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- /*
- * 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/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_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[DSS_FRAME_SIZE + 1];
-
- int packet_size;
- int dss_header_size;
- } DSSDemuxContext;
-
- static int dss_probe(const AVProbeData *p)
- {
- if ( AV_RL32(p->buf) != MKTAG(0x2, 'd', 's', 's')
- && AV_RL32(p->buf) != MKTAG(0x3, '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) {
- av_free(value);
- return ret < 0 ? ret : AVERROR_EOF;
- }
-
- return av_dict_set(&s->metadata, key, value, AV_DICT_DONT_STRDUP_VAL);
- }
-
- static int dss_read_header(AVFormatContext *s)
- {
- DSSDemuxContext *ctx = s->priv_data;
- AVIOContext *pb = s->pb;
- AVStream *st;
- int ret, version;
-
- st = avformat_new_stream(s, NULL);
- if (!st)
- return AVERROR(ENOMEM);
-
- version = avio_r8(pb);
- ctx->dss_header_size = version * DSS_BLOCK_SIZE;
-
- 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->codecpar->codec_id = AV_CODEC_ID_DSS_SP;
- st->codecpar->sample_rate = 11025;
- } else if (ctx->audio_codec == DSS_ACODEC_G723_1) {
- st->codecpar->codec_id = AV_CODEC_ID_G723_1;
- st->codecpar->sample_rate = 8000;
- } else {
- avpriv_request_sample(s, "Support for codec %x in DSS",
- ctx->audio_codec);
- return AVERROR_PATCHWELCOME;
- }
-
- st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
- st->codecpar->channel_layout = AV_CH_LAYOUT_MONO;
- st->codecpar->channels = 1;
-
- avpriv_set_pts_info(st, 64, 1, st->codecpar->sample_rate);
- st->start_time = 0;
-
- /* Jump over header */
-
- if (avio_seek(pb, ctx->dss_header_size, SEEK_SET) != ctx->dss_header_size)
- return AVERROR(EIO);
-
- ctx->counter = 0;
- ctx->swap = 0;
-
- 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->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->codecpar->sample_rate * 512 / (506 * pkt->duration);
-
- if (ctx->counter < read_size) {
- ret = avio_read(s->pb, ctx->dss_sp_buf + buff_offset,
- ctx->counter);
- if (ret < ctx->counter)
- goto error_eof;
-
- offset = ctx->counter;
- dss_skip_audio_header(s, pkt);
- }
- ctx->counter -= read_size;
-
- 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) {
- return AVERROR(EAGAIN);
- }
-
- return pkt->size;
-
- error_eof:
- 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--;
-
- 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->codecpar->sample_rate * 512 / (506 * pkt->duration);
-
- pkt->stream_index = 0;
-
- if (ctx->counter < size) {
- ret = avio_read(s->pb, pkt->data + offset,
- ctx->counter);
- if (ret < ctx->counter)
- return ret < 0 ? ret : AVERROR_EOF;
-
- offset += ctx->counter;
- size -= ctx->counter;
- ctx->counter = 0;
- dss_skip_audio_header(s, pkt);
- }
- ctx->counter -= size;
-
- ret = avio_read(s->pb, pkt->data + offset, size);
- if (ret < size)
- 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_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 += ctx->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_seek = dss_read_seek,
- .extensions = "dss"
- };
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