format conversion for 8->16 bis and mono->stereo
git-svn-id: https://svn.musicpd.org/mpd/trunk@973 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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@ -138,27 +138,92 @@ void pcm_mix(char * buffer1, char * buffer2, size_t bufferSize1,
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pcm_add(buffer1,buffer2,bufferSize1,bufferSize2,vol1,1000-vol1,format);
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pcm_add(buffer1,buffer2,bufferSize1,bufferSize2,vol1,1000-vol1,format);
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}
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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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void pcm_convertAudioFormat(AudioFormat * inFormat, char * inBuffer, size_t
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inSize, AudioFormat * outFormat, char * outBuffer)
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inSize, AudioFormat * outFormat, char * outBuffer)
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{
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{
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/*int inSampleSize = inFormat->bits*inFormat->channels/8;
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static char * bitConvBuffer = NULL;
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int outSampleSize = outFormat->bits*outFormat->channels/8;*/
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static int bitConvBufferLength = 0;
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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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int dataChannelLen;
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char * dataBitConv;
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int dataBitLen;
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assert(inFormat->bits==16);
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assert(outFormat->bits==16);
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assert(outFormat->bits==16);
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assert(inFormat->channels==2);
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assert(outFormat->channels==2);
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assert(outFormat->channels==2);
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if(inFormat->sampleRate == outFormat->sampleRate) return;
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/* converts */
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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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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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int 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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break;
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case 16:
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dataBitConv = inBuffer;
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dataBitLen = inSize;
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break;
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case 24:
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/* put dithering code from mp3_decode here */
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default:
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ERROR("only 8 or 16 bits are supported for conversion!\n");
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exit(EXIT_FAILURE);
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}
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/* converts only between 16 bit audio between mono and stereo */
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switch(inFormat->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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channelConvBuffer = realloc(channelConvBuffer,
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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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int i, inSamples = dataChannelLen >> 1;
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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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case 2:
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dataChannelConv = dataBitConv;
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dataChannelLen = dataBitLen;
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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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exit(EXIT_FAILURE);
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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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/* only works if outFormat is 16-bit stereo! */
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/* only works if outFormat is 16-bit stereo! */
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/* resampling code blatantly ripped from XMMS */
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/* resampling code blatantly ripped from XMMS */
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{
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const int shift = sizeof(mpd_sint16);
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const int shift = sizeof(mpd_sint16);
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mpd_sint32 i, in_samples, out_samples, x, delta;
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mpd_sint32 i, in_samples, out_samples, x, delta;
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mpd_sint16 * inptr = (mpd_sint16 *)inBuffer;
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mpd_sint16 * inptr = (mpd_sint16 *)dataChannelConv;
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mpd_sint16 * outptr = (mpd_sint16 *)outBuffer;
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mpd_sint16 * outptr = (mpd_sint16 *)outBuffer;
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mpd_uint32 nlen = (((inSize >> shift) *
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mpd_uint32 nlen = (((dataChannelLen >> shift) *
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(mpd_uint32)(outFormat->sampleRate)) /
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(mpd_uint32)(outFormat->sampleRate)) /
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inFormat->sampleRate);
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inFormat->sampleRate);
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nlen <<= shift;
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nlen <<= shift;
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@ -197,9 +262,7 @@ size_t pcm_sizeOfOutputBufferForAudioFormatConversion(AudioFormat * inFormat,
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nlen <<= shift;
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nlen <<= shift;
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assert(inFormat->bits==16);
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assert(outFormat->bits==16);
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assert(outFormat->bits==16);
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assert(inFormat->channels==2);
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assert(outFormat->channels==2);
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assert(outFormat->channels==2);
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return nlen;
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return nlen;
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