view stream/stream_rtp.c @ 25357:b265c001e64a

Add new audio filter for encoding multi-channel audio into ac3 at runtime. And if set first parameter of this filter to 1, it will do ac3 passthrough like hwac3 did.
author ulion
date Thu, 13 Dec 2007 12:38:17 +0000
parents c1d17bd6683c
children a26e50cae389
line wrap: on
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/*
 *  Copyright (C) 2006 Benjamin Zores
 *   Stream layer for MPEG over RTP, based on previous work from Dave Chapman
 *
 *   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 "config.h"

#include <stdlib.h>
#include <string.h>

#include "stream.h"
#include "url.h"
#include "udp.h"
#include "rtp.h"

static int
rtp_streaming_read (int fd, char *buffer,
                    int size, streaming_ctrl_t *streaming_ctrl)
{
  return read_rtp_from_server (fd, buffer, size);
}

static int
rtp_streaming_start (stream_t *stream)
{
  streaming_ctrl_t *streaming_ctrl;
  int fd;

  if (!stream)
    return -1;

  streaming_ctrl = stream->streaming_ctrl;
  fd = stream->fd;
	
  if (fd < 0)
  {
    fd = udp_open_socket (streaming_ctrl->url); 
    if (fd < 0)
      return -1;
    stream->fd = fd;
  }

  streaming_ctrl->streaming_read = rtp_streaming_read;
  streaming_ctrl->streaming_seek = nop_streaming_seek;
  streaming_ctrl->prebuffer_size = 64 * 1024; /* 64 KBytes */
  streaming_ctrl->buffering = 0;
  streaming_ctrl->status = streaming_playing_e;
  
  return 0;
}

static int
rtp_stream_open (stream_t *stream, int mode, void *opts, int *file_format)
{
  URL_t *url;
  extern int network_bandwidth;
  
  mp_msg (MSGT_OPEN, MSGL_INFO, "STREAM_RTP, 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);

  if (url->port == 0)
  {
    mp_msg (MSGT_NETWORK, MSGL_ERR,
            "You must enter a port number for RTP streams!\n");
    streaming_ctrl_free (stream->streaming_ctrl);
    stream->streaming_ctrl = NULL;
  
    return STREAM_UNSUPPORTED;
  }

  if (rtp_streaming_start (stream) < 0)
  {
    mp_msg (MSGT_NETWORK, MSGL_ERR, "rtp_streaming_start failed\n");
    streaming_ctrl_free (stream->streaming_ctrl);
    stream->streaming_ctrl = NULL;
  
    return STREAM_UNSUPPORTED;
  }

  stream->type = STREAMTYPE_STREAM;
  fixup_network_stream_cache (stream);
  
  return STREAM_OK;
}

const stream_info_t stream_info_rtp = {
  "MPEG over RTP streaming",
  "rtp",
  "Dave Chapman, Benjamin Zores",
  "native rtp support",
  rtp_stream_open,
  { "rtp", NULL},
  NULL,
  0 // Urls are an option string
};