Commit Graph

19626 Commits

Author SHA1 Message Date
ba68aabb06 Fix of integer overflow in WebRtcAecm_ProcessBlock / ApmTest.Process
This CL includes the patch from oprypin@webrtc.org, which is also applied
to the MIPS code (also affected), and the protobuf for ApmTest.Process
(audio_processing_unittest.cc), which used when WEBRTC_AUDIOPROC_FIXED_PROFILE
is set.

This change has been tested on mobile platforms.

Bug: webrtc:8200
Change-Id: Ic50a5ab57c16551397756b1fb473e1067b8e7ece
Reviewed-on: https://webrtc-review.googlesource.com/10811
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20394}
2017-10-23 14:25:37 +00:00
f92d871f4f Let ChanneOwner use scoped_refptr and RefCountedBase.
Eliminates manual reference counting logic, and one of the few
remaining uses of the Atomic32 type.

Bug: webrtc:8270
Change-Id: Ibbbf227c710f7e8f68a98bac46d3e24b7cb5ee2f
Reviewed-on: https://webrtc-review.googlesource.com/14600
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20393}
2017-10-23 14:22:17 +00:00
d92e0b5923 Fixing crash in Mac client when no cameras are available.
Bug: webrtc:8348
Change-Id: Ibf84ca76812d8c002fae9bd7bcf616abc53c78b1
Reviewed-on: https://webrtc-review.googlesource.com/7340
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Daniela Jovanoska Petrenko <denicija@webrtc.org>
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20392}
2017-10-23 13:56:08 +00:00
93bc308f31 MB: Add Android Perf (swarming)
TBR=phoglund@webrtc.org

No-Try: true
Bug: chromium:755660
Change-Id: I9ff6304142bfd3b0ab4d2b1cfa46466685d58e09
Reviewed-on: https://webrtc-review.googlesource.com/14060
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20391}
2017-10-23 13:02:07 +00:00
fc313dcda6 Add prefix to codec name constants.
Bug: webrtc:8401
Change-Id: I8cd4685df3609e8b91a79b19789aadef484138d4
Reviewed-on: https://webrtc-review.googlesource.com/14140
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20390}
2017-10-23 12:49:17 +00:00
47217364f5 Adding a KVO context to avoid issues with future super/sub-classing.
Bug: webrtc:8342
Change-Id: I457858056ffc7f33bbfb261153301ea2ccd71a51
Reviewed-on: https://webrtc-review.googlesource.com/6440
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Daniela Jovanoska Petrenko <denicija@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20389}
2017-10-23 12:33:17 +00:00
bd92d8dd2a Forgotten 'memset' in NoiseSuppression.
The 'parametricNoise' field is never initialized in the
'WebRtcNs_InitCore' function that initializes a 'NoiseSuppressionC'
struct.

This leads to use of unititialized value, which may affect the audio
output and result of the noise suppressor.

The issue was found by the Chrome fuzzer:
https://clusterfuzz.com/v2/testcase-detail/4749034115039232

Bug: chromium:776673
Change-Id: I1c3fd80cff178f2d5917064ad07f88c7b9a29e7d
Reviewed-on: https://webrtc-review.googlesource.com/14556
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20388}
2017-10-23 12:11:47 +00:00
8731176b92 NetEq: Fix an UBSan error
UBSan will trigger when shifting a negative value. This change avoids
that by replacing "x << 8" with "x * (1 << 8)".

Bug: chromium:666877
Change-Id: Ic89bd98e5a3feff35075df96b104b386cb4d8803
Reviewed-on: https://webrtc-review.googlesource.com/14552
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20387}
2017-10-23 11:56:47 +00:00
9155e4986d New classes RefCounter and RefCountedBase.
Bug: webrtc:8270
Change-Id: Ibdab81b3fcbe6cba9ae24033f56c84b13c868b21
Reviewed-on: https://webrtc-review.googlesource.com/2684
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20386}
2017-10-23 11:46:47 +00:00
ccdfccaa6f New PacketQueue2 behind WebRTC-RoundRobinPacing field trial.
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}
2017-10-23 11:39:57 +00:00
0c720505af Adding libFuzzer target for UlpFEC receiver.
Bug: none
Change-Id: I20e622455aee2f5aebad835e915d65f3475fbd17
Reviewed-on: https://webrtc-review.googlesource.com/14300
Commit-Queue: Mark Brand <markbrand@google.com>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20384}
2017-10-23 11:37:07 +00:00
4d9ac5886d Use C++11 static initialization, replacing Atomic32 CompareExchange.
Bug: webrtc:8270
Change-Id: I328cec46be2a017a518946d19d21c242a067747d
Reviewed-on: https://webrtc-review.googlesource.com/14220
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Zijie He <zijiehe@chromium.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20383}
2017-10-23 10:57:57 +00:00
d6314d9fc1 Delete unused method PacedSender::AverageQueueTimeMs
It was used only in tests.

Bug: webrtc:8422
Change-Id: I67b58663c171202240d1c5a7c230d6cd4cd6149b
Reviewed-on: https://webrtc-review.googlesource.com/13102
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20382}
2017-10-23 09:06:17 +00:00
b7e1788466 Fixing other unsafe conversions.
The bot "Win (more_configs)" has spotted another unsafe type conversion.

This CL is a follow-up of:
- https://webrtc-review.googlesource.com/c/src/+/12921
- https://webrtc-review.googlesource.com/c/src/+/13122
- https://webrtc-review.googlesource.com/c/src/+/13622

Bug: chromium:759980
Change-Id: I9a4268e7ea938cc85376211b40767fd8465f37fd
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/13623
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20381}
2017-10-23 07:59:39 +00:00
5526687dfc Add instructions for converting SCTP verbose log to pcap file.
This is pretty useful if you need to debug an SCTP issue. SCTP goes
over DTLS, which is encrypted, which means you need a private key
in order to decrypt a normal packet capture. We don't log this key,
for understandable reasons. So the alternative is to turn on SCTP
verbose logging, then turn the text log into a pcap file.

NOTRY=True
TBR=zhihuang@webrtc.org

Bug: None
Change-Id: If3380d7953ea829b3ae9945326d3c820ce7cc6a3
Reviewed-on: https://webrtc-review.googlesource.com/14561
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20380}
2017-10-22 23:13:55 +00:00
cb423c43b0 Adding comment about ice_connection_state really being ICE+DTLS.
NOTRY=True
TBR=zhihuang@webrtc.org

Bug: webrtc:6145
Change-Id: I02c75ddd96007d942cc4d3ee8313691cf0be6827
Reviewed-on: https://webrtc-review.googlesource.com/14560
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20379}
2017-10-22 20:46:05 +00:00
1c1a6815ae Revert "Add fine grained dropped video frames counters on sending side"
This reverts commit 4b1a363e4c238f2e1ec2d8a9ce1f819f59d710ce.

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
8d3444df2d Reland "Rewrite WebRtcSession media tests as PeerConnection tests"
This is a reland of 3df5dcac9b339ba4d3f4969602f094c2c8035b51
Original change's description:
> Rewrite WebRtcSession media tests as PeerConnection tests
> 
> Bug: webrtc:8222
> Change-Id: I782a3227e30de70eb8f6c26a48723cb3510a84ad
> Reviewed-on: https://webrtc-review.googlesource.com/6640
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20364}

Bug: webrtc:8222
Change-Id: I0a5398170d469eb9223bc781bfb417a85a72a2d2
Reviewed-on: https://webrtc-review.googlesource.com/14380
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20377}
2017-10-21 01:38:14 +00:00
160c02089e Implement Atomic32 using C++11's std::atomic
Chromoting is planning to bump up the iOS deployment target to 10.0
before but is blocked by WebRTC because it uses deprecated OSAtomic*
APIs.

This CL replaces the platform specific Atomic32 implementation and uses
C++11's std::atomic instead.

Bug: webrtc:8413
Change-Id: Icbd29bb92eec60066589bbabf53e0d50df2f9b0d
Reviewed-on: https://webrtc-review.googlesource.com/12740
Commit-Queue: Yuwei Huang <yuweih@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20376}
2017-10-20 23:25:04 +00:00
f2662f08e5 Revert "Rewrite WebRtcSession media tests as PeerConnection tests"
This reverts commit 3df5dcac9b339ba4d3f4969602f094c2c8035b51.

Reason for revert: suspected of breaking chromium.webrtc.fyi:
WebRtcBrowserTest.NegotiateUnsupportedVideoCodec
WebRtcBrowserTest.NegotiateNonCryptoCall

android https://uberchromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Android%20Tests%20%28dbg%29%20%28L%20Nexus5%29/builds/25506
linux https://uberchromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Linux%20Tester/builds/38809
mac
https://uberchromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Mac%20Tester/builds/44120
windows
https://uberchromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Win10%20Tester/builds/9236

Original change's description:
> Rewrite WebRtcSession media tests as PeerConnection tests
> 
> Bug: webrtc:8222
> Change-Id: I782a3227e30de70eb8f6c26a48723cb3510a84ad
> Reviewed-on: https://webrtc-review.googlesource.com/6640
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20364}

TBR=steveanton@webrtc.org,deadbeef@webrtc.org

Change-Id: Iaacc950d050ba2835d262908658dc045f234ef5b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8222
Reviewed-on: https://webrtc-review.googlesource.com/14160
Commit-Queue: Olga Sharonova <olka@webrtc.org>
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20375}
2017-10-20 13:00:45 +00:00
b49b66109e Revert "Rewrite WebRtcSession BUNDLE tests as PeerConnection tests"
This reverts commit 096e367bfd58f9c24199852dcfdb6447b71c4324.

Reason for revert:
suspected of breaking chromium.webrtc.fyi:
WebRtcBrowserTest.NegotiateUnsupportedVideoCodec
WebRtcBrowserTest.NegotiateNonCryptoCall

android https://uberchromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Android%20Tests%20%28dbg%29%20%28L%20Nexus5%29/builds/25506
linux https://uberchromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Linux%20Tester/builds/38809
mac
https://uberchromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Mac%20Tester/builds/44120
windows
https://uberchromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Win10%20Tester/builds/9236

Original change's description:
> Rewrite WebRtcSession BUNDLE tests as PeerConnection tests
> 
> Bug: webrtc:8222
> Change-Id: Id47e4544dc073564ad7e63d02865ca80dd5a85ff
> Reviewed-on: https://webrtc-review.googlesource.com/8280
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20365}

TBR=steveanton@webrtc.org,deadbeef@webrtc.org

Change-Id: I571d8c7fdce4b47137260e0f3276ea4eb04a496c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8222
Reviewed-on: https://webrtc-review.googlesource.com/14240
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Commit-Queue: Olga Sharonova <olka@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20374}
2017-10-20 12:57:06 +00:00
fd7c0a566a Avoid NetEq triggering a Framelength change when receiving an FEC packet.
Internally in NetEq, an FEC packet looks very similar to a split packet, which caused NetEq to miscalculate the frame length of FEC packets. This incorrect framelength calculation was incorrectly handled as a framelength change by NetEq.

Bug: webrtc:8410
Change-Id: Icaea961d055e49d7726b87811881db0b9149805b
Reviewed-on: https://webrtc-review.googlesource.com/12420
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20373}
2017-10-20 11:56:23 +00:00
9434240f7e Fix a crash in AppRTCMobile when videoCodec is set to null.
peerConnectionParameters.videoCodec can be null in some cases.
Previously, this would cause a crash in AppRTCMobile.

Bug: b/67938523
Change-Id: I30ebf0f91fad23a3cf34946736b9f4e6c266277f
Reviewed-on: https://webrtc-review.googlesource.com/14200
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20372}
2017-10-20 11:55:03 +00:00
36636a2d6c Removes StaticAccessedFromInstance warning in Android audio track.
Fixes the warning below:

WebRtcAudioTrack.java:364: warning: [StaticAccessedFromInstance] Static method getMaxVolume
should not be accessed from an object instance; instead use AudioTrack.getMaxVolume
+ "max gain: " + audioTrack.getMaxVolume());

Bug: NONE
Change-Id: I6247584b65ac972a6a3739fba718387873964f9f
Reviewed-on: https://webrtc-review.googlesource.com/14180
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20371}
2017-10-20 11:02:03 +00:00
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
78609d5b6b Reland of BWE allocation strategy
TBR=stefan@webrtc.org,alexnarest@webrtc.org

Bug: webrtc:8243
Change-Id: Ie68e4f414b2ac32ba4e64877cb250fabcb089a07
Reviewed-on: https://webrtc-review.googlesource.com/13940
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Reviewed-by: Alex Narest <alexnarest@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20369}
2017-10-20 10:16:15 +00:00
b82de30080 Add new neteq_signal_fuzzer
NetEq has been fuzzed by neteq_rtp_fuzzer for some time. That fuzzer
hammers the RTP data, but leaves much of the other data alone. This
new fuzzer instead alters the encoded audio, packet arrival timing,
clock drift, and packet losses.

Bug: webrtc:8421
Change-Id: Ie25b77590a66a7451f32a73c6b5b570944244027
Reviewed-on: https://webrtc-review.googlesource.com/13860
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20368}
2017-10-20 09:05:54 +00:00
6f38d25f11 Add Java support for AudioProcessing and PostProcessing injection
This allows injection of a user-defined post processing module from
the Android layer.

Bug: webrtc:8163
Change-Id: If3a6b4726c34c5f82d186b8cf95373c283cbd3f6
Reviewed-on: https://webrtc-review.googlesource.com/7610
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20367}
2017-10-20 08:37:23 +00:00
6f72f56b6c Change return types of refcount methods.
AddRef() now returns void, and Release() returns an enum
RefCountReleaseStatus, to indicate whether or not this Release
call implied deletion.

Bug: webrtc:8270
Change-Id: If2fb77f26118b61751b51c856af187c72112c630
Reviewed-on: https://webrtc-review.googlesource.com/3320
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20366}
2017-10-20 07:46:03 +00:00
096e367bfd Rewrite WebRtcSession BUNDLE tests as PeerConnection tests
Bug: webrtc:8222
Change-Id: Id47e4544dc073564ad7e63d02865ca80dd5a85ff
Reviewed-on: https://webrtc-review.googlesource.com/8280
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20365}
2017-10-20 02:31:13 +00:00
3df5dcac9b Rewrite WebRtcSession media tests as PeerConnection tests
Bug: webrtc:8222
Change-Id: I782a3227e30de70eb8f6c26a48723cb3510a84ad
Reviewed-on: https://webrtc-review.googlesource.com/6640
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20364}
2017-10-20 01:52:23 +00:00
930e1af76f Revert "Add suppression for issue 8405"
This reverts commit 2505346be2439af26b4c800b99784104672d4d03.

Reason for revert: Memory leak fixed by https://webrtc-review.googlesource.com/c/src/+/13401

Original change's description:
> Add suppression for issue 8405
> 
> Bug: webrtc:8405
> Change-Id: I517af29669f2313805f6aeaf117634b398c0a5f3
> Reviewed-on: https://webrtc-review.googlesource.com/13740
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20356}

TBR=phoglund@google.com,phoglund@webrtc.org,sprang@webrtc.org

Change-Id: I1e4a0e2009d28ae7059ad4d83f910ee166d11abe
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8405
Reviewed-on: https://webrtc-review.googlesource.com/14080
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20363}
2017-10-19 18:08:27 +00:00
3b80aace61 Fix flaky memory leak in RemoteAudioSource
Bug: webrtc:8405
Change-Id: Ie7c89214323678c6ea34e344bb1a5a33ad46b3f0
Reviewed-on: https://webrtc-review.googlesource.com/13401
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20362}
2017-10-19 18:01:52 +00:00
dc9ca9329b Revert "BWE allocation strategy"
This reverts commit a5fbc23379823d74b8cf4bc18887ff40237989e8.

Reason for revert: <INSERT REASONING HERE>

Original change's description:
> BWE allocation strategy
> 
> This is reland of https://webrtc-review.googlesource.com/c/src/+/4860 with the fixed RampUpTest test
> 
> Bug: webrtc:8243
> Change-Id: I4b90a449b00dd05feee974001e08fb40710b59ac
> Reviewed-on: https://webrtc-review.googlesource.com/13124
> Commit-Queue: Alex Narest <alexnarest@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20345}

TBR=stefan@webrtc.org,alexnarest@webrtc.org

Change-Id: I8ed12cd2115ef63204e384cc93c9f4473daa54d1
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8243
Reviewed-on: https://webrtc-review.googlesource.com/14020
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Reviewed-by: Alex Narest <alexnarest@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20361}
2017-10-19 15:34:52 +00:00
44582878d3 Final stage in removing unused ADM APIs
Bug: webrtc:7306
Change-Id: Id5ce43ed8c259ef343474c05b190f143be12e8aa
Reviewed-on: https://webrtc-review.googlesource.com/13861
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20360}
2017-10-19 14:36:31 +00:00
f74d641fc1 Simplify setting/unsetting REMB in RtcpSender
follow up of https://webrtc-review.googlesource.com/c/src/+/7983

Bug: None
Change-Id: I408c21408478d801a769e2e9d5f2eb9408430a4b
Reviewed-on: https://webrtc-review.googlesource.com/12520
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20359}
2017-10-19 14:07:31 +00:00
2d27fb5a33 Move TestConfig to separate file.
Move functions Set/PrintCodecSettings, NumberOfTemporalLayers to TestConfig.
Add function NumberOfCores.

Bug: none
Change-Id: Ic33d79681d59d62bf34d9c9ff056a751ed3f8da8
Reviewed-on: https://webrtc-review.googlesource.com/13120
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20358}
2017-10-19 13:56:31 +00:00
4696a159ef header_usage.sh: Keep leading directory for grep.
Makes the script a bit more accurate in identifying place-holder
header files.

Bug: None
Change-Id: Iba13562aed0d40bffef9c25d7345f3bb8af16c29
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/4381
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20357}
2017-10-19 13:46:51 +00:00
2505346be2 Add suppression for issue 8405
Bug: webrtc:8405
Change-Id: I517af29669f2313805f6aeaf117634b398c0a5f3
Reviewed-on: https://webrtc-review.googlesource.com/13740
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20356}
2017-10-19 13:16:41 +00:00
19f5be3e7b Fix bug that causes the visualization of RTCP fraction lost to be cropped.
The bug comes from computing the axis limits before adding the data.

Bug: webrtc:8417
Change-Id: I1e3860b4f655f40cd261d09b8357d563dbe51f1d
Reviewed-on: https://webrtc-review.googlesource.com/12922
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20355}
2017-10-19 13:15:01 +00:00
8818237538 voe::Channel: Don't use CodecManager and RentACodec
As a bonus, we also get rid of most uses of CodecInst in voe::Channel.

BUG=webrtc:8396

Change-Id: I20f82274dcec2ad8be6e046b987d7d65a19d0e46
Reviewed-on: https://webrtc-review.googlesource.com/10802
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20354}
2017-10-19 13:12:51 +00:00
5f5fc6845a Relax threading requirements in VideoEncoder.
Previous strict threading requirements we not always followed and this
causes crashes in some scenarios.

Bug: webrtc:7760
Change-Id: I4808edfff19c63a9c4a9585834d46f9c73bb1c4b
Reviewed-on: https://webrtc-review.googlesource.com/13660
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20353}
2017-10-19 12:44:11 +00:00
15c9cf732b Reland "Delete reference to webrtc_overides/rtc_base/task_queue.cc."
This is a reland of 17a0474379ebba7f48a59f2952b0a184071c47d8
Original change's description:
> Delete reference to webrtc_overides/rtc_base/task_queue.cc.
> 
> This change depends on Chromium cl
> https://chromium-review.googlesource.com/c/chromium/src/+/681814,
> which must be landed first.
> 
> Bug: webrtc:8166
> Change-Id: I8e1c96ad5927e1d9d4e45e6f6c0b5f712d58cef7
> Reviewed-on: https://webrtc-review.googlesource.com/3940
> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20071}

Bug: webrtc:8166
Change-Id: Ia011f82dde96e8f0c5c18ad7ac20fd6b7adcb1e8
Reviewed-on: https://webrtc-review.googlesource.com/13620
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20352}
2017-10-19 12:15:22 +00:00
1c66f9fcd1 Fix chrome tests by making SetBitrateAllocationStrategy not pure virtual
Bug: webrtc:8243
Change-Id: I6fedf00d6735895771d460299458c5f8b6face8d
Reviewed-on: https://webrtc-review.googlesource.com/13760
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20351}
2017-10-19 11:45:11 +00:00
7173cf20cc Add cpu measurements to VideoProcessorIntegrationTest.
Remove unused method ExcludeFrameTypesToStr.

Bug: webrtc:6634
Change-Id: I2816466ed428b8ce13f3073ca496c2891d5d6368
Reviewed-on: https://webrtc-review.googlesource.com/9400
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20350}
2017-10-19 11:37:51 +00:00
ea7a3f8225 Fixing unsafe conversion
The bot "Win (more_configs)" has spotted another unsafe type conversion.

This CL is a follow-up of:
- https://webrtc-review.googlesource.com/c/src/+/12921
- https://webrtc-review.googlesource.com/c/src/+/13122

Bug: chromium:759980
Change-Id: I3634c3e20fcd9f4e106914399ac40ca87d4c6137
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/13622
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20349}
2017-10-19 10:59:50 +00:00
712048c64f Replace raw new by MakeUnique, and improve declarations.
For unique_ptrs assigned at construction time, declare them const, and
use RTC_PT_GUARDED_BY rather than RTC_GUARDED_BY.

Bug: None
Change-Id: I8aa83e062a1550780ee07792c1fbb195267d5524
Reviewed-on: https://webrtc-review.googlesource.com/12923
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20348}
2017-10-19 10:53:40 +00:00
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
a8c08b1063 APM-QA annotations: incorrect type bugfix and level estimation with 1 ms frames.
TBR=

Bug: webrtc:7494
Change-Id: I2d4432d5b135e70b9abb5f2794a28228ec6808ba
Reviewed-on: https://webrtc-review.googlesource.com/13621
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20346}
2017-10-19 09:42:00 +00:00
a5fbc23379 BWE allocation strategy
This is reland of https://webrtc-review.googlesource.com/c/src/+/4860 with the fixed RampUpTest test

Bug: webrtc:8243
Change-Id: I4b90a449b00dd05feee974001e08fb40710b59ac
Reviewed-on: https://webrtc-review.googlesource.com/13124
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20345}
2017-10-19 09:30:00 +00:00