diff options
author | Michael Niedermayer <michaelni@gmx.at> | 2012-08-09 02:26:38 +0400 |
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committer | Michael Niedermayer <michaelni@gmx.at> | 2012-08-09 02:51:02 +0400 |
commit | 11a1033c9fcae380f4da06b2b0253ab0eb82b026 (patch) | |
tree | cee0379ce616e16ef342622b77fe417fa604c014 /libavformat/rtmpproto.c | |
parent | 4270d8c04d354b928d2329abd1037c6f0a72c07f (diff) | |
parent | 5864eb427f2f05342136f3bc9727d826e68d8dbf (diff) |
Merge remote-tracking branch 'qatar/master'
* qatar/master: (23 commits)
build: cosmetics: Reorder some lists in a more logical fashion
x86: pngdsp: Fix assembly for OS/2
fate: add test for RTjpeg in nuv with frameheader
rtmp: send check_bw as notification
g723_1: clip argument for 15-bit version of normalize_bits()
g723_1: use all LPC vectors in formant postfilter
id3v2: Support v2.2 PIC
avplay: fix build with lavfi disabled.
avconv: split configuring filter configuration to a separate file.
avconv: split option parsing into a separate file.
mpc8: do not leave padding after last frame in buffer for the next decode call
mpegaudioenc: list supported channel layouts.
mpegaudiodec: don't print an error on > 1 frame in a packet.
api-example: update to new audio encoding API.
configure: add --enable/disable-random option
doc: cygwin: Update list of FATE package requirements
build: Remove all installed headers and header directories on uninstall
build: change checkheaders to use regular build rules
rtmp: Add a new option 'rtmp_subscribe'
rtmp: Add support for subscribing live streams
...
Conflicts:
Makefile
common.mak
configure
doc/examples/decoding_encoding.c
ffmpeg.c
libavcodec/g723_1.c
libavcodec/mpegaudiodec.c
libavcodec/x86/pngdsp.asm
libavformat/version.h
library.mak
tests/fate/video.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Diffstat (limited to 'libavformat/rtmpproto.c')
-rw-r--r-- | libavformat/rtmpproto.c | 169 |
1 files changed, 64 insertions, 105 deletions
diff --git a/libavformat/rtmpproto.c b/libavformat/rtmpproto.c index 11ffca93d4..4120aa6b21 100644 --- a/libavformat/rtmpproto.c +++ b/libavformat/rtmpproto.c @@ -91,6 +91,7 @@ typedef struct RTMPContext { char* flashver; ///< version of the flash plugin char* swfurl; ///< url of the swf player char* pageurl; ///< url of the web page + char* subscribe; ///< name of live stream to subscribe int server_bw; ///< server bandwidth int client_buffer_time; ///< client buffer time in ms int flush_interval; ///< number of packets flushed in the same request (RTMPT only) @@ -572,7 +573,7 @@ static int gen_check_bw(URLContext *s, RTMPContext *rt) p = pkt.data; ff_amf_write_string(&p, "_checkbw"); - ff_amf_write_number(&p, ++rt->nb_invokes); + ff_amf_write_number(&p, RTMP_NOTIFICATION); ff_amf_write_null(&p); ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, @@ -604,6 +605,30 @@ static int gen_bytes_read(URLContext *s, RTMPContext *rt, uint32_t ts) return ret; } +static int gen_fcsubscribe_stream(URLContext *s, RTMPContext *rt, + const char *subscribe) +{ + RTMPPacket pkt; + uint8_t *p; + int ret; + + if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, + 0, 27 + strlen(subscribe))) < 0) + return ret; + + p = pkt.data; + ff_amf_write_string(&p, "FCSubscribe"); + ff_amf_write_number(&p, ++rt->nb_invokes); + ff_amf_write_null(&p); + ff_amf_write_string(&p, subscribe); + + ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, + rt->prev_pkt[1]); + ff_rtmp_packet_destroy(&pkt); + + return ret; +} + int ff_rtmp_calc_digest(const uint8_t *src, int len, int gap, const uint8_t *key, int keylen, uint8_t *dst) { @@ -1011,6 +1036,20 @@ static int handle_invoke(URLContext *s, RTMPPacket *pkt) } if ((ret = gen_create_stream(s, rt)) < 0) return ret; + + if (rt->is_input) { + /* Send the FCSubscribe command when the name of live + * stream is defined by the user or if it's a live stream. */ + if (rt->subscribe) { + if ((ret = gen_fcsubscribe_stream(s, rt, + rt->subscribe)) < 0) + return ret; + } else if (rt->live == -1) { + if ((ret = gen_fcsubscribe_stream(s, rt, + rt->playpath)) < 0) + return ret; + } + } break; case STATE_FCPUBLISH: rt->state = STATE_CONNECTING; @@ -1593,115 +1632,35 @@ static const AVOption rtmp_options[] = { {"recorded", "recorded stream", 0, AV_OPT_TYPE_CONST, {0}, 0, 0, DEC, "rtmp_live"}, {"rtmp_pageurl", "URL of the web page in which the media was embedded. By default no value will be sent.", OFFSET(pageurl), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC}, {"rtmp_playpath", "Stream identifier to play or to publish", OFFSET(playpath), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC}, + {"rtmp_subscribe", "Name of live stream to subscribe to. Defaults to rtmp_playpath.", OFFSET(subscribe), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC}, {"rtmp_swfurl", "URL of the SWF player. By default no value will be sent", OFFSET(swfurl), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC}, {"rtmp_tcurl", "URL of the target stream. Defaults to proto://host[:port]/app.", OFFSET(tcurl), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC}, { NULL }, }; -static const AVClass rtmp_class = { - .class_name = "rtmp", - .item_name = av_default_item_name, - .option = rtmp_options, - .version = LIBAVUTIL_VERSION_INT, -}; - -URLProtocol ff_rtmp_protocol = { - .name = "rtmp", - .url_open = rtmp_open, - .url_read = rtmp_read, - .url_write = rtmp_write, - .url_close = rtmp_close, - .priv_data_size = sizeof(RTMPContext), - .flags = URL_PROTOCOL_FLAG_NETWORK, - .priv_data_class= &rtmp_class, +#define RTMP_PROTOCOL(flavor) \ +static const AVClass flavor##_class = { \ + .class_name = #flavor, \ + .item_name = av_default_item_name, \ + .option = rtmp_options, \ + .version = LIBAVUTIL_VERSION_INT, \ +}; \ + \ +URLProtocol ff_##flavor##_protocol = { \ + .name = #flavor, \ + .url_open = rtmp_open, \ + .url_read = rtmp_read, \ + .url_write = rtmp_write, \ + .url_close = rtmp_close, \ + .priv_data_size = sizeof(RTMPContext), \ + .flags = URL_PROTOCOL_FLAG_NETWORK, \ + .priv_data_class= &flavor##_class, \ }; -static const AVClass rtmpe_class = { - .class_name = "rtmpe", - .item_name = av_default_item_name, - .option = rtmp_options, - .version = LIBAVUTIL_VERSION_INT, -}; - -URLProtocol ff_rtmpe_protocol = { - .name = "rtmpe", - .url_open = rtmp_open, - .url_read = rtmp_read, - .url_write = rtmp_write, - .url_close = rtmp_close, - .priv_data_size = sizeof(RTMPContext), - .flags = URL_PROTOCOL_FLAG_NETWORK, - .priv_data_class = &rtmpe_class, -}; -static const AVClass rtmps_class = { - .class_name = "rtmps", - .item_name = av_default_item_name, - .option = rtmp_options, - .version = LIBAVUTIL_VERSION_INT, -}; - -URLProtocol ff_rtmps_protocol = { - .name = "rtmps", - .url_open = rtmp_open, - .url_read = rtmp_read, - .url_write = rtmp_write, - .url_close = rtmp_close, - .priv_data_size = sizeof(RTMPContext), - .flags = URL_PROTOCOL_FLAG_NETWORK, - .priv_data_class = &rtmps_class, -}; - -static const AVClass rtmpt_class = { - .class_name = "rtmpt", - .item_name = av_default_item_name, - .option = rtmp_options, - .version = LIBAVUTIL_VERSION_INT, -}; - -URLProtocol ff_rtmpt_protocol = { - .name = "rtmpt", - .url_open = rtmp_open, - .url_read = rtmp_read, - .url_write = rtmp_write, - .url_close = rtmp_close, - .priv_data_size = sizeof(RTMPContext), - .flags = URL_PROTOCOL_FLAG_NETWORK, - .priv_data_class = &rtmpt_class, -}; - -static const AVClass rtmpte_class = { - .class_name = "rtmpte", - .item_name = av_default_item_name, - .option = rtmp_options, - .version = LIBAVUTIL_VERSION_INT, -}; - -URLProtocol ff_rtmpte_protocol = { - .name = "rtmpte", - .url_open = rtmp_open, - .url_read = rtmp_read, - .url_write = rtmp_write, - .url_close = rtmp_close, - .priv_data_size = sizeof(RTMPContext), - .flags = URL_PROTOCOL_FLAG_NETWORK, - .priv_data_class = &rtmpte_class, -}; - -static const AVClass rtmpts_class = { - .class_name = "rtmpts", - .item_name = av_default_item_name, - .option = rtmp_options, - .version = LIBAVUTIL_VERSION_INT, -}; - -URLProtocol ff_rtmpts_protocol = { - .name = "rtmpts", - .url_open = rtmp_open, - .url_read = rtmp_read, - .url_write = rtmp_write, - .url_close = rtmp_close, - .priv_data_size = sizeof(RTMPContext), - .flags = URL_PROTOCOL_FLAG_NETWORK, - .priv_data_class = &rtmpts_class, -}; +RTMP_PROTOCOL(rtmp) +RTMP_PROTOCOL(rtmpe) +RTMP_PROTOCOL(rtmps) +RTMP_PROTOCOL(rtmpt) +RTMP_PROTOCOL(rtmpte) +RTMP_PROTOCOL(rtmpts) |