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authorRoman Shaposhnik <roman@shaposhnik.org>2003-08-28 19:53:47 +0000
committerRoman Shaposhnik <roman@shaposhnik.org>2003-08-28 19:53:47 +0000
commit118e91f3d841435c50df4e110eb108d720cdcecf (patch)
tree7f9597bb3ac4f42fb15c4d15e67e718a2652af07 /libavformat/dvcore.c
parent8c5b5683aecb84100ca32a6f9ea15ffe8df1ac66 (diff)
downloadffmpeg-118e91f3d841435c50df4e110eb108d720cdcecf.tar.gz
* Phase 1 for DV encoding support. Muxing/demuxing of DV streams is now
possible. For example you can do: ffmpeg -i i.dv -i audio_track.mp3 -map 0.0 -map 1.0 \ -vcodec copy -acodec pcm_s16le out.dv * Preparations for getting rid of DVAUDIO codec, DV stream really contains PCM audio, so there's no codec needed if we have a decent demuxer. * Providing entry points for dv1394 write support. Originally committed as revision 2174 to svn://svn.ffmpeg.org/ffmpeg/trunk
Diffstat (limited to 'libavformat/dvcore.c')
-rw-r--r--libavformat/dvcore.c668
1 files changed, 668 insertions, 0 deletions
diff --git a/libavformat/dvcore.c b/libavformat/dvcore.c
new file mode 100644
index 0000000000..c6d5ed6a95
--- /dev/null
+++ b/libavformat/dvcore.c
@@ -0,0 +1,668 @@
+/*
+ * DV format muxer/demuxer
+ * Copyright (c) 2003 Roman Shaposhnik
+ *
+ * Many thanks to Dan Dennedy <dan@dennedy.org> for providing wealth
+ * of DV technical info, and SMPTE 314M specification.
+ *
+ * This library 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.
+ *
+ * This library 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
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+#include "avformat.h"
+#include "dvcore.h"
+
+/*
+ * The reason why the following three big ugly looking tables are
+ * here is my lack of DV spec IEC 61834. The tables were basically
+ * constructed to make code that places packs in SSYB, VAUX and
+ * AAUX blocks very simple and table-driven. They conform to the
+ * SMPTE 314M and the output of my personal DV camcorder, neither
+ * of which is sufficient for a reliable DV stream producing. Thus
+ * while code is still in development I'll be gathering input from
+ * people with different DV equipment and modifying the tables to
+ * accommodate all the quirks. Later on, if possible, some of them
+ * will be folded into smaller tables and/or switch-if logic. For
+ * now, my only excuse is -- they don't eat up that much of a space.
+ */
+
+static const int dv_ssyb_packs_dist[12][6] = {
+ { 0x13, 0x13, 0x13, 0x13, 0x13, 0x13 },
+ { 0x13, 0x13, 0x13, 0x13, 0x13, 0x13 },
+ { 0x13, 0x13, 0x13, 0x13, 0x13, 0x13 },
+ { 0x13, 0x13, 0x13, 0x13, 0x13, 0x13 },
+ { 0x13, 0x13, 0x13, 0x13, 0x13, 0x13 },
+ { 0x13, 0x13, 0x13, 0x13, 0x13, 0x13 },
+ { 0x13, 0x62, 0x63, 0x13, 0x62, 0x63 },
+ { 0x13, 0x62, 0x63, 0x13, 0x62, 0x63 },
+ { 0x13, 0x62, 0x63, 0x13, 0x62, 0x63 },
+ { 0x13, 0x62, 0x63, 0x13, 0x62, 0x63 },
+ { 0x13, 0x62, 0x63, 0x13, 0x62, 0x63 },
+ { 0x13, 0x62, 0x63, 0x13, 0x62, 0x63 },
+};
+
+static const int dv_vaux_packs_dist[12][15] = {
+ { 0x60, 0x61, 0x62, 0x63, 0xff, 0xff, 0xff, 0xff, 0xff,
+ 0x60, 0x61, 0x62, 0x63, 0xff, 0xff },
+ { 0x60, 0x61, 0x62, 0x63, 0xff, 0xff, 0xff, 0xff, 0xff,
+ 0x60, 0x61, 0x62, 0x63, 0xff, 0xff },
+ { 0x60, 0x61, 0x62, 0x63, 0xff, 0xff, 0xff, 0xff, 0xff,
+ 0x60, 0x61, 0x62, 0x63, 0xff, 0xff },
+ { 0x60, 0x61, 0x62, 0x63, 0xff, 0xff, 0xff, 0xff, 0xff,
+ 0x60, 0x61, 0x62, 0x63, 0xff, 0xff },
+ { 0x60, 0x61, 0x62, 0x63, 0xff, 0xff, 0xff, 0xff, 0xff,
+ 0x60, 0x61, 0x62, 0x63, 0xff, 0xff },
+ { 0x60, 0x61, 0x62, 0x63, 0xff, 0xff, 0xff, 0xff, 0xff,
+ 0x60, 0x61, 0x62, 0x63, 0xff, 0xff },
+ { 0x60, 0x61, 0x62, 0x63, 0xff, 0xff, 0xff, 0xff, 0xff,
+ 0x60, 0x61, 0x62, 0x63, 0xff, 0xff },
+ { 0x60, 0x61, 0x62, 0x63, 0xff, 0xff, 0xff, 0xff, 0xff,
+ 0x60, 0x61, 0x62, 0x63, 0xff, 0xff },
+ { 0x60, 0x61, 0x62, 0x63, 0xff, 0xff, 0xff, 0xff, 0xff,
+ 0x60, 0x61, 0x62, 0x63, 0xff, 0xff },
+ { 0x60, 0x61, 0x62, 0x63, 0xff, 0xff, 0xff, 0xff, 0xff,
+ 0x60, 0x61, 0x62, 0x63, 0xff, 0xff },
+ { 0x60, 0x61, 0x62, 0x63, 0xff, 0xff, 0xff, 0xff, 0xff,
+ 0x60, 0x61, 0x62, 0x63, 0xff, 0xff },
+ { 0x60, 0x61, 0x62, 0x63, 0xff, 0xff, 0xff, 0xff, 0xff,
+ 0x60, 0x61, 0x62, 0x63, 0xff, 0xff },
+};
+
+static const int dv_aaux_packs_dist[12][9] = {
+ { 0xff, 0xff, 0xff, 0x50, 0x51, 0x52, 0x53, 0xff, 0xff },
+ { 0x50, 0x51, 0x52, 0x53, 0xff, 0xff, 0xff, 0xff, 0xff },
+ { 0xff, 0xff, 0xff, 0x50, 0x51, 0x52, 0x53, 0xff, 0xff },
+ { 0x50, 0x51, 0x52, 0x53, 0xff, 0xff, 0xff, 0xff, 0xff },
+ { 0xff, 0xff, 0xff, 0x50, 0x51, 0x52, 0x53, 0xff, 0xff },
+ { 0x50, 0x51, 0x52, 0x53, 0xff, 0xff, 0xff, 0xff, 0xff },
+ { 0xff, 0xff, 0xff, 0x50, 0x51, 0x52, 0x53, 0xff, 0xff },
+ { 0x50, 0x51, 0x52, 0x53, 0xff, 0xff, 0xff, 0xff, 0xff },
+ { 0xff, 0xff, 0xff, 0x50, 0x51, 0x52, 0x53, 0xff, 0xff },
+ { 0x50, 0x51, 0x52, 0x53, 0xff, 0xff, 0xff, 0xff, 0xff },
+ { 0xff, 0xff, 0xff, 0x50, 0x51, 0x52, 0x53, 0xff, 0xff },
+ { 0x50, 0x51, 0x52, 0x53, 0xff, 0xff, 0xff, 0xff, 0xff },
+};
+
+enum dv_section_type {
+ dv_sect_header = 0x1f,
+ dv_sect_subcode = 0x3f,
+ dv_sect_vaux = 0x56,
+ dv_sect_audio = 0x76,
+ dv_sect_video = 0x96,
+};
+
+enum dv_pack_type {
+ dv_header525 = 0x3f, /* see dv_write_pack for important details on */
+ dv_header625 = 0xbf, /* these two packs */
+ dv_timecode = 0x13,
+ dv_audio_source = 0x50,
+ dv_audio_control = 0x51,
+ dv_audio_recdate = 0x52,
+ dv_audio_rectime = 0x53,
+ dv_video_source = 0x60,
+ dv_video_control = 0x61,
+ dv_viedo_recdate = 0x62,
+ dv_video_rectime = 0x63,
+ dv_unknown_pack = 0xff,
+};
+
+static const uint16_t dv_audio_shuffle525[10][9] = {
+ { 0, 30, 60, 20, 50, 80, 10, 40, 70 }, /* 1st channel */
+ { 6, 36, 66, 26, 56, 86, 16, 46, 76 },
+ { 12, 42, 72, 2, 32, 62, 22, 52, 82 },
+ { 18, 48, 78, 8, 38, 68, 28, 58, 88 },
+ { 24, 54, 84, 14, 44, 74, 4, 34, 64 },
+
+ { 1, 31, 61, 21, 51, 81, 11, 41, 71 }, /* 2nd channel */
+ { 7, 37, 67, 27, 57, 87, 17, 47, 77 },
+ { 13, 43, 73, 3, 33, 63, 23, 53, 83 },
+ { 19, 49, 79, 9, 39, 69, 29, 59, 89 },
+ { 25, 55, 85, 15, 45, 75, 5, 35, 65 },
+};
+
+static const uint16_t dv_audio_shuffle625[12][9] = {
+ { 0, 36, 72, 26, 62, 98, 16, 52, 88}, /* 1st channel */
+ { 6, 42, 78, 32, 68, 104, 22, 58, 94},
+ { 12, 48, 84, 2, 38, 74, 28, 64, 100},
+ { 18, 54, 90, 8, 44, 80, 34, 70, 106},
+ { 24, 60, 96, 14, 50, 86, 4, 40, 76},
+ { 30, 66, 102, 20, 56, 92, 10, 46, 82},
+
+ { 1, 37, 73, 27, 63, 99, 17, 53, 89}, /* 2nd channel */
+ { 7, 43, 79, 33, 69, 105, 23, 59, 95},
+ { 13, 49, 85, 3, 39, 75, 29, 65, 101},
+ { 19, 55, 91, 9, 45, 81, 35, 71, 107},
+ { 25, 61, 97, 15, 51, 87, 5, 41, 77},
+ { 31, 67, 103, 21, 57, 93, 11, 47, 83},
+};
+
+static const int dv_audio_frequency[3] = {
+ 48000, 44100, 32000,
+};
+
+const DVprofile dv_profiles[2] = {
+ { .dsf = 0,
+ .frame_size = 120000, /* 525/60 system (NTSC) */
+ .difseg_size = 10,
+ .frame_rate = 30000,
+ .ltc_divisor = 30,
+ .frame_rate_base = 1001,
+ .height = 480,
+/* .video_place = dv_place_411, */
+ .audio_stride = 90,
+ .audio_min_samples = { 1580, 1452, 1053 }, /* for 48, 44.1 and 32Khz */
+ .audio_samples_dist = { 1602, 1601, 1602, 1601, 1602 },
+ .audio_shuffle = dv_audio_shuffle525,
+ },
+ { .dsf = 1,
+ .frame_size = 144000, /* 625/50 system (PAL) */
+ .difseg_size = 12,
+ .frame_rate = 25,
+ .frame_rate_base = 1,
+ .ltc_divisor = 25,
+ .height = 576,
+/* .video_place = dv_place_420, */
+ .audio_stride = 108,
+ .audio_min_samples = { 1896, 1742, 1264 }, /* for 48, 44.1 and 32Khz */
+ .audio_samples_dist = { 1920, 1920, 1920, 1920, 1920 },
+ .audio_shuffle = dv_audio_shuffle525,
+ }
+};
+
+static inline uint16_t dv_audio_12to16(uint16_t sample)
+{
+ uint16_t shift, result;
+
+ sample = (sample < 0x800) ? sample : sample | 0xf000;
+ shift = (sample & 0xf00) >> 8;
+
+ if (shift < 0x2 || shift > 0xd) {
+ result = sample;
+ } else if (shift < 0x8) {
+ shift--;
+ result = (sample - (256 * shift)) << shift;
+ } else {
+ shift = 0xe - shift;
+ result = ((sample + ((256 * shift) + 1)) << shift) - 1;
+ }
+
+ return result;
+}
+
+static int dv_write_pack(enum dv_pack_type pack_id, DVMuxContext *c, uint8_t* buf)
+{
+ struct tm tc;
+ time_t ct;
+ int ltc_frame;
+
+ buf[0] = (uint8_t)pack_id;
+ switch (pack_id) {
+ case dv_header525: /* I can't imagine why these two weren't defined as real */
+ case dv_header625: /* packs in SMPTE314M -- they definitely look like ones */
+ buf[1] = 0xf8 | /* reserved -- always 1 */
+ (0 & 0x07); /* APT: Track application ID */
+ buf[2] = (0 << 7) | /* TF1: audio data is 0 - valid; 1 - invalid */
+ (0x0f << 3) | /* reserved -- always 1 */
+ (0 & 0x07); /* AP1: Audio application ID */
+ buf[3] = (0 << 7) | /* TF2: video data is 0 - valid; 1 - invalid */
+ (0x0f << 3) | /* reserved -- always 1 */
+ (0 & 0x07); /* AP2: Video application ID */
+ buf[4] = (0 << 7) | /* TF3: subcode(SSYB) is 0 - valid; 1 - invalid */
+ (0x0f << 3) | /* reserved -- always 1 */
+ (0 & 0x07); /* AP3: Subcode application ID */
+ break;
+ case dv_timecode:
+ ct = (time_t)(c->frames / ((float)c->sys->frame_rate /
+ (float)c->sys->frame_rate_base));
+ localtime_r(&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 fame: 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 */
+ break;
+ case dv_audio_source: /* AAUX source pack */
+ buf[1] = (0 << 7) | /* locked mode */
+ (1 << 6) | /* reserved -- always 1 */
+ (dv_audio_frame_size(c->sys, c->frames) -
+ c->sys->audio_min_samples[0]);
+ /* # of samples */
+ buf[2] = (0 << 7) | /* multi-stereo */
+ (0 << 5) | /* #of audio channels per block: 0 -- 1 channel */
+ (0 << 4) | /* pair bit: 0 -- one pair of channels */
+ 0; /* audio mode */
+ buf[3] = (1 << 7) | /* res */
+ (1 << 6) | /* multi-language flag */
+ (c->sys->dsf << 5) | /* system: 60fields/50fields */
+ 0; /* definition: 0 -- SD (525/625) */
+ buf[4] = (1 << 7) | /* emphasis: 1 -- off */
+ (0 << 6) | /* emphasis time constant: 0 -- reserved */
+ (0 << 3) | /* frequency: 0 -- 48Khz, 1 -- 44,1Khz, 2 -- 32Khz */
+ 0; /* quantization: 0 -- 16bit linear, 1 -- 12bit nonlinear */
+ break;
+ case dv_audio_control:
+ buf[1] = (0 << 6) | /* copy protection: 0 -- unrestricted */
+ (1 << 4) | /* input source: 1 -- digital input */
+ (3 << 2) | /* compression: 3 -- no information */
+ 0; /* misc. info/SMPTE emphasis off */
+ buf[2] = (1 << 7) | /* recording start point: 1 -- no */
+ (1 << 6) | /* recording end point: 1 -- no */
+ (1 << 3) | /* recording mode: 1 -- original */
+ 7;
+ buf[3] = (1 << 7) | /* direction: 1 -- forward */
+ 0x20; /* speed */
+ buf[4] = (1 << 7) | /* reserved -- always 1 */
+ 0x7f; /* genre category */
+ break;
+ case dv_audio_recdate:
+ case dv_viedo_recdate: /* VAUX recording date */
+ ct = c->start_time + (time_t)(c->frames /
+ ((float)c->sys->frame_rate / (float)c->sys->frame_rate_base));
+ localtime_r(&ct, &tc);
+ buf[1] = 0xff; /* ds, tm, tens of time zone, units of time zone */
+ /* 0xff is very likely to be "unknown" */
+ buf[2] = (3 << 6) | /* reserved -- always 1 */
+ ((tc.tm_mday / 10) << 4) | /* Tens of day */
+ (tc.tm_mday % 10); /* Units of day */
+ buf[3] = /* we set high 4 bits to 0, shouldn't we set them to week? */
+ (tc.tm_mon % 10); /* Units of month */
+ buf[4] = (((tc.tm_year % 100) / 10) << 4) | /* Tens of year */
+ (tc.tm_year % 10); /* Units of year */
+ break;
+ case dv_audio_rectime: /* AAUX recording time */
+ case dv_video_rectime: /* VAUX recording time */
+ ct = c->start_time + (time_t)(c->frames /
+ ((float)c->sys->frame_rate / (float)c->sys->frame_rate_base));
+ localtime_r(&ct, &tc);
+ buf[1] = (3 << 6) | /* reserved -- always 1 */
+ 0x3f; /* tens of frame, units of frame: 0x3f - "unknown" ? */
+ buf[2] = (1 << 7) | /* reserved -- always 1 */
+ ((tc.tm_sec / 10) << 4) | /* Tens of seconds */
+ (tc.tm_sec % 10); /* Units of seconds */
+ buf[3] = (1 << 7) | /* reserved -- always 1 */
+ ((tc.tm_min / 10) << 4) | /* Tens of minutes */
+ (tc.tm_min % 10); /* Units of minutes */
+ buf[4] = (3 << 6) | /* reserved -- always 1 */
+ ((tc.tm_hour / 10) << 4) | /* Tens of hours */
+ (tc.tm_hour % 10); /* Units of hours */
+ break;
+ case dv_video_source:
+ buf[1] = 0xff; /* reserved -- always 1 */
+ buf[2] = (1 << 7) | /* B/W: 0 - b/w, 1 - color */
+ (1 << 6) | /* following CLF is valid - 0, invalid - 1 */
+ (3 << 4) | /* CLF: color frames id (see ITU-R BT.470-4) */
+ 0xf; /* reserved -- always 1 */
+ buf[3] = (3 << 6) | /* reserved -- always 1 */
+ (c->sys->dsf << 5) | /* system: 60fields/50fields */
+ 0; /* signal type video compression */
+ buf[4] = 0xff; /* VISC: 0xff -- no information */
+ break;
+ case dv_video_control:
+ buf[1] = (0 << 6) | /* Copy generation management (CGMS) 0 -- free */
+ 0x3f; /* reserved -- always 1 */
+ buf[2] = 0xc8 | /* reserved -- always b11001xxx */
+ c->aspect;
+ buf[3] = (1 << 7) | /* Frame/field flag 1 -- frame, 0 -- field */
+ (1 << 6) | /* First/second field flag 0 -- field 2, 1 -- field 1 */
+ (1 << 5) | /* Frame change flag 0 -- same picture as before, 1 -- different */
+ (1 << 4) | /* 1 - interlaced, 0 - noninterlaced */
+ 0xc; /* reserved -- always b1100 */
+ buf[4] = 0xff; /* reserved -- always 1 */
+ break;
+ default:
+ buf[1] = buf[2] = buf[3] = buf[4] = 0xff;
+ }
+ return 5;
+}
+
+static inline int dv_write_dif_id(enum dv_section_type t, uint8_t seq_num,
+ uint8_t dif_num, uint8_t* buf)
+{
+ buf[0] = (uint8_t)t; /* Section type */
+ buf[1] = (seq_num<<4) | /* DIF seq number 0-9 for 525/60; 0-11 for 625/50 */
+ (0 << 3) | /* FSC: for 50Mb/s 0 - first channel; 1 - second */
+ 7; /* reserved -- always 1 */
+ buf[2] = dif_num; /* DIF block number Video: 0-134, Audio: 0-8 */
+ return 3;
+}
+
+static inline int dv_write_ssyb_id(uint8_t syb_num, uint8_t fr, uint8_t* buf)
+{
+ if (syb_num == 0 || syb_num == 6) {
+ buf[0] = (fr<<7) | /* FR ID 1 - first half of each channel; 0 - second */
+ (0<<4) | /* AP3 (Subcode application ID) */
+ 0x0f; /* reserved -- always 1 */
+ }
+ else if (syb_num == 11) {
+ buf[0] = (fr<<7) | /* FR ID 1 - first half of each channel; 0 - second */
+ 0x7f; /* reserved -- always 1 */
+ }
+ else {
+ buf[0] = (fr<<7) | /* FR ID 1 - first half of each channel; 0 - second */
+ (0<<4) | /* APT (Track application ID) */
+ 0x0f; /* reserved -- always 1 */
+ }
+ buf[1] = 0xf0 | /* reserved -- always 1 */
+ (syb_num & 0x0f); /* SSYB number 0 - 11 */
+ buf[2] = 0xff; /* reserved -- always 1 */
+ return 3;
+}
+
+void dv_format_frame(DVMuxContext *c, uint8_t* buf)
+{
+ int i, j, k;
+
+ for (i = 0; i < c->sys->difseg_size; i++) {
+ memset(buf, 0xff, 80 * 6); /* First 6 DIF blocks are for control data */
+
+ /* DV header: 1DIF */
+ buf += dv_write_dif_id(dv_sect_header, i, 0, buf);
+ buf += dv_write_pack((c->sys->dsf ? dv_header625 : dv_header525), c, buf);
+ buf += 72; /* unused bytes */
+
+ /* DV subcode: 2DIFs */
+ for (j = 0; j < 2; j++) {
+ buf += dv_write_dif_id( dv_sect_subcode, i, j, buf);
+ for (k = 0; k < 6; k++) {
+ buf += dv_write_ssyb_id(k, (i < c->sys->difseg_size/2), buf);
+ buf += dv_write_pack(dv_ssyb_packs_dist[i][k], c, buf);
+ }
+ buf += 29; /* unused bytes */
+ }
+
+ /* DV VAUX: 3DIFs */
+ for (j = 0; j < 3; j++) {
+ buf += dv_write_dif_id(dv_sect_vaux, i, j, buf);
+ for (k = 0; k < 15 ; k++)
+ buf += dv_write_pack(dv_vaux_packs_dist[i][k], c, buf);
+ buf += 2; /* unused bytes */
+ }
+
+ /* DV Audio/Video: 135 Video DIFs + 9 Audio DIFs */
+ for (j = 0; j < 135; j++) {
+ if (j%15 == 0) {
+ buf += dv_write_dif_id(dv_sect_audio, i, j/15, buf);
+ buf += dv_write_pack(dv_aaux_packs_dist[i][j/15], c, buf);
+ buf += 72; /* shuffled PCM audio */
+ }
+ buf += dv_write_dif_id(dv_sect_video, i, j, buf);
+ buf += 77; /* 1 video macro block: 1 bytes control
+ 4 * 14 bytes Y 8x8 data
+ 10 bytes Cr 8x8 data
+ 10 bytes Cb 8x8 data */
+ }
+ }
+}
+
+void dv_inject_audio(DVMuxContext *c, uint8_t* pcm, uint8_t* frame_ptr)
+{
+ int i, j, d, of;
+ for (i = 0; i < c->sys->difseg_size; i++) {
+ frame_ptr += 6 * 80; /* skip DIF segment header */
+ for (j = 0; j < 9; j++) {
+ for (d = 8; d < 80; d+=2) {
+ of = c->sys->audio_shuffle[i][j] + (d - 8)/2 * c->sys->audio_stride;
+ frame_ptr[d] = pcm[of*2+1]; // FIXME: may be we have to admit
+ frame_ptr[d+1] = pcm[of*2]; // that DV is a big endian PCM
+ }
+ frame_ptr += 16 * 80; /* 15 Video DIFs + 1 Audio DIF */
+ }
+ }
+}
+
+void dv_inject_video(DVMuxContext *c, uint8_t* video_data, uint8_t* frame_ptr)
+{
+ int i, j;
+ int ptr = 0;
+
+ for (i = 0; i < c->sys->difseg_size; i++) {
+ ptr += 6 * 80; /* skip DIF segment header */
+ for (j = 0; j < 135; j++) {
+ if (j%15 == 0)
+ ptr += 80; /* skip Audio DIF */
+ ptr += 3;
+ memcpy(frame_ptr + ptr, video_data + ptr, 77);
+ ptr += 77;
+ }
+ }
+}
+
+int dv_audio_frame_size(const DVprofile* sys, int frame)
+{
+ return sys->audio_samples_dist[frame % (sizeof(sys->audio_samples_dist)/
+ sizeof(sys->audio_samples_dist[0]))];
+}
+
+const DVprofile* dv_frame_profile(uint8_t* frame)
+{
+ return &dv_profiles[!!(frame[3] & 0x80)]; /* Header, DSF flag */
+}
+
+/*
+ * This is the dumbest implementation of all -- it simply looks at
+ * a fixed offset and if pack isn't there -- fails. We might want
+ * to have a fallback mechanism for complete search of missing packs.
+ */
+const uint8_t* dv_extract_pack(uint8_t* frame, enum dv_pack_type t)
+{
+ int offs;
+
+ switch (t) {
+ case dv_audio_source:
+ offs = (80*6 + 80*16*3 + 3);
+ break;
+ case dv_audio_control:
+ offs = (80*6 + 80*16*4 + 3);
+ break;
+ case dv_video_control:
+ offs = (80*5 + 48 + 5);
+ break;
+ default:
+ return NULL;
+ }
+
+ return (frame[offs] == t ? &frame[offs] : NULL);
+}
+
+/*
+ * There's a couple of assumptions being made here:
+ * 1. By default we silence erroneous (0x8000/16bit 0x800/12bit) audio samples.
+ * We can pass them upwards when ffmpeg will be ready to deal with them.
+ * 2. We don't do software emphasis.
+ * 3. Audio is always returned as 16bit linear samples: 12bit nonlinear samples
+ * are converted into 16bit linear ones.
+ */
+int dv_extract_audio(uint8_t* frame, uint8_t* pcm, AVCodecContext* avctx)
+{
+ int size, i, j, d, of, smpls, freq, quant;
+ uint16_t lc, rc;
+ const DVprofile* sys;
+ const uint8_t* as_pack;
+
+ as_pack = dv_extract_pack(frame, dv_audio_source);
+ if (!as_pack) /* No audio ? */
+ return 0;
+
+ sys = dv_frame_profile(frame);
+ 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 */
+
+ if (quant > 1)
+ return -1; /* Unsupported quantization */
+
+ avctx->sample_rate = dv_audio_frequency[freq];
+ avctx->channels = 2;
+ avctx->bit_rate = avctx->channels * avctx->sample_rate * 16;
+ // What about:
+ // avctx->frame_size =
+
+ size = (sys->audio_min_samples[freq] + smpls) * 4; /* 2ch, 2bytes */
+
+ /* for each DIF segment */
+ for (i = 0; i < sys->difseg_size; i++) {
+ frame += 6 * 80; /* skip DIF segment header */
+ 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;
+ 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
+ if (pcm[of*2+1] == 0x80 && pcm[of*2] == 0x00)
+ pcm[of*2+1] = 0;
+ } else { /* 12bit quantization */
+ if (i >= sys->difseg_size/2)
+ goto out; /* We're not doing 4ch at this time */
+
+ 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][j] + (d - 8)/3 * sys->audio_stride;
+ 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
+ of = sys->audio_shuffle[i+sys->difseg_size/2][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;
+ }
+ }
+
+ frame += 16 * 80; /* 15 Video DIFs + 1 Audio DIF */
+ }
+ }
+
+out:
+ return size;
+}
+
+/* FIXME: The following three functions could be underengineered ;-) */
+void dv_assemble_frame(DVMuxContext *c, uint8_t* video, uint8_t* audio, int asize)
+{
+ uint8_t pcm[8192];
+ uint8_t* frame = &c->frame_buf[0];
+ int fsize, reqasize;
+
+ if (c->has_audio && c->has_video) { /* must be a stale frame */
+ dv_format_frame(c, frame);
+ c->frames++;
+ c->has_audio = c->has_video = 0;
+ }
+
+ if (video) {
+ /* FIXME: we have to have more sensible approach than this one */
+ if (c->has_video)
+ fprintf(stderr, "Can't process DV frame #%d. Insufficient audio data or severe sync problem.\n", c->frames);
+
+ dv_inject_video(c, video, frame);
+ c->has_video = 1;
+ }
+ if (audio) {
+ reqasize = 4 * dv_audio_frame_size(c->sys, c->frames);
+ fsize = fifo_size(&c->audio_data, c->audio_data.rptr);
+ if (fsize + asize >= reqasize) {
+ if (fsize >= reqasize) {
+ fifo_read(&c->audio_data, &pcm[0], reqasize, &c->audio_data.rptr);
+ } else {
+ fifo_read(&c->audio_data, &pcm[0], fsize, &c->audio_data.rptr);
+ memcpy(&pcm[fsize], &audio[0], reqasize - fsize);
+ audio += reqasize - fsize;
+ asize -= reqasize - fsize;
+ }
+ dv_inject_audio(c, &pcm[0], frame);
+ c->has_audio = 1;
+ }
+
+ /* FIXME: we have to have more sensible approach than this one */
+ if (fifo_size(&c->audio_data, c->audio_data.rptr) + asize >= AVCODEC_MAX_AUDIO_FRAME_SIZE)
+ fprintf(stderr, "Can't process DV frame #%d. Insufficient video data or severe sync problem.\n", c->frames);
+ fifo_write(&c->audio_data, audio, asize, &c->audio_data.wptr);
+ }
+}
+
+int dv_core_init(DVMuxContext *c, AVStream *streams[])
+{
+ /* We have to sort out where audio and where video stream is */
+ if (streams[0]->codec.codec_type == CODEC_TYPE_VIDEO &&
+ streams[1]->codec.codec_type == CODEC_TYPE_AUDIO) {
+ c->vst = 0;
+ c->ast = 1;
+ }
+ else if (streams[1]->codec.codec_type == CODEC_TYPE_VIDEO &&
+ streams[0]->codec.codec_type == CODEC_TYPE_AUDIO) {
+ c->vst = 1;
+ c->ast = 0;
+ } else
+ goto bail_out;
+
+ /* Some checks -- DV format is very picky about its incoming streams */
+ if (streams[c->vst]->codec.codec_id != CODEC_ID_DVVIDEO ||
+ streams[c->ast]->codec.codec_id != CODEC_ID_PCM_S16LE)
+ goto bail_out;
+ if (streams[c->ast]->codec.sample_rate != 48000 ||
+ streams[c->ast]->codec.channels != 2)
+ goto bail_out;
+
+ if (streams[c->vst]->codec.frame_rate == 25 &&
+ streams[c->vst]->codec.frame_rate_base == 1) {
+ /* May be we have to pick sys for every frame */
+ c->sys = &dv_profiles[1];
+ }
+ else if (streams[c->vst]->codec.frame_rate == 30000 &&
+ streams[c->vst]->codec.frame_rate_base == 1001) {
+ /* May be we have to pick sys for every frame */
+ c->sys = &dv_profiles[0];
+ } else
+ goto bail_out;
+
+ /* Ok, everything seems to be in working order */
+ c->frames = 0;
+ c->has_audio = c->has_video = 0;
+ c->start_time = time(NULL);
+ c->aspect = 0; /* 4:3 is the default */
+ if (streams[c->vst]->codec.aspect_ratio == 16.0 / 9.0)
+ c->aspect = 0x07;
+
+ if (fifo_init(&c->audio_data, AVCODEC_MAX_AUDIO_FRAME_SIZE) < 0)
+ goto bail_out;
+
+ dv_format_frame(c, &c->frame_buf[0]);
+
+ return 0;
+
+bail_out:
+ return -1;
+}
+
+void dv_core_delete(DVMuxContext *c)
+{
+ fifo_free(&c->audio_data);
+}