some debug code
git-svn-id: https://svn.musicpd.org/mpd/trunk@978 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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@ -181,7 +181,7 @@ int ogg_decode(OutputBuffer * cb, DecoderControl * dc)
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int current_section;
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int current_section;
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int eof = 0;
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int eof = 0;
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long ret;
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long ret;
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#define OGG_CHUNK_SIZE 2048
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#define OGG_CHUNK_SIZE 64
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char chunk[OGG_CHUNK_SIZE];
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char chunk[OGG_CHUNK_SIZE];
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int chunkpos = 0;
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int chunkpos = 0;
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long bitRate = 0;
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long bitRate = 0;
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@ -220,6 +220,7 @@ void pcm_convertAudioFormat(AudioFormat * inFormat, char * inBuffer, size_t
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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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const int shift = sizeof(mpd_sint16);
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const int shift = sizeof(mpd_sint16);
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int x1 = 0, frac;
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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 *)dataChannelConv;
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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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@ -229,11 +230,17 @@ void pcm_convertAudioFormat(AudioFormat * inFormat, char * inBuffer, size_t
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nlen <<= shift;
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nlen <<= shift;
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in_samples = dataChannelLen >> shift;
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in_samples = dataChannelLen >> shift;
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out_samples = nlen >> shift;
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out_samples = nlen >> shift;
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delta = (in_samples << 12) / out_samples;
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//printf("in_samples=%i out_samples=%i\n",in_samples,out_samples);
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delta = ((in_samples-1) << 12) / (out_samples-1);
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for(x = 0, i = 0; i < out_samples; i++) {
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for(x = 0, i = 0; i < out_samples; i++) {
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int x1, frac;
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//int i1,i2,i3,i4;
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x1 = (x >> 12) << 12;
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x1 = (x >> 12) << 12;
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frac = x - x1;
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frac = x - x1;
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/* i1 = (x1 >> 12) << 1;
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i2 = ((x1 >> 12) + 1) << 1;
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i3 = ((x1 >> 12) << 1) + 1;
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i4 = (((x1 >> 12) + 1) << 1) + 1;
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printf("%i,%i,%i,%i\n",i1,i2,i3,i4);*/
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*outptr++ =
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*outptr++ =
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((inptr[(x1 >> 12) << 1] *
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((inptr[(x1 >> 12) << 1] *
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((1<<12) - frac) +
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((1<<12) - frac) +
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