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roq fixes, but timing is still way off

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an attempt to fix avi playback.. but it's still busted


git-svn-id: https://svn.code.sf.net/p/fteqw/code/trunk@1227 fc73d0e0-1445-4013-8a0c-d673dee63da5
This commit is contained in:
TimeServ 2005-08-12 19:37:24 +00:00
parent 0511e3bcbf
commit 027032a304
4 changed files with 34 additions and 32 deletions
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16
engine/client/m_mp3.c
View file

@ -1186,7 +1186,7 @@ qboolean Media_ShowFilm(void)
fwrite(&size, sizeof(size), 1, f);
fclose(f);
*/
S_RawAudio(-1, roqfilm->audio, 22050, roqfilm->audio_size/roqfilm->audio_channels/2, roqfilm->audio_channels, 2);
S_RawAudio(-1, roqfilm->audio, 22050, roqfilm->audio_size/roqfilm->audio_channels, roqfilm->audio_channels, 2);
}
return true;
@ -1236,18 +1236,18 @@ qboolean Media_ShowFilm(void)
{
LONG lSize;
LPBYTE pBuffer;
AVIStreamRead(pavisound, soundpos, AVISTREAMREAD_CONVENIENT,
NULL, 0, &lSize, NULL);
LONG samples;
soundpos+=lSize;
AVIStreamRead(pavisound, 0, AVISTREAMREAD_CONVENIENT,
NULL, 0, &lSize, &samples);
soundpos+=samples;
pBuffer = framedata;
AVIStreamRead(pavisound, soundpos, AVISTREAMREAD_CONVENIENT, pBuffer, lSize, NULL, &samples);
AVIStreamRead(pavisound, 0, AVISTREAMREAD_CONVENIENT, pBuffer, lSize, NULL, NULL);
S_RawAudio(-1, pBuffer, pWaveFormat->nSamplesPerSec, lSize, pWaveFormat->nChannels, 2);
S_RawAudio(-1, pBuffer, pWaveFormat->nSamplesPerSec, samples, pWaveFormat->nChannels, 2);
}
}
return true;

2
engine/client/roq.h
View file

@ -27,7 +27,7 @@ typedef struct {
unsigned char *buf;
roq_cell cells[256];
roq_qcell qcells[256];
short snd_sqr_arr[260];
short snd_sqr_arr[256];
long roq_start, aud_pos, vid_pos;
long *frame_offset;
unsigned long num_frames, num_audio_bytes;

20
engine/client/roq_read.c
View file

@ -569,7 +569,7 @@ int i, snd_left, snd_right;
fseek(fp, chunk_size, SEEK_CUR);
}
if(ri->audio_buf_size < chunk_size * 2)
if(ri->audio_buf_size < chunk_size*2)
{
if(ri->audio != NULL) BZ_Free(ri->audio);
ri->audio=NULL;
@ -583,12 +583,12 @@ int i, snd_left, snd_right;
if(chunk_id == RoQ_SOUND_MONO)
{
ri->audio_size = chunk_size * 2;
ri->audio_size = chunk_size;
snd_left = chunk_arg;
for(i = 0; i < chunk_size; i++)
{
snd_left += ri->snd_sqr_arr[(unsigned)fgetc(fp)];
*(unsigned short *)&ri->audio[i * 2] = snd_left;
snd_left += (int)ri->snd_sqr_arr[(unsigned)fgetc(fp)];
*(short *)&ri->audio[i * 2] = snd_left;
}
ri->aud_pos = ftell(fp);
return chunk_size;
@ -596,18 +596,18 @@ int i, snd_left, snd_right;
if(chunk_id == RoQ_SOUND_STEREO)
{
ri->audio_size = chunk_size * 2;
ri->audio_size = chunk_size;
snd_left = (chunk_arg & 0xFF00);
snd_right = (chunk_arg & 0xFF) << 8;
for(i = 0; i < chunk_size; i += 2)
{
snd_left += ri->snd_sqr_arr[(unsigned)fgetc(fp)];
snd_right += ri->snd_sqr_arr[(unsigned)fgetc(fp)];
*(unsigned short *)&ri->audio[i * 2] = snd_left;
*(unsigned short *)&ri->audio[i * 2 + 2] = snd_right;
snd_left += (int)ri->snd_sqr_arr[(unsigned)fgetc(fp)];
snd_right += (int)ri->snd_sqr_arr[(unsigned)fgetc(fp)];
*(short *)&ri->audio[i * 2] = snd_left;
*(short *)&ri->audio[i * 2 + 2] = snd_right;
}
ri->aud_pos = ftell(fp);
return chunk_size/2;
return chunk_size;
}
ri->aud_pos = ftell(fp);

28
engine/client/snd_dma.c
View file

@ -1559,6 +1559,7 @@ void S_RawAudio(int sourceid, qbyte *data, int speed, int samples, int channels,
int prepadl;
int oldlength;
int spare;
int outsamples;
float speedfactor;
sfxcache_t *newcache;
streaming_t *s, *free=NULL;
@ -1589,7 +1590,6 @@ void S_RawAudio(int sourceid, qbyte *data, int speed, int samples, int channels,
BZ_Free(s->sfxcache);
return;
}
samples/=channels;
if (i == MAX_RAW_SOURCES) //whoops.
{
if (!free)
@ -1620,6 +1620,9 @@ void S_RawAudio(int sourceid, qbyte *data, int speed, int samples, int channels,
// Con_Printf("Restarting raw stream\n");
}
speedfactor = (float)speed/snd_speed;
outsamples = samples/speedfactor;
oldlength = s->sfxcache->length;
prepadl = 0x7fffffff;
@ -1652,7 +1655,7 @@ void S_RawAudio(int sourceid, qbyte *data, int speed, int samples, int channels,
if (spare < 0) //remaining samples since last time
spare = 0;
if (s->sfxcache->length > snd_speed) //more than a second already buffered
if (s->sfxcache->length > snd_speed*2) // more than 2 seconds of sound
{
Con_Printf("Sacrificed raw sound stream\n");
spare = 0; //too far out. sacrifice it all
@ -1674,16 +1677,16 @@ void S_RawAudio(int sourceid, qbyte *data, int speed, int samples, int channels,
return; //let the slower sound cards catch up. (This shouldn't really happen, but it's possible two cards have slightly different timings but report the same speed)
}*/
newcache = BZ_Malloc(sizeof(sfxcache_t) + (spare+samples) * (s->sfxcache->stereo+1) * s->sfxcache->width);
newcache = BZ_Malloc(sizeof(sfxcache_t) + (spare+outsamples) * (s->sfxcache->stereo+1) * s->sfxcache->width);
memcpy(newcache, s->sfxcache, sizeof(sfxcache_t));
memcpy(newcache->data, s->sfxcache->data + prepadl * (s->sfxcache->stereo+1) * s->sfxcache->width, spare * (s->sfxcache->stereo+1) * s->sfxcache->width);
BZ_Free(s->sfxcache);
s->sfxcache = s->sfx.cache.data = newcache;
newcache->length = spare + samples;
newcache->length = spare + outsamples;
speedfactor = (float)snd_speed/speed;
// move this whole operation to a seperate function?
if (channels == 1)
{
if (width == 2)
@ -1704,9 +1707,9 @@ void S_RawAudio(int sourceid, qbyte *data, int speed, int samples, int channels,
int src=0;
int pos=0;
while (pos++ < samples)
while (pos++ < outsamples)
{
src = pos/speedfactor;
src = pos*speedfactor;
sample = indata[src];
*outpos++ = sample;
}
@ -1730,9 +1733,9 @@ void S_RawAudio(int sourceid, qbyte *data, int speed, int samples, int channels,
int src=0;
int pos=0;
while (pos++ < samples)
while (pos++ < outsamples)
{
src = pos/speedfactor;
src = pos*speedfactor;
sample = indata[src];
*outpos++ = (int)( (unsigned char)(sample) - 128);
}
@ -1764,9 +1767,9 @@ void S_RawAudio(int sourceid, qbyte *data, int speed, int samples, int channels,
int src=0;
int pos=0;
while (pos++ < samples)
while (pos++ < outsamples)
{
src = pos/speedfactor;
src = pos*speedfactor;
sample = indata[src*2];
*outpos++ = sample;
@ -1781,7 +1784,6 @@ void S_RawAudio(int sourceid, qbyte *data, int speed, int samples, int channels,
s->sfxcache->loopstart = s->sfxcache->length;
for (si = sndcardinfo; si; si=si->next)
{
for (i = 0; i < MAX_CHANNELS; i++)
@ -1789,7 +1791,7 @@ void S_RawAudio(int sourceid, qbyte *data, int speed, int samples, int channels,
{
si->channel[i].pos -= prepadl;
// si->channel[i].end -= prepadl;
si->channel[i].end += samples;
si->channel[i].end += outsamples;
if (si->channel[i].end < si->paintedtime)
{