admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/media/engine
History
Piotr (Peter) Slatala 946b968111 Add support for target rate constraints 
WebRTC video engine now configures bitrate on media transport
correctly.

Bug: webrtc:9719
Change-Id: I85884cd76644b7eca3763cec8ce9e31b5b64db27
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127941
Commit-Queue: Peter Slatala <psla@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27167}
2019-03-18 18:54:58 +00:00
..
adm_helpers.cc
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2.
2018-11-28 18:25:07 +00:00
adm_helpers.h
Delete unneeded includes of common_types.h and gn deps on webrtc_common.
2018-11-20 16:28:39 +00:00
apm_helpers_unittest.cc
Remove rtc_base/scoped_ref_ptr.h.
2019-01-25 20:29:58 +00:00
apm_helpers.cc
Make voiceengine/audio transport stop using voice_detection() interface
2019-01-11 12:31:29 +00:00
apm_helpers.h
Make voiceengine/audio transport stop using voice_detection() interface
2019-01-11 12:31:29 +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
encoder_simulcast_proxy_unittest.cc
Update more VideoEncoder implementations to drop CodecSpecificInfo input
2019-03-07 12:26:57 +00:00
encoder_simulcast_proxy.cc
Prepare for splitting FrameType into AudioFrameType and VideoFrameType
2019-03-07 10:12:57 +00:00
encoder_simulcast_proxy.h
Prepare for splitting FrameType into AudioFrameType and VideoFrameType
2019-03-07 10:12:57 +00:00
fake_video_codec_factory.cc
Fix heap use overrun in FakeEncoder
2019-02-01 11:26:57 +00:00
fake_video_codec_factory.h
Reland "Add a high bitrate full stack test with fake codec"
2019-01-16 21:03:22 +00:00
fake_webrtc_call.cc
Propagate base minimum delay from video jitter buffer to webrtc/api.
2019-02-27 15:08:34 +00:00
fake_webrtc_call.h
Propagate base minimum delay from video jitter buffer to webrtc/api.
2019-02-27 15:08:34 +00:00
fake_webrtc_video_engine.cc
Revert "Delete CodecSpecificInfo argument from VideoDecoder::Decode"
2019-03-07 19:40:17 +00:00
fake_webrtc_video_engine.h
Revert "Delete CodecSpecificInfo argument from VideoDecoder::Decode"
2019-03-07 19:40:17 +00:00
internal_decoder_factory_unittest.cc
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
2019-01-11 17:11:39 +00:00
internal_decoder_factory.cc
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
2019-01-11 17:11:39 +00:00
internal_decoder_factory.h
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
2019-01-11 17:11:39 +00:00
internal_encoder_factory.cc
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
2019-01-11 17:11:39 +00:00
internal_encoder_factory.h
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
2019-01-11 17:11:39 +00:00
multiplex_codec_factory_unittest.cc
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
2019-01-11 17:11:39 +00:00
multiplex_codec_factory.cc
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
2019-01-11 17:11:39 +00:00
multiplex_codec_factory.h
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
2019-01-11 17:11:39 +00:00
null_webrtc_video_engine_unittest.cc
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
2019-01-11 17:11:39 +00:00
null_webrtc_video_engine.h
Remove legacy video codec factories.
2019-01-14 14:56:40 +00:00
payload_type_mapper_unittest.cc
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
2019-01-11 17:11:39 +00:00
payload_type_mapper.cc
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
2019-01-11 17:11:39 +00:00
payload_type_mapper.h
Replace rtc::Optional with absl::optional in media, ortc, p2p
2018-06-16 07:09:59 +00:00
simulcast_encoder_adapter_unittest.cc
Update more VideoEncoder implementations to drop CodecSpecificInfo input
2019-03-07 12:26:57 +00:00
simulcast_encoder_adapter.cc
Prepare for splitting FrameType into AudioFrameType and VideoFrameType
2019-03-07 10:12:57 +00:00
simulcast_encoder_adapter.h
Prepare for splitting FrameType into AudioFrameType and VideoFrameType
2019-03-07 10:12:57 +00:00
simulcast_unittest.cc
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
2019-01-11 17:11:39 +00:00
simulcast.cc
Move some video codec constants to separate file.
2019-01-17 15:29:53 +00:00
simulcast.h
Implement the encoding RtpParameter scaleResolutionDownBy
2019-01-29 14:32:17 +00:00
unhandled_packets_buffer_unittest.cc
Improve handling of packets with unknown ssrc.
2019-03-18 11:34:43 +00:00
unhandled_packets_buffer.cc
Improve handling of packets with unknown ssrc.
2019-03-18 11:34:43 +00:00
unhandled_packets_buffer.h
Improve handling of packets with unknown ssrc.
2019-03-18 11:34:43 +00:00
webrtc_media_engine_unittest.cc
Always send abs-send-time when negotiated and do not filter it out.
2019-01-23 11:30:39 +00:00
webrtc_media_engine.cc
Use Abseil container algorithms in media/
2019-01-29 02:35:50 +00:00
webrtc_media_engine.h
Remove legacy video codec factories.
2019-01-14 14:56:40 +00:00
webrtc_video_engine_unittest.cc
Add support for target rate constraints
2019-03-18 18:54:58 +00:00
webrtc_video_engine.cc
Add support for target rate constraints
2019-03-18 18:54:58 +00:00
webrtc_video_engine.h
Improve handling of packets with unknown ssrc.
2019-03-18 11:34:43 +00:00
webrtc_voice_engine_unittest.cc
Increase the default maximum jitter buffer size to 200 packets.
2019-03-15 10:25:11 +00:00
webrtc_voice_engine.cc
Increase the default maximum jitter buffer size to 200 packets.
2019-03-15 10:25:11 +00:00
webrtc_voice_engine.h
Increase the default maximum jitter buffer size to 200 packets.
2019-03-15 10:25:11 +00:00