UBSan will trigger when time_stretched_samples overflows using a
big number. This change avoids this problem by storing the
intermediate result into a int64_t.
Bug: chromium:886904
Change-Id: Id09dc4b468f841f03b523d5f21763f610b163a42
Reviewed-on: https://webrtc-review.googlesource.com/c/103123
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Pablo Barrera González <barrerap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24977}
After the removal of field_trial_default, metrics_default and
runtime_enabled_features_default, this build target doesn't build
anything and can be safely removed.
Bug: webrtc:9631
Change-Id: Iee1111e065ffefe0b4b9a695ee67a594e6d82caa
Reviewed-on: https://webrtc-review.googlesource.com/c/103702
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24976}
This reverts commit 9e24dcff167c4eb3555bf0ce6eaba090c10fbe53.
Reason for revert: Breaks chromium.webrtc.fyi bots.
Original change's description:
> Export symbols needed by the Chromium component build (part 1).
>
> This CL uses RTC_EXPORT (defined in rtc_base/system/rtc_export.h)
> to mark WebRTC symbols as visible from a shared library, this doesn't
> mean these symbols are part of the public API (please continue to refer
> to [1] for info about what is considered public WebRTC API).
>
> [1] - https://webrtc.googlesource.com/src/+/HEAD/native-api.md
>
> Bug: webrtc:9419
> Change-Id: I802abd32874d42d3aa5ecd3c8022e7cf5e043d99
> Reviewed-on: https://webrtc-review.googlesource.com/c/103505
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24969}
TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org
Change-Id: I01f6e18f0d2c0f0309cdaa6c943c3927e1f1f49f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9419
Reviewed-on: https://webrtc-review.googlesource.com/c/103720
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24974}
It interferes with a future refactoring, and isn't all that useful
anyway.
Bug: webrtc:9810
Change-Id: I46163d7921d39249615b7eea60306c598f13c128
Reviewed-on: https://webrtc-review.googlesource.com/c/103443
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24970}
This CL uses RTC_EXPORT (defined in rtc_base/system/rtc_export.h)
to mark WebRTC symbols as visible from a shared library, this doesn't
mean these symbols are part of the public API (please continue to refer
to [1] for info about what is considered public WebRTC API).
[1] - https://webrtc.googlesource.com/src/+/HEAD/native-api.md
Bug: webrtc:9419
Change-Id: I802abd32874d42d3aa5ecd3c8022e7cf5e043d99
Reviewed-on: https://webrtc-review.googlesource.com/c/103505
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24969}
The RtpReceiver class is no longer used. Together with it, delete
RTPPayloadRegistry, RtpReceiverStrategy, and the tests under
modules/rtp_rtcp/test/testAPI/.
Bug: webrtc:8995
Change-Id: Ia9924d2f0f4315914a0dce6b7375ebb3601a6f96
Reviewed-on: https://webrtc-review.googlesource.com/c/103503
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24968}
The presubmit lint checks require GN but it is not downloaded because runhooks is avoided on the autoroller bot.
The trybots in Gerrit UI should catch the same errors anyway, if they somehow happen.
Also minor cleanup of obsolete flag
No-Try: True
Bug: chromium:836566
Change-Id: I8bf03b8e155343f723c6fdda37210d9161da984c
Reviewed-on: https://webrtc-review.googlesource.com/c/103620
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24967}
This method is added in API level 23, and is currently used in
NetworkMonitorAutoDetect to determine the underlying type of a VPN
network.
Bug: webrtc:9811
Change-Id: I7277cd9adb5b3d3d9b116f667bf533352f9b3bdf
Reviewed-on: https://webrtc-review.googlesource.com/c/103560
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@google.com>
Cr-Commit-Position: refs/heads/master@{#24961}
Rename to better match what it does,
Adjust to support two-byte header extension
Bug: webrtc:7990
Change-Id: I2786d70e7cf9cd3d722f54fb1d07c9cfaafab947
Reviewed-on: https://webrtc-review.googlesource.com/103201
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24958}
This refactoring merged PopulateCodecSpecific and FrameEncoded into a
single callback method. It also removes the FrameConfig parameter and
instead relies on the temporal layer to remember it internally.
Bug: webrtc:9012
Change-Id: I489b76821b534398ad452643f1322f411d3455b1
Reviewed-on: https://webrtc-review.googlesource.com/95681
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24957}
This CL adds the ability to toggle the AGC2 adaptive digital mode in
audioproc_f
Bug: webrtc:5298
Change-Id: If1567d8c87f88992dff89253edb293a56cee0a73
Reviewed-on: https://webrtc-review.googlesource.com/c/103361
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24954}
../../third_party/webrtc/modules/pacing/round_robin_packet_queue.h:70:5:
warning: explicitly defaulted default constructor is implicitly deleted
[-Wdefaulted-function-deleted]
StreamPrioKey() = default;
^
../../third_party/webrtc/modules/pacing/round_robin_packet_queue.h:80:37: note:
default constructor of 'StreamPrioKey' is implicitly deleted because field
'priority' of const-qualified type 'const RtpPacketSender::Priority' would not
be initialized
const RtpPacketSender::Priority priority;
^
Bug: chromium:890307
Change-Id: I58f21121fc9083a60ba1ad26492fdca6285d0447
Reviewed-on: https://webrtc-review.googlesource.com/c/103181
Commit-Queue: Nico Weber <thakis@chromium.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24952}
Calculate number of freezes per minute for a received video stream
and report this metric to UMA.
Bug: webrtc:9803
Change-Id: I6d72a2daf58b2f734a576fff469c1fead6cc69b3
Reviewed-on: https://webrtc-review.googlesource.com/c/103180
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24944}
This is a reland of 2fbb83b16b4c2c1712cbe898ca3ba42d6da3e96f
Original change's description:
> Remove APM-internal usage of EchoControlMobile
>
> This is a sibling CL to a similar one for EchoCancellation:
> https://webrtc-review.googlesource.com/c/src/+/97603
>
> - EchoControlMobileImpl will no longer inherit EchoControlMobile.
> - Removes usage of AudioProcessing::echo_control_mobile() inside most of
> the audio processing module and unit tests.
>
> The CL breaks audioproc_f backwards compatibility: It can no longer
> use all recorded settings (comfort noise, routing mode), but prints an
> error message when unsupported settings are encountered.
>
> Tested: audioproc_f with .wav and aecdump inputs.
> Bug: webrtc:9535
> Change-Id: I63c3c81bcaf44021315978e1a0f3e42173b988ce
> Reviewed-on: https://webrtc-review.googlesource.com/101621
> Reviewed-by: Alex Loiko <aleloi@webrtc.org>
> Commit-Queue: Sam Zackrisson <saza@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24888}
Bug: webrtc:9535
Change-Id: I172706c6729cac4eb6afde1ebd6fc8f3a289d6c7
Reviewed-on: https://webrtc-review.googlesource.com/c/102881
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24943}
This change adds bindings so that native FrameEncryptor and native FrameDecryptor
objects can be set on the objective C RTCRtpSender and RTCRtpReceiver objects.
Bug: webrtc:9681
Change-Id: Iec4006ea020d6ab6adcc0ad068dcd8fb2738063d
Reviewed-on: https://webrtc-review.googlesource.com/c/103020
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24936}
We're receiving SCTP messages one chunk at a time. The sender is supposed to set the MSG_EOR bit on the last chunk of a message. A crash (RTC_CHECK) happened when the sender started a new message without indicating the end of the previous message. This is likely due to an SCTP implementation that doesn't (correctly) support the MSG_EOR bit.
This change, rather than calling RTC_CHECK, delivers partial messages when SID is (unexpectedly) changed, which is what happened before we paid attention to the MSG_EOR bit ourselves.
TBR=pthatcher@webrtc.org
Bug: chromium:884926
Change-Id: I18c773bb60ae724b735a83933eedd030f6360a3c
Reviewed-on: https://webrtc-review.googlesource.com/c/103023
Commit-Queue: Jeroen de Borst <jeroendb@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24935}
In this CL the ERLE estimator is reset after a pre-amplifier gain change is communicated to APM.
Bug: webrtc:9805
Change-Id: I040f344e4607e862240250f9478d06de0d58a096
Reviewed-on: https://webrtc-review.googlesource.com/103222
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24933}
This change does three things:
- Move rtc_json into rtc_base/strings/, a non-API directory more fitting to
its purpose.
- Make a target for the currently unused json_unittest.
- Make the code available for use in non-test code again.
Bug: webrtc:9802
Change-Id: Id964a8a4b47b732a962a364894a4dbd3e7f4650f
Reviewed-on: https://webrtc-review.googlesource.com/103126
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24932}
This CL enables tests that were previously disabled and fixes the issues
that made them flaky.
Bug: webrtc:6889, webrtc:7888
Change-Id: I914b59200d7bf2973e8993b04de867cc3355b8a8
Reviewed-on: https://webrtc-review.googlesource.com/98381
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24930}
This CL makes the RtpGenericFrameDescriptor available in
RTPSenderVideo::SendVideo for encryption and in
RtpVideoStreamReceiver::OnReceivedFrame for decryption.
Bug: webrtc:9361
Change-Id: I5b6d10138c0874657862f103c8c9a2328e6d4a66
Reviewed-on: https://webrtc-review.googlesource.com/102720
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24929}
The PushResampler has a SincResampler per channel. Before this CL, it
was hard-coded to handle up to 2 channels. In this CL I made it handle
arbitrarily many.
Bug: webrtc:8649
Change-Id: Ia2f33e45535f8bbda59090f8a0847546ff7edd75
Reviewed-on: https://webrtc-review.googlesource.com/103000
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24928}
