forked from len0rd/rockbox
Cleaned up code a bit, removed all rb references.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@6825 a1c6a512-1295-4272-9138-f99709370657
This commit is contained in:
Thom Johansen
parent
54c1c667de
commit
10426dbe22
1 changed files with 103 additions and 115 deletions
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@ -25,12 +25,10 @@
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#include "mp3data.h"
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#include "lib/codeclib.h"
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static struct codec_api* rb;
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struct mad_stream Stream IDATA_ATTR;
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struct mad_frame Frame IDATA_ATTR;
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struct mad_synth Synth IDATA_ATTR;
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mad_timer_t Timer;
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struct mad_stream Stream IDATA_ATTR;
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struct mad_frame Frame IDATA_ATTR;
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struct mad_synth Synth IDATA_ATTR;
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mad_timer_t Timer;
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struct dither d0, d1;
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/* The following function is used inside libmad - let's hope it's never
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@ -44,8 +42,8 @@ void abort(void) {
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taken from the coolplayer project - coolplayer.sourceforge.net */
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struct dither {
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mad_fixed_t error[3];
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mad_fixed_t random;
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mad_fixed_t error[3];
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mad_fixed_t random;
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};
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# define SAMPLE_DEPTH 16
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@ -58,117 +56,112 @@ struct dither {
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static __inline
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unsigned long prng(unsigned long state)
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{
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return (state * 0x0019660dL + 0x3c6ef35fL) & 0xffffffffL;
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return (state * 0x0019660dL + 0x3c6ef35fL) & 0xffffffffL;
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}
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/*
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* NAME: dither()
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* DESCRIPTION: dither and scale sample
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*/
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static __inline
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signed int dither(mad_fixed_t sample, struct dither *dither)
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inline int dither(mad_fixed_t sample, struct dither *dither)
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{
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unsigned int scalebits;
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mad_fixed_t output, mask, random;
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unsigned int scalebits;
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mad_fixed_t output, mask, random;
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enum {
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MIN = -MAD_F_ONE,
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MAX = MAD_F_ONE - 1
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};
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enum {
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MIN = -MAD_F_ONE,
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MAX = MAD_F_ONE - 1
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};
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/* noise shape */
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sample += dither->error[0] - dither->error[1] + dither->error[2];
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/* noise shape */
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sample += dither->error[0] - dither->error[1] + dither->error[2];
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dither->error[2] = dither->error[1];
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dither->error[1] = dither->error[0] / 2;
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dither->error[2] = dither->error[1];
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dither->error[1] = dither->error[0]/2;
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/* bias */
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output = sample + (1L << (MAD_F_FRACBITS + 1 - SAMPLE_DEPTH - 1));
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/* bias */
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output = sample + (1L << (MAD_F_FRACBITS + 1 - SAMPLE_DEPTH - 1));
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scalebits = MAD_F_FRACBITS + 1 - SAMPLE_DEPTH;
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mask = (1L << scalebits) - 1;
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scalebits = MAD_F_FRACBITS + 1 - SAMPLE_DEPTH;
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mask = (1L << scalebits) - 1;
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/* dither */
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random = prng(dither->random);
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output += (random & mask) - (dither->random & mask);
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/* dither */
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random = prng(dither->random);
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output += (random & mask) - (dither->random & mask);
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//dither->random = random;
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//dither->random = random;
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/* clip */
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if (output > MAX) {
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output = MAX;
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/* clip */
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if (output > MAX) {
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output = MAX;
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if (sample > MAX)
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sample = MAX;
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} else if (output < MIN) {
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output = MIN;
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if (sample > MAX)
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sample = MAX;
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}
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else if (output < MIN) {
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output = MIN;
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if (sample < MIN)
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sample = MIN;
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}
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if (sample < MIN)
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sample = MIN;
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}
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/* quantize */
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output &= ~mask;
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/* quantize */
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output &= ~mask;
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/* error feedback */
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dither->error[0] = sample - output;
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/* error feedback */
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dither->error[0] = sample - output;
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/* scale */
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return output >> scalebits;
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/* scale */
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return output >> scalebits;
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}
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static __inline
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signed int detect_silence(mad_fixed_t sample)
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inline int detect_silence(mad_fixed_t sample)
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{
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unsigned int scalebits;
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mad_fixed_t output, mask;
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unsigned int scalebits;
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mad_fixed_t output, mask;
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enum {
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MIN = -MAD_F_ONE,
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MAX = MAD_F_ONE - 1
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};
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enum {
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MIN = -MAD_F_ONE,
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MAX = MAD_F_ONE - 1
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};
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/* bias */
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output = sample + (1L << (MAD_F_FRACBITS + 1 - SAMPLE_DEPTH - 1));
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/* bias */
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output = sample + (1L << (MAD_F_FRACBITS + 1 - SAMPLE_DEPTH - 1));
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scalebits = MAD_F_FRACBITS + 1 - SAMPLE_DEPTH;
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mask = (1L << scalebits) - 1;
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scalebits = MAD_F_FRACBITS + 1 - SAMPLE_DEPTH;
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mask = (1L << scalebits) - 1;
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/* clip */
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if (output > MAX) {
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output = MAX;
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/* clip */
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if (output > MAX) {
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output = MAX;
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if (sample > MAX)
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sample = MAX;
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}
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else if (output < MIN) {
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output = MIN;
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if (sample > MAX)
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sample = MAX;
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} else if (output < MIN) {
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output = MIN;
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if (sample < MIN)
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sample = MIN;
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}
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if (sample < MIN)
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sample = MIN;
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}
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/* quantize */
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output &= ~mask;
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/* quantize */
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output &= ~mask;
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/* scale */
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output >>= scalebits + 4;
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/* scale */
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output >>= scalebits + 4;
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if (output == 0x00 || output == 0xff)
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return 1;
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if (output == 0x00 || output == 0xff)
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return 1;
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return 0;
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return 0;
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}
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#define SHRT_MAX 32767
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#define INPUT_CHUNK_SIZE 8192
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#define OUTPUT_BUFFER_SIZE 65536 /* Must be an integer multiple of 4. */
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#define INPUT_CHUNK_SIZE 8192
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#define OUTPUT_BUFFER_SIZE 65536 /* Must be an integer multiple of 4. */
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unsigned char OutputBuffer[OUTPUT_BUFFER_SIZE];
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unsigned char *OutputPtr;
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unsigned char *GuardPtr=NULL;
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const unsigned char *OutputBufferEnd=OutputBuffer+OUTPUT_BUFFER_SIZE;
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unsigned char *GuardPtr = NULL;
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const unsigned char *OutputBufferEnd = OutputBuffer + OUTPUT_BUFFER_SIZE;
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long resampled_data[2][5000]; /* enough to cope with 11khz upsampling */
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mad_fixed_t mad_frame_overlap[2][32][18] IDATA_ATTR;
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@ -261,7 +254,7 @@ enum codec_status codec_start(struct codec_api* api)
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{
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struct codec_api *ci = api;
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struct mp3info *info;
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int Status=0;
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int Status = 0;
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size_t size;
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int file_end;
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unsigned short Sample;
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@ -281,10 +274,9 @@ enum codec_status codec_start(struct codec_api* api)
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/* Generic codec inititialisation */
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TEST_CODEC_API(api);
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rb = api;
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#ifdef USE_IRAM
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rb->memcpy(iramstart, iramcopy, iramend-iramstart);
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ci->memcpy(iramstart, iramcopy, iramend - iramstart);
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#endif
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/* This function sets up the buffers and reads the file into RAM */
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@ -298,10 +290,10 @@ enum codec_status codec_start(struct codec_api* api)
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ci->configure(CODEC_SET_FILEBUF_LIMIT, (int *)(1024*1024*2));
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ci->configure(CODEC_SET_FILEBUF_CHUNKSIZE, (int *)(1024*16));
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memset(&Stream, 0, sizeof(struct mad_stream));
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memset(&Frame, 0, sizeof(struct mad_frame));
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memset(&Synth, 0, sizeof(struct mad_synth));
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memset(&Timer, 0, sizeof(mad_timer_t));
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ci->memset(&Stream, 0, sizeof(struct mad_stream));
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ci->memset(&Frame, 0, sizeof(struct mad_frame));
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ci->memset(&Synth, 0, sizeof(struct mad_synth));
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ci->memset(&Timer, 0, sizeof(mad_timer_t));
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mad_stream_init(&Stream);
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mad_frame_init(&Frame);
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@ -319,9 +311,9 @@ enum codec_status codec_start(struct codec_api* api)
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#ifdef DEBUG_GAPLESS
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if (first)
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fd = rb->open("/first.pcm", O_WRONLY | O_CREAT);
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fd = ci->open("/first.pcm", O_WRONLY | O_CREAT);
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else
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fd = rb->open("/second.pcm", O_WRONLY | O_CREAT);
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fd = ci->open("/second.pcm", O_WRONLY | O_CREAT);
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first = false;
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#endif
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@ -331,7 +323,7 @@ enum codec_status codec_start(struct codec_api* api)
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OutputPtr = OutputBuffer;
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while (!*ci->taginfo_ready)
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rb->yield();
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ci->yield();
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ci->request_buffer(&size, ci->id3->first_frame_offset);
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ci->advance_buffer(size);
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@ -367,7 +359,7 @@ enum codec_status codec_start(struct codec_api* api)
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lr.delta = rr.delta = ci->id3->frequency*65536/44100;
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/* This is the decoding loop. */
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while (1) {
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rb->yield();
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ci->yield();
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if (ci->stop_codec || ci->reload_codec) {
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break ;
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}
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@ -401,7 +393,7 @@ enum codec_status codec_start(struct codec_api* api)
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if(mad_frame_decode(&Frame,&Stream))
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{
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if (Stream.error == MAD_FLAG_INCOMPLETE || Stream.error == MAD_ERROR_BUFLEN) {
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// rb->splash(HZ*1, true, "Incomplete");
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// ci->splash(HZ*1, true, "Incomplete");
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/* This makes the codec to support partially corrupted files too. */
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if (file_end == 30)
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break ;
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@ -434,10 +426,6 @@ enum codec_status codec_start(struct codec_api* api)
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/* ?? Do we need the timer module? */
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// mad_timer_add(&Timer,Frame.header.duration);
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/* DAVE: This can be used to attenuate the audio */
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// if(DoFilter)
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// ApplyFilter(&Frame);
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mad_synth_frame(&Synth,&Frame);
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//if (!first_frame) {
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@ -452,7 +440,7 @@ enum codec_status codec_start(struct codec_api* api)
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length = resample((long *)&Synth.pcm.samples[0][start_skip], resampled_data[0], Synth.pcm.length, &lr);
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if (MAD_NCHANNELS(&Frame.header) == 2)
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resample((long *)&Synth.pcm.samples[1][start_skip], resampled_data[1], Synth.pcm.length, &rr);
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for (i = 0;i<length;i++)
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for (i = 0; i < length; i++)
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{
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start_skip = 0; /* not very elegant, and might want to keep this value */
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samplesdone++;
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@ -467,42 +455,42 @@ enum codec_status codec_start(struct codec_api* api)
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}*/
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/* Left channel */
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Sample=scale(resampled_data[0][i],&d0);
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*(OutputPtr++)=Sample>>8;
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*(OutputPtr++)=Sample&0xff;
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Sample = scale(resampled_data[0][i], &d0);
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*(OutputPtr++) = Sample >> 8;
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*(OutputPtr++) = Sample & 0xff;
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/* Right channel. If the decoded stream is monophonic then
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* the right output channel is the same as the left one.
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*/
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if(MAD_NCHANNELS(&Frame.header)==2)
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Sample=scale(resampled_data[1][i],&d1);
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*(OutputPtr++)=Sample>>8;
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*(OutputPtr++)=Sample&0xff;
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if (MAD_NCHANNELS(&Frame.header) == 2)
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Sample = scale(resampled_data[1][i], &d1);
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*(OutputPtr++) = Sample >> 8;
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*(OutputPtr++) = Sample & 0xff;
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samplecount--;
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if (samplecount == 0) {
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#ifdef DEBUG_GAPLESS
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rb->write(fd, OutputBuffer, (int)OutputPtr-(int)OutputBuffer);
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ci->write(fd, OutputBuffer, (int)OutputPtr - (int)OutputBuffer);
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#endif
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while (!ci->audiobuffer_insert(OutputBuffer, (int)OutputPtr-(int)OutputBuffer))
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rb->yield();
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while (!ci->audiobuffer_insert(OutputBuffer, (int)OutputPtr - (int)OutputBuffer))
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ci->yield();
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goto song_end;
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}
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if (yieldcounter++ == 200) {
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rb->yield();
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ci->yield();
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yieldcounter = 0;
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}
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/* Flush the buffer if it is full. */
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if(OutputPtr==OutputBufferEnd)
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if (OutputPtr == OutputBufferEnd)
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{
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#ifdef DEBUG_GAPLESS
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rb->write(fd, OutputBuffer, OUTPUT_BUFFER_SIZE);
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ci->write(fd, OutputBuffer, OUTPUT_BUFFER_SIZE);
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#endif
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while (!ci->audiobuffer_insert(OutputBuffer, OUTPUT_BUFFER_SIZE))
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rb->yield();
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OutputPtr=OutputBuffer;
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ci->yield();
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OutputPtr = OutputBuffer;
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}
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}
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ci->set_elapsed(samplesdone / (ci->id3->frequency/1000));
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@ -510,7 +498,7 @@ enum codec_status codec_start(struct codec_api* api)
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song_end:
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#ifdef DEBUG_GAPLESS
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rb->close(fd);
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ci->close(fd);
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#endif
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Stream.error = 0;
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