[FFmpeg-devel] [PATCH] Support RTP/TCP in the RTSP muxer

[Ronald S. Bultje]

Hi,

On Mon, Mar 15, 2010 at 11:50 AM, Martin Storsj? <martin at martin.st> wrote:
> The third part actually adds support for the TCP interleaved packets.
> While things seem to work generally with darwin streaming server, some
> packets seem to get lost in some cases, which I don't totally understand
> at the moment. I don't know if this is DSS's fault or mine (the same
> setups usually work just fine over UDP).

> diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c
> +    if (handle)
>      url_fdopen(&rtpctx->pb, handle);
> +    else
> +        url_open_dyn_packet_buf(&rtpctx->pb, RTSP_TCP_MAX_PACKET_SIZE);

We use if (..) { .. } else, since it costs no extra line of code.

>      local_pkt = *pkt;
>      local_pkt.stream_index = 0;
> -    return av_write_frame(rtpctx, &local_pkt);
> +    ret = av_write_frame(rtpctx, &local_pkt);
> +    if (!ret && rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP)
> +        ret = tcp_write_packet(s, rtsp_st);
> +    return ret;
>  }

So what does this do exactly? It still writes the frame over the
non-existing UDP session, right? Should it do that? Alternatively, Can
we somehow change that so that av_write_frame() in fact internally
calls tcp_write_packet() for this case? Reason I'm asking is because
you add a lot of branching. I'm not so worried about performance, but
it's a lot of extra code that might be done otherwise.

Ronald