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c980be9e3a
* qatar/master: (21 commits) CDXL demuxer and decoder hls: Re-add legacy applehttp name to preserve interface compatibility. hlsproto: Rename the functions and context hlsproto: Encourage users to try the hls demuxer instead of the proto doc: Move the hls protocol section into the right place libavformat: Rename the applehttp protocol to hls hls: Rename the functions and context libavformat: Rename the applehttp demuxer to hls rtpdec: Support H263 in RFC 2190 format rv30: check block type validity ttadec: CRC checking movenc: Support muxing VC1 avconv: Don't split out inline sequence headers when stream copying VC1 rv34: handle size changes during frame multithreading rv40: prevent undefined signed overflow in rv40_loop_filter() rv34: use AVERROR return values in ff_rv34_decode_frame() rv34: use uint16_t for RV34DecContext.deblock_coefs librtmp: Add "lib" prefix to librtmp URLProtocol declarations. movenc: Use defines instead of hardcoded numbers for RTCP types smjpegdec: implement seeking ... Conflicts: Changelog doc/general.texi libavcodec/avcodec.h libavcodec/rv30.c libavcodec/tta.c libavcodec/version.h libavformat/Makefile libavformat/allformats.c libavformat/version.h libswscale/x86/swscale_mmx.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
219 lines
6.2 KiB
C
219 lines
6.2 KiB
C
/*
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* RTMP network protocol
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* Copyright (c) 2010 Howard Chu
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* RTMP protocol based on http://rtmpdump.mplayerhq.hu/ librtmp
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*/
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#include "libavutil/mathematics.h"
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#include "avformat.h"
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#include "url.h"
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#include <librtmp/rtmp.h>
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#include <librtmp/log.h>
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static void rtmp_log(int level, const char *fmt, va_list args)
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{
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switch (level) {
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default:
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case RTMP_LOGCRIT: level = AV_LOG_FATAL; break;
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case RTMP_LOGERROR: level = AV_LOG_ERROR; break;
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case RTMP_LOGWARNING: level = AV_LOG_WARNING; break;
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case RTMP_LOGINFO: level = AV_LOG_INFO; break;
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case RTMP_LOGDEBUG: level = AV_LOG_VERBOSE; break;
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case RTMP_LOGDEBUG2: level = AV_LOG_DEBUG; break;
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}
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av_vlog(NULL, level, fmt, args);
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av_log(NULL, level, "\n");
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}
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static int rtmp_close(URLContext *s)
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{
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RTMP *r = s->priv_data;
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RTMP_Close(r);
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return 0;
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}
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/**
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* Open RTMP connection and verify that the stream can be played.
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*
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* URL syntax: rtmp://server[:port][/app][/playpath][ keyword=value]...
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* where 'app' is first one or two directories in the path
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* (e.g. /ondemand/, /flash/live/, etc.)
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* and 'playpath' is a file name (the rest of the path,
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* may be prefixed with "mp4:")
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*
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* Additional RTMP library options may be appended as
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* space-separated key-value pairs.
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*/
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static int rtmp_open(URLContext *s, const char *uri, int flags)
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{
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RTMP *r = s->priv_data;
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int rc;
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switch (av_log_get_level()) {
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default:
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case AV_LOG_FATAL: rc = RTMP_LOGCRIT; break;
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case AV_LOG_ERROR: rc = RTMP_LOGERROR; break;
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case AV_LOG_WARNING: rc = RTMP_LOGWARNING; break;
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case AV_LOG_INFO: rc = RTMP_LOGINFO; break;
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case AV_LOG_VERBOSE: rc = RTMP_LOGDEBUG; break;
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case AV_LOG_DEBUG: rc = RTMP_LOGDEBUG2; break;
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}
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RTMP_LogSetLevel(rc);
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RTMP_LogSetCallback(rtmp_log);
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RTMP_Init(r);
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if (!RTMP_SetupURL(r, s->filename)) {
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rc = -1;
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goto fail;
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}
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if (flags & AVIO_FLAG_WRITE)
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RTMP_EnableWrite(r);
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if (!RTMP_Connect(r, NULL) || !RTMP_ConnectStream(r, 0)) {
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rc = -1;
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goto fail;
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}
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s->is_streamed = 1;
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return 0;
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fail:
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return rc;
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}
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static int rtmp_write(URLContext *s, const uint8_t *buf, int size)
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{
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RTMP *r = s->priv_data;
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return RTMP_Write(r, buf, size);
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}
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static int rtmp_read(URLContext *s, uint8_t *buf, int size)
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{
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RTMP *r = s->priv_data;
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return RTMP_Read(r, buf, size);
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}
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static int rtmp_read_pause(URLContext *s, int pause)
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{
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RTMP *r = s->priv_data;
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if (!RTMP_Pause(r, pause))
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return -1;
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return 0;
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}
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static int64_t rtmp_read_seek(URLContext *s, int stream_index,
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int64_t timestamp, int flags)
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{
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RTMP *r = s->priv_data;
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if (flags & AVSEEK_FLAG_BYTE)
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return AVERROR(ENOSYS);
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/* seeks are in milliseconds */
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if (stream_index < 0)
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timestamp = av_rescale_rnd(timestamp, 1000, AV_TIME_BASE,
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flags & AVSEEK_FLAG_BACKWARD ? AV_ROUND_DOWN : AV_ROUND_UP);
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if (!RTMP_SendSeek(r, timestamp))
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return -1;
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return timestamp;
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}
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static int rtmp_get_file_handle(URLContext *s)
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{
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RTMP *r = s->priv_data;
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return RTMP_Socket(r);
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}
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URLProtocol ff_librtmp_protocol = {
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.name = "rtmp",
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.url_open = rtmp_open,
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.url_read = rtmp_read,
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.url_write = rtmp_write,
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.url_close = rtmp_close,
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.url_read_pause = rtmp_read_pause,
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.url_read_seek = rtmp_read_seek,
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.url_get_file_handle = rtmp_get_file_handle,
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.priv_data_size = sizeof(RTMP),
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.flags = URL_PROTOCOL_FLAG_NETWORK,
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};
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URLProtocol ff_librtmpt_protocol = {
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.name = "rtmpt",
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.url_open = rtmp_open,
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.url_read = rtmp_read,
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.url_write = rtmp_write,
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.url_close = rtmp_close,
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.url_read_pause = rtmp_read_pause,
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.url_read_seek = rtmp_read_seek,
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.url_get_file_handle = rtmp_get_file_handle,
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.priv_data_size = sizeof(RTMP),
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.flags = URL_PROTOCOL_FLAG_NETWORK,
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};
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URLProtocol ff_librtmpe_protocol = {
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.name = "rtmpe",
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.url_open = rtmp_open,
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.url_read = rtmp_read,
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.url_write = rtmp_write,
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.url_close = rtmp_close,
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.url_read_pause = rtmp_read_pause,
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.url_read_seek = rtmp_read_seek,
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.url_get_file_handle = rtmp_get_file_handle,
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.priv_data_size = sizeof(RTMP),
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.flags = URL_PROTOCOL_FLAG_NETWORK,
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};
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URLProtocol ff_librtmpte_protocol = {
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.name = "rtmpte",
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.url_open = rtmp_open,
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.url_read = rtmp_read,
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.url_write = rtmp_write,
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.url_close = rtmp_close,
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.url_read_pause = rtmp_read_pause,
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.url_read_seek = rtmp_read_seek,
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.url_get_file_handle = rtmp_get_file_handle,
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.priv_data_size = sizeof(RTMP),
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.flags = URL_PROTOCOL_FLAG_NETWORK,
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};
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URLProtocol ff_librtmps_protocol = {
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.name = "rtmps",
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.url_open = rtmp_open,
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.url_read = rtmp_read,
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.url_write = rtmp_write,
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.url_close = rtmp_close,
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.url_read_pause = rtmp_read_pause,
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.url_read_seek = rtmp_read_seek,
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.url_get_file_handle = rtmp_get_file_handle,
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.priv_data_size = sizeof(RTMP),
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.flags = URL_PROTOCOL_FLAG_NETWORK,
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};
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