* Raw DV format
* Copyright (c) 2002 Fabrice Bellard.
*
- * 50 Mbps (DVCPRO50) support
+ * 50 Mbps (DVCPRO50) and 100 Mbps (DVCPRO HD) support
* Copyright (c) 2006 Daniel Maas <dmaas@maasdigital.com>
+ * Funded by BBC Research & Development
*
- * This library is free software; you can redistribute it and/or
+ * 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 of the License, or (at your option) any later version.
+ * version 2.1 of the License, or (at your option) any later version.
*
- * This library is distributed in the hope that it will be useful,
+ * 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 this library; if not, write to the Free Software
+ * License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <time.h>
#include "avformat.h"
-#include "dvdata.h"
+#include "libavcodec/dvdata.h"
#include "dv.h"
struct DVDemuxContext {
const DVprofile* sys; /* Current DV profile. E.g.: 525/60, 625/50 */
- AVFormatContext* fctx;
- AVStream* vst;
- AVStream* ast[2];
- AVPacket audio_pkt[2];
- uint8_t audio_buf[2][8192];
- int ach;
- int frames;
- uint64_t abytes;
+ AVFormatContext* fctx;
+ AVStream* vst;
+ AVStream* ast[4];
+ AVPacket audio_pkt[4];
+ uint8_t audio_buf[4][8192];
+ int ach;
+ int frames;
+ uint64_t abytes;
};
static inline uint16_t dv_audio_12to16(uint16_t sample)
uint16_t shift, result;
sample = (sample < 0x800) ? sample : sample | 0xf000;
- shift = (sample & 0xf00) >> 8;
+ shift = (sample & 0xf00) >> 8;
if (shift < 0x2 || shift > 0xd) {
result = sample;
switch (t) {
case dv_audio_source:
- offs = (80*6 + 80*16*3 + 3);
- break;
+ offs = (80*6 + 80*16*3 + 3);
+ break;
case dv_audio_control:
- offs = (80*6 + 80*16*4 + 3);
- break;
+ offs = (80*6 + 80*16*4 + 3);
+ break;
case dv_video_control:
- offs = (80*5 + 48 + 5);
- break;
+ offs = (80*5 + 48 + 5);
+ break;
default:
- return NULL;
+ return NULL;
}
- return (frame[offs] == t ? &frame[offs] : NULL);
+ return frame[offs] == t ? &frame[offs] : NULL;
}
/*
* 3. Audio is always returned as 16bit linear samples: 12bit nonlinear samples
* are converted into 16bit linear ones.
*/
-static int dv_extract_audio(uint8_t* frame, uint8_t* pcm, uint8_t* pcm2,
+static int dv_extract_audio(uint8_t* frame, uint8_t* ppcm[4],
const DVprofile *sys)
{
int size, chan, i, j, d, of, smpls, freq, quant, half_ch;
uint16_t lc, rc;
const uint8_t* as_pack;
+ uint8_t *pcm, ipcm;
as_pack = dv_extract_pack(frame, dv_audio_source);
if (!as_pack) /* No audio ? */
return 0;
- smpls = as_pack[1] & 0x3f; /* samples in this frame - min. samples */
- freq = (as_pack[4] >> 3) & 0x07; /* 0 - 48KHz, 1 - 44,1kHz, 2 - 32 kHz */
- quant = as_pack[4] & 0x07; /* 0 - 16bit linear, 1 - 12bit nonlinear */
+ smpls = as_pack[1] & 0x3f; /* samples in this frame - min. samples */
+ freq = (as_pack[4] >> 3) & 0x07; /* 0 - 48kHz, 1 - 44,1kHz, 2 - 32kHz */
+ quant = as_pack[4] & 0x07; /* 0 - 16bit linear, 1 - 12bit nonlinear */
if (quant > 1)
- return -1; /* Unsupported quantization */
+ return -1; /* unsupported quantization */
size = (sys->audio_min_samples[freq] + smpls) * 4; /* 2ch, 2bytes */
- half_ch = sys->difseg_size/2;
+ half_ch = sys->difseg_size / 2;
+
+ /* We work with 720p frames split in half, thus even frames have
+ * channels 0,1 and odd 2,3. */
+ ipcm = (sys->height == 720 && !(frame[1] & 0x0C)) ? 2 : 0;
+ pcm = ppcm[ipcm++];
/* for each DIF channel */
for (chan = 0; chan < sys->n_difchan; chan++) {
frame += 6 * 80; /* skip DIF segment header */
if (quant == 1 && i == half_ch) {
/* next stereo channel (12bit mode only) */
- if (!pcm2)
+ pcm = ppcm[ipcm++];
+ if (!pcm)
break;
- else
- pcm = pcm2;
}
/* for each AV sequence */
for (j = 0; j < 9; j++) {
for (d = 8; d < 80; d += 2) {
if (quant == 0) { /* 16bit quantization */
- of = sys->audio_shuffle[i][j] + (d - 8)/2 * sys->audio_stride;
+ of = sys->audio_shuffle[i][j] + (d - 8) / 2 * sys->audio_stride;
if (of*2 >= size)
continue;
- pcm[of*2] = frame[d+1]; // FIXME: may be we have to admit
- pcm[of*2+1] = frame[d]; // that DV is a big endian PCM
+ pcm[of*2] = frame[d+1]; // FIXME: maybe we have to admit
+ pcm[of*2+1] = frame[d]; // that DV is a big-endian PCM
if (pcm[of*2+1] == 0x80 && pcm[of*2] == 0x00)
pcm[of*2+1] = 0;
} else { /* 12bit quantization */
- lc = ((uint16_t)frame[d] << 4) |
+ lc = ((uint16_t)frame[d] << 4) |
((uint16_t)frame[d+2] >> 4);
rc = ((uint16_t)frame[d+1] << 4) |
((uint16_t)frame[d+2] & 0x0f);
lc = (lc == 0x800 ? 0 : dv_audio_12to16(lc));
rc = (rc == 0x800 ? 0 : dv_audio_12to16(rc));
- of = sys->audio_shuffle[i%half_ch][j] + (d - 8)/3 * sys->audio_stride;
+ of = sys->audio_shuffle[i%half_ch][j] + (d - 8) / 3 * sys->audio_stride;
if (of*2 >= size)
continue;
- pcm[of*2] = lc & 0xff; // FIXME: may be we have to admit
- pcm[of*2+1] = lc >> 8; // that DV is a big endian PCM
+ pcm[of*2] = lc & 0xff; // FIXME: maybe we have to admit
+ pcm[of*2+1] = lc >> 8; // that DV is a big-endian PCM
of = sys->audio_shuffle[i%half_ch+half_ch][j] +
- (d - 8)/3 * sys->audio_stride;
- pcm[of*2] = rc & 0xff; // FIXME: may be we have to admit
- pcm[of*2+1] = rc >> 8; // that DV is a big endian PCM
+ (d - 8) / 3 * sys->audio_stride;
+ pcm[of*2] = rc & 0xff; // FIXME: maybe we have to admit
+ pcm[of*2+1] = rc >> 8; // that DV is a big-endian PCM
++d;
}
}
}
}
- /* next stereo channel (50Mbps only) */
- if(!pcm2)
+ /* next stereo channel (50Mbps and 100Mbps only) */
+ pcm = ppcm[ipcm++];
+ if (!pcm)
break;
- pcm = pcm2;
}
return size;
return 0;
}
- smpls = as_pack[1] & 0x3f; /* samples in this frame - min. samples */
- freq = (as_pack[4] >> 3) & 0x07; /* 0 - 48KHz, 1 - 44,1kHz, 2 - 32 kHz */
- stype = (as_pack[3] & 0x1f); /* 0 - 2CH, 2 - 4CH */
- quant = as_pack[4] & 0x07; /* 0 - 16bit linear, 1 - 12bit nonlinear */
+ smpls = as_pack[1] & 0x3f; /* samples in this frame - min. samples */
+ freq = (as_pack[4] >> 3) & 0x07; /* 0 - 48kHz, 1 - 44,1kHz, 2 - 32kHz */
+ stype = (as_pack[3] & 0x1f); /* 0 - 2CH, 2 - 4CH, 3 - 8CH */
+ quant = as_pack[4] & 0x07; /* 0 - 16bit linear, 1 - 12bit nonlinear */
/* note: ach counts PAIRS of channels (i.e. stereo channels) */
- ach = (stype == 2 || (quant && (freq == 2))) ? 2 : 1;
+ ach = ((int[4]){ 1, 0, 2, 4})[stype];
+ if (ach == 1 && quant && freq == 2)
+ ach = 2;
/* Dynamic handling of the audio streams in DV */
- for (i=0; i<ach; i++) {
+ for (i = 0; i < ach; i++) {
if (!c->ast[i]) {
c->ast[i] = av_new_stream(c->fctx, 0);
if (!c->ast[i])
break;
av_set_pts_info(c->ast[i], 64, 1, 30000);
c->ast[i]->codec->codec_type = CODEC_TYPE_AUDIO;
- c->ast[i]->codec->codec_id = CODEC_ID_PCM_S16LE;
+ c->ast[i]->codec->codec_id = CODEC_ID_PCM_S16LE;
av_init_packet(&c->audio_pkt[i]);
- c->audio_pkt[i].size = 0;
- c->audio_pkt[i].data = c->audio_buf[i];
+ c->audio_pkt[i].size = 0;
+ c->audio_pkt[i].data = c->audio_buf[i];
c->audio_pkt[i].stream_index = c->ast[i]->index;
- c->audio_pkt[i].flags |= PKT_FLAG_KEY;
+ c->audio_pkt[i].flags |= PKT_FLAG_KEY;
}
c->ast[i]->codec->sample_rate = dv_audio_frequency[freq];
- c->ast[i]->codec->channels = 2;
- c->ast[i]->codec->bit_rate = 2 * dv_audio_frequency[freq] * 16;
- c->ast[i]->start_time = 0;
+ c->ast[i]->codec->channels = 2;
+ c->ast[i]->codec->bit_rate = 2 * dv_audio_frequency[freq] * 16;
+ c->ast[i]->start_time = 0;
}
c->ach = i;
if (c->sys) {
avctx = c->vst->codec;
- av_set_pts_info(c->vst, 64, c->sys->frame_rate_base, c->sys->frame_rate);
- avctx->time_base= (AVRational){c->sys->frame_rate_base, c->sys->frame_rate};
- if(!avctx->width){
+ av_set_pts_info(c->vst, 64, c->sys->time_base.num,
+ c->sys->time_base.den);
+ avctx->time_base= c->sys->time_base;
+ if (!avctx->width){
avctx->width = c->sys->width;
avctx->height = c->sys->height;
}
/* finding out SAR is a little bit messy */
vsc_pack = dv_extract_pack(frame, dv_video_control);
- apt = frame[4] & 0x07;
- is16_9 = (vsc_pack && ((vsc_pack[2] & 0x07) == 0x02 ||
- (!apt && (vsc_pack[2] & 0x07) == 0x07)));
- avctx->sample_aspect_ratio = c->sys->sar[is16_9];
- avctx->bit_rate = av_rescale(c->sys->frame_size * 8,
- c->sys->frame_rate,
- c->sys->frame_rate_base);
+ apt = frame[4] & 0x07;
+ is16_9 = (vsc_pack && ((vsc_pack[2] & 0x07) == 0x02 ||
+ (!apt && (vsc_pack[2] & 0x07) == 0x07)));
+ c->vst->sample_aspect_ratio = c->sys->sar[is16_9];
+ avctx->bit_rate = av_rescale_q(c->sys->frame_size, (AVRational){8,1},
+ c->sys->time_base);
size = c->sys->frame_size;
}
return size;
return NULL;
}
- c->sys = NULL;
+ c->sys = NULL;
c->fctx = s;
- c->ast[0] = c->ast[1] = NULL;
- c->ach = 0;
+ memset(c->ast, 0, sizeof(c->ast));
+ c->ach = 0;
c->frames = 0;
c->abytes = 0;
c->vst->codec->codec_type = CODEC_TYPE_VIDEO;
- c->vst->codec->codec_id = CODEC_ID_DVVIDEO;
- c->vst->codec->bit_rate = 25000000;
- c->vst->start_time = 0;
+ c->vst->codec->codec_id = CODEC_ID_DVVIDEO;
+ c->vst->codec->bit_rate = 25000000;
+ c->vst->start_time = 0;
return c;
}
int size = -1;
int i;
- for (i=0; i<c->ach; i++) {
+ for (i = 0; i < c->ach; i++) {
if (c->ast[i] && c->audio_pkt[i].size) {
*pkt = c->audio_pkt[i];
c->audio_pkt[i].size = 0;
uint8_t* buf, int buf_size)
{
int size, i;
+ uint8_t *ppcm[4] = {0};
if (buf_size < DV_PROFILE_BYTES ||
!(c->sys = dv_frame_profile(buf)) ||
/* Queueing audio packet */
/* FIXME: in case of no audio/bad audio we have to do something */
size = dv_extract_audio_info(c, buf);
- for (i=0; i<c->ach; i++) {
+ for (i = 0; i < c->ach; i++) {
c->audio_pkt[i].size = size;
c->audio_pkt[i].pts = c->abytes * 30000*8 / c->ast[i]->codec->bit_rate;
+ ppcm[i] = c->audio_buf[i];
}
- dv_extract_audio(buf, c->audio_buf[0], c->audio_buf[1], c->sys);
+ dv_extract_audio(buf, ppcm, c->sys);
c->abytes += size;
+ /* We work with 720p frames split in half, thus even frames have
+ * channels 0,1 and odd 2,3. */
+ if (c->sys->height == 720) {
+ if (buf[1] & 0x0C)
+ c->audio_pkt[2].size = c->audio_pkt[3].size = 0;
+ else
+ c->audio_pkt[0].size = c->audio_pkt[1].size = 0;
+ }
+
/* Now it's time to return video packet */
size = dv_extract_video_info(c, buf);
av_init_packet(pkt);
- pkt->data = buf;
- pkt->size = size;
- pkt->flags |= PKT_FLAG_KEY;
+ pkt->data = buf;
+ pkt->size = size;
+ pkt->flags |= PKT_FLAG_KEY;
pkt->stream_index = c->vst->id;
- pkt->pts = c->frames;
+ pkt->pts = c->frames;
c->frames++;
// FIXME: sys may be wrong if last dv_read_packet() failed (buffer is junk)
const DVprofile* sys = dv_codec_profile(c->vst->codec);
int64_t offset;
- int64_t size = url_fsize(&s->pb);
+ int64_t size = url_fsize(s->pb);
int64_t max_offset = ((size-1) / sys->frame_size) * sys->frame_size;
offset = sys->frame_size * timestamp;
return offset;
}
-void dv_flush_audio_packets(DVDemuxContext *c)
+void dv_offset_reset(DVDemuxContext *c, int64_t frame_offset)
{
+ c->frames= frame_offset;
+ if (c->ach)
+ c->abytes= av_rescale_q(c->frames, c->sys->time_base,
+ (AVRational){8, c->ast[0]->codec->bit_rate});
c->audio_pkt[0].size = c->audio_pkt[1].size = 0;
+ c->audio_pkt[2].size = c->audio_pkt[3].size = 0;
}
/************************************************************
if (!c->dv_demux)
return -1;
- if (get_buffer(&s->pb, c->buf, DV_PROFILE_BYTES) <= 0 ||
- url_fseek(&s->pb, -DV_PROFILE_BYTES, SEEK_CUR) < 0)
- return AVERROR_IO;
+ if (get_buffer(s->pb, c->buf, DV_PROFILE_BYTES) <= 0 ||
+ url_fseek(s->pb, -DV_PROFILE_BYTES, SEEK_CUR) < 0)
+ return AVERROR(EIO);
c->dv_demux->sys = dv_frame_profile(c->buf);
- s->bit_rate = av_rescale(c->dv_demux->sys->frame_size * 8,
- c->dv_demux->sys->frame_rate,
- c->dv_demux->sys->frame_rate_base);
+ if (!c->dv_demux->sys) {
+ av_log(s, AV_LOG_ERROR, "Can't determine profile of DV input stream.\n");
+ return -1;
+ }
+
+ s->bit_rate = av_rescale_q(c->dv_demux->sys->frame_size, (AVRational){8,1},
+ c->dv_demux->sys->time_base);
return 0;
}
if (size < 0) {
size = c->dv_demux->sys->frame_size;
- if (get_buffer(&s->pb, c->buf, size) <= 0)
- return AVERROR_IO;
+ if (get_buffer(s->pb, c->buf, size) <= 0)
+ return AVERROR(EIO);
size = dv_produce_packet(c->dv_demux, pkt, c->buf, size);
}
static int dv_read_seek(AVFormatContext *s, int stream_index,
int64_t timestamp, int flags)
{
- RawDVContext *r = s->priv_data;
+ RawDVContext *r = s->priv_data;
DVDemuxContext *c = r->dv_demux;
- int64_t offset= dv_frame_offset(s, c, timestamp, flags);
+ int64_t offset = dv_frame_offset(s, c, timestamp, flags);
- c->frames= offset / c->sys->frame_size;
- if (c->ach)
- c->abytes= av_rescale(c->frames,
- c->ast[0]->codec->bit_rate * (int64_t)c->sys->frame_rate_base,
- 8*c->sys->frame_rate);
+ dv_offset_reset(c, offset / c->sys->frame_size);
- dv_flush_audio_packets(c);
- return url_fseek(&s->pb, offset, SEEK_SET);
+ offset = url_fseek(s->pb, offset, SEEK_SET);
+ return (offset < 0) ? offset : 0;
}
static int dv_read_close(AVFormatContext *s)
#ifdef CONFIG_DV_DEMUXER
AVInputFormat dv_demuxer = {
"dv",
- "DV video format",
+ NULL_IF_CONFIG_SMALL("DV video format"),
sizeof(RawDVContext),
NULL,
dv_read_header,