forked from len0rd/rockbox
Cleaned up code. Now passes full precision samples to the playback engine, and DSP is enabled.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@7620 a1c6a512-1295-4272-9138-f99709370657
This commit is contained in:
Thom Johansen
parent
b91e9fdfa9
commit
6762810ba6
1 changed files with 112 additions and 161 deletions
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@ -18,63 +18,41 @@
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****************************************************************************/
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#include "codec.h"
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#include "lib/codeclib.h"
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#include <inttypes.h> /* Needed by a52.h */
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#include <codecs/liba52/config-a52.h>
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#include <codecs/liba52/a52.h>
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#include "playback.h"
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#include "dsp.h"
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#include "lib/codeclib.h"
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#define BUFFER_SIZE 4096
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struct codec_api* rb;
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struct codec_api* ci;
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struct codec_api *ci;
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static float gain = 1;
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static a52_state_t * state;
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static a52_state_t *state;
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unsigned long samplesdone;
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unsigned long frequency;
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/* Two buffers used outside liba52 */
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/* used outside liba52 */
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static uint8_t buf[3840] IDATA_ATTR;
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static int16_t int16_samples[256*2] IDATA_ATTR;
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static inline int16_t convert (int32_t i)
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void output_audio(sample_t *samples, int flags)
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{
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i >>= 15;
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return (i > 32767) ? 32767 : ((i < -32768) ? -32768 : i);
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}
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void output_audio(sample_t* samples,int flags) {
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int i;
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flags &= A52_CHANNEL_MASK | A52_LFE;
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/* We may need to check the output format in flags - I'm not sure... */
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for (i = 0; i < 256; i++) {
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int16_samples[2*i] = convert (samples[i]);
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int16_samples[2*i+1] = convert (samples[i+256]);
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}
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rb->yield();
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while(!ci->pcmbuf_insert((unsigned char*)int16_samples,256*2*2))
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rb->yield();
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do {
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ci->yield();
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} while (!ci->pcmbuf_insert_split(&samples[0], &samples[256],
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256*sizeof(sample_t)));
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}
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void a52_decode_data (uint8_t * start, uint8_t * end)
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void a52_decode_data(uint8_t *start, uint8_t *end)
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{
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static uint8_t * bufptr = buf;
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static uint8_t * bufpos = buf + 7;
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static uint8_t *bufptr = buf;
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static uint8_t *bufpos = buf + 7;
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/*
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* sample_rate and flags are static because this routine could
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* exit between the a52_syncinfo() and the ao_setup(), and we want
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* to have the same values when we get back !
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*/
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static int sample_rate;
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static int flags;
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int bit_rate;
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@ -86,61 +64,48 @@ void a52_decode_data (uint8_t * start, uint8_t * end)
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break;
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if (len > bufpos - bufptr)
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len = bufpos - bufptr;
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memcpy (bufptr, start, len);
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memcpy(bufptr, start, len);
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bufptr += len;
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start += len;
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if (bufptr == bufpos) {
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if (bufpos == buf + 7) {
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int length;
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length = a52_syncinfo (buf, &flags, &sample_rate, &bit_rate);
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length = a52_syncinfo(buf, &flags, &sample_rate, &bit_rate);
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if (!length) {
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DEBUGF("skip\n");
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//DEBUGF("skip\n");
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for (bufptr = buf; bufptr < buf + 6; bufptr++)
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bufptr[0] = bufptr[1];
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continue;
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}
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bufpos = buf + length;
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} else {
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// The following two defaults are taken from audio_out_oss.c:
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level_t level;
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sample_t bias;
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/* The following two defaults are taken from audio_out_oss.c: */
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level_t level = 1 << 26;
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sample_t bias = 0;
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int i;
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/* This is the configuration for the downmixing: */
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flags=A52_STEREO|A52_ADJUST_LEVEL|A52_LFE;
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level=(1 << 26);
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bias=0;
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flags = A52_STEREO | A52_ADJUST_LEVEL | A52_LFE;
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level = (level_t) (level * gain);
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if (a52_frame (state, buf, &flags, &level, bias)) {
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if (a52_frame(state, buf, &flags, &level, bias))
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goto error;
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}
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// file_info->frames_decoded++;
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frequency = sample_rate;
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// /* We assume this never changes */
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// file_info->samplerate=sample_rate;
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frequency=sample_rate;
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// An A52 frame consists of 6 blocks of 256 samples
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// So we decode and output them one block at a time
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/* An A52 frame consists of 6 blocks of 256 samples
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So we decode and output them one block at a time */
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for (i = 0; i < 6; i++) {
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if (a52_block (state)) {
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if (a52_block(state))
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goto error;
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}
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output_audio(a52_samples (state),flags);
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samplesdone+=256;
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output_audio(a52_samples(state), flags);
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samplesdone += 256;
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}
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ci->set_elapsed(samplesdone/(frequency/1000));
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bufptr = buf;
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bufpos = buf + 7;
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continue;
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error:
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//logf("Error decoding A52 stream\n");
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bufptr = buf;
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bufpos = buf + 7;
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@ -149,77 +114,63 @@ void a52_decode_data (uint8_t * start, uint8_t * end)
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}
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}
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#ifndef SIMULATOR
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#ifdef USE_IRAM
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extern char iramcopy[];
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extern char iramstart[];
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extern char iramend[];
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#endif
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/* this is the codec entry point */
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enum codec_status codec_start(struct codec_api* api)
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enum codec_status codec_start(struct codec_api *api)
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{
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size_t n;
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unsigned char* filebuf;
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unsigned char *filebuf;
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/* Generic codec initialisation */
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TEST_CODEC_API(api);
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ci = api;
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rb = api;
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ci = (struct codec_api*)api;
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#ifndef SIMULATOR
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rb->memcpy(iramstart, iramcopy, iramend-iramstart);
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#endif
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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*128));
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#ifdef USE_IRAM
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ci->memcpy(iramstart, iramcopy, iramend - iramstart);
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#endif
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ci->configure(CODEC_DSP_ENABLE, (bool *)true);
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ci->configure(DSP_DITHER, (bool *)false);
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ci->configure(DSP_SET_STEREO_MODE, (int *)STEREO_INTERLEAVED);
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ci->configure(DSP_SET_SAMPLE_DEPTH, (int *)(16));
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ci->configure(DSP_SET_STEREO_MODE, (long *)STEREO_NONINTERLEAVED);
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ci->configure(DSP_SET_SAMPLE_DEPTH, (long *)30);
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ci->configure(DSP_SET_CLIP_MAX, (long *)((1 << 30) - 1));
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ci->configure(DSP_SET_CLIP_MIN, (long *)-(1 << 30));
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ci->configure(CODEC_SET_FILEBUF_LIMIT, (long *)(1024*1024*2));
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ci->configure(CODEC_SET_FILEBUF_CHUNKSIZE, (long *)(1024*128));
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next_track:
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if (codec_init(api)) {
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next_track:
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if (codec_init(api))
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return CODEC_ERROR;
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}
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while (!rb->taginfo_ready)
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rb->yield();
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while (!ci->taginfo_ready)
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ci->yield();
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if (rb->id3->frequency != NATIVE_FREQUENCY) {
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rb->configure(DSP_SET_FREQUENCY, (long *)(rb->id3->frequency));
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rb->configure(CODEC_DSP_ENABLE, (bool *)true);
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} else {
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rb->configure(CODEC_DSP_ENABLE, (bool *)false);
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}
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ci->configure(DSP_SET_FREQUENCY, (long *)(ci->id3->frequency));
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/* Intialise the A52 decoder and check for success */
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state = a52_init (0); // Parameter is "accel"
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state = a52_init(0);
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/* The main decoding loop */
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samplesdone=0;
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samplesdone = 0;
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while (1) {
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if (ci->stop_codec || ci->reload_codec) {
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if (ci->stop_codec || ci->reload_codec)
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break;
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}
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filebuf=ci->request_buffer(&n,BUFFER_SIZE);
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filebuf = ci->request_buffer(&n, BUFFER_SIZE);
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if (n==0) { /* End of Stream */
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if (n == 0) /* End of Stream */
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break;
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}
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a52_decode_data(filebuf,filebuf+n);
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a52_decode_data(filebuf, filebuf + n);
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ci->advance_buffer(n);
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}
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if (ci->request_next_track())
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goto next_track;
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//NOT NEEDED??: a52_free (state);
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a52_free(state);
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return CODEC_OK;
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}
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