view oma.c @ 4723:a2390c6a35e6 libavformat

Fix index generation in the way that it was supposed to be used. See the discussion in the ML thread "[PATCH] rmdec.c: merge old/new packet reading code". Over time, this code broke somewhat, e.g. seq was never actually written into (and was thus always 1, therefore the seq condition was always true), whereas it was supposed to be set to the sequence number of the video slice in case the video frame is divided over multiple RM packets (slices). The problem of this is that packets other than those containing the beginning of a video frame would be indexed as well. Secondly, flags&2 is supposed to be true for video keyframes and for these audio packets containing the start of a block. For some codecs (e.g. AAC), that is every single packet, whereas for others (e.g. cook), that is the packet containing the first of a series of scrambled packets that are to be descrambled together. Indexing any of the following would lead to incomplete and thus useless frames. Problem here is that flags would be reset to 2 to indicate that the first packet is ready to be returned, and in addition if no data was left to be returned (which is always true for the first packet), then we wouldn't actually write the index entry anyway. All in all, the idea was good and it probably worked at some point, but that is long ago. This patch should at the very least make it likely for this code to be executed again at the right times, i.e. the way it was originally intended to be used.

/*
 * 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    = 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},
};