[FFmpeg-devel] [PATCH 1/2] doc/examples/muxing: Remove unnecessary ret

[FFmpeg-devel] [PATCH 1/2] doc/examples/muxing: Remove unnecessary ret

[Jun Zhao]

From: Jun Zhao <barryjzhao@tencent.com>

Remove unnecessary ret and make the code more compact

Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
---
 doc/examples/muxing.c | 5 +----
 1 file changed, 1 insertion(+), 4 deletions(-)

diff --git a/doc/examples/muxing.c b/doc/examples/muxing.c
index 3acb778322..cd997d5431 100644
--- a/doc/examples/muxing.c
+++ b/doc/examples/muxing.c
@@ -219,8 +219,6 @@ static AVFrame *alloc_audio_frame(enum AVSampleFormat sample_fmt,
                                   int sample_rate, int nb_samples)
 {
     AVFrame *frame = av_frame_alloc();
-    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);
         }
-- 
2.25.1

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[FFmpeg-devel] [PATCH 2/2] doc/protocols: update rtsp options

[Jun Zhao]

From: Jun Zhao <barryjzhao@tencent.com>

Split the rtsp options to muxer/demuxer, and update the options.

Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
---
 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
-- 
2.25.1

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[FFmpeg-devel] [PATCH 2/2] doc/protocols: update rtsp options

[Jun Zhao]

From: Jun Zhao <barryjzhao@tencent.com>

Split the rtsp options to muxer/demuxer, and update the options.

Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
---
 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
-- 
2.25.1

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