annotate oma.c @ 5330:936a03bbd757 libavformat

Suppress ?params in the rtsp uri Right now rtsp demuxer receives it's ffmpeg specific params encoded in the url That made the server receiving requests with the url ending with "?udp", "?multicast" and "?tcp". That may or may not cause problems to servers with overly strict or overly simple uri parsers Patch from Armand Bendanan (name.surnameATfreeDOTfr)

/*
 * Sony OpenMG (OMA) demuxer
 *
 * Copyright (c) 2008 Maxim Poliakovski
 *               2008 Benjamin Larsson
 *
 * 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 libavformat/oma.c
 * This is a demuxer for Sony OpenMG Music files
 *
 * Known file extensions: ".oma", "aa3"
 * The format of such files consists of three parts:
 * - "ea3" header carrying overall info and metadata.
 * - "EA3" header is a Sony-specific header containing information about
 *   the OpenMG file: codec type (usually ATRAC, can also be MP3 or WMA),
 *   codec specific info (packet size, sample rate, channels and so on)
 *   and DRM related info (file encryption, content id).
 * - Sound data organized in packets follow the EA3 header
 *   (can be encrypted using the Sony DRM!).
 *
 * LIMITATIONS: This version supports only plain (unencrypted) OMA files.
 * If any DRM-protected (encrypted) file is encountered you will get the
 * corresponding error message. Try to remove the encryption using any
 * Sony software (for example SonicStage).
 * CODEC SUPPORT: Only ATRAC3 codec is currently supported!
 */

#include "avformat.h"
#include "libavutil/intreadwrite.h"
#include "raw.h"
#include "riff.h"

#define EA3_HEADER_SIZE 96

enum {
    OMA_CODECID_ATRAC3  = 0,
    OMA_CODECID_ATRAC3P = 1,
    OMA_CODECID_MP3     = 3,
    OMA_CODECID_LPCM    = 4,
    OMA_CODECID_WMA     = 5,
};

static const AVCodecTag codec_oma_tags[] = {
    { CODEC_ID_ATRAC3,  OMA_CODECID_ATRAC3 },
    { CODEC_ID_ATRAC3P, OMA_CODECID_ATRAC3P },
    { CODEC_ID_MP3,     OMA_CODECID_MP3 },
};

static int oma_read_header(AVFormatContext *s,
                           AVFormatParameters *ap)
{
    static const uint16_t srate_tab[6] = {320,441,480,882,960,0};
    int     ret, ea3_taglen, EA3_pos, framesize, jsflag, samplerate;
    uint32_t codec_params;
    int16_t eid;
    uint8_t buf[EA3_HEADER_SIZE];
    uint8_t *edata;
    AVStream *st;

    ret = get_buffer(s->pb, buf, 10);
    if (ret != 10)
        return -1;

    ea3_taglen = ((buf[6] & 0x7f) << 21) | ((buf[7] & 0x7f) << 14) | ((buf[8] & 0x7f) << 7) | (buf[9] & 0x7f);

    EA3_pos = ea3_taglen + 10;
    if (buf[5] & 0x10)
        EA3_pos += 10;

    url_fseek(s->pb, EA3_pos, SEEK_SET);
    ret = get_buffer(s->pb, buf, EA3_HEADER_SIZE);
    if (ret != EA3_HEADER_SIZE)
        return -1;

    if (memcmp(buf, ((const uint8_t[]){'E', 'A', '3'}),3) || buf[4] != 0 || buf[5] != EA3_HEADER_SIZE) {
        av_log(s, AV_LOG_ERROR, "Couldn't find the EA3 header !\n");
        return -1;
    }

    eid = AV_RB16(&buf[6]);
    if (eid != -1 && eid != -128) {
        av_log(s, AV_LOG_ERROR, "Encrypted file! Eid: %d\n", eid);
        return -1;
    }

    codec_params = AV_RB24(&buf[33]);

    st = av_new_stream(s, 0);
    if (!st)
        return AVERROR(ENOMEM);

    st->start_time = 0;
    st->codec->codec_type  = CODEC_TYPE_AUDIO;
    st->codec->codec_tag   = buf[32];
    st->codec->codec_id    = ff_codec_get_id(codec_oma_tags, st->codec->codec_tag);

    switch (buf[32]) {
        case OMA_CODECID_ATRAC3:
            samplerate = srate_tab[(codec_params >> 13) & 7]*100;
            if (samplerate != 44100)
                av_log(s, AV_LOG_ERROR, "Unsupported sample rate, send sample file to developers: %d\n", samplerate);

            framesize = (codec_params & 0x3FF) * 8;
            jsflag = (codec_params >> 17) & 1; /* get stereo coding mode, 1 for joint-stereo */
            st->codec->channels    = 2;
            st->codec->sample_rate = samplerate;
            st->codec->bit_rate    = st->codec->sample_rate * framesize * 8 / 1024;

            /* fake the atrac3 extradata (wav format, makes stream copy to wav work) */
            st->codec->extradata_size = 14;
            edata = av_mallocz(14 + FF_INPUT_BUFFER_PADDING_SIZE);
            if (!edata)
                return AVERROR(ENOMEM);

            st->codec->extradata = edata;
            AV_WL16(&edata[0],  1);             // always 1
            AV_WL32(&edata[2],  samplerate);    // samples rate
            AV_WL16(&edata[6],  jsflag);        // coding mode
            AV_WL16(&edata[8],  jsflag);        // coding mode
            AV_WL16(&edata[10], 1);             // always 1
            // AV_WL16(&edata[12], 0);          // always 0

            av_set_pts_info(st, 64, 1, st->codec->sample_rate);
            break;
        case OMA_CODECID_ATRAC3P:
            st->codec->channels = (codec_params >> 10) & 7;
            framesize = ((codec_params & 0x3FF) * 8) + 8;
            st->codec->sample_rate = srate_tab[(codec_params >> 13) & 7]*100;
            st->codec->bit_rate    = st->codec->sample_rate * framesize * 8 / 1024;
            av_set_pts_info(st, 64, 1, st->codec->sample_rate);
            av_log(s, AV_LOG_ERROR, "Unsupported codec ATRAC3+!\n");
            break;
        case OMA_CODECID_MP3:
            st->need_parsing = AVSTREAM_PARSE_FULL;
            framesize = 1024;
            break;
        default:
            av_log(s, AV_LOG_ERROR, "Unsupported codec %d!\n",buf[32]);
            return -1;
            break;
    }

    st->codec->block_align = framesize;
    url_fseek(s->pb, EA3_pos + EA3_HEADER_SIZE, SEEK_SET);

    return 0;
}


static int oma_read_packet(AVFormatContext *s, AVPacket *pkt)
{
    int ret = av_get_packet(s->pb, pkt, s->streams[0]->codec->block_align);

    pkt->stream_index = 0;
    if (ret <= 0)
        return AVERROR(EIO);

    return ret;
}

static int oma_read_probe(AVProbeData *p)
{
    if (!memcmp(p->buf, ((const uint8_t[]){'e', 'a', '3', 3, 0}),5))
        return AVPROBE_SCORE_MAX;
    else
        return 0;
}


AVInputFormat oma_demuxer = {
    "oma",
    NULL_IF_CONFIG_SMALL("Sony OpenMG audio"),
    0,
    oma_read_probe,
    oma_read_header,
    oma_read_packet,
    0,
    pcm_read_seek,
    .flags= AVFMT_GENERIC_INDEX,
    .extensions = "oma,aa3",
    .codec_tag= (const AVCodecTag* const []){codec_oma_tags, 0},
};