add fine grained dropped video frames counters on sending side

4 new counters added to SendStatisticsProxy and reported to UMA and logs.

Bug: webrtc:8355
Change-Id: I1f9bdfea9cbf17cf38b3cb2f55d406ffdb06614f
Reviewed-on: https://webrtc-review.googlesource.com/14580
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20421}
2017-10-25 09:32:15 +00:00
Ilya Nikolaevskiy
1c1a6815ae Revert "Add fine grained dropped video frames counters on sending side"
This reverts commit 4b1a363e4c.

Reason for revert: Breaks dependent android projects.

Original change's description:
> Add fine grained dropped video frames counters on sending side
> 
> 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}

TBR=deadbeef@webrtc.org,ilnik@webrtc.org,sprang@webrtc.org,stefan@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:8355
Change-Id: I59b02f4eb77abad7ff1fbcbfa61844918c95d723
Reviewed-on: https://webrtc-review.googlesource.com/14500
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20378}
2017-10-21 09:23:54 +00:00
Åsa Persson
d29b54c93a Set start time for encoded framerate tracker on first incoming frame (instead of
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}
2017-10-20 10:20:43 +00:00
Ilya Nikolaevskiy
4b1a363e4c Add fine grained dropped video frames counters on sending side
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}
2017-10-19 10:37:12 +00:00
Åsa Persson
0122e8443b Reland "Remove sent framerate and bitrate calculations from MediaOptimization."
TBR=sprang@webrtc.org

This is a reland of af721b72cc
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}
2017-10-16 12:43:07 +00:00
Åsa Persson
ca0ed63c19 Revert "Remove sent framerate and bitrate calculations from MediaOptimization."
This reverts commit af721b72cc.

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}
2017-10-11 12:59:15 +00:00
Åsa Persson
18945c35c2 Revert "Reduce max possible size of map that holds encoded frame info."
This reverts commit 2ff7ecfceb.

Reason for revert: <INSERT REASONING HERE>

Original change's description:
> Reduce max possible size of map that holds encoded frame info.
> 
> Bug: webrtc:8375
> Change-Id: Idc57e68dc44fd73e5c0aa85d82c1e3659d8ea292
> Reviewed-on: https://webrtc-review.googlesource.com/8301
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Åsa Persson <asapersson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20232}

TBR=brandtr@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org

Change-Id: I1dcb7ab588e5ab3eb79bec2c39f615480e31c3bb
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8375
Reviewed-on: https://webrtc-review.googlesource.com/8460
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20247}
2017-10-11 12:58:11 +00:00
Åsa Persson
2ff7ecfceb Reduce max possible size of map that holds encoded frame info.
Bug: webrtc:8375
Change-Id: Idc57e68dc44fd73e5c0aa85d82c1e3659d8ea292
Reviewed-on: https://webrtc-review.googlesource.com/8301
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20232}
2017-10-11 07:34:30 +00:00
Åsa Persson
af721b72cc 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}
2017-10-10 15:36:08 +00:00
asapersson
8d75ac7e3f Add stats for forced software encoder fallback for VP8.
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}
2017-09-15 13:41:15 +00:00
Mirko Bonadei
7120742701 Adding NOLINT for typedefs.h and common_types.h
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}
2017-09-15 13:03:51 +00:00
Mirko Bonadei
92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
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}
2017-09-15 05:02:56 +00:00
Mirko Bonadei
bb547203bf Moving src/webrtc into src/.
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}
2017-09-15 04:25:06 +00:00
Renamed from webrtc/video/send_statistics_proxy.cc (Browse further)