From 872af2077706da859b5a64c2e06bfd06de80387b Mon Sep 17 00:00:00 2001
From: Warren Dukes <warren.dukes@gmail.com>
Date: Mon, 10 May 2004 15:21:40 +0000
Subject: [PATCH] resampling code blatantly ripped from xmms, needs testing and
need to right conversion routines for bit conversion and channel conversion
git-svn-id: https://svn.musicpd.org/mpd/trunk@971 09075e82-0dd4-0310-85a5-a0d7c8717e4f
---
src/audio.c | 4 ++--
src/pcm_utils.c | 58 ++++++++++++++++++++++++++++++++++++++++++++++---
2 files changed, 57 insertions(+), 5 deletions(-)
diff --git a/src/audio.c b/src/audio.c
index de5296f2e..5de64b447 100644
--- a/src/audio.c
+++ b/src/audio.c
@@ -159,7 +159,7 @@ void initAudioConfig() {
exit(EXIT_FAILURE);
}
- audio_configFormat->bits = strtol(test,&test,10);
+ audio_configFormat->bits = strtol(test+1,&test,10);
if(*test!=':') {
ERROR("error parsing audio output format: %s\n",conf);
@@ -175,7 +175,7 @@ void initAudioConfig() {
exit(EXIT_FAILURE);
}
- audio_configFormat->channels = strtol(test,&test,10);
+ audio_configFormat->channels = strtol(test+1,&test,10);
if(*test!='\0') {
ERROR("error parsing audio output format: %s\n",conf);
diff --git a/src/pcm_utils.c b/src/pcm_utils.c
index 741cb3d7b..8ef431841 100644
--- a/src/pcm_utils.c
+++ b/src/pcm_utils.c
@@ -24,6 +24,7 @@
#include <string.h>
#include <math.h>
+#include <assert.h>
void pcm_changeBufferEndianness(char * buffer, int bufferSize, int bits) {
char temp;
@@ -140,7 +141,47 @@ void pcm_mix(char * buffer1, char * buffer2, size_t bufferSize1,
void pcm_convertAudioFormat(AudioFormat * inFormat, char * inBuffer, size_t
inSize, AudioFormat * outFormat, char * outBuffer)
{
- abort();
+ /*int inSampleSize = inFormat->bits*inFormat->channels/8;
+ int outSampleSize = outFormat->bits*outFormat->channels/8;*/
+
+ assert(inFormat->bits==16);
+ assert(outFormat->bits==16);
+ assert(inFormat->channels==2);
+ assert(outFormat->channels==2);
+
+ if(inFormat->sampleRate == outFormat->sampleRate) return;
+
+ /* only works if outFormat is 16-bit stereo! */
+ /* resampling code blatantly ripped from XMMS */
+ {
+ const int shift = sizeof(mpd_sint16);
+ mpd_sint32 i, in_samples, out_samples, x, delta;
+ mpd_sint16 * inptr = (mpd_sint16 *)inBuffer;
+ mpd_sint16 * outptr = (mpd_sint16 *)outBuffer;
+ mpd_uint32 nlen = (((inSize >> shift) *
+ (mpd_uint32)(outFormat->sampleRate)) /
+ inFormat->sampleRate);
+ nlen <<= shift;
+ in_samples = inSize >> shift;
+ out_samples = nlen >> shift;
+ delta = (in_samples << 12) / out_samples;
+ for(x = 0, i = 0; i < out_samples; i++) {
+ int x1, frac;
+ x1 = (x >> 12) << 12;
+ frac = x - x1;
+ *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;
}
@@ -148,9 +189,20 @@ void pcm_convertAudioFormat(AudioFormat * inFormat, char * inBuffer, size_t
size_t pcm_sizeOfOutputBufferForAudioFormatConversion(AudioFormat * inFormat,
char * inBuffer, size_t inSize, AudioFormat * outFormat)
{
- abort();
+ const int shift = sizeof(mpd_sint16);
+ mpd_uint32 nlen = (((inSize >> shift) *
+ (mpd_uint32)(outFormat->sampleRate)) /
+ inFormat->sampleRate);
- return 0;
+ nlen <<= shift;
+
+
+ assert(inFormat->bits==16);
+ assert(outFormat->bits==16);
+ assert(inFormat->channels==2);
+ assert(outFormat->channels==2);
+
+ return nlen;
}
/* vim:set shiftwidth=8 tabstop=8 expandtab: */