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Very excited to see that the one way media scene has been added to the latest webrtc nv use case.
Regarding the scenarios inside, I have a few questions.
About N40, N41, it seems that what we want to add is support for pre-encoded video frames,
but does not include pre-encoded audio frames? ,N40 does not describe whether it is an audio
encoder or a video encoder?
Live streaming is a typical one way media scene, and live streaming usually requires video
to support B-frame transmission and audio to support AAC encoding. Do we need to add such
capability requirements to one way media? The advantage of adding these two capability requirements
is that when pre-encoded audio and video are injected through the one way media interface and stream
to media server via rtp then distributed through the RTMP/HTTP-FLV/HLS/DASH protocol, the media distribution side
does not need to do audio conversion (opus->aac), support Video B frames can be used to improve video quality,
which is also a common practice in streaming media scenarios, Supporting this capability requires some modification of the
protocol layer(rtp, sdp) of webrtc, but it will make webrtc more suitable for live streaming scenarios and more compatible in media part with other live streaming protocols.
The text was updated successfully, but these errors were encountered:
aboba
changed the title
A little thought about the one way media use case
Section 3.10: A little thought about the one way media use case
May 15, 2023
Very excited to see that the one way media scene has been added to the latest webrtc nv use case.
Regarding the scenarios inside, I have a few questions.
but does not include pre-encoded audio frames? ,N40 does not describe whether it is an audio
encoder or a video encoder?
to support B-frame transmission and audio to support AAC encoding. Do we need to add such
capability requirements to one way media? The advantage of adding these two capability requirements
is that when pre-encoded audio and video are injected through the one way media interface and stream
to media server via rtp then distributed through the RTMP/HTTP-FLV/HLS/DASH protocol, the media distribution side
does not need to do audio conversion (opus->aac), support Video B frames can be used to improve video quality,
which is also a common practice in streaming media scenarios, Supporting this capability requires some modification of the
protocol layer(rtp, sdp) of webrtc, but it will make webrtc more suitable for live streaming scenarios and more compatible in media part with other live streaming protocols.
The text was updated successfully, but these errors were encountered: