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trash XMMS resampling, use ESD's instead, don't understand it, but it works 

git-svn-id: https://svn.musicpd.org/mpd/trunk@979 09075e82-0dd4-0310-85a5-a0d7c8717e4f
This commit is contained in:
Warren Dukes 2004-05-10 22:31:23 +00:00
parent 33d112499d
commit 69a0b86173
4 changed files with 28 additions and 37 deletions
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2
src/audio.c
View File

@ -152,6 +152,8 @@ void initAudioConfig() {
switch(audio_configFormat->sampleRate) { switch(audio_configFormat->sampleRate) {
case 48000: case 48000:
case 44100: case 44100:
case 32000:
case 16000:
break; break;
default: default:
ERROR("sample rate %i can not be used for audio output\n", ERROR("sample rate %i can not be used for audio output\n",

5
src/flac_decode.c
View File

@ -35,7 +35,8 @@
#include <FLAC/metadata.h> #include <FLAC/metadata.h>
typedef struct { typedef struct {
unsigned char chunk[CHUNK_SIZE]; #define FLAC_CHUNK_SIZE 4080
unsigned char chunk[FLAC_CHUNK_SIZE];
int chunk_length; int chunk_length;
float time; float time;
int bitRate; int bitRate;
@ -417,7 +418,7 @@ FLAC__StreamDecoderWriteStatus flacWrite(const FLAC__SeekableStreamDecoder *dec,
u16 = buf[c_chan][c_samp]; u16 = buf[c_chan][c_samp];
uc = (unsigned char *)&u16; uc = (unsigned char *)&u16;
for(i=0;i<(data->dc->audioFormat.bits/8);i++) { for(i=0;i<(data->dc->audioFormat.bits/8);i++) {
if(data->chunk_length>=CHUNK_SIZE) { if(data->chunk_length>=FLAC_CHUNK_SIZE) {
if(flacSendChunk(data)<0) { if(flacSendChunk(data)<0) {
return FLAC__STREAM_DECODER_WRITE_STATUS_ABORT; return FLAC__STREAM_DECODER_WRITE_STATUS_ABORT;
} }

2
src/ogg_decode.c
View File

@ -181,7 +181,7 @@ int ogg_decode(OutputBuffer * cb, DecoderControl * dc)
int current_section; int current_section;
int eof = 0; int eof = 0;
long ret; long ret;
#define OGG_CHUNK_SIZE 64 #define OGG_CHUNK_SIZE 4096
char chunk[OGG_CHUNK_SIZE]; char chunk[OGG_CHUNK_SIZE];
int chunkpos = 0; int chunkpos = 0;
long bitRate = 0; long bitRate = 0;

56
src/pcm_utils.c
View File

@ -218,41 +218,29 @@ void pcm_convertAudioFormat(AudioFormat * inFormat, char * inBuffer, size_t
} }
else { else {
/* only works if outFormat is 16-bit stereo! */ /* only works if outFormat is 16-bit stereo! */
/* resampling code blatantly ripped from XMMS */ /* resampling code blatantly ripped from ESD */
const int shift = sizeof(mpd_sint16); mpd_sint32 rd_dat = 0;
int x1 = 0, frac; mpd_uint32 wr_dat = 0;
mpd_sint32 i, in_samples, out_samples, x, delta; mpd_sint16 lsample, rsample;
mpd_sint16 * inptr = (mpd_sint16 *)dataChannelConv; register mpd_sint16 * out = (mpd_sint16 *)outBuffer;
mpd_sint16 * outptr = (mpd_sint16 *)outBuffer; register mpd_sint16 * in = (mpd_sint16 *)dataChannelConv;
mpd_uint32 nlen = (((dataChannelLen >> shift) * const int shift = sizeof(mpd_sint16);
(outFormat->sampleRate)) / mpd_uint32 nlen = ((( dataChannelLen >> shift) *
(mpd_uint32)(outFormat->sampleRate)) /
inFormat->sampleRate); inFormat->sampleRate);
nlen <<= shift; nlen <<= shift;
in_samples = dataChannelLen >> shift;
out_samples = nlen >> shift; while( wr_dat < nlen / shift) {
//printf("in_samples=%i out_samples=%i\n",in_samples,out_samples); rd_dat = wr_dat * inFormat->sampleRate /
delta = ((in_samples-1) << 12) / (out_samples-1); outFormat->sampleRate;
for(x = 0, i = 0; i < out_samples; i++) { rd_dat &= ~1;
//int i1,i2,i3,i4;
x1 = (x >> 12) << 12; lsample = in[ rd_dat++ ];
frac = x - x1; rsample = in[ rd_dat++ ];
/* i1 = (x1 >> 12) << 1;
i2 = ((x1 >> 12) + 1) << 1; out[ wr_dat++ ] = lsample;
i3 = ((x1 >> 12) << 1) + 1; out[ wr_dat++ ] = rsample;
i4 = (((x1 >> 12) + 1) << 1) + 1; }
printf("%i,%i,%i,%i\n",i1,i2,i3,i4);*/
*outptr++ =
((inptr[(x1 >> 12) << 1] *
((1<<12) - frac) +
inptr[((x1 >> 12) + 1) << 1 ] *
frac) >> 12);
*outptr++ =
((inptr[((x1 >> 12) << 1) + 1] *
((1<<12) - frac) +
inptr[(((x1 >> 12) + 1) << 1) + 1] *
frac) >> 12);
x += delta;
}
} }
return; return;