Add mpd-indent.sh
Indent the entire tree, hopefully we can keep it indented. git-svn-id: https://svn.musicpd.org/mpd/trunk@4410 09075e82-0dd4-0310-85a5-a0d7c8717e4f
This commit is contained in:
Avuton Olrich
239
src/pcm_utils.c
239
src/pcm_utils.c
@@ -16,7 +16,6 @@
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* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
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*/
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#include "pcm_utils.h"
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#include "mpd_types.h"
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@@ -26,134 +25,143 @@
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#include <math.h>
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#include <assert.h>
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void pcm_volumeChange(char * buffer, int bufferSize, AudioFormat * format,
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int volume)
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void pcm_volumeChange(char *buffer, int bufferSize, AudioFormat * format,
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int volume)
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{
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mpd_sint32 temp32;
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mpd_sint8 * buffer8 = (mpd_sint8 *)buffer;
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mpd_sint16 * buffer16 = (mpd_sint16 *)buffer;
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mpd_sint8 *buffer8 = (mpd_sint8 *) buffer;
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mpd_sint16 *buffer16 = (mpd_sint16 *) buffer;
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if(volume>=1000) return;
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if(volume<=0) {
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memset(buffer,0,bufferSize);
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if (volume >= 1000)
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return;
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if (volume <= 0) {
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memset(buffer, 0, bufferSize);
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return;
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}
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switch(format->bits) {
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switch (format->bits) {
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case 16:
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while(bufferSize>0) {
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while (bufferSize > 0) {
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temp32 = *buffer16;
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temp32*= volume;
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temp32/=1000;
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*buffer16 = temp32>32767 ? 32767 :
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(temp32<-32768 ? -32768 : temp32);
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temp32 *= volume;
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temp32 /= 1000;
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*buffer16 = temp32 > 32767 ? 32767 :
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(temp32 < -32768 ? -32768 : temp32);
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buffer16++;
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bufferSize-=2;
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bufferSize -= 2;
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}
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break;
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case 8:
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while(bufferSize>0) {
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while (bufferSize > 0) {
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temp32 = *buffer8;
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temp32*= volume;
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temp32/=1000;
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*buffer8 = temp32>127 ? 127 :
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(temp32<-128 ? -128 : temp32);
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temp32 *= volume;
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temp32 /= 1000;
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*buffer8 = temp32 > 127 ? 127 :
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(temp32 < -128 ? -128 : temp32);
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buffer8++;
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bufferSize--;
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}
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break;
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default:
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ERROR("%i bits not supported by pcm_volumeChange!\n",
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format->bits);
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format->bits);
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exit(EXIT_FAILURE);
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}
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}
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static void pcm_add(char * buffer1, char * buffer2, size_t bufferSize1,
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size_t bufferSize2, int vol1, int vol2, AudioFormat * format)
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static void pcm_add(char *buffer1, char *buffer2, size_t bufferSize1,
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size_t bufferSize2, int vol1, int vol2,
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AudioFormat * format)
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{
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mpd_sint32 temp32;
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mpd_sint8 * buffer8_1 = (mpd_sint8 *)buffer1;
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mpd_sint8 * buffer8_2 = (mpd_sint8 *)buffer2;
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mpd_sint16 * buffer16_1 = (mpd_sint16 *)buffer1;
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mpd_sint16 * buffer16_2 = (mpd_sint16 *)buffer2;
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mpd_sint8 *buffer8_1 = (mpd_sint8 *) buffer1;
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mpd_sint8 *buffer8_2 = (mpd_sint8 *) buffer2;
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mpd_sint16 *buffer16_1 = (mpd_sint16 *) buffer1;
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mpd_sint16 *buffer16_2 = (mpd_sint16 *) buffer2;
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switch(format->bits) {
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switch (format->bits) {
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case 16:
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while(bufferSize1>0 && bufferSize2>0) {
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temp32 = (vol1*(*buffer16_1)+vol2*(*buffer16_2))/1000;
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*buffer16_1 = temp32>32767 ? 32767 :
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(temp32<-32768 ? -32768 : temp32);
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while (bufferSize1 > 0 && bufferSize2 > 0) {
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temp32 =
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(vol1 * (*buffer16_1) +
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vol2 * (*buffer16_2)) / 1000;
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*buffer16_1 =
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temp32 > 32767 ? 32767 : (temp32 <
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-32768 ? -32768 : temp32);
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buffer16_1++;
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buffer16_2++;
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bufferSize1-=2;
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bufferSize2-=2;
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bufferSize1 -= 2;
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bufferSize2 -= 2;
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}
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if(bufferSize2>0) memcpy(buffer16_1,buffer16_2,bufferSize2);
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if (bufferSize2 > 0)
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memcpy(buffer16_1, buffer16_2, bufferSize2);
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break;
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case 8:
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while(bufferSize1>0 && bufferSize2>0) {
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temp32 = (vol1*(*buffer8_1)+vol2*(*buffer8_2))/1000;
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*buffer8_1 = temp32>127 ? 127 :
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(temp32<-128 ? -128 : temp32);
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while (bufferSize1 > 0 && bufferSize2 > 0) {
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temp32 =
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(vol1 * (*buffer8_1) + vol2 * (*buffer8_2)) / 1000;
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*buffer8_1 =
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temp32 > 127 ? 127 : (temp32 <
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-128 ? -128 : temp32);
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buffer8_1++;
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buffer8_2++;
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bufferSize1--;
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bufferSize2--;
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}
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if(bufferSize2>0) memcpy(buffer8_1,buffer8_2,bufferSize2);
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if (bufferSize2 > 0)
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memcpy(buffer8_1, buffer8_2, bufferSize2);
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break;
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default:
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ERROR("%i bits not supported by pcm_add!\n",format->bits);
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ERROR("%i bits not supported by pcm_add!\n", format->bits);
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exit(EXIT_FAILURE);
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}
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}
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void pcm_mix(char * buffer1, char * buffer2, size_t bufferSize1,
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size_t bufferSize2, AudioFormat * format, float portion1)
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void pcm_mix(char *buffer1, char *buffer2, size_t bufferSize1,
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size_t bufferSize2, AudioFormat * format, float portion1)
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{
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int vol1;
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float s = sin(M_PI_2*portion1);
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s*=s;
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vol1 = s*1000+0.5;
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vol1 = vol1>1000 ? 1000 : ( vol1<0 ? 0 : vol1 );
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float s = sin(M_PI_2 * portion1);
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s *= s;
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pcm_add(buffer1,buffer2,bufferSize1,bufferSize2,vol1,1000-vol1,format);
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vol1 = s * 1000 + 0.5;
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vol1 = vol1 > 1000 ? 1000 : (vol1 < 0 ? 0 : vol1);
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pcm_add(buffer1, buffer2, bufferSize1, bufferSize2, vol1, 1000 - vol1,
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format);
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}
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/* outFormat bits must be 16 and channels must be 2! */
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void pcm_convertAudioFormat(AudioFormat * inFormat, char * inBuffer, size_t
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inSize, AudioFormat * outFormat, char * outBuffer)
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void pcm_convertAudioFormat(AudioFormat * inFormat, char *inBuffer, size_t
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inSize, AudioFormat * outFormat, char *outBuffer)
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{
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static char * bitConvBuffer = NULL;
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static char *bitConvBuffer = NULL;
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static int bitConvBufferLength = 0;
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static char * channelConvBuffer = NULL;
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static char *channelConvBuffer = NULL;
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static int channelConvBufferLength = 0;
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char * dataChannelConv;
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char *dataChannelConv;
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int dataChannelLen;
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char * dataBitConv;
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char *dataBitConv;
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int dataBitLen;
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assert(outFormat->bits==16);
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assert(outFormat->channels==2 || outFormat->channels==1);
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assert(outFormat->bits == 16);
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assert(outFormat->channels == 2 || outFormat->channels == 1);
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/* converts */
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switch(inFormat->bits) {
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switch (inFormat->bits) {
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case 8:
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dataBitLen = inSize << 1;
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if(dataBitLen > bitConvBufferLength) {
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if (dataBitLen > bitConvBufferLength) {
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bitConvBuffer = realloc(bitConvBuffer, dataBitLen);
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bitConvBufferLength = dataBitLen;
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}
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dataBitConv = bitConvBuffer;
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{
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mpd_sint8 * in = (mpd_sint8 *)inBuffer;
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mpd_sint16 * out = (mpd_sint16 *)dataBitConv;
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mpd_sint8 *in = (mpd_sint8 *) inBuffer;
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mpd_sint16 *out = (mpd_sint16 *) dataBitConv;
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int i;
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for(i=0; i<inSize; i++) {
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for (i = 0; i < inSize; i++) {
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*out++ = (*in++) << 8;
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}
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}
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@@ -170,96 +178,96 @@ void pcm_convertAudioFormat(AudioFormat * inFormat, char * inBuffer, size_t
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}
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/* converts only between 16 bit audio between mono and stereo */
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if(inFormat->channels == outFormat->channels)
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{
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if (inFormat->channels == outFormat->channels) {
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dataChannelConv = dataBitConv;
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dataChannelLen = dataBitLen;
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}
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else {
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switch(inFormat->channels) {
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/* convert from 1 -> 2 channels */
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} else {
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switch (inFormat->channels) {
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/* convert from 1 -> 2 channels */
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case 1:
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dataChannelLen = (dataBitLen >> 1) << 2;
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if(dataChannelLen > channelConvBufferLength) {
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if (dataChannelLen > channelConvBufferLength) {
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channelConvBuffer = realloc(channelConvBuffer,
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dataChannelLen);
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dataChannelLen);
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channelConvBufferLength = dataChannelLen;
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}
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dataChannelConv = channelConvBuffer;
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{
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mpd_sint16 * in = (mpd_sint16 *)dataBitConv;
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mpd_sint16 * out = (mpd_sint16 *)dataChannelConv;
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mpd_sint16 *in = (mpd_sint16 *) dataBitConv;
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mpd_sint16 *out =
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(mpd_sint16 *) dataChannelConv;
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int i, inSamples = dataBitLen >> 1;
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for(i=0;i<inSamples;i++) {
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for (i = 0; i < inSamples; i++) {
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*out++ = *in;
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*out++ = *in++;
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}
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}
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break;
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/* convert from 2 -> 1 channels */
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/* convert from 2 -> 1 channels */
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case 2:
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dataChannelLen = dataBitLen >> 1;
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if(dataChannelLen > channelConvBufferLength) {
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if (dataChannelLen > channelConvBufferLength) {
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channelConvBuffer = realloc(channelConvBuffer,
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dataChannelLen);
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dataChannelLen);
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channelConvBufferLength = dataChannelLen;
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}
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dataChannelConv = channelConvBuffer;
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{
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mpd_sint16 * in = (mpd_sint16 *)dataBitConv;
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mpd_sint16 * out = (mpd_sint16 *)dataChannelConv;
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mpd_sint16 *in = (mpd_sint16 *) dataBitConv;
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mpd_sint16 *out =
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(mpd_sint16 *) dataChannelConv;
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int i, inSamples = dataBitLen >> 2;
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for(i=0;i<inSamples;i++) {
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*out = (*in++)/2;
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*out++ += (*in++)/2;
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for (i = 0; i < inSamples; i++) {
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*out = (*in++) / 2;
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*out++ += (*in++) / 2;
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}
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}
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break;
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default:
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ERROR("only 1 or 2 channels are supported for conversion!\n");
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ERROR
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("only 1 or 2 channels are supported for conversion!\n");
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exit(EXIT_FAILURE);
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}
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}
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if(inFormat->sampleRate == outFormat->sampleRate) {
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memcpy(outBuffer,dataChannelConv,dataChannelLen);
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}
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else {
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if (inFormat->sampleRate == outFormat->sampleRate) {
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memcpy(outBuffer, dataChannelConv, dataChannelLen);
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} else {
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/* only works if outFormat is 16-bit stereo! */
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/* resampling code blatantly ripped from ESD */
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mpd_uint32 rd_dat = 0;
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mpd_uint32 wr_dat = 0;
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mpd_sint16 lsample, rsample;
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mpd_sint16 * out = (mpd_sint16 *)outBuffer;
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mpd_sint16 * in = (mpd_sint16 *)dataChannelConv;
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const int shift = sizeof(mpd_sint16)*outFormat->channels;
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mpd_uint32 nlen = ((( dataChannelLen / shift) *
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(mpd_uint32)(outFormat->sampleRate)) /
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inFormat->sampleRate);
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mpd_sint16 *out = (mpd_sint16 *) outBuffer;
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mpd_sint16 *in = (mpd_sint16 *) dataChannelConv;
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const int shift = sizeof(mpd_sint16) * outFormat->channels;
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mpd_uint32 nlen = (((dataChannelLen / shift) *
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(mpd_uint32) (outFormat->sampleRate)) /
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inFormat->sampleRate);
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nlen *= outFormat->channels;
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switch(outFormat->channels) {
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switch (outFormat->channels) {
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case 1:
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while( wr_dat < nlen) {
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rd_dat = wr_dat * inFormat->sampleRate /
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outFormat->sampleRate;
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while (wr_dat < nlen) {
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rd_dat = wr_dat * inFormat->sampleRate /
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outFormat->sampleRate;
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lsample = in[ rd_dat++ ];
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lsample = in[rd_dat++];
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out[ wr_dat++ ] = lsample;
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out[wr_dat++] = lsample;
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}
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break;
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case 2:
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while( wr_dat < nlen) {
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rd_dat = wr_dat * inFormat->sampleRate /
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outFormat->sampleRate;
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while (wr_dat < nlen) {
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rd_dat = wr_dat * inFormat->sampleRate /
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outFormat->sampleRate;
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rd_dat &= ~1;
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lsample = in[ rd_dat++ ];
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rsample = in[ rd_dat++ ];
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lsample = in[rd_dat++];
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rsample = in[rd_dat++];
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out[ wr_dat++ ] = lsample;
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out[ wr_dat++ ] = rsample;
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out[wr_dat++] = lsample;
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out[wr_dat++] = rsample;
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}
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break;
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}
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@@ -269,12 +277,13 @@ void pcm_convertAudioFormat(AudioFormat * inFormat, char * inBuffer, size_t
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}
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size_t pcm_sizeOfOutputBufferForAudioFormatConversion(AudioFormat * inFormat,
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size_t inSize, AudioFormat * outFormat)
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size_t inSize,
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AudioFormat * outFormat)
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{
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const int shift = sizeof(mpd_sint16)*outFormat->channels;
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const int shift = sizeof(mpd_sint16) * outFormat->channels;
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size_t outSize = inSize;
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switch(inFormat->bits) {
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switch (inFormat->bits) {
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case 8:
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outSize = outSize << 1;
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break;
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@@ -285,8 +294,8 @@ size_t pcm_sizeOfOutputBufferForAudioFormatConversion(AudioFormat * inFormat,
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exit(EXIT_FAILURE);
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}
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if(inFormat->channels != outFormat->channels) {
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switch(inFormat->channels) {
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if (inFormat->channels != outFormat->channels) {
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switch (inFormat->channels) {
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case 1:
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outSize = (outSize >> 1) << 2;
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break;
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@@ -295,9 +304,9 @@ size_t pcm_sizeOfOutputBufferForAudioFormatConversion(AudioFormat * inFormat,
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break;
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}
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}
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outSize = (((outSize / shift) * (mpd_uint32)(outFormat->sampleRate)) /
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inFormat->sampleRate);
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outSize = (((outSize / shift) * (mpd_uint32) (outFormat->sampleRate)) /
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inFormat->sampleRate);
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outSize *= shift;
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Block a user