libavformat/dss.c - nest-android-app/ffmpeg - Git at Google

 /*
  * Digital Speech Standard (DSS) demuxer
  * Copyright (c) 2014 Oleksij Rempel <linux@rempel-privat.de>
  *
  * 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/attributes.h"
 #include "libavutil/bswap.h"
 #include "libavutil/channel_layout.h"
 #include "libavutil/intreadwrite.h"

 #include "avformat.h"
 #include "internal.h"

 #define DSS_HEAD_OFFSET_AUTHOR        0xc
 #define DSS_AUTHOR_SIZE               16

 #define DSS_HEAD_OFFSET_START_TIME    0x26
 #define DSS_HEAD_OFFSET_END_TIME      0x32
 #define DSS_TIME_SIZE                 12

 #define DSS_HEAD_OFFSET_ACODEC        0x2a4
 #define DSS_ACODEC_DSS_SP             0x0    /* SP mode */
 #define DSS_ACODEC_G723_1             0x2    /* LP mode */

 #define DSS_HEAD_OFFSET_COMMENT       0x31e
 #define DSS_COMMENT_SIZE              64

 #define DSS_BLOCK_SIZE                512
 #define DSS_HEADER_SIZE              (DSS_BLOCK_SIZE * 2)
 #define DSS_AUDIO_BLOCK_HEADER_SIZE   6
 #define DSS_FRAME_SIZE                42

 static const uint8_t frame_size[4] = { 24, 20, 4, 1 };

 typedef struct DSSDemuxContext {
     unsigned int audio_codec;
     int counter;
     int swap;
     int dss_sp_swap_byte;
     int8_t *dss_sp_buf;

     int packet_size;
 } DSSDemuxContext;

 static int dss_probe(AVProbeData *p)
 {
     if (AV_RL32(p->buf) != MKTAG(0x2, 'd', 's', 's'))
         return 0;

     return AVPROBE_SCORE_MAX;
 }

 static int dss_read_metadata_date(AVFormatContext *s, unsigned int offset,
                                   const char *key)
 {
     AVIOContext *pb = s->pb;
     char datetime[64], string[DSS_TIME_SIZE + 1] = { 0 };
     int y, month, d, h, minute, sec;
     int ret;

     avio_seek(pb, offset, SEEK_SET);

     ret = avio_read(s->pb, string, DSS_TIME_SIZE);
     if (ret < DSS_TIME_SIZE)
         return ret < 0 ? ret : AVERROR_EOF;

     if (sscanf(string, "%2d%2d%2d%2d%2d%2d", &y, &month, &d, &h, &minute, &sec) != 6)
         return AVERROR_INVALIDDATA;
     /* We deal with a two-digit year here, so set the default date to 2000
      * and hope it will never be used in the next century. */
     snprintf(datetime, sizeof(datetime), "%.4d-%.2d-%.2dT%.2d:%.2d:%.2d",
              y + 2000, month, d, h, minute, sec);
     return av_dict_set(&s->metadata, key, datetime, 0);
 }

 static int dss_read_metadata_string(AVFormatContext *s, unsigned int offset,
                                     unsigned int size, const char *key)
 {
     AVIOContext *pb = s->pb;
     char *value;
     int ret;

     avio_seek(pb, offset, SEEK_SET);

     value = av_mallocz(size + 1);
     if (!value)
         return AVERROR(ENOMEM);

     ret = avio_read(s->pb, value, size);
     if (ret < size) {
         ret = ret < 0 ? ret : AVERROR_EOF;
         goto exit;
     }

     ret = av_dict_set(&s->metadata, key, value, 0);

 exit:
     av_free(value);
     return ret;
 }

 static int dss_read_header(AVFormatContext *s)
 {
     DSSDemuxContext *ctx = s->priv_data;
     AVIOContext *pb = s->pb;
     AVStream *st;
     int ret;

     st = avformat_new_stream(s, NULL);
     if (!st)
         return AVERROR(ENOMEM);

     ret = dss_read_metadata_string(s, DSS_HEAD_OFFSET_AUTHOR,
                                    DSS_AUTHOR_SIZE, "author");
     if (ret)
         return ret;

     ret = dss_read_metadata_date(s, DSS_HEAD_OFFSET_END_TIME, "date");
     if (ret)
         return ret;

     ret = dss_read_metadata_string(s, DSS_HEAD_OFFSET_COMMENT,
                                    DSS_COMMENT_SIZE, "comment");
     if (ret)
         return ret;

     avio_seek(pb, DSS_HEAD_OFFSET_ACODEC, SEEK_SET);
     ctx->audio_codec = avio_r8(pb);

     if (ctx->audio_codec == DSS_ACODEC_DSS_SP) {
         st->codec->codec_id    = AV_CODEC_ID_DSS_SP;
         st->codec->sample_rate = 11025;
     } else if (ctx->audio_codec == DSS_ACODEC_G723_1) {
         st->codec->codec_id    = AV_CODEC_ID_G723_1;
         st->codec->sample_rate = 8000;
     } else {
         avpriv_request_sample(s, "Support for codec %x in DSS",
                               ctx->audio_codec);
         return AVERROR_PATCHWELCOME;
     }

     st->codec->codec_type     = AVMEDIA_TYPE_AUDIO;
     st->codec->channel_layout = AV_CH_LAYOUT_MONO;
     st->codec->channels       = 1;

     avpriv_set_pts_info(st, 64, 1, st->codec->sample_rate);
     st->start_time = 0;

     /* Jump over header */

     if (avio_seek(pb, DSS_HEADER_SIZE, SEEK_SET) != DSS_HEADER_SIZE)
         return AVERROR(EIO);

     ctx->counter = 0;
     ctx->swap    = 0;

     ctx->dss_sp_buf = av_malloc(DSS_FRAME_SIZE + 1);
     if (!ctx->dss_sp_buf)
         return AVERROR(ENOMEM);

     return 0;
 }

 static void dss_skip_audio_header(AVFormatContext *s, AVPacket *pkt)
 {
     DSSDemuxContext *ctx = s->priv_data;
     AVIOContext *pb = s->pb;

     avio_skip(pb, DSS_AUDIO_BLOCK_HEADER_SIZE);
     ctx->counter += DSS_BLOCK_SIZE - DSS_AUDIO_BLOCK_HEADER_SIZE;
 }

 static void dss_sp_byte_swap(DSSDemuxContext *ctx,
                              uint8_t *dst, const uint8_t *src)
 {
     int i;

     if (ctx->swap) {
         for (i = 3; i < DSS_FRAME_SIZE; i += 2)
             dst[i] = src[i];

         for (i = 0; i < DSS_FRAME_SIZE - 2; i += 2)
             dst[i] = src[i + 4];

         dst[1] = ctx->dss_sp_swap_byte;
     } else {
         memcpy(dst, src, DSS_FRAME_SIZE);
         ctx->dss_sp_swap_byte = src[DSS_FRAME_SIZE - 2];
     }

     /* make sure byte 40 is always 0 */
     dst[DSS_FRAME_SIZE - 2] = 0;
     ctx->swap             ^= 1;
 }

 static int dss_sp_read_packet(AVFormatContext *s, AVPacket *pkt)
 {
     DSSDemuxContext *ctx = s->priv_data;
     AVStream *st = s->streams[0];
     int read_size, ret, offset = 0, buff_offset = 0;
     int64_t pos = avio_tell(s->pb);

     if (ctx->counter == 0)
         dss_skip_audio_header(s, pkt);

     if (ctx->swap) {
         read_size   = DSS_FRAME_SIZE - 2;
         buff_offset = 3;
     } else
         read_size = DSS_FRAME_SIZE;

     ctx->counter -= read_size;
     ctx->packet_size = DSS_FRAME_SIZE - 1;

     ret = av_new_packet(pkt, DSS_FRAME_SIZE);
     if (ret < 0)
         return ret;

     pkt->duration     = 264;
     pkt->pos = pos;
     pkt->stream_index = 0;
     s->bit_rate = 8LL * ctx->packet_size * st->codec->sample_rate * 512 / (506 * pkt->duration);

     if (ctx->counter < 0) {
         int size2 = ctx->counter + read_size;

         ret = avio_read(s->pb, ctx->dss_sp_buf + offset + buff_offset,
                         size2 - offset);
         if (ret < size2 - offset)
             goto error_eof;

         dss_skip_audio_header(s, pkt);
         offset = size2;
     }

     ret = avio_read(s->pb, ctx->dss_sp_buf + offset + buff_offset,
                     read_size - offset);
     if (ret < read_size - offset)
         goto error_eof;

     dss_sp_byte_swap(ctx, pkt->data, ctx->dss_sp_buf);

     if (ctx->dss_sp_swap_byte < 0) {
         ret = AVERROR(EAGAIN);
         goto error_eof;
     }

     if (pkt->data[0] == 0xff)
         return AVERROR_INVALIDDATA;

     return pkt->size;

 error_eof:
     av_packet_unref(pkt);
     return ret < 0 ? ret : AVERROR_EOF;
 }

 static int dss_723_1_read_packet(AVFormatContext *s, AVPacket *pkt)
 {
     DSSDemuxContext *ctx = s->priv_data;
     AVStream *st = s->streams[0];
     int size, byte, ret, offset;
     int64_t pos = avio_tell(s->pb);

     if (ctx->counter == 0)
         dss_skip_audio_header(s, pkt);

     /* We make one byte-step here. Don't forget to add offset. */
     byte = avio_r8(s->pb);
     if (byte == 0xff)
         return AVERROR_INVALIDDATA;

     size = frame_size[byte & 3];

     ctx->packet_size = size;
     ctx->counter -= size;

     ret = av_new_packet(pkt, size);
     if (ret < 0)
         return ret;
     pkt->pos = pos;

     pkt->data[0]  = byte;
     offset        = 1;
     pkt->duration = 240;
     s->bit_rate = 8LL * size * st->codec->sample_rate * 512 / (506 * pkt->duration);

     pkt->stream_index = 0;

     if (ctx->counter < 0) {
         int size2 = ctx->counter + size;

         ret = avio_read(s->pb, pkt->data + offset,
                         size2 - offset);
         if (ret < size2 - offset) {
             av_packet_unref(pkt);
             return ret < 0 ? ret : AVERROR_EOF;
         }

         dss_skip_audio_header(s, pkt);
         offset = size2;
     }

     ret = avio_read(s->pb, pkt->data + offset, size - offset);
     if (ret < size - offset) {
         av_packet_unref(pkt);
         return ret < 0 ? ret : AVERROR_EOF;
     }

     return pkt->size;
 }

 static int dss_read_packet(AVFormatContext *s, AVPacket *pkt)
 {
     DSSDemuxContext *ctx = s->priv_data;

     if (ctx->audio_codec == DSS_ACODEC_DSS_SP)
         return dss_sp_read_packet(s, pkt);
     else
         return dss_723_1_read_packet(s, pkt);
 }

 static int dss_read_close(AVFormatContext *s)
 {
     DSSDemuxContext *ctx = s->priv_data;

     av_freep(&ctx->dss_sp_buf);

     return 0;
 }

 static int dss_read_seek(AVFormatContext *s, int stream_index,
                          int64_t timestamp, int flags)
 {
     DSSDemuxContext *ctx = s->priv_data;
     int64_t ret, seekto;
     uint8_t header[DSS_AUDIO_BLOCK_HEADER_SIZE];
     int offset;

     if (ctx->audio_codec == DSS_ACODEC_DSS_SP)
         seekto = timestamp / 264 * 41 / 506 * 512;
     else
         seekto = timestamp / 240 * ctx->packet_size / 506 * 512;

     if (seekto < 0)
         seekto = 0;

     seekto += DSS_HEADER_SIZE;

     ret = avio_seek(s->pb, seekto, SEEK_SET);
     if (ret < 0)
         return ret;

     avio_read(s->pb, header, DSS_AUDIO_BLOCK_HEADER_SIZE);
     ctx->swap = !!(header[0] & 0x80);
     offset = 2*header[1] + 2*ctx->swap;
     if (offset < DSS_AUDIO_BLOCK_HEADER_SIZE)
         return AVERROR_INVALIDDATA;
     if (offset == DSS_AUDIO_BLOCK_HEADER_SIZE) {
         ctx->counter = 0;
         offset = avio_skip(s->pb, -DSS_AUDIO_BLOCK_HEADER_SIZE);
     } else {
         ctx->counter = DSS_BLOCK_SIZE - offset;
         offset = avio_skip(s->pb, offset - DSS_AUDIO_BLOCK_HEADER_SIZE);
     }
     ctx->dss_sp_swap_byte = -1;
     return 0;
 }


 AVInputFormat ff_dss_demuxer = {
     .name           = "dss",
     .long_name      = NULL_IF_CONFIG_SMALL("Digital Speech Standard (DSS)"),
     .priv_data_size = sizeof(DSSDemuxContext),
     .read_probe     = dss_probe,
     .read_header    = dss_read_header,
     .read_packet    = dss_read_packet,
     .read_close     = dss_read_close,
     .read_seek      = dss_read_seek,
     .extensions     = "dss"
 };
	/*
	* Digital Speech Standard (DSS) demuxer
	* Copyright (c) 2014 Oleksij Rempel <linux@rempel-privat.de>
	*
	* 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/attributes.h"
	#include "libavutil/bswap.h"
	#include "libavutil/channel_layout.h"
	#include "libavutil/intreadwrite.h"

	#include "avformat.h"
	#include "internal.h"

	#define DSS_HEAD_OFFSET_AUTHOR 0xc
	#define DSS_AUTHOR_SIZE 16

	#define DSS_HEAD_OFFSET_START_TIME 0x26
	#define DSS_HEAD_OFFSET_END_TIME 0x32
	#define DSS_TIME_SIZE 12

	#define DSS_HEAD_OFFSET_ACODEC 0x2a4
	#define DSS_ACODEC_DSS_SP 0x0 /* SP mode */
	#define DSS_ACODEC_G723_1 0x2 /* LP mode */

	#define DSS_HEAD_OFFSET_COMMENT 0x31e
	#define DSS_COMMENT_SIZE 64

	#define DSS_BLOCK_SIZE 512
	#define DSS_HEADER_SIZE (DSS_BLOCK_SIZE * 2)
	#define DSS_AUDIO_BLOCK_HEADER_SIZE 6
	#define DSS_FRAME_SIZE 42

	static const uint8_t frame_size[4] = { 24, 20, 4, 1 };

	typedef struct DSSDemuxContext {
	unsigned int audio_codec;
	int counter;
	int swap;
	int dss_sp_swap_byte;
	int8_t *dss_sp_buf;

	int packet_size;
	} DSSDemuxContext;

	static int dss_probe(AVProbeData *p)
	{
	if (AV_RL32(p->buf) != MKTAG(0x2, 'd', 's', 's'))
	return 0;

	return AVPROBE_SCORE_MAX;
	}

	static int dss_read_metadata_date(AVFormatContext *s, unsigned int offset,
	const char *key)
	{
	AVIOContext *pb = s->pb;
	char datetime[64], string[DSS_TIME_SIZE + 1] = { 0 };
	int y, month, d, h, minute, sec;
	int ret;

	avio_seek(pb, offset, SEEK_SET);

	ret = avio_read(s->pb, string, DSS_TIME_SIZE);
	if (ret < DSS_TIME_SIZE)
	return ret < 0 ? ret : AVERROR_EOF;

	if (sscanf(string, "%2d%2d%2d%2d%2d%2d", &y, &month, &d, &h, &minute, &sec) != 6)
	return AVERROR_INVALIDDATA;
	/* We deal with a two-digit year here, so set the default date to 2000
	* and hope it will never be used in the next century. */
	snprintf(datetime, sizeof(datetime), "%.4d-%.2d-%.2dT%.2d:%.2d:%.2d",
	y + 2000, month, d, h, minute, sec);
	return av_dict_set(&s->metadata, key, datetime, 0);
	}

	static int dss_read_metadata_string(AVFormatContext *s, unsigned int offset,
	unsigned int size, const char *key)
	{
	AVIOContext *pb = s->pb;
	char *value;
	int ret;

	avio_seek(pb, offset, SEEK_SET);

	value = av_mallocz(size + 1);
	if (!value)
	return AVERROR(ENOMEM);

	ret = avio_read(s->pb, value, size);
	if (ret < size) {
	ret = ret < 0 ? ret : AVERROR_EOF;
	goto exit;
	}

	ret = av_dict_set(&s->metadata, key, value, 0);

	exit:
	av_free(value);
	return ret;
	}

	static int dss_read_header(AVFormatContext *s)
	{
	DSSDemuxContext *ctx = s->priv_data;
	AVIOContext *pb = s->pb;
	AVStream *st;
	int ret;

	st = avformat_new_stream(s, NULL);
	if (!st)
	return AVERROR(ENOMEM);

	ret = dss_read_metadata_string(s, DSS_HEAD_OFFSET_AUTHOR,
	DSS_AUTHOR_SIZE, "author");
	if (ret)
	return ret;

	ret = dss_read_metadata_date(s, DSS_HEAD_OFFSET_END_TIME, "date");
	if (ret)
	return ret;

	ret = dss_read_metadata_string(s, DSS_HEAD_OFFSET_COMMENT,
	DSS_COMMENT_SIZE, "comment");
	if (ret)
	return ret;

	avio_seek(pb, DSS_HEAD_OFFSET_ACODEC, SEEK_SET);
	ctx->audio_codec = avio_r8(pb);

	if (ctx->audio_codec == DSS_ACODEC_DSS_SP) {
	st->codec->codec_id = AV_CODEC_ID_DSS_SP;
	st->codec->sample_rate = 11025;
	} else if (ctx->audio_codec == DSS_ACODEC_G723_1) {
	st->codec->codec_id = AV_CODEC_ID_G723_1;
	st->codec->sample_rate = 8000;
	} else {
	avpriv_request_sample(s, "Support for codec %x in DSS",
	ctx->audio_codec);
	return AVERROR_PATCHWELCOME;
	}

	st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
	st->codec->channel_layout = AV_CH_LAYOUT_MONO;
	st->codec->channels = 1;

	avpriv_set_pts_info(st, 64, 1, st->codec->sample_rate);
	st->start_time = 0;

	/* Jump over header */

	if (avio_seek(pb, DSS_HEADER_SIZE, SEEK_SET) != DSS_HEADER_SIZE)
	return AVERROR(EIO);

	ctx->counter = 0;
	ctx->swap = 0;

	ctx->dss_sp_buf = av_malloc(DSS_FRAME_SIZE + 1);
	if (!ctx->dss_sp_buf)
	return AVERROR(ENOMEM);

	return 0;
	}

	static void dss_skip_audio_header(AVFormatContext s, AVPacket pkt)
	{
	DSSDemuxContext *ctx = s->priv_data;
	AVIOContext *pb = s->pb;

	avio_skip(pb, DSS_AUDIO_BLOCK_HEADER_SIZE);
	ctx->counter += DSS_BLOCK_SIZE - DSS_AUDIO_BLOCK_HEADER_SIZE;
	}

	static void dss_sp_byte_swap(DSSDemuxContext *ctx,
	uint8_t dst, const uint8_t src)
	{
	int i;

	if (ctx->swap) {
	for (i = 3; i < DSS_FRAME_SIZE; i += 2)
	dst[i] = src[i];

	for (i = 0; i < DSS_FRAME_SIZE - 2; i += 2)
	dst[i] = src[i + 4];

	dst[1] = ctx->dss_sp_swap_byte;
	} else {
	memcpy(dst, src, DSS_FRAME_SIZE);
	ctx->dss_sp_swap_byte = src[DSS_FRAME_SIZE - 2];
	}

	/* make sure byte 40 is always 0 */
	dst[DSS_FRAME_SIZE - 2] = 0;
	ctx->swap ^= 1;
	}

	static int dss_sp_read_packet(AVFormatContext s, AVPacket pkt)
	{
	DSSDemuxContext *ctx = s->priv_data;
	AVStream *st = s->streams[0];
	int read_size, ret, offset = 0, buff_offset = 0;
	int64_t pos = avio_tell(s->pb);

	if (ctx->counter == 0)
	dss_skip_audio_header(s, pkt);

	if (ctx->swap) {
	read_size = DSS_FRAME_SIZE - 2;
	buff_offset = 3;
	} else
	read_size = DSS_FRAME_SIZE;

	ctx->counter -= read_size;
	ctx->packet_size = DSS_FRAME_SIZE - 1;

	ret = av_new_packet(pkt, DSS_FRAME_SIZE);
	if (ret < 0)
	return ret;

	pkt->duration = 264;
	pkt->pos = pos;
	pkt->stream_index = 0;
	s->bit_rate = 8LL * ctx->packet_size * st->codec->sample_rate * 512 / (506 * pkt->duration);

	if (ctx->counter < 0) {
	int size2 = ctx->counter + read_size;

	ret = avio_read(s->pb, ctx->dss_sp_buf + offset + buff_offset,
	size2 - offset);
	if (ret < size2 - offset)
	goto error_eof;

	dss_skip_audio_header(s, pkt);
	offset = size2;
	}

	ret = avio_read(s->pb, ctx->dss_sp_buf + offset + buff_offset,
	read_size - offset);
	if (ret < read_size - offset)
	goto error_eof;

	dss_sp_byte_swap(ctx, pkt->data, ctx->dss_sp_buf);

	if (ctx->dss_sp_swap_byte < 0) {
	ret = AVERROR(EAGAIN);
	goto error_eof;
	}

	if (pkt->data[0] == 0xff)
	return AVERROR_INVALIDDATA;

	return pkt->size;

	error_eof:
	av_packet_unref(pkt);
	return ret < 0 ? ret : AVERROR_EOF;
	}

	static int dss_723_1_read_packet(AVFormatContext s, AVPacket pkt)
	{
	DSSDemuxContext *ctx = s->priv_data;
	AVStream *st = s->streams[0];
	int size, byte, ret, offset;
	int64_t pos = avio_tell(s->pb);

	if (ctx->counter == 0)
	dss_skip_audio_header(s, pkt);

	/* We make one byte-step here. Don't forget to add offset. */
	byte = avio_r8(s->pb);
	if (byte == 0xff)
	return AVERROR_INVALIDDATA;

	size = frame_size[byte & 3];

	ctx->packet_size = size;
	ctx->counter -= size;

	ret = av_new_packet(pkt, size);
	if (ret < 0)
	return ret;
	pkt->pos = pos;

	pkt->data[0] = byte;
	offset = 1;
	pkt->duration = 240;
	s->bit_rate = 8LL * size * st->codec->sample_rate * 512 / (506 * pkt->duration);

	pkt->stream_index = 0;

	if (ctx->counter < 0) {
	int size2 = ctx->counter + size;

	ret = avio_read(s->pb, pkt->data + offset,
	size2 - offset);
	if (ret < size2 - offset) {
	av_packet_unref(pkt);
	return ret < 0 ? ret : AVERROR_EOF;
	}

	dss_skip_audio_header(s, pkt);
	offset = size2;
	}

	ret = avio_read(s->pb, pkt->data + offset, size - offset);
	if (ret < size - offset) {
	av_packet_unref(pkt);
	return ret < 0 ? ret : AVERROR_EOF;
	}

	return pkt->size;
	}

	static int dss_read_packet(AVFormatContext s, AVPacket pkt)
	{
	DSSDemuxContext *ctx = s->priv_data;

	if (ctx->audio_codec == DSS_ACODEC_DSS_SP)
	return dss_sp_read_packet(s, pkt);
	else
	return dss_723_1_read_packet(s, pkt);
	}

	static int dss_read_close(AVFormatContext *s)
	{
	DSSDemuxContext *ctx = s->priv_data;

	av_freep(&ctx->dss_sp_buf);

	return 0;
	}

	static int dss_read_seek(AVFormatContext *s, int stream_index,
	int64_t timestamp, int flags)
	{
	DSSDemuxContext *ctx = s->priv_data;
	int64_t ret, seekto;
	uint8_t header[DSS_AUDIO_BLOCK_HEADER_SIZE];
	int offset;

	if (ctx->audio_codec == DSS_ACODEC_DSS_SP)
	seekto = timestamp / 264 * 41 / 506 * 512;
	else
	seekto = timestamp / 240 * ctx->packet_size / 506 * 512;

	if (seekto < 0)
	seekto = 0;

	seekto += DSS_HEADER_SIZE;

	ret = avio_seek(s->pb, seekto, SEEK_SET);
	if (ret < 0)
	return ret;

	avio_read(s->pb, header, DSS_AUDIO_BLOCK_HEADER_SIZE);
	ctx->swap = !!(header[0] & 0x80);
	offset = 2header[1] + 2ctx->swap;
	if (offset < DSS_AUDIO_BLOCK_HEADER_SIZE)
	return AVERROR_INVALIDDATA;
	if (offset == DSS_AUDIO_BLOCK_HEADER_SIZE) {
	ctx->counter = 0;
	offset = avio_skip(s->pb, -DSS_AUDIO_BLOCK_HEADER_SIZE);
	} else {
	ctx->counter = DSS_BLOCK_SIZE - offset;
	offset = avio_skip(s->pb, offset - DSS_AUDIO_BLOCK_HEADER_SIZE);
	}
	ctx->dss_sp_swap_byte = -1;
	return 0;
	}


	AVInputFormat ff_dss_demuxer = {
	.name = "dss",
	.long_name = NULL_IF_CONFIG_SMALL("Digital Speech Standard (DSS)"),
	.priv_data_size = sizeof(DSSDemuxContext),
	.read_probe = dss_probe,
	.read_header = dss_read_header,
	.read_packet = dss_read_packet,
	.read_close = dss_read_close,
	.read_seek = dss_read_seek,
	.extensions = "dss"
	};