view stream/stream_rtsp.c @ 24590:2c238fa777ff

ao_alsa: Fix get_space() return values larger than buffersize After a buffer underrun the ALSA get_space() function sometimes returned values larger than the ao had set in ao_data.buffersize. Fix this by replacing the old check against MAX_OUTBURST by one against ao_data.buffersize. There should be no need for the MAX_OUTBURST check; the current MPlayer side should no longer have any constant limit on the amount of data an ao can buffer or request at once. The get_space() values larger than ao_data.buffersize triggered errors in audio decoding causing the current attempt to fill audio buffers to be aborted. I'm not sure how often that caused behavior noticeably worse then an underrun already is.

/*
 *  Copyright (C) 2006 Benjamin Zores
 *   based on previous Real RTSP support from Roberto Togni and xine team.
 *
 *   This program is free software; you can redistribute it and/or modify
 *  it under the terms of the GNU General Public License as published by
 *  the Free Software Foundation; either version 2 of the License, or
 *  (at your option) any later version.
 *
 *   This program 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 General Public License for more details.
 *
 *   You should have received a copy of the GNU General Public License
 *  along with this program; if not, write to the Free Software Foundation,
 *  Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */

#include <stdlib.h>
#include <string.h>
#include <unistd.h>
#include <stdlib.h>
#include <stdio.h>
#include <sys/types.h>
#include <ctype.h>
#include "config.h"
#ifndef HAVE_WINSOCK2
#include <netinet/in.h>
#include <sys/socket.h>
#include <arpa/inet.h>
#define closesocket close
#else
#include <winsock2.h>
#include <ws2tcpip.h>
#endif
#include <errno.h>

#include "stream.h"
#include "tcp.h"
#include "librtsp/rtsp.h"
#include "librtsp/rtsp_session.h"

#define RTSP_DEFAULT_PORT 554

extern int network_bandwidth;

static int
rtsp_streaming_read (int fd, char *buffer,
                     int size, streaming_ctrl_t *stream_ctrl)
{
  return rtsp_session_read (stream_ctrl->data, buffer, size);
}

static int
rtsp_streaming_start (stream_t *stream)
{
  int fd;
  rtsp_session_t *rtsp;
  char *mrl;
  char *file;
  int port;
  int redirected, temp;

  if (!stream)
    return -1;

  /* counter so we don't get caught in infinite redirections */
  temp = 5;

  do {
    redirected = 0;

    fd = connect2Server (stream->streaming_ctrl->url->hostname,
                         port = (stream->streaming_ctrl->url->port ?
                                 stream->streaming_ctrl->url->port :
                                 RTSP_DEFAULT_PORT), 1);
    
    if (fd < 0 && !stream->streaming_ctrl->url->port)
      fd = connect2Server (stream->streaming_ctrl->url->hostname,
                           port = 7070, 1);

    if (fd < 0)
      return -1;
    
    file = stream->streaming_ctrl->url->file;
    if (file[0] == '/')
      file++;

    mrl = malloc (strlen (stream->streaming_ctrl->url->hostname)
                  + strlen (file) + 16);
    
    sprintf (mrl, "rtsp://%s:%i/%s",
             stream->streaming_ctrl->url->hostname, port, file);

    rtsp = rtsp_session_start (fd, &mrl, file,
                               stream->streaming_ctrl->url->hostname,
                               port, &redirected,
                               stream->streaming_ctrl->bandwidth,
                               stream->streaming_ctrl->url->username,
                               stream->streaming_ctrl->url->password);

    if (redirected == 1)
    {
      url_free (stream->streaming_ctrl->url);
      stream->streaming_ctrl->url = url_new (mrl);
      closesocket (fd);
    }

    free (mrl);
    temp--;
  } while ((redirected != 0) && (temp > 0));    

  if (!rtsp)
    return -1;

  stream->fd = fd;
  stream->streaming_ctrl->data = rtsp;
  
  stream->streaming_ctrl->streaming_read = rtsp_streaming_read;
  stream->streaming_ctrl->streaming_seek = NULL;
  stream->streaming_ctrl->prebuffer_size = 128*1024;  // 640 KBytes
  stream->streaming_ctrl->buffering = 1;
  stream->streaming_ctrl->status = streaming_playing_e;
  
  return 0;
}

static void
rtsp_streaming_close (struct stream_st *s)
{
  rtsp_session_t *rtsp = NULL;
  
  rtsp = (rtsp_session_t *) s->streaming_ctrl->data;
  if (rtsp)
    rtsp_session_end (rtsp);
}

static int
rtsp_streaming_open (stream_t *stream, int mode, void *opts, int *file_format)
{
  URL_t *url;
  extern int index_mode;
  
  mp_msg (MSGT_OPEN, MSGL_V, "STREAM_RTSP, URL: %s\n", stream->url);
  stream->streaming_ctrl = streaming_ctrl_new ();
  if (!stream->streaming_ctrl)
    return STREAM_ERROR;

  stream->streaming_ctrl->bandwidth = network_bandwidth;
  url = url_new (stream->url);
  stream->streaming_ctrl->url = check4proxies (url);

  stream->fd = -1;
  index_mode = -1; /* prevent most RTSP streams from locking due to -idx */
  if (rtsp_streaming_start (stream) < 0)
  {
    streaming_ctrl_free (stream->streaming_ctrl);
    stream->streaming_ctrl = NULL;
    return STREAM_UNSUPPORTED;
  }

  fixup_network_stream_cache (stream);
  stream->type = STREAMTYPE_STREAM;
  stream->close = rtsp_streaming_close;

  return STREAM_OK;
}

stream_info_t stream_info_rtsp = {
  "RTSP streaming",
  "rtsp",
  "Benjamin Zores, Roberto Togni",
  "ported from xine",
  rtsp_streaming_open,
  {"rtsp", NULL},
  NULL,
  0 /* Urls are an option string */
};