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3e9af7fe05
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Insert audio frame transformer between depacketizer and decoder.
The frame transformer is passed from RTPReceiverInterface through the
library to be eventually set in ChannelReceive, where the frame
transformation will occur in the follow-up CL.
Insertable Streams Web API explainer:
https://github.com/alvestrand/webrtc-media-streams/blob/master/explainer.md
Design doc for WebRTC library changes:
http://doc/1eiLkjNUkRy2FssCPLUp6eH08BZuXXoHfbbBP1ZN7EVk
Bug: webrtc:11380
Change-Id: I5af06d1431047ef50d00e304cf95e92a832b4220
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171872
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30956}
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2020-04-01 08:15:53 +00:00 |
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934afc6ba1
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Deprecate RtpReceiver's SetParameters method
This removes the SetParameters method from AudioRtpReceiver and Video
RtpReceiver, which is currently not used and is not part of the
specifications.
Bug: webrtc:11111
Change-Id: I6f67773bfef2d4b51e9ab670bde17b5fbf5f94c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159307
Reviewed-by: Patrik Höglund <phoglund@google.com>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Daniela Jovanoska Petrenko <denicija@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Saurav Das <dinosaurav@chromium.org>
Cr-Commit-Position: refs/heads/master@{#29995}
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2019-12-03 19:50:42 +00:00 |
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7262fc29a0
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Refactor Rtp Receivers to accept SSRC 0.
Changes Rtp Receivers to use a null value of ssrc to mean a default
receive stream.
Bug: webrtc:8694
Change-Id: I835199345f7add993b9078c8b0e7988d5cdd6646
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152425
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Saurav Das <dinosaurav@chromium.org>
Cr-Commit-Position: refs/heads/master@{#29201}
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2019-09-16 21:29:58 +00:00 |
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a4d873786f
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Format almost everything.
This CL was generated by running
git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \
grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \
grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \
grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \
grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \
| xargs clang-format -i ; git cl format
Most of these changes are clang-format grouping and reordering includes
differently.
Bug: webrtc:9340
Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28505}
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2019-07-08 13:45:15 +00:00 |
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428dcb2517
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Remove SetLatency/GetLatency from MediaSourceInterface API level
Bug: webrtc:10287
Change-Id: I74fad31db98b75791085688438064f9510b0b6fa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133165
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27692}
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2019-04-18 19:11:31 +00:00 |
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4bac79ece2
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Add SetJitterBufferMinimumDelay method to RtpReceiverInterface.
This change is required to allow modification of Jitter Buffer delay
in javascript via Origin Trial Experiment.
Link to experiment description:
https://groups.google.com/a/chromium.org/forum/#!topic/blink-dev/Tgm4qiNepJc
Bug: webrtc:10287
Change-Id: I4f21380aad5982a4a60c55683b5173ce72ce0392
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131144
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27444}
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2019-04-04 09:00:16 +00:00 |
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501bfba0cb
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Split rtp_receiver for readability.
Bug: webrtc:10304
Change-Id: I85e421060d8560cf36cdb5970ae190efc77e7709
Reviewed-on: https://webrtc-review.googlesource.com/c/122085
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Cr-Commit-Position: refs/heads/master@{#26634}
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2019-02-11 12:47:51 +00:00 |
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