52130b6412Revert 8635 "Make LS_ logging constants to match Chromium's logg..." LibjingleLoggingTests in Chromium started failing so more thought needs to be applied here. Would be good to get he perf improvement in though.
tommi@webrtc.org
2015-03-07 12:09:04 +00:00
92696cd0c6Speculative revert of 8631 "Remove lock from Bitrate() and FrameRate() in Video..."
tommi@webrtc.org
2015-03-07 09:26:16 +00:00
dc08a230daFix H.264 start code position search.
glaznev@webrtc.org
2015-03-06 23:32:20 +00:00
c2008a0e8cRTCOpenGLVideoRenderer: Add support for padded frames
magjed@webrtc.org
2015-03-06 16:58:54 +00:00
b4cd093f41Change the unintentioal CHECK to DCHECK in DtlsIdentityStore.
jiayl@webrtc.org
2015-03-06 16:31:20 +00:00
66f153f89fMake LS_ logging constants to match Chromium's logging constants when building with Chrome. This was causing logging to be done at incorrect levels and filters not work as expected.
tommi@webrtc.org
2015-03-06 15:56:30 +00:00
a2a6fe66a3Reconfigure default streams on AddRecvStream.
pbos@webrtc.org
2015-03-06 15:35:19 +00:00
bcead305a2Make the entry point for VideoFrames to webrtc const ref I420VideoFrame.
perkj@webrtc.org
2015-03-06 12:37:19 +00:00
75e850e192Enable isac NEON building on Aarch64
kjellander@webrtc.org
2015-03-06 12:27:50 +00:00
0d5ea21325Remove lock from Bitrate() and FrameRate() in VideoSender. These methods are called on the VideoSender's construction thread, which is the same thread as modifies the value of _encoder. It's therefore safe to not require a lock to access _encoder on this thread.
tommi@webrtc.org
2015-03-06 12:21:30 +00:00
f98030b029Add intermediate TextureVideoFrame typedef for Chromium
magjed@webrtc.org
2015-03-06 11:18:23 +00:00
a743f6f17fWidening memcheck suppressions for libjingle_peerconnection_unittest
kjellander@webrtc.org
2015-03-06 07:16:29 +00:00
818c4984e4Modify the simulcast encoder factory adapter to allow external encoder factories that support more than one codec.
pthatcher@webrtc.org
2015-03-06 02:20:58 +00:00
16a87b97f9Add VP9 denoiser test to videoprocessor_integrationtest.
marpan@webrtc.org
2015-03-05 22:19:00 +00:00
1d88394bcbAdd support for arbitrary array geometries in Beamformer
aluebs@webrtc.org
2015-03-05 20:38:21 +00:00
0933d01d09Enabling common_audio building with NEON on ARM64
andrew@webrtc.org
2015-03-05 19:13:46 +00:00
d7a212e8b9audio_processing/aec: Increased delay metrics aggregation window to five seconds
bjornv@webrtc.org
2015-03-05 16:14:18 +00:00
c3f15c08bcFix scoped_ptrs in bwe_simulations.
stefan@webrtc.org
2015-03-05 16:05:53 +00:00
74304330dfPrint better information during Chromium sync.
kjellander@webrtc.org
2015-03-05 14:38:09 +00:00
67a9e40286Prevent encoding frames with wrong resolution.
pbos@webrtc.org
2015-03-05 13:57:37 +00:00
03054486f5Adding basic support for posting tasks to a process thread.
tommi@webrtc.org
2015-03-05 13:13:42 +00:00
658d2015f3Allow VideoSender to be constructed on one thread but initialized and used for doing registrations, on another.
tommi@webrtc.org
2015-03-05 12:21:54 +00:00
7008f2227cRevert Clang roll in r8596 + add memcheck suppression.
kjellander@webrtc.org
2015-03-05 08:53:12 +00:00
5af41aabaeFix uninitialized variable. If FindConstraint() returns false, we check |value| in two places and at that point, it can hold an uninitialized value. Caught by Linux Memcheck builder.
tommi@webrtc.org
2015-03-05 08:42:01 +00:00
fa67463d37skip isac_neon if neon is not supported
andrew@webrtc.org
2015-03-05 06:07:25 +00:00
bbce5efaa6Turn on IPv6 for WebRTC as default as required before ramping the experiment to 30%.
guoweis@webrtc.org
2015-03-05 04:38:29 +00:00
d43b2c098dRevert "Turn on IPv6 for WebRTC as default as required before ramping the experiment to 30%."
guoweis@webrtc.org
2015-03-05 04:03:10 +00:00
86c33e3a94Turn on IPv6 for WebRTC as default as required before ramping the experiment to 30%.
guoweis@webrtc.org
2015-03-05 03:40:08 +00:00
4536289353Add CVO support to RTP sender side.
guoweis@webrtc.org
2015-03-04 22:55:15 +00:00
61e00b0bcaCreate a in-memory DTLS identity store that keeps a free identity generated in the background.
jiayl@webrtc.org
2015-03-04 22:17:38 +00:00
6daacbc8aeSet cpu_speed parameter for low resolutions, for non-simulcast.
marpan@webrtc.org
2015-03-04 21:47:06 +00:00
7b93ea1667Remove DCHECK from common_types.cc
kjellander@webrtc.org
2015-03-04 20:09:40 +00:00
4a4e688438Remove dependecy on win32.h in criticalsection.h. This was causing build errors that we haven't fully figured out yet but somehow this caused override files to include the files they're supposed to override, which in turn included webrtc build files that then conflict with Chromium's configuration.
tommi@webrtc.org
2015-03-04 20:09:37 +00:00
f7abb12aa9Fix OVERRIDE->override again after reverting video frame cl.
tommi@webrtc.org
2015-03-04 17:43:14 +00:00
c86bbbaa93Add speech flag to EncodedInfo
henrik.lundin@webrtc.org
2015-03-04 16:02:42 +00:00
92f4018d80Start using std::map for Values in the statscollector. This is in preparaton for more work which will cut down on the string copying work we do.
tommi@webrtc.org
2015-03-04 15:25:19 +00:00
fc2f146af2Revert "Turn on IPv6 for WebRTC as default as required before ramping the experiment to 30%."
guoweis@webrtc.org
2015-03-04 04:50:15 +00:00
7bea1ffe77Expose negotiated ciphers through stats API.
pthatcher@webrtc.org
2015-03-04 01:38:30 +00:00
be77872d2cRevert "Create a in-memory DTLS identity store that keeps a free identity generated in the background."
jiayl@webrtc.org
2015-03-04 00:18:51 +00:00
bbbdeed2bfTurn on IPv6 for WebRTC as default as required before ramping the experiment to 30%.
guoweis@webrtc.org
2015-03-03 23:27:17 +00:00
369f68255fCreate a in-memory DTLS identity store that keeps a free identity generated in the background.
jiayl@webrtc.org
2015-03-03 23:13:54 +00:00
c8895aa2f3Unify underlying frame buffer in I420VideoFrame and WebRtcVideoFrame
magjed@webrtc.org
2015-03-03 21:21:28 +00:00
8ad96605c1Revert "Create a in-memory DTLS identity store that keeps a free identity generated in the background."
jiayl@webrtc.org
2015-03-03 20:34:38 +00:00
bcef431902Revert r8577 "Collapse AudioEncoderDecoderIsacRed into ..."
henrik.lundin@webrtc.org
2015-03-03 20:13:11 +00:00
1fc28f2305Collapse AudioEncoderDecoderIsacRed into AudioEncoderDecoderIsac
henrik.lundin@webrtc.org
2015-03-03 19:30:45 +00:00
df512cc8b7Create a in-memory DTLS identity store that keeps a free identity generated in the background.
jiayl@webrtc.org
2015-03-03 16:41:45 +00:00
982cd2a94cFilter receiver-side DataCountersUpdated on SSRC.
pbos@webrtc.org
2015-03-03 15:56:56 +00:00
b144b4b74eFixed bug in SendTimeHistory, where deleting packets via the getter would not update the oldest suence number.
sprang@webrtc.org
2015-03-03 15:44:15 +00:00
0561716ae2Adding Opus DTX support in ACM.
minyue@webrtc.org
2015-03-03 12:02:30 +00:00
a1c9803e32Fix crash in setPictureSize on Galaxy Nexus. This cl tries to find the best supported pictureSize before setting it. BUG=4197 R=magjed@webrtc.org
perkj@webrtc.org
2015-03-03 10:54:07 +00:00
1d82813961Reland "Fix CVO in androidvideocapturer".
perkj@webrtc.org
2015-03-03 06:44:06 +00:00
c9ce07ed87Add Config option to enable 48kHz support in AudioProcessing
aluebs@webrtc.org
2015-03-02 20:07:31 +00:00
0482d01902Implement TraceCallback in a nested class of WebRtcVideoEngine. This is to fix a race that occurs in unit tests when the tests inherit from the engine class that also implements the callback interface for tracing. If tracing happens while the most derived class is still being constructed, we're in trouble.
tommi@webrtc.org
2015-03-02 17:51:27 +00:00
97ed2a4b70I420VideoFrame: Remove function ResetSize
magjed@webrtc.org
2015-03-02 17:33:26 +00:00
43f4a47c28Add more Android peer connection client unit tests:
glaznev@webrtc.org
2015-03-02 17:32:06 +00:00
976c0f3043audio_processing/aec: NEON code should not be invoked if it is detectable, but is not NEON
bjornv@webrtc.org
2015-03-02 16:25:08 +00:00
48ac226b9aAdd support for writing h264 decoder input to file and parsing interleaved length/packet RTP dumps.
stefan@webrtc.org
2015-03-02 16:18:56 +00:00
3fe17d1598Adjust a few thresholds for VP9 tests.
marpan@webrtc.org
2015-03-02 15:34:00 +00:00
fd33293d58I420VideoFrame: Remove functions set_width and set_height
magjed@webrtc.org
2015-03-02 13:57:22 +00:00
25dd1dbb9fFixed bug in test frame generator, causing incorrect reuse of frame object, in turn causing performance regression.
sprang@webrtc.org
2015-03-02 11:55:45 +00:00
60f9d6f959Revert "Add default implementation to VideoSourceInterface." Chrome test mock has been updated so VideoSourceInterface can now be pure virtual again. This reverts commit ed8d52378c43a7a93e0d2ca586486ca06db9eabe.
perkj@webrtc.org
2015-03-02 11:33:20 +00:00
afa6d16a05Add a ToString() method to StatsReport::Value. This is an interface change only at this point which will be followed up by a matching change in Chromium that removes the dependency on the 'value' member variable. Once that's been done, I'll add native support for non-string types in the Value class.
tommi@webrtc.org
2015-03-02 11:27:48 +00:00
50b2295091cricket::VideoFrameFactory: Don't overwrite frames in use
magjed@webrtc.org
2015-03-02 10:03:47 +00:00
24485eb3ccRemove last pieces of libjingle_unittest
kjellander@webrtc.org
2015-03-02 09:40:44 +00:00
5cd6828ee6Remove stale isolate files.
kjellander@webrtc.org
2015-03-02 09:37:35 +00:00
f35e4bc694Introduce a send time history class, keeping track of packet send times.
sprang@webrtc.org
2015-03-02 09:05:47 +00:00
59ae5ff310Filter logic for ip leak misses ::ffff:0.0.0.0
guoweis@webrtc.org
2015-03-01 23:45:16 +00:00
2f6ae0de5baudio_coding/codec/ilbc: Removed usage of WEBRTC_SPL_MUL_16_16_RSFT
bjornv@webrtc.org
2015-03-01 19:50:41 +00:00
e1b84a0b2bFix a race reported by tsan. TSAN complains about this variable not having synchronized access, so I'm using atomic operations on it instead. There's no functional difference really though.
tommi@webrtc.org
2015-03-01 08:29:23 +00:00
d68fa65d76Improve cleaning for Android demo applications
kjellander@webrtc.org
2015-02-28 11:17:33 +00:00
f7bb6e723bUse new API from BoringSSL to get RFC name of cipher.
pthatcher@webrtc.org
2015-02-28 01:41:07 +00:00
d31250518aTest to try to track down the alignment problem on Mac 10.9. There's no code change here, I'm rearranging member variables of the trace class and adding a sizeof check to the CriticalSection class + alignment attribute for the mutex, on Mac only.
tommi@webrtc.org
2015-02-28 00:01:11 +00:00