This CL simplifies the VideoCapturer interface from 'String getSupportedFormatsAsJson() throws JSONException' to 'List<CaptureFormat> getSupportedFormats()'. The intermediate conversion to/from a JSON string is removed, and AndroidVideoCapturerJni converts the Java list to a C++ vector directly instead.
BUG=webrtc:5519
R=perkj@webrtc.org
Review URL: https://codereview.webrtc.org/1702603002 .
Cr-Commit-Position: refs/heads/master@{#11669}
This CL adds a check to see if the return value of GLES20.glCreateShader() is zero. Also, shaders are flagged for deletion immediately after glLinkProgram() instead of doing it in release().
BUG=b/27197590
Review URL: https://codereview.webrtc.org/1702953002
Cr-Commit-Position: refs/heads/master@{#11668}
This appears to be dead code because GetTransport() is not used by WebRTC. It also adds dead code to DtlsTransportChannelWrapper and P2PTransportChannel.
BUG=
Review URL: https://codereview.webrtc.org/1691673002
Cr-Commit-Position: refs/heads/master@{#11662}
Also introduce a pair of scoped_ptr <-> unique_ptr conversion
functions. By using them judiciously, we can keep these CL:s small and
avoid having to convert enormous amounts of code at once.
BUG=webrtc:5520
Review URL: https://codereview.webrtc.org/1702983002
Cr-Commit-Position: refs/heads/master@{#11658}
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}
For audio, the flag is apparently unused. For video, the flag is moved to
VideoSendParameters, with the intention to keep only per-stream flags in
VideoOptions. The flag is used for the webrtcvideoengine2 logic commented like
// Conference mode screencast uses 2 temporal layers split at 100kbit.
// For screenshare in conference mode, tl0 and tl1 bitrates are piggybacked
// on the VideoCodec struct as target and max bitrates, respectively.
// See eg. webrtc::VP8EncoderImpl::SetRates().
BUG=webrtc:5426
Review URL: https://codereview.webrtc.org/1697163002
Cr-Commit-Position: refs/heads/master@{#11651}
In https://codereview.webrtc.org/1691463002 the legacy objc tests
were moved into a new GYP file, which was conditioned so it's not
included for platforms that cannot build it. This condition contained
an error which makes the GYP file being processed even if include_tests=0,
which causes errors in downstream code.
BUG=webrtc:5419
NOTRY=True
Review URL: https://codereview.webrtc.org/1701053005
Cr-Commit-Position: refs/heads/master@{#11650}
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}
Makes DecodesRetransmittedFrame not flake/fail due to sent padding when
probing, which is correct behavior. Also removes hack that accepted this
only during the first n packets.
BUG=
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1698343003 .
Cr-Commit-Position: refs/heads/master@{#11648}
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}
Meaning "a=msid:...", instead of "a=ssrc:X msid:...".
An additional option to SdpSerialize determines if the
"a=msid" attribute is used.
Review URL: https://codereview.webrtc.org/1688383002
Cr-Commit-Position: refs/heads/master@{#11644}
For now, the network cost is purely based on the network type (cellular has cost 0xFFFF and everything else has cost 0).
Add cost to the candidate signaling and the stun request signaling (which is needed for peer reflexive candidates).
BUG=webrtc:4325
Review URL: https://codereview.webrtc.org/1668073002
Cr-Commit-Position: refs/heads/master@{#11642}
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}
This started flaking due to allowing probes to restart if they were aborted due to insufficient packets. This is reasonable behavior.
TBR=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1701033002 .
Cr-Commit-Position: refs/heads/master@{#11638}
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}
When the input to WebRtcSpl_Sqrt was the maximum negative value
(-2147483648), the calculations would overflow. This is now solved by
nudging this particular input value one step.
BUG=webrtc:5512
Review URL: https://codereview.webrtc.org/1685743003
Cr-Commit-Position: refs/heads/master@{#11631}
Previous suppressions for libjingle_media_unittest stopped working when
the target was renamed rtc_media_unittests causing the bots to start
flaking.
BUG=
TBR=kjellander@webrtc.org
Review URL: https://codereview.webrtc.org/1698873002 .
Cr-Commit-Position: refs/heads/master@{#11624}
