The array is reset in Init() but not the indexer. This makes the start point undefined after Init() for re-initializing an AudioLoop. This can be fixed.
BUG=
Review URL: https://codereview.webrtc.org/1727353002
Cr-Commit-Position: refs/heads/master@{#11739}
initialization and errors.
The stats are counts using enumeration, an instance of
H264EncoderImpl/H264DecoderImpl will report at most 1 Init
and 1 Error for its entire lifetime. This is to avoid
spamming reports if initialization or coding fails and it
retries in a loop. The Init stats will give us an idea of
usage counts for the encoder/decoder. The Error stats will
give us an idea of how many of these usages encounters some
type of problem, such as encode or decode errors.
- WebRTC.Video.H264EncoderImpl.Event:
* kH264EncoderEventInit: Occurs at InitEncode.
* kH264EncoderEventError: Occurs if any type of error
occurs during initialization or encoding.
- WebRTC.Video.H264DecoderImpl.Event:
* kH264DecoderEventInit: Occurs at InitDecode.
* kH264DecoderEventError: Occurs if any type of error
occurs during initialization, AVGetBuffer2 or decoding.
Chromium sibling CL:
https://codereview.chromium.org/1719273002/
BUG=chromium:500605, chromium:468365
Review URL: https://codereview.webrtc.org/1716173002
Cr-Commit-Position: refs/heads/master@{#11736}
Previously, we relied on the encoded stream to come to an end before
the end of the buffer. This is a bad idea, since it is possible to
craft a stream that fills the buffer while decoding to less than the
expected amount of data; without the new checks introduced here, this
causes the decoder to read past the end of the input buffer.
BUG=chromium:582471, chromium:587852
Review URL: https://codereview.webrtc.org/1721593004
Cr-Commit-Position: refs/heads/master@{#11734}
In order for the change to be reviewable, the
move was made into two steps consisting of the
first two patches in this CL.
Step 1 (patch set 1):
-Changed file types to use .cc
-Changed buildfiles to use the new files
-Changed C code inclusion to properly match the changed
file formats (removed and added extern "C" declarations).
-Changed implicit void-> nonvoid casts that are
illegal in C++ to be explicit.
Step 2 (patch set 2):
-Changed all the warnings reported when uploading the CL.
-The warnings about formatting of the assembly optimized
code were not addressed though.
BUG=webrtc:5201
Review URL: https://codereview.webrtc.org/1713923002
Cr-Commit-Position: refs/heads/master@{#11727}
Reason for revert:
Breaks downstream compilation using webrtc/common_audio/real_fourier.h. Let's chat tomorrow on how to coordinate a re-land.
Original issue's description:
> Replace scoped_ptr with unique_ptr in webrtc/common_audio/
>
> BUG=webrtc:5520
>
> Committed: https://crrev.com/79d7a499c0c3e1de8f5ad1138236f0386701053f
> Cr-Commit-Position: refs/heads/master@{#11716}
TBR=henrik.lundin@webrtc.org,kwiberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5520
Review URL: https://codereview.webrtc.org/1726043002
Cr-Commit-Position: refs/heads/master@{#11726}
This allows other projects to more easily depend on this.
The plan is to move remote_bitrate_estimator and bitrate_controller into this module and reduce the exposed interface to only a simplified version of congestion_controller.h.
No functional changes in this CL.
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1718473002 .
Cr-Commit-Position: refs/heads/master@{#11718}
Compact NTP representation was designed exactly for that purpose: calculate RTT. No need to map to ms before doing arithmetic on this values.
Because of this change there is no need to keep mapping between compact ntp presentation and milliseconds in the RTCPSender.
BUG=webrtc:5565
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1491843004 .
Cr-Commit-Position: refs/heads/master@{#11710}
Also move some stats reporting from vie_channel to send stats proxy
BUG=
Review URL: https://codereview.webrtc.org/1669623004
Cr-Commit-Position: refs/heads/master@{#11688}
Prevents allocating sequence numbers for packets that go out on the
network even though sending media is disabled.
This race caused a replay of sequence numbers when GetRtpState() on a
stopped stream would not return the last sequence number sent, since the
pacer thread could request and send padding on a later sequence number
before the modules are disconnected from the pacer.
BUG=webrtc:5543
R=stefan@webrtc.org
TEST=Repeating EndToEndTest.RestartingSendStreamPreservesRtpState 1000 times under TSan.
Review URL: https://codereview.webrtc.org/1715703002 .
Cr-Commit-Position: refs/heads/master@{#11685}
There were two different structures named RtpPacket in webrtc namespace:
RtpPacket defined in fec_test_helper renamed to test::RawRtpPacket
RtpPacket defined in rtp_sender_video and producer_fec removed as unused
BUG=webrtc:5261
R=sprang@google.com, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1710103004 .
Cr-Commit-Position: refs/heads/master@{#11682}
It was hardly used, making the code more complex than needed and caused problems on iOS because it uses system.
BUG=webrtc:5549
R=kjellander@webrtc.org
Review URL: https://codereview.webrtc.org/1708353002 .
Cr-Commit-Position: refs/heads/master@{#11677}
Instead of excluding the whole test binaries, only exclude the parts that cause the
compilation to fail for modules_unittests and common_audio_unittests.
BUG=webrtc:4752, webrtc:4755, webrtc:5544
TESTED=Successful build with:
GYP_DEFINES='OS=ios target_arch=x64' webrtc/build/gyp_webrtc
ninja -C out/Debug-iphonesimulator modules_unittests common_audio_unittests
NOTRY=True
Review URL: https://codereview.webrtc.org/1698033002
Cr-Commit-Position: refs/heads/master@{#11675}
When composing a RTCP packet, if there is a BYE
to be appended, preserve it and append it at the
end after all other packet types are added.
BUG=webrtc:5498
NOTRY=true
Review URL: https://codereview.webrtc.org/1674963004
Cr-Commit-Position: refs/heads/master@{#11672}
The legacy objc API is not included in the GYP generation if include_tests=0.
This causes problems downstream in some cases, so it's changed in this CL.
The libyuv dependency needs to be possible to disable using the build_libyuv
GYP variable.
NOTRY=True
Review URL: https://codereview.webrtc.org/1705733002
Cr-Commit-Position: refs/heads/master@{#11652}
rtcp::RawPacket is rtc::Buffer, it had no extra functionality.
rtc::Buffer is a movable class - no point to wrap it into rtc::scoped_ptr
change is large, but straightforward:
rtc::scoped_ptr<rtcp::RawPacket> replaced with rtc::Buffer
->Buffer() replaced with .data()
->Length() replaced with .size()
BUG=webrtc:5260
Review URL: https://codereview.webrtc.org/1696203002
Cr-Commit-Position: refs/heads/master@{#11649}
With this change the following tests have been successfully
passing in the iOS Simulator for iPhone 5 and iOS 9:
* audio_decoder_unittests
* common_video_unittests
* modules_tests
* rtc_api_objc_tests
* rtc_pc_unittests
* system_wrappers_unittests
* voice_engine_unittests
The modules_unittests and common_audio_unittests are
handled in https://codereview.webrtc.org/1698033002/
BUG=webrtc:4755
NOTRY=True
Review URL: https://codereview.webrtc.org/1694353003
Cr-Commit-Position: refs/heads/master@{#11646}
In some cases, the decoder can write outside of an allocated array. See
the new comment in the code for more details.
BUG=chromium:568885, webrtc:5305
Review URL: https://codereview.webrtc.org/1704463002
Cr-Commit-Position: refs/heads/master@{#11641}
TMMBN was capped by configured max bitrate for no apparent reason.
Removing this to not require payload-type reconfiguration on new
video-codec settings. Actual removal of payload-type reconfiguration
will happen in a pending CL.
BUG=webrtc:5494
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1702043002 .
Cr-Commit-Position: refs/heads/master@{#11639}
In some cases, the decoder can read outside of an allocated array. See
the new comment in the code for more details.
BUG=chromium:568889, webrtc:5305
Review URL: https://codereview.webrtc.org/1700973002
Cr-Commit-Position: refs/heads/master@{#11637}
Also cleans up some unused code and makes sure the min bitrate of the BWE can't be set to anything lower than 10 kbps.
BUG=webrtc:5474
R=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1699903003 .
Cr-Commit-Position: refs/heads/master@{#11636}
This is needed when synthesizing a call based on
48 kHz audio files as otherwise an error is
generated about the wrong sample rate is generated.
That error is in turned caused by the sample rate
being changed from the default 16 kHz
at the first Capture API call event.
BUG=
Review URL: https://codereview.webrtc.org/1698243003
Cr-Commit-Position: refs/heads/master@{#11635}
Skip accounting for small packets and suspend the prober if no
large-enough packets have been sent for some time. This especially seems
to have triggered in audio-only calls where all packets are too small,
making TimeUntilNextProbe return 0 forever, causing the module process
thread to wake up forever.
BUG=webrtc:5506
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1688703002 .
Cr-Commit-Position: refs/heads/master@{#11634}
Apart from being motivated in order to make the source files shorter
this is needed when separating the APM submodules structs into
separate files.
BUG=
Review URL: https://codereview.webrtc.org/1678813002
Cr-Commit-Position: refs/heads/master@{#11611}
