[FFmpeg-devel] [PATCH] RTP/Vorbis payload implementation (GSoC qual task)

[Colin McQuillan]

2009/4/12 Ronald S. Bultje <rsbultje at gmail.com>:
> Hi Colin,
>
> On Sun, Apr 12, 2009 at 2:27 AM, Colin McQuillan <m.niloc at googlemail.com> wrote:
>> This patch implements Vorbis over RTP.
>
>> + ? ? ? ?// remove the protocol identifier..
>> + ? ? ? ?while (*p && *p == ' ')
>> + ? ? ? ? ? ?p++; ? ? ? ? ? ? ? ?// strip spaces.
>> + ? ? ? ?while (*p && *p != ' ')
>> + ? ? ? ? ? ?p++; ? ? ? ? ? ? ? ?// eat protocol identifier
>> + ? ? ? ?while (*p && *p == ' ')
>> + ? ? ? ? ? ?p++; ? ? ? ? ? ? ? ?// strip trailing spaces.
>
> We (you, if we can all agree on a location before the deadline) should
> export get_word(), get_word_sep() and skip_spaces() so that all rtp
> modules can share it with rtsp.c.
>
>> + ? ?if (fragmented != 0 || vdt != 0 || num_pkts != 1) {
>> + ? ? ? ?av_log(ctx, AV_LOG_ERROR,
>> + ? ? ? ? ? ? ? "Unimplemented RTP Vorbis packet settings (%d,%d,%d)\n",
>> + ? ? ? ? ? ? ? fragmented, vdt, num_pkts);
>> + ? ? ? ?return -1;
>> + ? ?}
>
> I won't ask this for the qualification task, because the task was to
> play that Feng stream, but it'd be nice to eventually support
> fragmented packets also, it's likely going to be used eventually for
> larger packets over UDP.
>
>> On my sample ogg my patched ffmpeg stutters but I think that's just Feng
>
> Did you test over UDP or TCP?

UDP. Using Gstreamer and TCP does sound better, for example
gst-launch-0.10 rtspsrc
location=rtsp://localhost:8554/knalgan_theme.ogg 'protocols=TCP
interleaved mode' ! decodebin ! autoaudiosink

Do you know whether ffplay can use RTSP over TCP with Feng?

--
Colin McQuillan