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split surround delay buf into Ls and Rs in prep for active decoding stuff, and fiddled a bit more with surround level 

git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@3421 b3059339-0415-0410-9bf9-f77b7e298cf2
This commit is contained in:
steve 2001-12-10 00:10:47 +00:00
parent 1df44996db
commit 507a79aa45
1 changed files with 19 additions and 18 deletions
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37
libao2/pl_surround.c
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@ -51,7 +51,7 @@ typedef struct pl_surround_s
int passthrough; // Just be a "NO-OP"
int msecs; // Rear channel delay in milliseconds
int16_t* databuf; // Output audio buffer
int16_t* delaybuf; // circular buffer to be used for delaying audio signal
int16_t* delaybuf; // circular buffer to be used for delaying Ls and Rs audio
int delaybuf_len; // local buffer length in samples
int delaybuf_ptr; // offset in buffer where we are reading/writing
int rate; // input data rate
@ -109,11 +109,11 @@ static int init(){
ao_plugin_data.sz_mult /= 2;
// Figure out buffer space needed for the 15msec delay
pl_surround.delaybuf_len = pl_surround.rate * pl_surround.msecs / 1000;
pl_surround.delaybuf_len = 2 * (pl_surround.rate * pl_surround.msecs / 1000);
// Allocate delay buffer
pl_surround.delaybuf=(void*)calloc(pl_surround.delaybuf_len,sizeof(int16_t));
fprintf(stderr, "pl_surround: %dmsec surround delay, rate %d - buffer is %d samples\n",
pl_surround.msecs,pl_surround.rate, pl_surround.delaybuf_len);
fprintf(stderr, "pl_surround: %dmsec surround delay, rate %d - buffer is %d bytes\n",
pl_surround.msecs,pl_surround.rate, pl_surround.delaybuf_len*sizeof(int16_t));
pl_surround.delaybuf_ptr = 0;
return 1;
@ -149,7 +149,7 @@ static int play(){
if (pl_surround.passthrough) return 1;
// fprintf(stderr, "pl_surround: play %d bytes, %d samples\n", ao_plugin_data.len, samples);
// fprintf(stderr, "pl_surround: play %d bytes, %d samples\n", ao_plugin_data.len, samples);
samples = ao_plugin_data.len / sizeof(int16_t) / pl_surround.input_channels;
@ -159,23 +159,24 @@ static int play(){
// About volume balancing...
// Surround encoding does the following:
// Lt=L+.707*C+.707*S, Rt=R+.707*C-.707*S
// So S should to be extracted as:
// .707*(Lt-Rt)
// So S should be extracted as:
// (Lt-Rt)
// But we are splitting the S to two output channels, so we
// must take another 3dB off as we split it:
// Ls=Rs=.707*.707*(Lt-Rt)
// = .5*(Lt-Rt)
// This result is handy as it is also sure not to clip, even
// though L could be full scale +ve, R full scale -ve
// must take 3dB off as we split it:
// Ls=Rs=.707*(Lt-Rt)
// Trouble is, Lt could be +32767, Rt -32768, so possibility that S will
// clip. So to compensate, we cut L/R by 3dB (*.707), and S by 6dB (/2).
// front left and right
out[0] = in[0];
out[1] = in[1];
// surround - from 15msec ago
// output front left and right
out[0] = in[0]*.707;
out[1] = in[1]*.707;
// output Ls and Rs - from 15msec ago
out[2] = pl_surround.delaybuf[pl_surround.delaybuf_ptr];
out[3] = -out[2];
out[3] = pl_surround.delaybuf[pl_surround.delaybuf_ptr+1];
// calculate and save surround for 15msecs time
pl_surround.delaybuf[pl_surround.delaybuf_ptr++] = (in[0]/2 - in[1]/2);
surround = (in[0]/2 - in[1]/2);
pl_surround.delaybuf[pl_surround.delaybuf_ptr++] = surround;
pl_surround.delaybuf[pl_surround.delaybuf_ptr++] = - surround;
pl_surround.delaybuf_ptr %= pl_surround.delaybuf_len;
// next samples...
in = &in[pl_surround.input_channels]; out = &out[4];