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webrtc_m130/media/engine
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Johannes Kron d38a2b860b Increase the UDP receive buffer for video 
Lost packets have been seen in high-bitrate applications and increasing
the UDP receive buffer reduced the problems.

Bug: b/115713113
Change-Id: I671f528afeaea525150fdc2013f2b245778e5d16
Reviewed-on: https://webrtc-review.googlesource.com/c/107580
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25328}
2018-10-24 07:54:12 +00:00
..
adm_helpers.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
adm_helpers.h
Move ADM initialization into WebRtcVoiceEngine
2017-11-21 20:48:07 +00:00
apm_helpers_unittest.cc
Reland "Remove the HighPassFilter interface"
2018-10-23 07:44:09 +00:00
apm_helpers.cc
Remove echo_cancellation() and echo_control_mobile() interface access outside APM
2018-09-27 12:28:12 +00:00
apm_helpers.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
constants.cc
Increase the UDP receive buffer for video
2018-10-24 07:54:12 +00:00
constants.h
Increase the UDP receive buffer for video
2018-10-24 07:54:12 +00:00
convert_legacy_video_factory.cc
Revert "Delete CodecNamesEq, replaced with absl::EqualsIgnoreCase"
2018-10-23 13:21:27 +00:00
convert_legacy_video_factory.h
Export symbols needed by the Chromium component build (part 3).
2018-10-16 12:57:04 +00:00
fakewebrtccall.cc
Move FecController to RtpVideoSender.
2018-10-05 14:39:01 +00:00
fakewebrtccall.h
Move FecController to RtpVideoSender.
2018-10-05 14:39:01 +00:00
fakewebrtcdeviceinfo.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
fakewebrtcvcmfactory.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
fakewebrtcvideocapturemodule.h
Delete unused code from media/base/testutils.{cc,h}
2018-10-18 16:06:33 +00:00
fakewebrtcvideoengine.cc
Revert "Delete CodecNamesEq, replaced with absl::EqualsIgnoreCase"
2018-10-23 13:21:27 +00:00
fakewebrtcvideoengine.h
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
internaldecoderfactory_unittest.cc
Reland "Update internal video decoder factory to new interface"
2017-11-09 14:29:12 +00:00
internaldecoderfactory.cc
Revert "Delete CodecNamesEq, replaced with absl::EqualsIgnoreCase"
2018-10-23 13:21:27 +00:00
internaldecoderfactory.h
Reland "Update internal video decoder factory to new interface"
2017-11-09 14:29:12 +00:00
internalencoderfactory.cc
Revert "Delete CodecNamesEq, replaced with absl::EqualsIgnoreCase"
2018-10-23 13:21:27 +00:00
internalencoderfactory.h
Reland "Update internal encoder factory to new interface"
2017-11-18 10:48:28 +00:00
multiplexcodecfactory_unittest.cc
Reland "Rename stereo video codec to multiplex"
2018-01-29 20:37:59 +00:00
multiplexcodecfactory.cc
Revert "Delete CodecNamesEq, replaced with absl::EqualsIgnoreCase"
2018-10-23 13:21:27 +00:00
multiplexcodecfactory.h
Export symbols needed by the Chromium component build (part 3).
2018-10-16 12:57:04 +00:00
nullwebrtcvideoengine_unittest.cc
Use AudioProcessingBuilder everywhere AudioProcessing is created.
2018-01-09 13:45:20 +00:00
nullwebrtcvideoengine.h
Adds new CryptoOption crypto_options.frame.require_frame_encryption.
2018-10-17 17:44:19 +00:00
payload_type_mapper_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
payload_type_mapper.cc
Replace _stricmp with absl::EqualsIgnoreCase
2018-10-19 14:17:31 +00:00
payload_type_mapper.h
Replace rtc::Optional with absl::optional in media, ortc, p2p
2018-06-16 07:09:59 +00:00
scopedvideodecoder.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
scopedvideodecoder.h
…
scopedvideoencoder.cc
Revert "Add ability to specify if rate controller of video encoder is trusted."
2018-10-11 15:37:40 +00:00
scopedvideoencoder.h
…
simulcast_encoder_adapter_unittest.cc
Split out a separate target for SimulcastEncoderAdapter
2018-10-22 22:36:59 +00:00
simulcast_encoder_adapter.cc
Split out a separate target for SimulcastEncoderAdapter
2018-10-22 22:36:59 +00:00
simulcast_encoder_adapter.h
Split out a separate target for SimulcastEncoderAdapter
2018-10-22 22:36:59 +00:00
simulcast_unittest.cc
Add field trial for normalized simulcast size.
2018-10-15 08:05:38 +00:00
simulcast.cc
Add field trial for normalized simulcast size.
2018-10-15 08:05:38 +00:00
simulcast.h
Reland "Add initial support for RtpEncodingParameters max_framerate."
2018-09-13 10:06:33 +00:00
vp8_encoder_simulcast_proxy_unittest.cc
Move TemporalLayers to api/video_codecs.
2018-10-12 09:15:21 +00:00
vp8_encoder_simulcast_proxy.cc
Split out a separate target for SimulcastEncoderAdapter
2018-10-22 22:36:59 +00:00
vp8_encoder_simulcast_proxy.h
Revert "Add ability to specify if rate controller of video encoder is trusted."
2018-10-11 15:37:40 +00:00
webrtcmediaengine_unittest.cc
Parse two-byte header extensions.
2018-09-28 08:32:17 +00:00
webrtcmediaengine.cc
Parse two-byte header extensions.
2018-09-28 08:32:17 +00:00
webrtcmediaengine.h
Add missing ifdefs to header files for SW video codecs.
2018-08-21 16:20:23 +00:00
webrtcvideocapturer_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
webrtcvideocapturer.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
webrtcvideocapturer.h
Enable cpplint in media/
2017-10-31 17:46:42 +00:00
webrtcvideocapturerfactory.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
webrtcvideocapturerfactory.h
Enable cpplint in media/
2017-10-31 17:46:42 +00:00
webrtcvideodecoderfactory.cc
Remove double declaration of 2 conversion functions.
2018-09-11 06:00:05 +00:00
webrtcvideodecoderfactory.h
Export symbols needed by the Chromium component build (part 3).
2018-10-16 12:57:04 +00:00
webrtcvideoencoderfactory_unittest.cc
…
webrtcvideoencoderfactory.cc
Clean up some bad constructs in media/
2017-09-29 12:22:57 +00:00
webrtcvideoencoderfactory.h
Export symbols needed by the Chromium component build (part 4).
2018-10-18 08:42:22 +00:00
webrtcvideoengine_unittest.cc
Increase the UDP receive buffer for video
2018-10-24 07:54:12 +00:00
webrtcvideoengine.cc
Increase the UDP receive buffer for video
2018-10-24 07:54:12 +00:00
webrtcvideoengine.h
Media engine and channel support for per-channel dscp values, specified by RtpParameter
2018-10-19 21:47:55 +00:00
webrtcvoiceengine_unittest.cc
Delete use of STR_CASE_CMP, replaced with absl::EqualsIgnoreCase.
2018-10-23 09:24:15 +00:00
webrtcvoiceengine.cc
Delete use of STR_CASE_CMP, replaced with absl::EqualsIgnoreCase.
2018-10-23 09:24:15 +00:00
webrtcvoiceengine.h
Media engine and channel support for per-channel dscp values, specified by RtpParameter
2018-10-19 21:47:55 +00:00