ffmpeg/libavformat/librtmp.c
Michael Niedermayer 7b0b10ce41 Merge remote-tracking branch 'qatar/master'
* qatar/master: (25 commits)
  rtpenc: Add support for G726 audio
  rtpdec: Interpret the different G726 names as bits_per_coded_sample
  rtpenc: Change rtp_send_samples to handle sample sizes other than even bytes
  rtpenc: Cast a rescaling parameter to int64_t
  h264: cap max has_b_frames at MAX_DELAYED_PIC_COUNT - 1.
  ARM: fix indentation in ff_dsputil_init_neon()
  ARM: NEON put/avg_pixels8/16 cosmetics
  ARM: add remaining NEON avg_pixels8/16 functions
  ARM: clean up NEON put/avg_pixels macros
  fate: split acodec-pcm into individual tests
  swscale: #include "libavutil/mathematics.h"
  pmpdec: don't use deprecated av_set_pts_info.
  rv34: align temporary block of "dct" coefs
  Add PlayStation Portable PMP format demuxer
  proto: Realign struct initializers
  proto: Use .priv_data_size to allocate the private context
  mmsh: Properly clean up if the second ffurl_alloc failed
  rtmp: Clean up properly if the handshake failed
  md5proto: Remove the get_file_handle function
  applehttpproto: Use the close function if the open function fails
  ...

Conflicts:
	libavcodec/vble.c
	libavformat/mmsh.c
	libavformat/pmpdec.c
	libavformat/udp.c
	tests/ref/acodec/pcm

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-12-02 00:51:11 +01:00

214 lines
6.0 KiB
C

/*
* RTMP network protocol
* Copyright (c) 2010 Howard Chu
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* RTMP protocol based on http://rtmpdump.mplayerhq.hu/ librtmp
*/
#include "libavutil/mathematics.h"
#include "avformat.h"
#include "url.h"
#include <librtmp/rtmp.h>
#include <librtmp/log.h>
static void rtmp_log(int level, const char *fmt, va_list args)
{
switch (level) {
default:
case RTMP_LOGCRIT: level = AV_LOG_FATAL; break;
case RTMP_LOGERROR: level = AV_LOG_ERROR; break;
case RTMP_LOGWARNING: level = AV_LOG_WARNING; break;
case RTMP_LOGINFO: level = AV_LOG_INFO; break;
case RTMP_LOGDEBUG: level = AV_LOG_VERBOSE; break;
case RTMP_LOGDEBUG2: level = AV_LOG_DEBUG; break;
}
av_vlog(NULL, level, fmt, args);
av_log(NULL, level, "\n");
}
static int rtmp_close(URLContext *s)
{
RTMP *r = s->priv_data;
RTMP_Close(r);
return 0;
}
/**
* Open RTMP connection and verify that the stream can be played.
*
* URL syntax: rtmp://server[:port][/app][/playpath][ keyword=value]...
* where 'app' is first one or two directories in the path
* (e.g. /ondemand/, /flash/live/, etc.)
* and 'playpath' is a file name (the rest of the path,
* may be prefixed with "mp4:")
*
* Additional RTMP library options may be appended as
* space-separated key-value pairs.
*/
static int rtmp_open(URLContext *s, const char *uri, int flags)
{
RTMP *r = s->priv_data;
int rc;
switch (av_log_get_level()) {
default:
case AV_LOG_FATAL: rc = RTMP_LOGCRIT; break;
case AV_LOG_ERROR: rc = RTMP_LOGERROR; break;
case AV_LOG_WARNING: rc = RTMP_LOGWARNING; break;
case AV_LOG_INFO: rc = RTMP_LOGINFO; break;
case AV_LOG_VERBOSE: rc = RTMP_LOGDEBUG; break;
case AV_LOG_DEBUG: rc = RTMP_LOGDEBUG2; break;
}
RTMP_LogSetLevel(rc);
RTMP_LogSetCallback(rtmp_log);
RTMP_Init(r);
if (!RTMP_SetupURL(r, s->filename)) {
rc = -1;
goto fail;
}
if (flags & AVIO_FLAG_WRITE)
RTMP_EnableWrite(r);
if (!RTMP_Connect(r, NULL) || !RTMP_ConnectStream(r, 0)) {
rc = -1;
goto fail;
}
s->is_streamed = 1;
return 0;
fail:
return rc;
}
static int rtmp_write(URLContext *s, const uint8_t *buf, int size)
{
RTMP *r = s->priv_data;
return RTMP_Write(r, buf, size);
}
static int rtmp_read(URLContext *s, uint8_t *buf, int size)
{
RTMP *r = s->priv_data;
return RTMP_Read(r, buf, size);
}
static int rtmp_read_pause(URLContext *s, int pause)
{
RTMP *r = s->priv_data;
if (!RTMP_Pause(r, pause))
return -1;
return 0;
}
static int64_t rtmp_read_seek(URLContext *s, int stream_index,
int64_t timestamp, int flags)
{
RTMP *r = s->priv_data;
if (flags & AVSEEK_FLAG_BYTE)
return AVERROR(ENOSYS);
/* seeks are in milliseconds */
if (stream_index < 0)
timestamp = av_rescale_rnd(timestamp, 1000, AV_TIME_BASE,
flags & AVSEEK_FLAG_BACKWARD ? AV_ROUND_DOWN : AV_ROUND_UP);
if (!RTMP_SendSeek(r, timestamp))
return -1;
return timestamp;
}
static int rtmp_get_file_handle(URLContext *s)
{
RTMP *r = s->priv_data;
return RTMP_Socket(r);
}
URLProtocol ff_rtmp_protocol = {
.name = "rtmp",
.url_open = rtmp_open,
.url_read = rtmp_read,
.url_write = rtmp_write,
.url_close = rtmp_close,
.url_read_pause = rtmp_read_pause,
.url_read_seek = rtmp_read_seek,
.url_get_file_handle = rtmp_get_file_handle,
.priv_data_size = sizeof(RTMP),
};
URLProtocol ff_rtmpt_protocol = {
.name = "rtmpt",
.url_open = rtmp_open,
.url_read = rtmp_read,
.url_write = rtmp_write,
.url_close = rtmp_close,
.url_read_pause = rtmp_read_pause,
.url_read_seek = rtmp_read_seek,
.url_get_file_handle = rtmp_get_file_handle,
.priv_data_size = sizeof(RTMP),
};
URLProtocol ff_rtmpe_protocol = {
.name = "rtmpe",
.url_open = rtmp_open,
.url_read = rtmp_read,
.url_write = rtmp_write,
.url_close = rtmp_close,
.url_read_pause = rtmp_read_pause,
.url_read_seek = rtmp_read_seek,
.url_get_file_handle = rtmp_get_file_handle,
.priv_data_size = sizeof(RTMP),
};
URLProtocol ff_rtmpte_protocol = {
.name = "rtmpte",
.url_open = rtmp_open,
.url_read = rtmp_read,
.url_write = rtmp_write,
.url_close = rtmp_close,
.url_read_pause = rtmp_read_pause,
.url_read_seek = rtmp_read_seek,
.url_get_file_handle = rtmp_get_file_handle,
.priv_data_size = sizeof(RTMP),
};
URLProtocol ff_rtmps_protocol = {
.name = "rtmps",
.url_open = rtmp_open,
.url_read = rtmp_read,
.url_write = rtmp_write,
.url_close = rtmp_close,
.url_read_pause = rtmp_read_pause,
.url_read_seek = rtmp_read_seek,
.url_get_file_handle = rtmp_get_file_handle,
.priv_data_size = sizeof(RTMP),
};