Re: [Live-devel] Streaming G711 audio
It works! Thanks a lot! On Nov 15, 2024, at 2:53 AM, d.gorde...@ngslab.ru wrote: Hi, Now I have a streamer that streams video with AAC audio: audio_sink_ = MPEG4GenericRTPSink::createNew(*env_, audio_rtp_socket_.get(), payload, 16000, "audio", "AAC-hbr", "", 1); and it works well. I need also stream G711U or G711A audio. How to make it? Try the following: SimpleRTPSink::createNew(*env_, audio_rtp_socket_.get(), payload, 8000, “audio”, “PCMU”, 1); for G711U. (For G711A, replace “PCMU” with “PCMA”.) Ross Finlayson Live Networks, Inc. http://www.live555.com/ ___ live-devel mailing list live-devel@lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel ___ live-devel mailing list live-devel@lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel
[Live-devel] Streaming G711 audio
Hi, Now I have a streamer that streams video with AAC audio: audio_sink_ = MPEG4GenericRTPSink::createNew(*env_, audio_rtp_socket_.get(), payload, 16000, "audio", "AAC-hbr", "", 1); and it works well. I need also stream G711U or G711A audio. How to make it? I see there are some classes derived from AudioRTPSink, but doubt which one should I use. ___ live-devel mailing list live-devel@lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel
Re: [Live-devel] Receiving slice-based frames
It works, thanks a lot!
On Feb 22, 2025, at 1:10 AM, d.gorde...@ngslab.ru wrote:
Hello
I need to receive slice-based frames from a video server. These frames, as you
know come by slices. The last slice (packet) comes with Mark=True in the RTP
header.
I have a class which receives the frames:
class VideoSink final : public MediaSink {}
It has afterGettingFrame() method which gets the frames and works well for
usual frames. After getting frames I save them in archive and later send them
to a receiver. So, when I send them, I need to set the Mark flag to say the
receiver what is it - a intermediate slice or a last slice or a usual frame.
But I don't know nothing about the Mark, because I receive only video payload.
So, how to get this flag, or is there other method to know about which slice
comes?
In general, it's the job of the decoder to figure out how to render the
incoming NAL units - which includes deciding when a received slice is the last
slice in a video frame.
However, as a hint, you can use the value of the RTP packet's 'M' (i.e., 'marker') bit,
which is (supposed to be) set for the last RTP packet of an 'access unit' (i.e., video
frame). I.e., you can call "RTPSource::curPacketMarkerBit()" to test this.
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
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[Live-devel] Receiving slice-based frames
Hello
I need to receive slice-based frames from a video server. These frames,
as you know come by slices. The last slice (packet) comes with Mark=True
in the RTP header.
I have a class which receives the frames:
class VideoSink final : public MediaSink {}
It has afterGettingFrame() method which gets the frames and works well
for usual frames. After getting frames I save them in archive and later
send them to a receiver. So, when I send them, I need to set the Mark
flag to say the receiver what is it - a intermediate slice or a last
slice or a usual frame. But I don't know nothing about the Mark, because
I receive only video payload.
So, how to get this flag, or is there other method to know about which
slice comes?
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[Live-devel] Troubles with playing backward
I recorded a number of frames and want to play them backward (forward playing works). The problems are: video doesn't play or video stops after 10 - 20 seconds and is slower twice. How I do the playing. I read the archive backward and find previous I-frame. After I read 2 frames before it, there are SPS and PPS. After I make SEI, set all its frames a unique timestamp = I-frame timestamp, set gettimeofday(&fPresentationTime, NULL) and send them. What I tried. Merging these frames into one frame. In this case Wireshark shows only SEI and FU-A Start:PPS. I'm not sure what is it. Are there PPS and SPS in this frame alongside with I-frame? Why did I merge them? RTP timestamps are different in Wireshark. I set them the same in my code, but there are different anyways! If I play forward there the same, ok, but the backward playing changes something in Live555? RTP timestamps are increasing also. ___ live-devel mailing list live-devel@lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel
