diff options
Diffstat (limited to 'libavformat/pmpdec.c')
-rw-r--r-- | libavformat/pmpdec.c | 172 |
1 files changed, 172 insertions, 0 deletions
diff --git a/libavformat/pmpdec.c b/libavformat/pmpdec.c new file mode 100644 index 0000000000..ba40003359 --- /dev/null +++ b/libavformat/pmpdec.c @@ -0,0 +1,172 @@ +/* + * PMP demuxer. + * Copyright (c) 2011 Reimar Döffinger + * + * 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 "libavutil/intreadwrite.h" +#include "avformat.h" + +typedef struct { + int cur_stream; + int num_streams; + int audio_packets; + int current_packet; + uint32_t *packet_sizes; + int packet_sizes_alloc; +} PMPContext; + +static int pmp_probe(AVProbeData *p) { + if (AV_RN32(p->buf) == AV_RN32("pmpm") && + AV_RL32(p->buf + 4) == 1) + return AVPROBE_SCORE_MAX; + return 0; +} + +static int pmp_header(AVFormatContext *s, AVFormatParameters *ap) { + PMPContext *pmp = s->priv_data; + AVIOContext *pb = s->pb; + int tb_num, tb_den; + int index_cnt; + int audio_codec_id = CODEC_ID_NONE; + int srate, channels; + int i; + uint64_t pos; + AVStream *vst = av_new_stream(s, 0); + if (!vst) + return AVERROR(ENOMEM); + vst->codec->codec_type = AVMEDIA_TYPE_VIDEO; + avio_skip(pb, 8); + switch (avio_rl32(pb)) { + case 0: + vst->codec->codec_id = CODEC_ID_MPEG4; + break; + case 1: + vst->codec->codec_id = CODEC_ID_H264; + break; + default: + av_log(s, AV_LOG_ERROR, "Unsupported video format\n"); + break; + } + index_cnt = avio_rl32(pb); + vst->codec->width = avio_rl32(pb); + vst->codec->height = avio_rl32(pb); + + tb_num = avio_rl32(pb); + tb_den = avio_rl32(pb); + av_set_pts_info(vst, 32, tb_num, tb_den); + vst->nb_frames = index_cnt; + vst->duration = index_cnt; + + switch (avio_rl32(pb)) { + case 0: + audio_codec_id = CODEC_ID_MP3; + break; + case 1: + av_log(s, AV_LOG_ERROR, "AAC not yet correctly supported\n"); + audio_codec_id = CODEC_ID_AAC; + break; + default: + av_log(s, AV_LOG_ERROR, "Unsupported audio format\n"); + break; + } + pmp->num_streams = avio_rl16(pb) + 1; + avio_skip(pb, 10); + srate = avio_rl32(pb); + channels = avio_rl32(pb) + 1; + for (i = 1; i < pmp->num_streams; i++) { + AVStream *ast = av_new_stream(s, i); + if (!ast) + return AVERROR(ENOMEM); + ast->codec->codec_type = AVMEDIA_TYPE_AUDIO; + ast->codec->codec_id = audio_codec_id; + ast->codec->channels = channels; + ast->codec->sample_rate = srate; + av_set_pts_info(ast, 32, 1, srate); + } + pos = avio_tell(pb) + 4*index_cnt; + for (i = 0; i < index_cnt; i++) { + int size = avio_rl32(pb); + int flags = size & 1 ? AVINDEX_KEYFRAME : 0; + size >>= 1; + av_add_index_entry(vst, pos, i, size, 0, flags); + pos += size; + } + return 0; +} + +static int pmp_packet(AVFormatContext *s, AVPacket *pkt) { + PMPContext *pmp = s->priv_data; + AVIOContext *pb = s->pb; + int ret = 0; + int i; + + if (url_feof(pb)) + return AVERROR_EOF; + if (pmp->cur_stream == 0) { + int num_packets; + pmp->audio_packets = avio_r8(pb); + num_packets = (pmp->num_streams - 1) * pmp->audio_packets + 1; + avio_skip(pb, 8); + pmp->current_packet = 0; + av_fast_malloc(&pmp->packet_sizes, + &pmp->packet_sizes_alloc, + num_packets * sizeof(*pmp->packet_sizes)); + for (i = 0; i < num_packets; i++) + pmp->packet_sizes[i] = avio_rl32(pb); + } + ret = av_get_packet(pb, pkt, pmp->packet_sizes[pmp->current_packet]); + if (ret >= 0) { + ret = 0; + // FIXME: this is a hack that should be remove once + // compute_pkt_fields can handle + if (pmp->cur_stream == 0) + pkt->dts = s->streams[0]->cur_dts++; + pkt->stream_index = pmp->cur_stream; + } + if (pmp->current_packet % pmp->audio_packets == 0) + pmp->cur_stream = (pmp->cur_stream + 1) % pmp->num_streams; + pmp->current_packet++; + return ret; +} + +static int pmp_seek(AVFormatContext *s, int stream_index, + int64_t ts, int flags) { + PMPContext *pmp = s->priv_data; + pmp->cur_stream = 0; + // fallback to default seek now + return -1; +} + +static int pmp_close(AVFormatContext *s) +{ + PMPContext *pmp = s->priv_data; + av_freep(&pmp->packet_sizes); + return 0; +} + +AVInputFormat ff_pmp_demuxer = { + .name = "pmp", + .long_name = NULL_IF_CONFIG_SMALL("Playstation Portable PMP format"), + .priv_data_size = sizeof(PMPContext), + .read_probe = pmp_probe, + .read_header = pmp_header, + .read_packet = pmp_packet, + .read_seek = pmp_seek, + .read_close = pmp_close, +}; |