[FFmpeg-devel] [avfomat/rtp: source ips lost when specified as URL options] Patch for ffmpeg using rtp protocol where sources option is not retained

[Ross Nicholson]

Hey Jun Li,

I noticed you have submitted some patches which work around the same code
area's that I submitted for. Your patches look quite tidy and well thought
out so I was wondering if you could look at this patch and see if I'm going
about it in the right way.

I'm not sure this area of ffmpeg currently has a maintainer currently so
the patches may be difficult to progress.

Ross

On Tue, 11 Feb 2020 at 22:42, Ross Nicholson <phunkyfish at gmail.com> wrote:

> The patch was created as a workaround to an issue from in kodi (apologies,
> it's a rather long thread):
> https://forum.kodi.tv/showthread.php?tid=350901&pid=2923550#pid2923550
>
> As an example, here is a URL: rtp://87.141.215.251@232.0.10.234:10000
>
> Taking this URL we should be able to either reformat it to: rtp://
> 232.0.10.234:10000?sources=87.141.215.251 or pass the sources as an
> av_dict to avfomat_open_input.
>
> Neither option works however. Instead the above workaround was created but
> it's not really the right way to fix this. Would be great to get some
> guidance on the right place to fix this in the right way.
>
> Thanks in advance.
>
> On Tue, 11 Feb 2020 at 22:30, phunkyfish <phunkyfish at gmail.com> wrote:
>
>> ---
>>  libavformat/rtsp.c | 26 ++++++++++++++++++++++++--
>>  1 file changed, 24 insertions(+), 2 deletions(-)
>>
>> diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c
>> index 859defa592..f922055134 100644
>> --- a/libavformat/rtsp.c
>> +++ b/libavformat/rtsp.c
>> @@ -2334,7 +2334,9 @@ static int sdp_read_header(AVFormatContext *s)
>>      RTSPStream *rtsp_st;
>>      int size, i, err;
>>      char *content;
>> +    const char *p, *sp="", *sources="", *sp2, *sources2;
>>      char url[1024];
>> +    char sources_buf[1024];
>>
>>      if (!ff_network_init())
>>          return AVERROR(EIO);
>> @@ -2360,6 +2362,16 @@ static int sdp_read_header(AVFormatContext *s)
>>      av_freep(&content);
>>      if (err) goto fail;
>>
>> +    /* Search for sources= tag in original URL for rtp protocol only */
>> +    if (strncmp(s->url, "rtp://", 6) == 0) {
>> +        p = strchr(s->url, '?');
>> +        if (p && av_find_info_tag(sources_buf, sizeof(sources_buf),
>> "sources", p)) {
>> +            /* av_log(s, AV_LOG_VERBOSE, "sdp_read_header found sources
>> %s\n", sources_buf);  */
>> +            sp = sources_buf;
>> +            sources = "&sources=";
>> +        }
>> +    }
>> +
>>      /* open each RTP stream */
>>      for (i = 0; i < rt->nb_rtsp_streams; i++) {
>>          char namebuf[50];
>> @@ -2377,12 +2389,22 @@ static int sdp_read_header(AVFormatContext *s)
>>                  av_dict_free(&opts);
>>                  goto fail;
>>              }
>> +
>> +            /* Prepare to add sources to the url to be opened.
>> +               Otherwise the join to the source specific muliticast will
>> be missing */
>> +            sources2 = sources;
>> +            sp2 = sp;
>> +            /* ignore sources from original URL, when sources are
>> already set in rtsp_st */
>> +            if (rtsp_st->nb_include_source_addrs > 0)
>> +                sources2 = sp2 = "";
>> +
>>              ff_url_join(url, sizeof(url), "rtp", NULL,
>>                          namebuf, rtsp_st->sdp_port,
>> -
>> "?localport=%d&ttl=%d&connect=%d&write_to_source=%d",
>> +
>> "?localport=%d&ttl=%d&connect=%d&write_to_source=%d%s%s",
>>                          rtsp_st->sdp_port, rtsp_st->sdp_ttl,
>>                          rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0,
>> -                        rt->rtsp_flags & RTSP_FLAG_RTCP_TO_SOURCE ? 1 :
>> 0);
>> +                        rt->rtsp_flags & RTSP_FLAG_RTCP_TO_SOURCE ? 1 :
>> 0,
>> +                        sources2, sp2);
>>
>>              append_source_addrs(url, sizeof(url), "sources",
>>                                  rtsp_st->nb_include_source_addrs,
>> --
>> 2.20.1 (Apple Git-117)
>>
>>