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}
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}
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}
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}
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}
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}
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}
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}
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}
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}
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}
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}
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}
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}
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}
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}
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}
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}
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}
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}