view rtp.c @ 5836:2997c88028cd libavformat

Move the probe loop from av_open_input_file() into its own method av_probe_input_buffer() so that it can be reused. Here are a few differences to the original way things were probed: - maximum probe buffer size can be specified as a parameter. - offset within the stream to probe from can be specified as a parameter. - instead of seeking back to the start each time a probe fails, stream data is appended to the reallocated buffer. This lowers the amount of data read from the stream (there is no repetition) and results in fewer closed and reopened streams (when seeking fails). New attempt after r22296, which was revert in r22315 due to a FATE failure. See the thread: Subject: [FFmpeg-devel] [PATCH] Move av_open_input_file probe loop to its own method Date: 2010-03-05 03:23:57 GMT Patch by Micah F. Galizia printf("%s%s@%s.%s", "micah", "galizia", "gmail", "com").

/*
 * RTP input/output format
 * Copyright (c) 2002 Fabrice Bellard
 *
 * 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
 */

#include "avformat.h"

#include "rtp.h"

//#define DEBUG

/* from http://www.iana.org/assignments/rtp-parameters last updated 05 January 2005 */
/* payload types >= 96 are dynamic;
 * payload types between 72 and 76 are reserved for RTCP conflict avoidance;
 * all the other payload types not present in the table are unassigned or
 * reserved
 */
static const struct
{
    int pt;
    const char enc_name[6];
    enum CodecType codec_type;
    enum CodecID codec_id;
    int clock_rate;
    int audio_channels;
} AVRtpPayloadTypes[]=
{
  {0, "PCMU",        CODEC_TYPE_AUDIO,   CODEC_ID_PCM_MULAW, 8000, 1},
  {3, "GSM",         CODEC_TYPE_AUDIO,   CODEC_ID_NONE, 8000, 1},
  {4, "G723",        CODEC_TYPE_AUDIO,   CODEC_ID_NONE, 8000, 1},
  {5, "DVI4",        CODEC_TYPE_AUDIO,   CODEC_ID_NONE, 8000, 1},
  {6, "DVI4",        CODEC_TYPE_AUDIO,   CODEC_ID_NONE, 16000, 1},
  {7, "LPC",         CODEC_TYPE_AUDIO,   CODEC_ID_NONE, 8000, 1},
  {8, "PCMA",        CODEC_TYPE_AUDIO,   CODEC_ID_PCM_ALAW, 8000, 1},
  {9, "G722",        CODEC_TYPE_AUDIO,   CODEC_ID_NONE, 8000, 1},
  {10, "L16",        CODEC_TYPE_AUDIO,   CODEC_ID_PCM_S16BE, 44100, 2},
  {11, "L16",        CODEC_TYPE_AUDIO,   CODEC_ID_PCM_S16BE, 44100, 1},
  {12, "QCELP",      CODEC_TYPE_AUDIO,   CODEC_ID_QCELP, 8000, 1},
  {13, "CN",         CODEC_TYPE_AUDIO,   CODEC_ID_NONE, 8000, 1},
  {14, "MPA",        CODEC_TYPE_AUDIO,   CODEC_ID_MP2, -1, -1},
  {14, "MPA",        CODEC_TYPE_AUDIO,   CODEC_ID_MP3, -1, -1},
  {15, "G728",       CODEC_TYPE_AUDIO,   CODEC_ID_NONE, 8000, 1},
  {16, "DVI4",       CODEC_TYPE_AUDIO,   CODEC_ID_NONE, 11025, 1},
  {17, "DVI4",       CODEC_TYPE_AUDIO,   CODEC_ID_NONE, 22050, 1},
  {18, "G729",       CODEC_TYPE_AUDIO,   CODEC_ID_NONE, 8000, 1},
  {25, "CelB",       CODEC_TYPE_VIDEO,   CODEC_ID_NONE, 90000, -1},
  {26, "JPEG",       CODEC_TYPE_VIDEO,   CODEC_ID_MJPEG, 90000, -1},
  {28, "nv",         CODEC_TYPE_VIDEO,   CODEC_ID_NONE, 90000, -1},
  {31, "H261",       CODEC_TYPE_VIDEO,   CODEC_ID_H261, 90000, -1},
  {32, "MPV",        CODEC_TYPE_VIDEO,   CODEC_ID_MPEG1VIDEO, 90000, -1},
  {32, "MPV",        CODEC_TYPE_VIDEO,   CODEC_ID_MPEG2VIDEO, 90000, -1},
  {33, "MP2T",       CODEC_TYPE_DATA,    CODEC_ID_MPEG2TS, 90000, -1},
  {34, "H263",       CODEC_TYPE_VIDEO,   CODEC_ID_H263, 90000, -1},
  {-1, "",           CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}
};

int ff_rtp_get_codec_info(AVCodecContext *codec, int payload_type)
{
    int i = 0;

    for (i = 0; AVRtpPayloadTypes[i].pt >= 0; i++)
        if (AVRtpPayloadTypes[i].pt == payload_type) {
            if (AVRtpPayloadTypes[i].codec_id != CODEC_ID_NONE) {
                codec->codec_type = AVRtpPayloadTypes[i].codec_type;
                codec->codec_id = AVRtpPayloadTypes[i].codec_id;
                if (AVRtpPayloadTypes[i].audio_channels > 0)
                    codec->channels = AVRtpPayloadTypes[i].audio_channels;
                if (AVRtpPayloadTypes[i].clock_rate > 0)
                    codec->sample_rate = AVRtpPayloadTypes[i].clock_rate;
                return 0;
            }
        }
    return -1;
}

int ff_rtp_get_payload_type(AVCodecContext *codec)
{
    int i, payload_type;

    /* compute the payload type */
    for (payload_type = -1, i = 0; AVRtpPayloadTypes[i].pt >= 0; ++i)
        if (AVRtpPayloadTypes[i].codec_id == codec->codec_id) {
            if (codec->codec_id == CODEC_ID_H263)
                continue;
            if (codec->codec_id == CODEC_ID_PCM_S16BE)
                if (codec->channels != AVRtpPayloadTypes[i].audio_channels)
                    continue;
            payload_type = AVRtpPayloadTypes[i].pt;
        }
    return payload_type;
}

const char *ff_rtp_enc_name(int payload_type)
{
    int i;

    for (i = 0; AVRtpPayloadTypes[i].pt >= 0; i++)
        if (AVRtpPayloadTypes[i].pt == payload_type) {
            return AVRtpPayloadTypes[i].enc_name;
        }

    return "";
}

enum CodecID ff_rtp_codec_id(const char *buf, enum CodecType codec_type)
{
    int i;

    for (i = 0; AVRtpPayloadTypes[i].pt >= 0; i++)
        if (!strcmp(buf, AVRtpPayloadTypes[i].enc_name) && (codec_type == AVRtpPayloadTypes[i].codec_type)){
            return AVRtpPayloadTypes[i].codec_id;
        }

    return CODEC_ID_NONE;
}