[Bf-blender-cvs] SVN commit: /data/svn/bf-blender [20805] trunk/blender/source/blender: == SEQUENCER ==
Peter Schlaile
peter at schlaile.de
Thu Jun 11 13:44:47 CEST 2009
Revision: 20805
http://projects.blender.org/plugins/scmsvn/viewcvs.php?view=rev&root=bf-blender&revision=20805
Author: schlaile
Date: 2009-06-11 13:44:47 +0200 (Thu, 11 Jun 2009)
Log Message:
-----------
== SEQUENCER ==
This fixes
* some issues with Scene strips containing audio by removing
the curpos pointer from sequence structure. (the same scene
strip can now be used in a row)
That also makes the code a lot cleaner.
* fixed a corner case on the beginning of a strip, where audio was
not mixed in, depending of current audio buffer state.
* Also: made some hardwired variables macros to enhance readability.
Problem remaining: mixing the same scene strip several times (read
put it into a stack instead of into a row) has
problems with HD-audio since the same HD-audio state structure is
used and therefore the system will seek permanently, which leads to
audio distortions...
Modified Paths:
--------------
trunk/blender/source/blender/makesdna/DNA_sequence_types.h
trunk/blender/source/blender/src/seqaudio.c
Modified: trunk/blender/source/blender/makesdna/DNA_sequence_types.h
===================================================================
--- trunk/blender/source/blender/makesdna/DNA_sequence_types.h 2009-06-11 10:46:13 UTC (rev 20804)
+++ trunk/blender/source/blender/makesdna/DNA_sequence_types.h 2009-06-11 11:44:47 UTC (rev 20805)
@@ -159,7 +159,7 @@
struct bSound *sound; /* the linked "bSound" object */
struct hdaudio *hdaudio; /* external hdaudio object */
float level, pan; /* level in dB (0=full), pan -1..1 */
- int curpos; /* last sample position in audio_fill() */
+ int scenenr; /* for scene selection */
float strobe;
void *effectdata; /* Struct pointer for effect settings */
@@ -170,8 +170,6 @@
int blend_mode;
float blend_opacity;
- int scenenr; /* for scene selection */
- int pad;
} Sequence;
typedef struct MetaStack {
Modified: trunk/blender/source/blender/src/seqaudio.c
===================================================================
--- trunk/blender/source/blender/src/seqaudio.c 2009-06-11 10:46:13 UTC (rev 20804)
+++ trunk/blender/source/blender/src/seqaudio.c 2009-06-11 11:44:47 UTC (rev 20805)
@@ -104,6 +104,10 @@
#define AFRA2TIME(a) ((((double) audio_scene->r.frs_sec_base) * (a)) / audio_scene->r.frs_sec)
#define ATIME2FRA(a) ((((double) audio_scene->r.frs_sec) * (a)) / audio_scene->r.frs_sec_base)
+/* we do currently stereo 16 bit mixing only */
+#define AUDIO_CHANNELS 2
+#define SAMPLE_SIZE (AUDIO_CHANNELS * sizeof(short))
+
/////
//
/* local protos ------------------- */
@@ -149,7 +153,8 @@
strcpy(buf, "RIFFlengWAVEfmt fmln01ccRATEbsecBP16dataDLEN");
totframe = (EFRA - SFRA + 1);
- totlen = (int) ( FRA2TIME(totframe) * (float)G.scene->audio.mixrate * 4.0);
+ totlen = (int) ( FRA2TIME(totframe)
+ * (float)G.scene->audio.mixrate * SAMPLE_SIZE);
printf(" totlen %d\n", totlen+36+8);
totlen+= 36; /* len is filesize-8 in WAV spec, total header is 44 bytes */
@@ -159,7 +164,7 @@
buf[16] = 0x10; buf[17] = buf[18] = buf[19] = 0; buf[20] = 1; buf[21] = 0;
buf[22] = 2; buf[23]= 0;
memcpy(buf+24, &G.scene->audio.mixrate, 4);
- i = G.scene->audio.mixrate * 4;
+ i = G.scene->audio.mixrate * SAMPLE_SIZE;
memcpy(buf+28, &i, 4);
buf[32] = 4; buf[33] = 0; buf[34] = 16; buf[35] = 0;
i = totlen;
@@ -192,7 +197,8 @@
memset(buf+i, 0, 64);
- CFRA=(int) ( ((float)(audio_pos-64)/( G.scene->audio.mixrate*4 ))*FPS );
+ CFRA=(int) ( ((float)(audio_pos-64)
+ / ( G.scene->audio.mixrate*SAMPLE_SIZE ))*FPS );
audio_fill(buf+i, NULL, 64);
if (G.order == B_ENDIAN) {
@@ -226,7 +232,7 @@
#ifndef DISABLE_SDL
for (i = 0; i < len; i += 64) {
CFRA = (int) ( ((float)(audio_pos-64)
- /( audio_scene->audio.mixrate*4 ))
+ /( audio_scene->audio.mixrate * SAMPLE_SIZE ))
* FPS );
audio_fill(mixdown + i, NULL,
@@ -251,7 +257,7 @@
fac = pow(10.0, ((-(db+audio_scene->audio.main))/20.0));
- for (i=0; i<len; i+=4) {
+ for (i = 0; i < len; i += SAMPLE_SIZE) {
float facf = facf_start + ((double) i) * m;
float f_l = facl / (fac / facf);
float f_r = facr / (fac / facf);
@@ -276,31 +282,52 @@
return;
}
ratio = (float)G.scene->audio.mixrate / (float)sound->sample->rate;
- sound->streamlen = (int) ( (float)sound->sample->len * ratio * 2.0/((float)sound->sample->channels) );
+ sound->streamlen = (int) ( (float)sound->sample->len * ratio
+ * AUDIO_CHANNELS
+ / ((float)sound->sample->channels) );
sound->stream = malloc((int) ((float)sound->streamlen * 1.05));
if (sound->sample->rate == G.scene->audio.mixrate) {
- if (sound->sample->channels == 2) {
- memcpy(sound->stream, sound->sample->data, sound->streamlen);
+ if (sound->sample->channels == AUDIO_CHANNELS) {
+ memcpy(sound->stream,
+ sound->sample->data, sound->streamlen);
return;
- } else {
+ } else if (sound->sample->channels == 1) {
for (source = (signed short*)(sound->sample->data),
dest = (signed short*)(sound->stream),
i=0;
- i<sound->streamlen/4;
- dest += 2, source++, i++) dest[0] = dest[1] = source[0];
+ i<sound->streamlen/SAMPLE_SIZE;
+ dest += 2, source++, i++) {
+ int j;
+ for (j = 0; j < AUDIO_CHANNELS; j++) {
+ dest[j] = source[0];
+ }
+ }
return;
+ } else {
+ fprintf(stderr, "audio_makestream: "
+ "FIXME: can't handle number of channels %d\n",
+ sound->sample->channels);
+ return;
}
}
if (sound->sample->channels == 1) {
- for (dest=(signed short*)(sound->stream), i=0, source=(signed short*)(sound->sample->data);
- i<(sound->streamlen/4); dest+=2, i++)
- dest[0] = dest[1] = source[(int)((float)i/ratio)];
+ for (dest = (signed short*)(sound->stream), i=0,
+ source = (signed short*)(sound->sample->data);
+ i<(sound->streamlen/SAMPLE_SIZE);
+ dest += AUDIO_CHANNELS, i++) {
+ int j;
+ int s = source[(int)((float)i/ratio)];
+ for (j = 0; j < AUDIO_CHANNELS; j++) {
+ dest[j] = s;
+ }
+ }
}
else if (sound->sample->channels == 2) {
- for (dest=(signed short*)(sound->stream), i=0, source=(signed short*)(sound->sample->data);
- i<(sound->streamlen/2); dest+=2, i+=2) {
+ for (dest=(signed short*)(sound->stream), i=0,
+ source = (signed short*)(sound->sample->data);
+ i<(sound->streamlen / 2); dest += AUDIO_CHANNELS, i+=2) {
dest[1] = source[(int)((float)i/ratio)];
- dest[0] = source[(int)((float)i/ratio)+1];
+ dest[0] = source[(int)((float)i/ratio)+1];
}
}
}
@@ -315,24 +342,37 @@
seq->anim_startofs))
* ((float)audio_scene
->audio.mixrate)
- * 4 ));
+ * SAMPLE_SIZE));
}
static int curpos2fra(Sequence * seq, int curpos)
{
return ((int) floor(
ATIME2FRA(
- ((double) curpos) / 4
+ ((double) curpos) / SAMPLE_SIZE
/audio_scene->audio.mixrate)))
- seq->anim_startofs + seq->start;
}
+static int get_curpos(Sequence * seq, int cfra)
+{
+ return audio_pos +
+ (((int)((FRA2TIME(((double) cfra)
+ - ((double) audio_scene->r.cfra)
+ - ((double) seq->start)
+ + ((double) seq->anim_startofs))
+ * ((float)audio_scene->audio.mixrate)
+ * SAMPLE_SIZE )))
+ & (~(SAMPLE_SIZE - 1))); /* has to be sample aligned! */
+}
+
static void do_audio_seq_ipo(Sequence * seq, int len, float * facf_start,
- float * facf_end)
+ float * facf_end, int cfra)
{
- int cfra_start = curpos2fra(seq, seq->curpos);
+ int seq_curpos = get_curpos(seq, cfra);
+ int cfra_start = curpos2fra(seq, seq_curpos);
int cfra_end = cfra_start + 1;
- int ipo_curpos_start = fra2curpos(seq, curpos2fra(seq, seq->curpos));
+ int ipo_curpos_start = fra2curpos(seq, curpos2fra(seq, seq_curpos));
int ipo_curpos_end = fra2curpos(seq, cfra_end);
double ipo_facf_start;
double ipo_facf_end;
@@ -346,8 +386,8 @@
m = (ipo_facf_end- ipo_facf_start)/(ipo_curpos_end - ipo_curpos_start);
- *facf_start = ipo_facf_start + (seq->curpos - ipo_curpos_start) * m;
- *facf_end = ipo_facf_start + (seq->curpos + len-ipo_curpos_start) * m;
+ *facf_start = ipo_facf_start + (seq_curpos - ipo_curpos_start) * m;
+ *facf_end = ipo_facf_start + (seq_curpos + len-ipo_curpos_start) * m;
}
#endif
@@ -361,20 +401,30 @@
bSound* sound;
float facf_start;
float facf_end;
+ int seq_curpos = get_curpos(seq, cfra);
+ /* catch corner case at the beginning of strip */
+ if (seq_curpos < 0 && (seq_curpos + len > 0)) {
+ seq_curpos *= -1;
+ len -= seq_curpos;
+ sstream += seq_curpos;
+ seq_curpos = 0;
+ }
+
sound = seq->sound;
audio_makestream(sound);
- if ((seq->curpos<sound->streamlen -len) && (seq->curpos>=0) &&
+ if ((seq_curpos < sound->streamlen -len) && (seq_curpos >= 0) &&
(seq->startdisp <= cfra) && ((seq->enddisp) > cfra))
{
if(seq->ipo && seq->ipo->curve.first) {
- do_audio_seq_ipo(seq, len, &facf_start, &facf_end);
+ do_audio_seq_ipo(seq, len, &facf_start, &facf_end,
+ cfra);
} else {
facf_start = 1.0;
facf_end = 1.0;
}
cvtbuf = malloc(len);
- memcpy(cvtbuf, ((uint8_t*)sound->stream)+(seq->curpos & (~3)), len);
+ memcpy(cvtbuf, ((uint8_t*)sound->stream)+(seq_curpos), len);
audio_levels(cvtbuf, len, seq->level, facf_start, facf_end,
seq->pan);
if (!mixdown) {
@@ -384,7 +434,6 @@
}
free(cvtbuf);
}
- seq->curpos += len;
}
#endif
@@ -396,12 +445,22 @@
uint8_t* cvtbuf;
float facf_start;
float facf_end;
+ int seq_curpos = get_curpos(seq, cfra);
- if ((seq->curpos >= 0) &&
+ /* catch corner case at the beginning of strip */
+ if (seq_curpos < 0 && (seq_curpos + len > 0)) {
+ seq_curpos *= -1;
+ len -= seq_curpos;
+ sstream += seq_curpos;
+ seq_curpos = 0;
+ }
+
+ if ((seq_curpos >= 0) &&
(seq->startdisp <= cfra) && ((seq->enddisp) > cfra))
{
if(seq->ipo && seq->ipo->curve.first) {
- do_audio_seq_ipo(seq, len, &facf_start, &facf_end);
+ do_audio_seq_ipo(seq, len, &facf_start, &facf_end,
+ cfra);
} else {
facf_start = 1.0;
facf_end = 1.0;
@@ -409,10 +468,10 @@
cvtbuf = malloc(len);
sound_hdaudio_extract(seq->hdaudio, (short*) cvtbuf,
- seq->curpos / 4,
+ seq_curpos / SAMPLE_SIZE,
audio_scene->audio.mixrate,
- 2,
- len / 4);
+ AUDIO_CHANNELS,
+ len / SAMPLE_SIZE);
audio_levels(cvtbuf, len, seq->level, facf_start, facf_end,
seq->pan);
if (!mixdown) {
@@ -424,18 +483,15 @@
}
free(cvtbuf);
}
- seq->curpos += len;
}
#endif
#ifndef DISABLE_SDL
static void audio_fill_seq(Sequence * seq, void * mixdown,
- uint8_t *sstream, int len, int cfra,
- int advance_only);
+ uint8_t *sstream, int len, int cfra);
static void audio_fill_scene_strip(Sequence * seq, void * mixdown,
- uint8_t *sstream, int len, int cfra,
- int advance_only)
+ uint8_t *sstream, int len, int cfra)
{
Editing *ed;
@@ -450,8 +506,7 @@
audio_fill_seq(ed->seqbasep->first,
mixdown,
- sstream, len, sce_cfra,
- advance_only);
+ sstream, len, sce_cfra);
}
/* restore */
@@ -461,8 +516,7 @@
#ifndef DISABLE_SDL
static void audio_fill_seq(Sequence * seq, void * mixdown,
- uint8_t *sstream, int len, int cfra,
- int advance_only)
+ uint8_t *sstream, int len, int cfra)
{
while(seq) {
if (seq->type == SEQ_META &&
@@ -470,11 +524,7 @@
if (seq->startdisp <= cfra && seq->enddisp > cfra) {
audio_fill_seq(seq->seqbase.first,
@@ Diff output truncated at 10240 characters. @@
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