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	 ca467f2a93
			
		
	
	ca467f2a93
	
	
	
		
			
			Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com> Signed-off-by: James Almer <jamrial@gmail.com>
		
			
				
	
	
		
			366 lines
		
	
	
		
			10 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
			
		
		
	
	
			366 lines
		
	
	
		
			10 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
| /*
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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 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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| 
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| #include "libavutil/channel_layout.h"
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| #include "libavutil/intreadwrite.h"
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| 
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| #include "avformat.h"
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| #include "internal.h"
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| 
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| #define DSS_HEAD_OFFSET_AUTHOR        0xc
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| #define DSS_AUTHOR_SIZE               16
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| 
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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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| 
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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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| 
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| #define DSS_HEAD_OFFSET_COMMENT       0x31e
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| #define DSS_COMMENT_SIZE              64
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| 
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| #define DSS_BLOCK_SIZE                512
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| #define DSS_AUDIO_BLOCK_HEADER_SIZE   6
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| #define DSS_FRAME_SIZE                42
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| 
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| static const uint8_t frame_size[4] = { 24, 20, 4, 1 };
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| 
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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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| 
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|     int packet_size;
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|     int dss_header_size;
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| } DSSDemuxContext;
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| 
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| static int dss_probe(const AVProbeData *p)
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| {
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|     if (   AV_RL32(p->buf) != MKTAG(0x2, 'd', 's', 's')
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|         && AV_RL32(p->buf) != MKTAG(0x3, 'd', 's', 's'))
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|         return 0;
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| 
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|     return AVPROBE_SCORE_MAX;
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| }
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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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| 
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|     avio_seek(pb, offset, SEEK_SET);
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| 
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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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| 
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|     if (sscanf(string, "%2d%2d%2d%2d%2d%2d", &y, &month, &d, &h, &minute, &sec) != 6)
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|         return AVERROR_INVALIDDATA;
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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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| 
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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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| 
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|     avio_seek(pb, offset, SEEK_SET);
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| 
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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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| 
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|     ret = avio_read(s->pb, value, size);
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|     if (ret < size) {
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|         av_free(value);
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|         return ret < 0 ? ret : AVERROR_EOF;
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|     }
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| 
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|     return av_dict_set(&s->metadata, key, value, AV_DICT_DONT_STRDUP_VAL);
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| }
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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, version;
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| 
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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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| 
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|     version = avio_r8(pb);
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|     ctx->dss_header_size = version * DSS_BLOCK_SIZE;
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| 
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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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| 
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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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| 
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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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| 
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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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| 
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|     if (ctx->audio_codec == DSS_ACODEC_DSS_SP) {
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|         st->codecpar->codec_id    = AV_CODEC_ID_DSS_SP;
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|         st->codecpar->sample_rate = 11025;
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|         s->bit_rate = 8 * (DSS_FRAME_SIZE - 1) * st->codecpar->sample_rate
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|                         * 512 / (506 * 264);
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|     } else if (ctx->audio_codec == DSS_ACODEC_G723_1) {
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|         st->codecpar->codec_id    = AV_CODEC_ID_G723_1;
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|         st->codecpar->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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| 
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|     st->codecpar->codec_type     = AVMEDIA_TYPE_AUDIO;
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|     st->codecpar->ch_layout      = (AVChannelLayout)AV_CHANNEL_LAYOUT_MONO;
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| 
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|     avpriv_set_pts_info(st, 64, 1, st->codecpar->sample_rate);
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|     st->start_time = 0;
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| 
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|     /* Jump over header */
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| 
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|     if (avio_seek(pb, ctx->dss_header_size, SEEK_SET) != ctx->dss_header_size)
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|         return AVERROR(EIO);
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| 
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|     ctx->counter = 0;
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|     ctx->swap    = 0;
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| 
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|     return 0;
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| }
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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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| 
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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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| 
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| static void dss_sp_byte_swap(DSSDemuxContext *ctx, uint8_t *data)
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| {
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|     int i;
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| 
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|     if (ctx->swap) {
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|         for (i = 0; i < DSS_FRAME_SIZE - 2; i += 2)
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|             data[i] = data[i + 4];
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| 
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|         /* Zero the padding. */
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|         data[DSS_FRAME_SIZE] = 0;
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|         data[1] = ctx->dss_sp_swap_byte;
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|     } else {
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|         ctx->dss_sp_swap_byte = data[DSS_FRAME_SIZE - 2];
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|     }
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| 
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|     /* make sure byte 40 is always 0 */
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|     data[DSS_FRAME_SIZE - 2] = 0;
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|     ctx->swap             ^= 1;
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| }
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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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|     int64_t pos = avio_tell(s->pb);
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| 
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|     if (ctx->counter == 0)
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|         dss_skip_audio_header(s, pkt);
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| 
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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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| 
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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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| 
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|     pkt->duration     = 264;
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|     pkt->pos = pos;
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|     pkt->stream_index = 0;
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| 
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|     if (ctx->counter < read_size) {
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|         ret = avio_read(s->pb, pkt->data + buff_offset,
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|                         ctx->counter);
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|         if (ret < ctx->counter)
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|             goto error_eof;
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| 
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|         offset = ctx->counter;
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|         dss_skip_audio_header(s, pkt);
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|     }
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|     ctx->counter -= read_size;
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| 
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|     /* This will write one byte into pkt's padding if buff_offset == 3 */
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|     ret = avio_read(s->pb, pkt->data + 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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| 
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|     dss_sp_byte_swap(ctx, pkt->data);
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| 
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|     if (ctx->dss_sp_swap_byte < 0) {
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|         return AVERROR(EAGAIN);
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|     }
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| 
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|     return 0;
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| 
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| error_eof:
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|     return ret < 0 ? ret : AVERROR_EOF;
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| }
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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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|     AVStream *st = s->streams[0];
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|     int size, byte, ret, offset;
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|     int64_t pos = avio_tell(s->pb);
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| 
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|     if (ctx->counter == 0)
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|         dss_skip_audio_header(s, pkt);
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| 
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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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| 
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|     size = frame_size[byte & 3];
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| 
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|     ctx->packet_size = size;
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|     ctx->counter--;
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| 
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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->pos = pos;
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| 
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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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|     s->bit_rate = 8LL * size-- * st->codecpar->sample_rate * 512 / (506 * pkt->duration);
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| 
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|     pkt->stream_index = 0;
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| 
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|     if (ctx->counter < size) {
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|         ret = avio_read(s->pb, pkt->data + offset,
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|                         ctx->counter);
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|         if (ret < ctx->counter)
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|             return ret < 0 ? ret : AVERROR_EOF;
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| 
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|         offset += ctx->counter;
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|         size   -= ctx->counter;
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|         ctx->counter = 0;
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|         dss_skip_audio_header(s, pkt);
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|     }
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|     ctx->counter -= size;
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| 
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|     ret = avio_read(s->pb, pkt->data + offset, size);
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|     if (ret < size)
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|         return ret < 0 ? ret : AVERROR_EOF;
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| 
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|     return 0;
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| }
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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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| 
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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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| 
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| static int dss_read_seek(AVFormatContext *s, int stream_index,
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|                          int64_t timestamp, int flags)
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| {
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|     DSSDemuxContext *ctx = s->priv_data;
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|     int64_t ret, seekto;
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|     uint8_t header[DSS_AUDIO_BLOCK_HEADER_SIZE];
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|     int offset;
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| 
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|     if (ctx->audio_codec == DSS_ACODEC_DSS_SP)
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|         seekto = timestamp / 264 * 41 / 506 * 512;
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|     else
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|         seekto = timestamp / 240 * ctx->packet_size / 506 * 512;
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| 
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|     if (seekto < 0)
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|         seekto = 0;
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| 
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|     seekto += ctx->dss_header_size;
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| 
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|     ret = avio_seek(s->pb, seekto, SEEK_SET);
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|     if (ret < 0)
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|         return ret;
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| 
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|     avio_read(s->pb, header, DSS_AUDIO_BLOCK_HEADER_SIZE);
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|     ctx->swap = !!(header[0] & 0x80);
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|     offset = 2*header[1] + 2*ctx->swap;
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|     if (offset < DSS_AUDIO_BLOCK_HEADER_SIZE)
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|         return AVERROR_INVALIDDATA;
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|     if (offset == DSS_AUDIO_BLOCK_HEADER_SIZE) {
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|         ctx->counter = 0;
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|         offset = avio_skip(s->pb, -DSS_AUDIO_BLOCK_HEADER_SIZE);
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|     } else {
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|         ctx->counter = DSS_BLOCK_SIZE - offset;
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|         offset = avio_skip(s->pb, offset - DSS_AUDIO_BLOCK_HEADER_SIZE);
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|     }
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|     ctx->dss_sp_swap_byte = -1;
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|     return 0;
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| }
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| 
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| 
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| const 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_seek      = dss_read_seek,
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|     .extensions     = "dss"
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| };
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