Mercurial > mplayer.hg
changeset 3420:cfc10bc948c4
split surround delay buf into Ls and Rs in prep for active decoding stuff, and fiddled a bit more with surround level
author | steve |
---|---|
date | Mon, 10 Dec 2001 00:10:47 +0000 |
parents | dfdc17415fb9 |
children | 3478654d2230 |
files | libao2/pl_surround.c |
diffstat | 1 files changed, 19 insertions(+), 18 deletions(-) [+] |
line wrap: on
line diff
--- a/libao2/pl_surround.c Sun Dec 09 22:15:51 2001 +0000 +++ b/libao2/pl_surround.c Mon Dec 10 00:10:47 2001 +0000 @@ -51,7 +51,7 @@ 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 @@ 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 @@ 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 @@ // 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];