libavformat.hg: sierravmd.c comparison

comparison sierravmd.c @ 387:f2760852ed18 libavformat

minor VMD system update; still not perfect, but should not crash either

author	melanson
date	Sun, 14 Mar 2004 04:08:11 +0000
parents	6f50cb0ead51
children	b69898ffc92a

comparison

equal deleted inserted replaced

-:c152849ee643
+:f2760852ed18
 /**
 * @file sierravmd.c
 * Sierra VMD file demuxer
 * by Vladimir "VAG" Gneushev (vagsoft at mail.ru)
+* for more information on the Sierra VMD file format, visit:
+*   http://www.pcisys.net/~melanson/codecs/
 */
 #include "avformat.h"
-#define LE_16(x)  ((((uint8_t*)(x))[1] << 8) | ((uint8_t*)(x))[0])
-#define LE_32(x)  ((((uint8_t*)(x))[3] << 24) | \
-(((uint8_t*)(x))[2] << 16) | \
-(((uint8_t*)(x))[1] << 8) | \
-((uint8_t*)(x))[0])
 #define VMD_HEADER_SIZE 0x0330
 #define BYTES_PER_FRAME_RECORD 16
 typedef struct {
 unsigned int frame_count;
 vmd_frame_t *frame_table;
 unsigned int current_frame;
+int sample_rate;
+int64_t audio_sample_counter;
+int audio_frame_divisor;
+int audio_block_align;
 unsigned char vmd_header[VMD_HEADER_SIZE];
 } VmdDemuxContext;
 static int vmd_probe(AVProbeData *p)
 {
 if (LE_16(&p->buf[0]) != VMD_HEADER_SIZE - 2)
 return 0;
 /* only return half certainty since this check is a bit sketchy */
 return AVPROBE_SCORE_MAX / 2;
+}
+/* This is a support function to determine the duration, in sample
+* frames, of a particular audio chunk, taking into account silent
+* encodings. */
+static int vmd_calculate_audio_duration(unsigned char *audio_chunk,
+int audio_chunk_size, int block_align)
+{
+unsigned char *p = audio_chunk + 16;
+unsigned char *p_end = audio_chunk + audio_chunk_size;
+int total_samples = 0;
+unsigned int sound_flags;
+if (audio_chunk_size < 16)
+return 0;
+sound_flags = LE_32(p);
+p += 4;
+while (p < p_end) {
+total_samples += block_align;
+if ((sound_flags & 0x01) == 0)
+p += block_align;
+sound_flags >>= 1;
+}
+return total_samples;
 }
 static int vmd_read_header(AVFormatContext *s,
 AVFormatParameters *ap)
 {
 unsigned char *raw_frame_table;
 int raw_frame_table_size;
 unsigned char *current_frame_record;
 offset_t current_offset;
 int i;
-int sample_rate;
 unsigned int total_frames;
-int video_frame_count = 0;
+int64_t video_pts_inc;
+int64_t current_video_pts = 0;
 /* fetch the main header, including the 2 header length bytes */
 url_fseek(pb, 0, SEEK_SET);
 if (get_buffer(pb, vmd->vmd_header, VMD_HEADER_SIZE) != VMD_HEADER_SIZE)
 return -EIO;
+vmd->audio_sample_counter = 0;
+vmd->audio_frame_divisor = 1;
+vmd->audio_block_align = 1;
 /* start up the decoders */
 st = av_new_stream(s, 0);
 if (!st)
 return AVERROR_NOMEM;
 st->codec.extradata_size = VMD_HEADER_SIZE;
 st->codec.extradata = av_malloc(VMD_HEADER_SIZE);
 memcpy(st->codec.extradata, vmd->vmd_header, VMD_HEADER_SIZE);
 /* if sample rate is 0, assume no audio */
-sample_rate = LE_16(&vmd->vmd_header[804]);
+vmd->sample_rate = LE_16(&vmd->vmd_header[804]);
-if (sample_rate) {
+if (vmd->sample_rate) {
 st = av_new_stream(s, 0);
 if (!st)
 return AVERROR_NOMEM;
 vmd->audio_stream_index = st->index;
 st->codec.codec_type = CODEC_TYPE_AUDIO;
 st->codec.codec_id = CODEC_ID_VMDAUDIO;
 st->codec.codec_tag = 0;  /* no codec tag */
 st->codec.channels = (vmd->vmd_header[811] & 0x80) ? 2 : 1;
-st->codec.sample_rate = sample_rate;
+st->codec.sample_rate = vmd->sample_rate;
-st->codec.bit_rate = st->codec.sample_rate *
+st->codec.block_align = vmd->audio_block_align =
-st->codec.bits_per_sample * st->codec.channels;
+LE_16(&vmd->vmd_header[806]);
-st->codec.block_align = LE_16(&vmd->vmd_header[806]);
 if (st->codec.block_align & 0x8000) {
 st->codec.bits_per_sample = 16;
 st->codec.block_align = -(st->codec.block_align - 0x10000);
 } else
-st->codec.bits_per_sample = 8;
+st->codec.bits_per_sample = 16;
+//            st->codec.bits_per_sample = 8;
+st->codec.bit_rate = st->codec.sample_rate *
+st->codec.bits_per_sample * st->codec.channels;
+/* for calculating pts */
+vmd->audio_frame_divisor = st->codec.bits_per_sample / 8 /
+st->codec.channels;
+video_pts_inc = 90000;
+video_pts_inc *= st->codec.block_align;
+video_pts_inc /= st->codec.sample_rate;
+} else {
+/* if no audio, assume 10 frames/second */
+video_pts_inc = 90000 / 10;
 }
 /* skip over the offset table and load the table of contents; don't
 * care about the offset table since demuxer will calculate those
 * independently */
 vmd->frame_table[i].frame_offset = current_offset;
 current_offset += vmd->frame_table[i].frame_size;
 memcpy(vmd->frame_table[i].frame_record, current_frame_record,
 BYTES_PER_FRAME_RECORD);
-if (current_frame_record[0] == 0x02) {
+/* figure out the pts for this frame */
-/* assume 15 fps for now */
+if (current_frame_record[0] == 0x02) {
-vmd->frame_table[i].pts = video_frame_count++;
+vmd->frame_table[i].pts = current_video_pts;
-vmd->frame_table[i].pts *= 90000;
+current_video_pts += video_pts_inc;
-vmd->frame_table[i].pts /= 15;
+} else if (current_frame_record[0] == 0x01) {
-}
+/* figure out the pts during the dispatch phase */
+vmd->frame_table[i].pts = 0;
+}
 current_frame_record += BYTES_PER_FRAME_RECORD;
 i++;
 }
 return AVERROR_NOMEM;
 memcpy(pkt->data, frame->frame_record, BYTES_PER_FRAME_RECORD);
 ret = get_buffer(pb, pkt->data + BYTES_PER_FRAME_RECORD,
 frame->frame_size);
-if (ret != frame->frame_size)
+if (ret != frame->frame_size) {
+av_free_packet(pkt);
 ret = -EIO;
+}
 pkt->stream_index = frame->stream_index;
-pkt->pts = frame->pts;
+if (frame->frame_record[0] == 0x02)
+pkt->pts = frame->pts;
+else {
+pkt->pts = vmd->audio_sample_counter;
+pkt->pts *= 90000;
+pkt->pts /= vmd->sample_rate;
+//        pkt->pts /= vmd->audio_frame_divisor;
+vmd->audio_sample_counter += vmd_calculate_audio_duration(
+pkt->data, pkt->size, vmd->audio_block_align);
+}
+printf (" dispatching %s frame with %d bytes and pts %lld (%0.1f sec)\n",
+(frame->frame_record[0] == 0x02) ? "video" : "audio",
+frame->frame_size + BYTES_PER_FRAME_RECORD,
+pkt->pts, (float)(pkt->pts / 90000.0));
 vmd->current_frame++;
 return ret;
 }

Mercurial > libavformat.hg

comparison sierravmd.c @ 387:f2760852ed18 libavformat