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-rw-r--r--libavformat/dvenc.c118
1 files changed, 75 insertions, 43 deletions
diff --git a/libavformat/dvenc.c b/libavformat/dvenc.c
index a33973f192..18fdf9fc86 100644
--- a/libavformat/dvenc.c
+++ b/libavformat/dvenc.c
@@ -11,20 +11,20 @@
* 50 Mbps (DVCPRO50) support
* Copyright (c) 2006 Daniel Maas <dmaas@maasdigital.com>
*
- * This file is part of Libav.
+ * This file is part of FFmpeg.
*
- * Libav is free software; you can redistribute it and/or
+ * 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.
*
- * Libav 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 Libav; 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>
@@ -37,10 +37,14 @@
#include "dv.h"
#include "libavutil/fifo.h"
#include "libavutil/mathematics.h"
+#include "libavutil/intreadwrite.h"
+#include "libavutil/opt.h"
+#include "libavutil/timecode.h"
#define MAX_AUDIO_FRAME_SIZE 192000 // 1 second of 48khz 32bit audio
struct DVMuxContext {
+ AVClass *av_class;
const AVDVProfile* sys; /* current DV profile, e.g.: 525/60, 625/50 */
int n_ast; /* number of stereo audio streams (up to 2) */
AVStream *ast[2]; /* stereo audio streams */
@@ -50,6 +54,7 @@ struct DVMuxContext {
int has_audio; /* frame under construction has audio */
int has_video; /* frame under construction has video */
uint8_t frame_buf[DV_MAX_FRAME_SIZE]; /* frame under construction */
+ AVTimecode tc; /* timecode context */
};
static const int dv_aaux_packs_dist[12][9] = {
@@ -67,8 +72,16 @@ static const int dv_aaux_packs_dist[12][9] = {
{ 0x50, 0x51, 0x52, 0x53, 0xff, 0xff, 0xff, 0xff, 0xff },
};
-static int dv_audio_frame_size(const AVDVProfile* sys, int frame)
+static int dv_audio_frame_size(const AVDVProfile* sys, int frame, int sample_rate)
{
+ if ((sys->time_base.den == 25 || sys->time_base.den == 50) && sys->time_base.num == 1) {
+ if (sample_rate == 32000) return 1280;
+ else if (sample_rate == 44100) return 1764;
+ else return 1920;
+ }
+
+ av_assert0(sample_rate == 48000);
+
return sys->audio_samples_dist[frame % (sizeof(sys->audio_samples_dist) /
sizeof(sys->audio_samples_dist[0]))];
}
@@ -77,41 +90,29 @@ static int dv_write_pack(enum dv_pack_type pack_id, DVMuxContext *c, uint8_t* bu
{
struct tm tc;
time_t ct;
- int ltc_frame;
+ uint32_t timecode;
va_list ap;
+ int audio_type = 0;
+ int channel;
buf[0] = (uint8_t)pack_id;
switch (pack_id) {
case dv_timecode:
- ct = (time_t)av_rescale_rnd(c->frames, c->sys->time_base.num,
- c->sys->time_base.den, AV_ROUND_DOWN);
- ff_brktimegm(ct, &tc);
- /*
- * LTC drop-frame frame counter drops two frames (0 and 1) every
- * minute, unless it is exactly divisible by 10
- */
- ltc_frame = (c->frames + 2 * ct / 60 - 2 * ct / 600) % c->sys->ltc_divisor;
- buf[1] = (0 << 7) | /* color frame: 0 - unsync; 1 - sync mode */
- (1 << 6) | /* drop frame timecode: 0 - nondrop; 1 - drop */
- ((ltc_frame / 10) << 4) | /* tens of frames */
- (ltc_frame % 10); /* units of frames */
- buf[2] = (1 << 7) | /* biphase mark polarity correction: 0 - even; 1 - odd */
- ((tc.tm_sec / 10) << 4) | /* tens of seconds */
- (tc.tm_sec % 10); /* units of seconds */
- buf[3] = (1 << 7) | /* binary group flag BGF0 */
- ((tc.tm_min / 10) << 4) | /* tens of minutes */
- (tc.tm_min % 10); /* units of minutes */
- buf[4] = (1 << 7) | /* binary group flag BGF2 */
- (1 << 6) | /* binary group flag BGF1 */
- ((tc.tm_hour / 10) << 4) | /* tens of hours */
- (tc.tm_hour % 10); /* units of hours */
+ timecode = av_timecode_get_smpte_from_framenum(&c->tc, c->frames);
+ timecode |= 1<<23 | 1<<15 | 1<<7 | 1<<6; // biphase and binary group flags
+ AV_WB32(buf + 1, timecode);
break;
case dv_audio_source: /* AAUX source pack */
va_start(ap, buf);
+ channel = va_arg(ap, int);
+ if (c->ast[channel]->codec->sample_rate == 44100) {
+ audio_type = 1;
+ } else if (c->ast[channel]->codec->sample_rate == 32000)
+ audio_type = 2;
buf[1] = (1 << 7) | /* locked mode -- SMPTE only supports locked mode */
(1 << 6) | /* reserved -- always 1 */
- (dv_audio_frame_size(c->sys, c->frames) -
- c->sys->audio_min_samples[0]);
+ (dv_audio_frame_size(c->sys, c->frames, c->ast[channel]->codec->sample_rate) -
+ c->sys->audio_min_samples[audio_type]);
/* # of samples */
buf[2] = (0 << 7) | /* multi-stereo */
(0 << 5) | /* #of audio channels per block: 0 -- 1 channel */
@@ -123,8 +124,9 @@ static int dv_write_pack(enum dv_pack_type pack_id, DVMuxContext *c, uint8_t* bu
(c->sys->n_difchan & 2); /* definition: 0 -- 25Mbps, 2 -- 50Mbps */
buf[4] = (1 << 7) | /* emphasis: 1 -- off */
(0 << 6) | /* emphasis time constant: 0 -- reserved */
- (0 << 3) | /* frequency: 0 -- 48kHz, 1 -- 44,1kHz, 2 -- 32kHz */
+ (audio_type << 3) | /* frequency: 0 -- 48kHz, 1 -- 44,1kHz, 2 -- 32kHz */
0; /* quantization: 0 -- 16bit linear, 1 -- 12bit nonlinear */
+
va_end(ap);
break;
case dv_audio_control:
@@ -184,12 +186,12 @@ static int dv_write_pack(enum dv_pack_type pack_id, DVMuxContext *c, uint8_t* bu
static void dv_inject_audio(DVMuxContext *c, int channel, uint8_t* frame_ptr)
{
int i, j, d, of, size;
- size = 4 * dv_audio_frame_size(c->sys, c->frames);
+ size = 4 * dv_audio_frame_size(c->sys, c->frames, c->ast[channel]->codec->sample_rate);
frame_ptr += channel * c->sys->difseg_size * 150 * 80;
for (i = 0; i < c->sys->difseg_size; i++) {
frame_ptr += 6 * 80; /* skip DIF segment header */
for (j = 0; j < 9; j++) {
- dv_write_pack(dv_aaux_packs_dist[i][j], c, &frame_ptr[3], i >= c->sys->difseg_size/2);
+ dv_write_pack(dv_aaux_packs_dist[i][j], c, &frame_ptr[3], channel, i >= c->sys->difseg_size/2);
for (d = 8; d < 80; d+=2) {
of = c->sys->audio_shuffle[i][j] + (d - 8)/2 * c->sys->audio_stride;
if (of*2 >= size)
@@ -242,7 +244,6 @@ static int dv_assemble_frame(DVMuxContext *c, AVStream* st,
int i, reqasize;
*frame = &c->frame_buf[0];
- reqasize = 4 * dv_audio_frame_size(c->sys, c->frames);
switch (st->codec->codec_type) {
case AVMEDIA_TYPE_VIDEO:
@@ -261,6 +262,8 @@ static int dv_assemble_frame(DVMuxContext *c, AVStream* st,
av_log(st->codec, AV_LOG_ERROR, "Can't process DV frame #%d. Insufficient video data or severe sync problem.\n", c->frames);
av_fifo_generic_write(c->audio_data[i], data, data_size, NULL);
+ reqasize = 4 * dv_audio_frame_size(c->sys, c->frames, st->codec->sample_rate);
+
/* Let us see if we've got enough audio for one DV frame. */
c->has_audio |= ((reqasize <= av_fifo_size(c->audio_data[i])) << i);
@@ -275,6 +278,7 @@ static int dv_assemble_frame(DVMuxContext *c, AVStream* st,
c->has_audio = 0;
for (i=0; i < c->n_ast; i++) {
dv_inject_audio(c, i, *frame);
+ reqasize = 4 * dv_audio_frame_size(c->sys, c->frames, c->ast[i]->codec->sample_rate);
av_fifo_drain(c->audio_data[i], reqasize);
c->has_audio |= ((reqasize <= av_fifo_size(c->audio_data[i])) << i);
}
@@ -323,15 +327,27 @@ static DVMuxContext* dv_init_mux(AVFormatContext* s)
if (!vst || vst->codec->codec_id != AV_CODEC_ID_DVVIDEO)
goto bail_out;
for (i=0; i<c->n_ast; i++) {
- if (c->ast[i] && (c->ast[i]->codec->codec_id != AV_CODEC_ID_PCM_S16LE ||
- c->ast[i]->codec->sample_rate != 48000 ||
- c->ast[i]->codec->channels != 2))
- goto bail_out;
+ if (c->ast[i]) {
+ if(c->ast[i]->codec->codec_id != AV_CODEC_ID_PCM_S16LE ||
+ c->ast[i]->codec->channels != 2)
+ goto bail_out;
+ if (c->ast[i]->codec->sample_rate != 48000 &&
+ c->ast[i]->codec->sample_rate != 44100 &&
+ c->ast[i]->codec->sample_rate != 32000 )
+ goto bail_out;
+ }
}
c->sys = av_dv_codec_profile(vst->codec->width, vst->codec->height, vst->codec->pix_fmt);
if (!c->sys)
goto bail_out;
+ if ((c->sys->time_base.den != 25 && c->sys->time_base.den != 50) || c->sys->time_base.num != 1) {
+ if (c->ast[0] && c->ast[0]->codec->sample_rate != 48000)
+ goto bail_out;
+ if (c->ast[1] && c->ast[1]->codec->sample_rate != 48000)
+ goto bail_out;
+ }
+
if ((c->n_ast > 1) && (c->sys->n_difchan < 2)) {
/* only 1 stereo pair is allowed in 25Mbps mode */
goto bail_out;
@@ -345,10 +361,10 @@ static DVMuxContext* dv_init_mux(AVFormatContext* s)
c->start_time = ff_iso8601_to_unix_time(t->value);
for (i=0; i < c->n_ast; i++) {
- if (c->ast[i] && !(c->audio_data[i]=av_fifo_alloc(100*MAX_AUDIO_FRAME_SIZE))) {
+ if (c->ast[i] && !(c->audio_data[i]=av_fifo_alloc_array(100, MAX_AUDIO_FRAME_SIZE))) {
while (i > 0) {
i--;
- av_fifo_free(c->audio_data[i]);
+ av_fifo_freep(&c->audio_data[i]);
}
goto bail_out;
}
@@ -364,19 +380,35 @@ static void dv_delete_mux(DVMuxContext *c)
{
int i;
for (i=0; i < c->n_ast; i++)
- av_fifo_free(c->audio_data[i]);
+ av_fifo_freep(&c->audio_data[i]);
}
static int dv_write_header(AVFormatContext *s)
{
+ AVRational rate;
+ DVMuxContext *dvc = s->priv_data;
+ AVDictionaryEntry *tcr = av_dict_get(s->metadata, "timecode", NULL, 0);
+
if (!dv_init_mux(s)) {
av_log(s, AV_LOG_ERROR, "Can't initialize DV format!\n"
"Make sure that you supply exactly two streams:\n"
- " video: 25fps or 29.97fps, audio: 2ch/48kHz/PCM\n"
+ " video: 25fps or 29.97fps, audio: 2ch/48|44|32kHz/PCM\n"
" (50Mbps allows an optional second audio stream)\n");
return -1;
}
- return 0;
+ rate.num = dvc->sys->ltc_divisor;
+ rate.den = 1;
+ if (!tcr) { // no global timecode, look into the streams
+ int i;
+ for (i = 0; i < s->nb_streams; i++) {
+ tcr = av_dict_get(s->streams[i]->metadata, "timecode", NULL, 0);
+ if (tcr)
+ break;
+ }
+ }
+ if (tcr && av_timecode_init_from_string(&dvc->tc, rate, tcr->value, s) >= 0)
+ return 0;
+ return av_timecode_init(&dvc->tc, rate, 0, 0, s);
}
static int dv_write_packet(struct AVFormatContext *s, AVPacket *pkt)
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-06 04:43:38 +0000