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- int ret; - if (!frame) { fprintf(stderr, "Error allocating an audio frame\n"); exit(1); @@ -232,8 +230,7 @@ static AVFrame *alloc_audio_frame(enum AVSampleFormat sample_fmt, frame->nb_samples = nb_samples; if (nb_samples) { - ret = av_frame_get_buffer(frame, 0); - if (ret < 0) { + if (av_frame_get_buffer(frame, 0) < 0) { fprintf(stderr, "Error allocating an audio buffer\n"); exit(1); } From patchwork Sat Oct 22 10:49:20 2022 Content-Type: text/plain; charset="utf-8" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit X-Patchwork-Submitter: Jun Zhao X-Patchwork-Id: 38906 Delivered-To: ffmpegpatchwork2@gmail.com Received: by 2002:a05:6a20:4a86:b0:9d:28a3:170e with SMTP id fn6csp1227929pzb; Sat, 22 Oct 2022 03:50:19 -0700 (PDT) X-Google-Smtp-Source: AMsMyM5BsdDL+lV2WJLIXsNuU60OgpNrST5HPS0wd870awxzvzUfdo1rPtpiDuBwz57oTLVgkbmw X-Received: by 2002:a17:906:9bca:b0:78d:bc5a:9137 with SMTP id de10-20020a1709069bca00b0078dbc5a9137mr20527093ejc.25.1666435819246; Sat, 22 Oct 2022 03:50:19 -0700 (PDT) ARC-Seal: i=1; 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Signed-off-by: Jun Zhao --- doc/protocols.texi | 64 ++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 64 insertions(+) diff --git a/doc/protocols.texi b/doc/protocols.texi index 0df38d790c..5e9198e67c 100644 --- a/doc/protocols.texi +++ b/doc/protocols.texi @@ -1178,6 +1178,59 @@ Options can be set on the @command{ffmpeg}/@command{ffplay} command line, or set in code via @code{AVOption}s or in @code{avformat_open_input}. +@subsection Muxer +The following options are supported. + +@table @option +@item rtsp_transport +Set RTSP transport protocols. + +It accepts the following values: +@table @samp +@item udp +Use UDP as lower transport protocol. + +@item tcp +Use TCP (interleaving within the RTSP control channel) as lower +transport protocol. +@end table + +Default value is @samp{0}. + +@item rtsp_flags +Set RTSP flags. + +The following values are accepted: +@table @samp +@item latm +Use MP4A-LATM packetization instead of MPEG4-GENERIC for AAC. +@item rfc2190 +Use RFC 2190 packetization instead of RFC 4629 for H.263. +@item skip_rtcp +Don't send RTCP sender reports. +@item h264_mode0 +Use mode 0 for H.264 in RTP. +@item send_bye +Send RTCP BYE packets when finishing. +@end table + +Default value is @samp{0}. + + +@item min_port +Set minimum local UDP port. Default value is 5000. + +@item max_port +Set maximum local UDP port. Default value is 65000. + +@item buffer_size +Set the maximum socket buffer size in bytes. + +@item pkt_size +Set max send packet size (in bytes). Default value is 1472. +@end table + +@subsection Demuxer The following options are supported. @table @option @@ -1203,6 +1256,10 @@ Use UDP multicast as lower transport protocol. @item http Use HTTP tunneling as lower transport protocol, which is useful for passing proxies. + +@item https +Use HTTPs tunneling as lower transport protocol, which is useful for +passing proxies and widely used for security consideration. @end table Multiple lower transport protocols may be specified, in that case they are @@ -1220,6 +1277,9 @@ Accept packets only from negotiated peer address and port. Act as a server, listening for an incoming connection. @item prefer_tcp Try TCP for RTP transport first, if TCP is available as RTSP RTP transport. +@item satip_raw +Export raw MPEG-TS stream instead of demuxing. The flag will simply write out +the raw stream, with the original PAT/PMT/PIDs intact. @end table Default value is @samp{none}. @@ -1232,6 +1292,7 @@ The following flags are accepted: @item video @item audio @item data +@item subtitle @end table By default it accepts all media types. @@ -1256,6 +1317,9 @@ Set socket TCP I/O timeout in microseconds. @item user_agent Override User-Agent header. If not specified, it defaults to the libavformat identifier string. + +@item buffer_size +Set the maximum socket buffer size in bytes. @end table When receiving data over UDP, the demuxer tries to reorder received packets