mirror of https://github.com/FFmpeg/FFmpeg.git
Signed-off-by: Oleksij Rempel <linux@rempel-privat.de> Signed-off-by: Luca Barbato <lu_zero@gentoo.org> Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>pull/113/head^2
Oleksij Rempel
10 years ago
committed by
Vittorio Giovara
parent
c56b9b1eb2
commit
062cd5a975
6 changed files with 347 additions and 2 deletions
@ -0,0 +1,342 @@ |
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/*
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* Digital Speech Standard (DSS) demuxer |
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* Copyright (c) 2014 Oleksij Rempel <linux@rempel-privat.de> |
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* |
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* This file is part of Libav. |
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* |
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* Libav 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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* Libav 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 Libav; 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/attributes.h" |
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#include "libavutil/bswap.h" |
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#include "libavutil/channel_layout.h" |
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#include "libavutil/intreadwrite.h" |
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#include "avformat.h" |
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#include "internal.h" |
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#define DSS_HEAD_OFFSET_AUTHOR 0xc |
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#define DSS_AUTHOR_SIZE 16 |
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#define DSS_HEAD_OFFSET_START_TIME 0x26 |
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#define DSS_HEAD_OFFSET_END_TIME 0x32 |
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#define DSS_TIME_SIZE 12 |
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#define DSS_HEAD_OFFSET_ACODEC 0x2a4 |
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#define DSS_ACODEC_DSS_SP 0x0 /* SP mode */ |
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#define DSS_ACODEC_G723_1 0x2 /* LP mode */ |
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#define DSS_HEAD_OFFSET_COMMENT 0x31e |
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#define DSS_COMMENT_SIZE 64 |
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#define DSS_BLOCK_SIZE 512 |
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#define DSS_HEADER_SIZE (DSS_BLOCK_SIZE * 2) |
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#define DSS_AUDIO_BLOCK_HEADER_SIZE 6 |
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#define DSS_FRAME_SIZE 42 |
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static const uint8_t frame_size[4] = { 24, 20, 4, 1 }; |
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typedef struct DSSDemuxContext { |
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unsigned int audio_codec; |
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int counter; |
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int swap; |
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int dss_sp_swap_byte; |
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int8_t *dss_sp_buf; |
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} DSSDemuxContext; |
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static int dss_probe(AVProbeData *p) |
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{ |
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if (AV_RL32(p->buf) != MKTAG(0x2, 'd', 's', 's')) |
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return 0; |
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return AVPROBE_SCORE_MAX; |
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} |
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static int dss_read_metadata_date(AVFormatContext *s, unsigned int offset, |
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const char *key) |
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{ |
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AVIOContext *pb = s->pb; |
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char datetime[64], string[DSS_TIME_SIZE + 1] = { 0 }; |
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int y, month, d, h, minute, sec; |
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int ret; |
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avio_seek(pb, offset, SEEK_SET); |
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ret = avio_read(s->pb, string, DSS_TIME_SIZE); |
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if (ret < DSS_TIME_SIZE) |
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return ret < 0 ? ret : AVERROR_EOF; |
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sscanf(string, "%2d%2d%2d%2d%2d%2d", &y, &month, &d, &h, &minute, &sec); |
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/* We deal with a two-digit year here, so set the default date to 2000
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* and hope it will never be used in the next century. */ |
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snprintf(datetime, sizeof(datetime), "%.4d-%.2d-%.2dT%.2d:%.2d:%.2d", |
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y + 2000, month, d, h, minute, sec); |
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return av_dict_set(&s->metadata, key, datetime, 0); |
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} |
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static int dss_read_metadata_string(AVFormatContext *s, unsigned int offset, |
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unsigned int size, const char *key) |
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{ |
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AVIOContext *pb = s->pb; |
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char *value; |
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int ret; |
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avio_seek(pb, offset, SEEK_SET); |
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value = av_mallocz(size + 1); |
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if (!value) |
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return AVERROR(ENOMEM); |
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ret = avio_read(s->pb, value, size); |
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if (ret < size) { |
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ret = ret < 0 ? ret : AVERROR_EOF; |
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goto exit; |
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} |
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ret = av_dict_set(&s->metadata, key, value, 0); |
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exit: |
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av_free(value); |
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return ret; |
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} |
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static int dss_read_header(AVFormatContext *s) |
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{ |
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DSSDemuxContext *ctx = s->priv_data; |
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AVIOContext *pb = s->pb; |
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AVStream *st; |
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int ret; |
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st = avformat_new_stream(s, NULL); |
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if (!st) |
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return AVERROR(ENOMEM); |
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ret = dss_read_metadata_string(s, DSS_HEAD_OFFSET_AUTHOR, |
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DSS_AUTHOR_SIZE, "author"); |
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if (ret) |
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return ret; |
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ret = dss_read_metadata_date(s, DSS_HEAD_OFFSET_END_TIME, "date"); |
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if (ret) |
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return ret; |
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ret = dss_read_metadata_string(s, DSS_HEAD_OFFSET_COMMENT, |
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DSS_COMMENT_SIZE, "comment"); |
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if (ret) |
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return ret; |
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avio_seek(pb, DSS_HEAD_OFFSET_ACODEC, SEEK_SET); |
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ctx->audio_codec = avio_r8(pb); |
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if (ctx->audio_codec == DSS_ACODEC_DSS_SP) { |
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st->codec->codec_id = AV_CODEC_ID_DSS_SP; |
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st->codec->sample_rate = 12000; |
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} else if (ctx->audio_codec == DSS_ACODEC_G723_1) { |
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st->codec->codec_id = AV_CODEC_ID_G723_1; |
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st->codec->sample_rate = 8000; |
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} else { |
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avpriv_request_sample(s, "Support for codec %x in DSS", |
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ctx->audio_codec); |
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return AVERROR_PATCHWELCOME; |
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} |
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st->codec->codec_type = AVMEDIA_TYPE_AUDIO; |
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st->codec->channel_layout = AV_CH_LAYOUT_MONO; |
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st->codec->channels = 1; |
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avpriv_set_pts_info(st, 64, 1, st->codec->sample_rate); |
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st->start_time = 0; |
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/* Jump over header */ |
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if (avio_seek(pb, DSS_HEADER_SIZE, SEEK_SET) != DSS_HEADER_SIZE) |
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return AVERROR(EIO); |
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ctx->counter = 0; |
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ctx->swap = 0; |
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ctx->dss_sp_buf = av_malloc(DSS_FRAME_SIZE + 1); |
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if (!ctx->dss_sp_buf) |
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return AVERROR(ENOMEM); |
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return 0; |
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} |
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static void dss_skip_audio_header(AVFormatContext *s, AVPacket *pkt) |
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{ |
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DSSDemuxContext *ctx = s->priv_data; |
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AVIOContext *pb = s->pb; |
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avio_skip(pb, DSS_AUDIO_BLOCK_HEADER_SIZE); |
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ctx->counter += DSS_BLOCK_SIZE - DSS_AUDIO_BLOCK_HEADER_SIZE; |
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} |
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static void dss_sp_byte_swap(DSSDemuxContext *ctx, |
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uint8_t *dst, const uint8_t *src) |
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{ |
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int i; |
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if (ctx->swap) { |
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for (i = 3; i < DSS_FRAME_SIZE; i += 2) |
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dst[i] = src[i]; |
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for (i = 0; i < DSS_FRAME_SIZE - 2; i += 2) |
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dst[i] = src[i + 4]; |
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dst[1] = ctx->dss_sp_swap_byte; |
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} else { |
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memcpy(dst, src, DSS_FRAME_SIZE); |
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ctx->dss_sp_swap_byte = src[DSS_FRAME_SIZE - 2]; |
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} |
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/* make sure byte 40 is always 0 */ |
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dst[DSS_FRAME_SIZE - 2] = 0; |
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ctx->swap ^= 1; |
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} |
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static int dss_sp_read_packet(AVFormatContext *s, AVPacket *pkt) |
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{ |
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DSSDemuxContext *ctx = s->priv_data; |
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int read_size, ret, offset = 0, buff_offset = 0; |
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if (ctx->counter == 0) |
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dss_skip_audio_header(s, pkt); |
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pkt->pos = avio_tell(s->pb); |
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if (ctx->swap) { |
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read_size = DSS_FRAME_SIZE - 2; |
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buff_offset = 3; |
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} else |
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read_size = DSS_FRAME_SIZE; |
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ctx->counter -= read_size; |
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ret = av_new_packet(pkt, DSS_FRAME_SIZE); |
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if (ret < 0) |
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return ret; |
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pkt->duration = 0; |
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pkt->stream_index = 0; |
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if (ctx->counter < 0) { |
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int size2 = ctx->counter + read_size; |
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ret = avio_read(s->pb, ctx->dss_sp_buf + offset + buff_offset, |
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size2 - offset); |
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if (ret < size2 - offset) |
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goto error_eof; |
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dss_skip_audio_header(s, pkt); |
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offset = size2; |
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} |
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ret = avio_read(s->pb, ctx->dss_sp_buf + offset + buff_offset, |
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read_size - offset); |
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if (ret < read_size - offset) |
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goto error_eof; |
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dss_sp_byte_swap(ctx, pkt->data, ctx->dss_sp_buf); |
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if (pkt->data[0] == 0xff) |
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return AVERROR_INVALIDDATA; |
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return pkt->size; |
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error_eof: |
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av_free_packet(pkt); |
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return ret < 0 ? ret : AVERROR_EOF; |
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} |
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static int dss_723_1_read_packet(AVFormatContext *s, AVPacket *pkt) |
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{ |
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DSSDemuxContext *ctx = s->priv_data; |
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int size, byte, ret, offset; |
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if (ctx->counter == 0) |
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dss_skip_audio_header(s, pkt); |
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pkt->pos = avio_tell(s->pb); |
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/* We make one byte-step here. Don't forget to add offset. */ |
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byte = avio_r8(s->pb); |
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if (byte == 0xff) |
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return AVERROR_INVALIDDATA; |
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size = frame_size[byte & 3]; |
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ctx->counter -= size; |
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ret = av_new_packet(pkt, size); |
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if (ret < 0) |
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return ret; |
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pkt->data[0] = byte; |
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offset = 1; |
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pkt->duration = 240; |
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pkt->stream_index = 0; |
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if (ctx->counter < 0) { |
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int size2 = ctx->counter + size; |
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ret = avio_read(s->pb, pkt->data + offset, |
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size2 - offset); |
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if (ret < size2 - offset) { |
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av_free_packet(pkt); |
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return ret < 0 ? ret : AVERROR_EOF; |
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} |
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dss_skip_audio_header(s, pkt); |
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offset = size2; |
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} |
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ret = avio_read(s->pb, pkt->data + offset, size - offset); |
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if (ret < size - offset) { |
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av_free_packet(pkt); |
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return ret < 0 ? ret : AVERROR_EOF; |
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} |
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return pkt->size; |
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} |
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static int dss_read_packet(AVFormatContext *s, AVPacket *pkt) |
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{ |
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DSSDemuxContext *ctx = s->priv_data; |
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if (ctx->audio_codec == DSS_ACODEC_DSS_SP) |
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return dss_sp_read_packet(s, pkt); |
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else |
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return dss_723_1_read_packet(s, pkt); |
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} |
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static int dss_read_close(AVFormatContext *s) |
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{ |
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DSSDemuxContext *ctx = s->priv_data; |
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av_free(ctx->dss_sp_buf); |
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return 0; |
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} |
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AVInputFormat ff_dss_demuxer = { |
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.name = "dss", |
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.long_name = NULL_IF_CONFIG_SMALL("Digital Speech Standard (DSS)"), |
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.priv_data_size = sizeof(DSSDemuxContext), |
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.read_probe = dss_probe, |
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.read_header = dss_read_header, |
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.read_packet = dss_read_packet, |
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.read_close = dss_read_close, |
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.extensions = "dss" |
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}; |
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