From 872af2077706da859b5a64c2e06bfd06de80387b Mon Sep 17 00:00:00 2001 From: Warren Dukes Date: Mon, 10 May 2004 15:21:40 +0000 Subject: 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 #include +#include 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); + + nlen <<= shift; + + + assert(inFormat->bits==16); + assert(outFormat->bits==16); + assert(inFormat->channels==2); + assert(outFormat->channels==2); - return 0; + return nlen; } /* vim:set shiftwidth=8 tabstop=8 expandtab: */ -- cgit v1.2.3