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webrtc_m130/media/base
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Florent Castelli f4673f97ed Move webrtc::AudioDeviceModule include to api/ folder 
Bug: webrtc:15874
Change-Id: I5bdb19d5e710838b41e6ca283d406c9f1f21286b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/348060
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42137}
2024-04-22 08:56:31 +00:00
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adapted_video_track_source.cc
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adapted_video_track_source.h
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audio_source.h
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codec_unittest.cc
Document how codec comparisons happen
2024-02-20 16:38:51 +00:00
codec.cc
Sync definitions of IsSameCodecSpecific
2024-02-20 14:27:28 +00:00
codec.h
Document how codec comparisons happen
2024-02-20 16:38:51 +00:00
fake_frame_source.cc
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fake_frame_source.h
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fake_media_engine.cc
Rename Set(Send|Recv)Parameters Set(Sender|Receiver)Parameters
2023-09-01 08:12:55 +00:00
fake_media_engine.h
Move webrtc::AudioProcessing include to api/ folder
2024-04-20 07:02:50 +00:00
fake_network_interface.h
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fake_rtp.cc
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fake_rtp.h
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fake_video_renderer.cc
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fake_video_renderer.h
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media_channel_impl.cc
Remove pointless Set*Encryptor functions
2023-12-07 13:55:52 +00:00
media_channel_impl.h
Remove pointless Set*Encryptor functions
2023-12-07 13:55:52 +00:00
media_channel.h
Move webrtc::AudioProcessing include to api/ folder
2024-04-20 07:02:50 +00:00
media_config.h
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media_constants.cc
Increase WebRTC default receive buffer size to 1MB.
2024-02-27 12:35:45 +00:00
media_constants.h
Add nonstandard x-google-per-layer-pli fmtp for enabling per-layer keyFrames in response to PLIs
2024-02-26 07:11:45 +00:00
media_engine_unittest.cc
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media_engine.cc
Reland "Make setCodecPreferences only look at receive codecs"
2024-02-26 10:52:23 +00:00
media_engine.h
Move webrtc::AudioDeviceModule include to api/ folder
2024-04-22 08:56:31 +00:00
rid_description.cc
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rid_description.h
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rtp_utils_unittest.cc
Change RtpTransport and DsctTransport to receives packets through ReceivedPacketCallback
2024-02-27 15:55:02 +00:00
rtp_utils.cc
Change RtpTransport and DsctTransport to receives packets through ReceivedPacketCallback
2024-02-27 15:55:02 +00:00
rtp_utils.h
Change RtpTransport and DsctTransport to receives packets through ReceivedPacketCallback
2024-02-27 15:55:02 +00:00
sdp_video_format_utils_unittest.cc
Add nonstandard x-google-per-layer-pli fmtp for enabling per-layer keyFrames in response to PLIs
2024-02-26 07:11:45 +00:00
sdp_video_format_utils.cc
Add nonstandard x-google-per-layer-pli fmtp for enabling per-layer keyFrames in response to PLIs
2024-02-26 07:11:45 +00:00
sdp_video_format_utils.h
Add nonstandard x-google-per-layer-pli fmtp for enabling per-layer keyFrames in response to PLIs
2024-02-26 07:11:45 +00:00
stream_params_unittest.cc
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stream_params.cc
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stream_params.h
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test_utils.cc
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test_utils.h
Remove more codec-related templating
2023-10-12 15:36:42 +00:00
turn_utils_unittest.cc
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turn_utils.cc
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turn_utils.h
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video_adapter_unittest.cc
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video_adapter.cc
Delete expired field trial WebRTC-Video-VariableStartScaleFactor
2024-04-18 15:41:42 +00:00
video_adapter.h
Delete expired field trial WebRTC-Video-VariableStartScaleFactor
2024-04-18 15:41:42 +00:00
video_broadcaster_unittest.cc
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video_broadcaster.cc
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video_broadcaster.h
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video_common_unittest.cc
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video_common.cc
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video_common.h
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video_source_base.cc
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video_source_base.h
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