mirror of https://github.com/FFmpeg/FFmpeg.git
Not yet complete, for demuxing AAC the AAC header must be generated manually. Possibly the decoder could accept the header as extradata to simplify this.oldabi
Reimar Döffinger
14 years ago
parent
d6c858658f
commit
1265395b5b
6 changed files with 177 additions and 1 deletions
@ -0,0 +1,172 @@ |
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/*
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* PMP demuxer. |
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* Copyright (c) 2011 Reimar Döffinger |
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* |
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* This file is part of FFmpeg. |
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* |
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* FFmpeg is free software; you can redistribute it and/or |
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* modify it under the terms of the GNU Lesser General Public |
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* License as published by the Free Software Foundation; either |
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* version 2.1 of the License, or (at your option) any later version. |
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* |
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* FFmpeg is distributed in the hope that it will be useful, |
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* but WITHOUT ANY WARRANTY; without even the implied warranty of |
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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* Lesser General Public License for more details. |
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* |
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* You should have received a copy of the GNU Lesser General Public |
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* License along with FFmpeg; if not, write to the Free Software |
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
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*/ |
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#include "libavutil/intreadwrite.h" |
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#include "avformat.h" |
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typedef struct { |
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int cur_stream; |
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int num_streams; |
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int audio_packets; |
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int current_packet; |
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uint32_t *packet_sizes; |
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int packet_sizes_alloc; |
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} PMPContext; |
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static int pmp_probe(AVProbeData *p) { |
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if (AV_RN32(p->buf) == AV_RN32("pmpm") && |
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AV_RL32(p->buf + 4) == 1) |
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return AVPROBE_SCORE_MAX; |
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return 0; |
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} |
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static int pmp_header(AVFormatContext *s, AVFormatParameters *ap) { |
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PMPContext *pmp = s->priv_data; |
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AVIOContext *pb = s->pb; |
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int tb_num, tb_den; |
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int index_cnt; |
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int audio_codec_id = CODEC_ID_NONE; |
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int srate, channels; |
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int i; |
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uint64_t pos; |
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AVStream *vst = av_new_stream(s, 0); |
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if (!vst) |
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return AVERROR(ENOMEM); |
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vst->codec->codec_type = AVMEDIA_TYPE_VIDEO; |
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avio_skip(pb, 8); |
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switch (avio_rl32(pb)) { |
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case 0: |
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vst->codec->codec_id = CODEC_ID_MPEG4; |
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break; |
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case 1: |
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vst->codec->codec_id = CODEC_ID_H264; |
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break; |
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default: |
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av_log(s, AV_LOG_ERROR, "Unsupported video format\n"); |
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break; |
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} |
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index_cnt = avio_rl32(pb); |
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vst->codec->width = avio_rl32(pb); |
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vst->codec->height = avio_rl32(pb); |
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tb_num = avio_rl32(pb); |
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tb_den = avio_rl32(pb); |
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av_set_pts_info(vst, 32, tb_num, tb_den); |
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vst->nb_frames = index_cnt; |
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vst->duration = index_cnt; |
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switch (avio_rl32(pb)) { |
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case 0: |
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audio_codec_id = CODEC_ID_MP3; |
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break; |
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case 1: |
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av_log(s, AV_LOG_ERROR, "AAC not yet correctly supported\n"); |
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audio_codec_id = CODEC_ID_AAC; |
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break; |
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default: |
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av_log(s, AV_LOG_ERROR, "Unsupported audio format\n"); |
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break; |
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} |
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pmp->num_streams = avio_rl16(pb) + 1; |
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avio_skip(pb, 10); |
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srate = avio_rl32(pb); |
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channels = avio_rl32(pb) + 1; |
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for (i = 1; i < pmp->num_streams; i++) { |
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AVStream *ast = av_new_stream(s, i); |
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if (!ast) |
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return AVERROR(ENOMEM); |
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ast->codec->codec_type = AVMEDIA_TYPE_AUDIO; |
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ast->codec->codec_id = audio_codec_id; |
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ast->codec->channels = channels; |
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ast->codec->sample_rate = srate; |
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av_set_pts_info(ast, 32, 1, srate); |
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} |
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pos = avio_tell(pb) + 4*index_cnt; |
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for (i = 0; i < index_cnt; i++) { |
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int size = avio_rl32(pb); |
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int flags = size & 1 ? AVINDEX_KEYFRAME : 0; |
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size >>= 1; |
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av_add_index_entry(vst, pos, i, size, 0, flags); |
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pos += size; |
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} |
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return 0; |
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} |
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static int pmp_packet(AVFormatContext *s, AVPacket *pkt) { |
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PMPContext *pmp = s->priv_data; |
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AVIOContext *pb = s->pb; |
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int ret = 0; |
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int i; |
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if (url_feof(pb)) |
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return AVERROR_EOF; |
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if (pmp->cur_stream == 0) { |
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int num_packets; |
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pmp->audio_packets = avio_r8(pb); |
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num_packets = (pmp->num_streams - 1) * pmp->audio_packets + 1; |
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avio_skip(pb, 8); |
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pmp->current_packet = 0; |
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av_fast_malloc(&pmp->packet_sizes, |
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&pmp->packet_sizes_alloc, |
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num_packets * sizeof(*pmp->packet_sizes)); |
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for (i = 0; i < num_packets; i++) |
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pmp->packet_sizes[i] = avio_rl32(pb); |
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} |
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ret = av_get_packet(pb, pkt, pmp->packet_sizes[pmp->current_packet]); |
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if (ret >= 0) { |
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ret = 0; |
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// FIXME: this is a hack that should be remove once
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// compute_pkt_fields can handle
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if (pmp->cur_stream == 0) |
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pkt->dts = s->streams[0]->cur_dts++; |
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pkt->stream_index = pmp->cur_stream; |
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} |
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if (pmp->current_packet % pmp->audio_packets == 0) |
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pmp->cur_stream = (pmp->cur_stream + 1) % pmp->num_streams; |
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pmp->current_packet++; |
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return ret; |
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} |
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static int pmp_seek(AVFormatContext *s, int stream_index, |
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int64_t ts, int flags) { |
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PMPContext *pmp = s->priv_data; |
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pmp->cur_stream = 0; |
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// fallback to default seek now
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return -1; |
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} |
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static int pmp_close(AVFormatContext *s) |
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{ |
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PMPContext *pmp = s->priv_data; |
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av_freep(&pmp->packet_sizes); |
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return 0; |
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} |
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AVInputFormat ff_pmp_demuxer = { |
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.name = "pmp", |
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.long_name = NULL_IF_CONFIG_SMALL("Playstation Portable PMP format"), |
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.priv_data_size = sizeof(PMPContext), |
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.read_probe = pmp_probe, |
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.read_header = pmp_header, |
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.read_packet = pmp_packet, |
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.read_seek = pmp_seek, |
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.read_close = pmp_close, |
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}; |
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