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First audio codec playback attempt by Miikka Pekkarinen
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@6574 a1c6a512-1295-4272-9138-f99709370657
This commit is contained in:
Linus Nielsen Feltzing
parent
b1e1e44041
commit
1c497e6045
20 changed files with 2266 additions and 153 deletions
404
apps/plugins/codecmpa.c
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404
apps/plugins/codecmpa.c
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/***************************************************************************
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* __________ __ ___.
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* Open \______ \ ____ ____ | | _\_ |__ _______ ___
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* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
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* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
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* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
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* \/ \/ \/ \/ \/
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* $Id$
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*
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* Copyright (C) 2005 Dave Chapman
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*
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* All files in this archive are subject to the GNU General Public License.
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* See the file COPYING in the source tree root for full license agreement.
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*
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* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
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* KIND, either express or implied.
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*
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****************************************************************************/
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#include "plugin.h"
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#include <codecs/libmad/mad.h>
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#include "playback.h"
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#include "mp3data.h"
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#include "lib/codeclib.h"
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static struct plugin_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 dither d0, d1;
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/* The following function is used inside libmad - let's hope it's never
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called.
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*/
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void abort(void) {
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}
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/* The "dither" code to convert the 24-bit samples produced by libmad was
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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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};
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# define SAMPLE_DEPTH 16
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# define scale(x, y) dither((x), (y))
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/*
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* NAME: prng()
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* DESCRIPTION: 32-bit pseudo-random number generator
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*/
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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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}
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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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{
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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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/* 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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/* 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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/* 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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/* 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 < 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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/* 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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}
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static __inline
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signed 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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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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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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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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/* quantize */
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output &= ~mask;
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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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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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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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mad_fixed_t mad_frame_overlap[2][32][18] IDATA_ATTR;
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unsigned char mad_main_data[MAD_BUFFER_MDLEN] IDATA_ATTR;
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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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#undef DEBUG_GAPLESS
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/* this is the plugin entry point */
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enum plugin_status plugin_start(struct plugin_api* api, void* parm)
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{
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struct codec_api *ci = (struct codec_api *)parm;
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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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char *InputBuffer;
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unsigned int samplecount;
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unsigned int samplesdone;
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bool first_frame;
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#ifdef DEBUG_GAPLESS
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bool first = true;
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int fd;
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#endif
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int i;
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/* Generic plugin inititialisation */
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TEST_PLUGIN_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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#endif
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/* This function sets up the buffers and reads the file into RAM */
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if (codec_init(api, ci)) {
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return PLUGIN_ERROR;
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}
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/* Create a decoder instance */
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next_track:
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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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mad_stream_init(&Stream);
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mad_frame_init(&Frame);
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mad_synth_init(&Synth);
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mad_timer_reset(&Timer);
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/* We do this so libmad doesn't try to call codec_calloc() */
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memset(mad_frame_overlap, 0, sizeof(mad_frame_overlap));
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Frame.overlap = &mad_frame_overlap;
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Stream.main_data = &mad_main_data;
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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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else
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fd = rb->open("/second.pcm", O_WRONLY | O_CREAT);
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first = false;
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#endif
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samplesdone = 0;
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first_frame = false;
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file_end = 0;
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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->request_buffer(&size, ci->id3->first_frame_offset);
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ci->advance_buffer(size);
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samplecount = ci->id3->length * (ci->id3->frequency / 100) / 10;
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/* rb->snprintf(buf2, sizeof(buf2), "sc: %d", samplecount);
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rb->splash(0, true, buf2);
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rb->snprintf(buf2, sizeof(buf2), "length: %d", ci->id3->length);
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rb->splash(HZ*5, true, buf2);
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rb->snprintf(buf2, sizeof(buf2), "frequency: %d", ci->id3->frequency);
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rb->splash(HZ*5, true, buf2); */
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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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if (ci->stop_codec || ci->reload_codec) {
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break ;
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}
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if (ci->seek_time) {
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unsigned int sample_loc;
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int newpos;
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sample_loc = ci->seek_time/1000 * ci->id3->frequency;
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newpos = ci->mp3_get_filepos(ci->seek_time-1);
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if (ci->seek_buffer(newpos)) {
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ci->seek_time = 0;
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if (sample_loc >= samplecount + samplesdone)
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break ;
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samplecount += samplesdone - sample_loc;
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samplesdone = sample_loc;
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}
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}
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/* Lock buffers */
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if (Stream.error == 0) {
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InputBuffer = ci->request_buffer(&size, INPUT_CHUNK_SIZE);
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if (size == 0 || InputBuffer == NULL)
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break ;
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mad_stream_buffer(&Stream, InputBuffer, size);
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}
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//if ((int)ci->curpos >= ci->id3->first_frame_offset)
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//first_frame = true;
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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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/* 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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/* Fill the buffer */
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Stream.error = 0;
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file_end++;
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continue ;
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}
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else if(MAD_RECOVERABLE(Stream.error))
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{
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if(Stream.error!=MAD_ERROR_LOSTSYNC || Stream.this_frame!=GuardPtr)
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{
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// rb->splash(HZ*1, true, "Recoverable...!");
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}
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continue;
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}
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else if(Stream.error==MAD_ERROR_BUFLEN) {
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//rb->splash(HZ*1, true, "Buflen error");
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break ;
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} else {
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//rb->splash(HZ*1, true, "Unrecoverable error");
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Status=1;
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break;
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}
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}
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if (Stream.next_frame)
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ci->advance_buffer_loc((void *)Stream.next_frame);
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file_end = false;
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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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//samplecount -= Synth.pcm.length;
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//continue ;
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//}
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/* Convert MAD's numbers to an array of 16-bit LE signed integers */
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for(i=0;i<Synth.pcm.length;i++)
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{
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samplesdone++;
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//if (ci->mp3data->padding > 0) {
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// ci->mp3data->padding--;
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// continue ;
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//}
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if (!first_frame) {
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if (detect_silence(Synth.pcm.samples[0][i]))
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continue ;
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first_frame = true;
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}
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/* Left channel */
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Sample=scale(Synth.pcm.samples[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(Synth.pcm.samples[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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#endif
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while (!ci->audiobuffer_insert(OutputBuffer, (int)OutputPtr-(int)OutputBuffer))
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rb->yield();
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goto song_end;
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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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{
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#ifdef DEBUG_GAPLESS
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rb->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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}
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}
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ci->set_elapsed(samplesdone / (ci->id3->frequency/1000));
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}
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song_end:
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#ifdef DEBUG_GAPLESS
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rb->close(fd);
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#endif
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Stream.error = 0;
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if (ci->request_next_track())
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goto next_track;
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return PLUGIN_OK;
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}
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