4392506d59
Don't crossfade if audio is paused when selecting a new track. git-svn-id: svn://svn.rockbox.org/rockbox/trunk@7711 a1c6a512-1295-4272-9138-f99709370657
2325 lines
63 KiB
C
2325 lines
63 KiB
C
/***************************************************************************
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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 Miika Pekkarinen
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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 <stdio.h>
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#include <string.h>
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#include <stdlib.h>
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#include <ctype.h>
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#include "system.h"
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#include "thread.h"
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#include "file.h"
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#include "lcd.h"
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#include "font.h"
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#include "backlight.h"
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#include "button.h"
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#include "kernel.h"
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#include "tree.h"
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#include "debug.h"
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#include "sprintf.h"
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#include "settings.h"
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#include "codecs.h"
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#include "wps.h"
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#include "wps-display.h"
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#include "audio.h"
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#include "logf.h"
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#include "mp3_playback.h"
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#include "usb.h"
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#include "status.h"
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#include "main_menu.h"
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#include "ata.h"
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#include "screens.h"
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#include "playlist.h"
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#include "playback.h"
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#include "pcmbuf.h"
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#include "pcm_playback.h"
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#include "buffer.h"
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#include "dsp.h"
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#ifdef HAVE_LCD_BITMAP
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#include "icons.h"
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#include "peakmeter.h"
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#include "action.h"
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#endif
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#include "lang.h"
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#include "bookmark.h"
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#include "misc.h"
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#include "sound.h"
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#include "metadata.h"
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#include "talk.h"
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static volatile bool audio_codec_loaded;
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static volatile bool voice_codec_loaded;
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static volatile bool playing;
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static volatile bool paused;
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#define CODEC_VORBIS "/.rockbox/codecs/vorbis.codec"
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#define CODEC_MPA_L3 "/.rockbox/codecs/mpa.codec"
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#define CODEC_FLAC "/.rockbox/codecs/flac.codec"
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#define CODEC_WAV "/.rockbox/codecs/wav.codec"
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#define CODEC_A52 "/.rockbox/codecs/a52.codec"
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#define CODEC_MPC "/.rockbox/codecs/mpc.codec"
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#define CODEC_WAVPACK "/.rockbox/codecs/wavpack.codec"
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#define CODEC_ALAC "/.rockbox/codecs/alac.codec"
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#define CODEC_AAC "/.rockbox/codecs/aac.codec"
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#define AUDIO_FILL_CYCLE (1024*256)
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#define AUDIO_DEFAULT_WATERMARK (1024*512)
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#define AUDIO_DEFAULT_FILECHUNK (1024*32)
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#define AUDIO_PLAY 1
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#define AUDIO_STOP 2
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#define AUDIO_PAUSE 3
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#define AUDIO_RESUME 4
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#define AUDIO_NEXT 5
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#define AUDIO_PREV 6
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#define AUDIO_FF_REWIND 7
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#define AUDIO_FLUSH_RELOAD 8
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#define AUDIO_CODEC_DONE 9
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#define AUDIO_FLUSH 10
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#define AUDIO_TRACK_CHANGED 11
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#define CODEC_LOAD 1
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#define CODEC_LOAD_DISK 2
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/* As defined in plugins/lib/xxx2wav.h */
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#define MALLOC_BUFSIZE (512*1024)
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#define GUARD_BUFSIZE (32*1024)
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/* As defined in plugin.lds */
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#define CODEC_IRAM_ORIGIN 0x1000c000
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#define CODEC_IRAM_SIZE 0xc000
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extern bool audio_is_initialized;
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/* Buffer control thread. */
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static struct event_queue audio_queue;
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static long audio_stack[(DEFAULT_STACK_SIZE + 0x1000)/sizeof(long)];
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static const char audio_thread_name[] = "audio";
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/* Codec thread. */
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static struct event_queue codec_queue;
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static long codec_stack[(DEFAULT_STACK_SIZE + 0x2000)/sizeof(long)] IBSS_ATTR;
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static const char codec_thread_name[] = "codec";
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/* Voice codec thread. */
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static struct event_queue voice_codec_queue;
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/* Not enough IRAM for this. */
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static long voice_codec_stack[(DEFAULT_STACK_SIZE + 0x2000)/sizeof(long)] IBSS_ATTR;
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static const char voice_codec_thread_name[] = "voice codec";
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static struct mutex mutex_bufferfill;
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static struct mutex mutex_codecthread;
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static struct mp3entry id3_voice;
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#define CODEC_IDX_AUDIO 0
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#define CODEC_IDX_VOICE 1
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static char *voicebuf;
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static int voice_remaining;
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static bool voice_is_playing;
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static void (*voice_getmore)(unsigned char** start, int* size);
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/* Is file buffer currently being refilled? */
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static volatile bool filling;
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volatile int current_codec;
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extern unsigned char codecbuf[];
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/* Ring buffer where tracks and codecs are loaded. */
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static char *filebuf;
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/* Total size of the ring buffer. */
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int filebuflen;
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/* Bytes available in the buffer. */
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int filebufused;
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/* Ring buffer read and write indexes. */
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static volatile int buf_ridx;
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static volatile int buf_widx;
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/* Step count to the next unbuffered track. */
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static int last_peek_offset;
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/* Index of the last buffered track. */
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static int last_index;
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/* Track information (count in file buffer, read/write indexes for
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track ring structure. */
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int track_count;
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static volatile int track_ridx;
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static volatile int track_widx;
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static bool track_changed;
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/* Partially loaded song's file handle to continue buffering later. */
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static int current_fd;
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/* Information about how many bytes left on the buffer re-fill run. */
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static long fill_bytesleft;
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/* Track info structure about songs in the file buffer. */
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static struct track_info tracks[MAX_TRACK];
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/* Pointer to track info structure about current song playing. */
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static struct track_info *cur_ti;
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/* Codec API including function callbacks. */
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extern struct codec_api ci;
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extern struct codec_api ci_voice;
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/* When we change a song and buffer is not in filling state, this
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variable keeps information about whether to go a next/previous track. */
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static int new_track;
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/* Callback function to call when current track has really changed. */
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void (*track_changed_callback)(struct mp3entry *id3);
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void (*track_buffer_callback)(struct mp3entry *id3, bool last_track);
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void (*track_unbuffer_callback)(struct mp3entry *id3, bool last_track);
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/* Configuration */
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static int conf_bufferlimit;
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static int conf_watermark;
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static int conf_filechunk;
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static int buffer_margin;
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static bool v1first = false;
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static void mp3_set_elapsed(struct mp3entry* id3);
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int mp3_get_file_pos(void);
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static void do_swap(int idx_old, int idx_new)
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{
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#ifndef SIMULATOR
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unsigned char *iram_p = (unsigned char *)(CODEC_IRAM_ORIGIN);
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unsigned char *iram_buf[2];
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#endif
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unsigned char *dram_buf[2];
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#ifndef SIMULATOR
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iram_buf[0] = &filebuf[filebuflen];
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iram_buf[1] = &filebuf[filebuflen+CODEC_IRAM_SIZE];
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memcpy(iram_buf[idx_old], iram_p, CODEC_IRAM_SIZE);
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memcpy(iram_p, iram_buf[idx_new], CODEC_IRAM_SIZE);
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#endif
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dram_buf[0] = &filebuf[filebuflen+CODEC_IRAM_SIZE*2];
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dram_buf[1] = &filebuf[filebuflen+CODEC_IRAM_SIZE*2+CODEC_SIZE];
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memcpy(dram_buf[idx_old], codecbuf, CODEC_SIZE);
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memcpy(codecbuf, dram_buf[idx_new], CODEC_SIZE);
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}
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static void swap_codec(void)
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{
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int last_codec;
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logf("swapping codec:%d", current_codec);
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/* We should swap codecs' IRAM contents and code space. */
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do_swap(current_codec, !current_codec);
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last_codec = current_codec;
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current_codec = !current_codec;
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/* Release the semaphore and force a task switch. */
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mutex_unlock(&mutex_codecthread);
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sleep(1);
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/* Waiting until we are ready to run again. */
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mutex_lock(&mutex_codecthread);
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/* Check if codec swap did not happen. */
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if (current_codec != last_codec)
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{
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logf("no codec switch happened!");
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do_swap(current_codec, !current_codec);
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current_codec = !current_codec;
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}
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invalidate_icache();
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logf("codec resuming:%d", current_codec);
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}
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#ifdef HAVE_ADJUSTABLE_CPU_FREQ
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static void voice_boost_cpu(bool state)
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{
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static bool voice_cpu_boosted = false;
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if (state != voice_cpu_boosted)
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{
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cpu_boost(state);
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voice_cpu_boosted = state;
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}
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}
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#else
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#define voice_boost_cpu(state) do { } while(0)
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#endif
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bool codec_pcmbuf_insert_split_callback(void *ch1, void *ch2,
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long length)
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{
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char* src[2];
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char *dest;
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long input_size;
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long output_size;
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src[0] = ch1;
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src[1] = ch2;
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while (paused)
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{
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if (pcm_is_playing())
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pcm_play_pause(false);
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sleep(1);
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if (ci.stop_codec || ci.reload_codec || ci.seek_time)
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return true;
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}
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if (dsp_stereo_mode() == STEREO_NONINTERLEAVED)
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{
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length *= 2; /* Length is per channel */
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}
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while (length > 0) {
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/* This will prevent old audio from playing when skipping tracks. */
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if ((ci.reload_codec || ci.stop_codec) && current_codec != CODEC_IDX_VOICE)
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return true;
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while ((dest = pcmbuf_request_buffer(dsp_output_size(length),
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&output_size)) == NULL) {
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sleep(1);
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if ((ci.reload_codec || ci.stop_codec) && current_codec != CODEC_IDX_VOICE)
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return true;
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}
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/* Get the real input_size for output_size bytes, guarding
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* against resampling buffer overflows. */
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input_size = dsp_input_size(output_size);
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if (input_size > length) {
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DEBUGF("Error: dsp_input_size(%ld=dsp_output_size(%ld))=%ld > %ld\n",
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output_size, length, input_size, length);
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input_size = length;
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}
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if (input_size <= 0) {
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pcmbuf_flush_buffer(0);
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DEBUGF("Warning: dsp_input_size(%ld=dsp_output_size(%ld))=%ld <= 0\n",
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output_size, length, input_size);
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/* should we really continue, or should we break?
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* We should probably continue because calling pcmbuf_flush_buffer(0)
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* will wrap the buffer if it was fully filled and so next call to
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* pcmbuf_request_buffer should give the requested output_size. */
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continue;
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}
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output_size = dsp_process(dest, src, input_size);
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/* Hotswap between audio and voice codecs as necessary. */
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switch (current_codec)
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{
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case CODEC_IDX_AUDIO:
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pcmbuf_flush_buffer(output_size);
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if (voice_is_playing && pcmbuf_usage() > 30
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&& pcmbuf_mix_usage() < 20)
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{
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voice_boost_cpu(true);
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swap_codec();
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voice_boost_cpu(false);
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}
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break ;
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case CODEC_IDX_VOICE:
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if (audio_codec_loaded) {
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pcmbuf_mix(dest, output_size);
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if ((pcmbuf_usage() < 10)
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|| pcmbuf_mix_usage() > 70)
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swap_codec();
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} else {
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pcmbuf_flush_buffer(output_size);
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}
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break ;
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}
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length -= input_size;
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}
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return true;
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}
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bool codec_pcmbuf_insert_callback(char *buf, long length)
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{
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/* TODO: The audiobuffer API should probably be updated, and be based on
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* pcmbuf_insert_split().
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*/
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long real_length = length;
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if (dsp_stereo_mode() == STEREO_NONINTERLEAVED)
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{
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length /= 2; /* Length is per channel */
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}
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/* Second channel is only used for non-interleaved stereo. */
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return codec_pcmbuf_insert_split_callback(buf, buf + (real_length / 2),
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length);
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}
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void* get_codec_memory_callback(long *size)
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{
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*size = MALLOC_BUFSIZE;
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if (voice_codec_loaded)
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return &audiobuf[talk_get_bufsize()];
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return &audiobuf[0];
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}
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void codec_set_elapsed_callback(unsigned int value)
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{
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unsigned int latency;
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if (ci.stop_codec || current_codec == CODEC_IDX_VOICE)
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return ;
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latency = pcmbuf_get_latency();
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if (value < latency) {
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cur_ti->id3.elapsed = 0;
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} else if (value - latency > cur_ti->id3.elapsed
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|| value - latency < cur_ti->id3.elapsed - 2) {
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cur_ti->id3.elapsed = value - latency;
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}
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}
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void codec_set_offset_callback(unsigned int value)
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{
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unsigned int latency;
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if (ci.stop_codec || current_codec == CODEC_IDX_VOICE)
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return ;
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latency = pcmbuf_get_latency() * cur_ti->id3.bitrate / 8;
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if (value < latency) {
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cur_ti->id3.offset = 0;
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} else {
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cur_ti->id3.offset = value - latency;
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}
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}
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long codec_filebuf_callback(void *ptr, long size)
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{
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char *buf = (char *)ptr;
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int copy_n;
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int part_n;
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if (ci.stop_codec || !playing || current_codec == CODEC_IDX_VOICE)
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return 0;
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copy_n = MIN((off_t)size, (off_t)cur_ti->available + cur_ti->filerem);
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while (copy_n > cur_ti->available) {
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yield();
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if (ci.stop_codec || ci.reload_codec)
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return 0;
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}
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if (copy_n == 0)
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return 0;
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part_n = MIN(copy_n, filebuflen - buf_ridx);
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memcpy(buf, &filebuf[buf_ridx], part_n);
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if (part_n < copy_n) {
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memcpy(&buf[part_n], &filebuf[0], copy_n - part_n);
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}
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buf_ridx += copy_n;
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if (buf_ridx >= filebuflen)
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buf_ridx -= filebuflen;
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ci.curpos += copy_n;
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cur_ti->available -= copy_n;
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filebufused -= copy_n;
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return copy_n;
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}
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void* voice_request_data(long *realsize, long reqsize)
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{
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while (queue_empty(&voice_codec_queue) && (voice_remaining == 0
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|| voicebuf == NULL) && !ci_voice.stop_codec)
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{
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yield();
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if (audio_codec_loaded && (pcmbuf_usage() < 30
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|| !voice_is_playing || voicebuf == NULL))
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{
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swap_codec();
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}
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else if (!voice_is_playing)
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{
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voice_boost_cpu(false);
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if (!pcm_is_playing())
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pcmbuf_boost(false);
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sleep(HZ/16);
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}
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if (voice_remaining)
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{
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voice_is_playing = true;
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break ;
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}
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if (voice_getmore != NULL)
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{
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voice_getmore((unsigned char **)&voicebuf, (int *)&voice_remaining);
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if (!voice_remaining)
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{
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voice_is_playing = false;
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/* Force pcm playback. */
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pcmbuf_play_start();
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}
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}
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}
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if (reqsize < 0)
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reqsize = 0;
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voice_is_playing = true;
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*realsize = voice_remaining;
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if (*realsize > reqsize)
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*realsize = reqsize;
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if (*realsize == 0)
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return NULL;
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return voicebuf;
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}
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void* codec_request_buffer_callback(long *realsize, long reqsize)
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{
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long part_n;
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/* Voice codec. */
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if (current_codec == CODEC_IDX_VOICE) {
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return voice_request_data(realsize, reqsize);
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}
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if (ci.stop_codec || !playing) {
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*realsize = 0;
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return NULL;
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}
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*realsize = MIN((off_t)reqsize, (off_t)cur_ti->available + cur_ti->filerem);
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if (*realsize == 0) {
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return NULL;
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}
|
|
|
|
while ((int)*realsize > cur_ti->available) {
|
|
yield();
|
|
if (ci.stop_codec || ci.reload_codec) {
|
|
*realsize = 0;
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
part_n = MIN((int)*realsize, filebuflen - buf_ridx);
|
|
if (part_n < *realsize) {
|
|
part_n += GUARD_BUFSIZE;
|
|
if (part_n < *realsize)
|
|
*realsize = part_n;
|
|
memcpy(&filebuf[filebuflen], &filebuf[0], *realsize -
|
|
(filebuflen - buf_ridx));
|
|
}
|
|
|
|
return (char *)&filebuf[buf_ridx];
|
|
}
|
|
|
|
static bool rebuffer_and_seek(int newpos)
|
|
{
|
|
int fd;
|
|
|
|
logf("Re-buffering song");
|
|
mutex_lock(&mutex_bufferfill);
|
|
|
|
/* (Re-)open current track's file handle. */
|
|
fd = open(playlist_peek(0), O_RDONLY);
|
|
if (fd < 0) {
|
|
logf("Open failed!");
|
|
mutex_unlock(&mutex_bufferfill);
|
|
return false;
|
|
}
|
|
if (current_fd >= 0)
|
|
close(current_fd);
|
|
current_fd = fd;
|
|
|
|
/* Clear codec buffer. */
|
|
audio_invalidate_tracks();
|
|
filebufused = 0;
|
|
buf_ridx = buf_widx = 0;
|
|
cur_ti->filerem = cur_ti->filesize - newpos;
|
|
cur_ti->filepos = newpos;
|
|
cur_ti->start_pos = newpos;
|
|
ci.curpos = newpos;
|
|
cur_ti->available = 0;
|
|
lseek(current_fd, newpos, SEEK_SET);
|
|
|
|
mutex_unlock(&mutex_bufferfill);
|
|
|
|
while (cur_ti->available == 0 && cur_ti->filerem > 0) {
|
|
yield();
|
|
if (ci.stop_codec || ci.reload_codec)
|
|
return false;
|
|
}
|
|
|
|
return true;
|
|
}
|
|
|
|
void codec_advance_buffer_callback(long amount)
|
|
{
|
|
if (current_codec == CODEC_IDX_VOICE) {
|
|
//logf("voice ad.buf:%d", amount);
|
|
amount = MAX(0, MIN(amount, voice_remaining));
|
|
voicebuf += amount;
|
|
voice_remaining -= amount;
|
|
|
|
return ;
|
|
}
|
|
|
|
if (amount > cur_ti->available + cur_ti->filerem)
|
|
amount = cur_ti->available + cur_ti->filerem;
|
|
|
|
if (amount > cur_ti->available) {
|
|
if (!rebuffer_and_seek(ci.curpos + amount))
|
|
ci.stop_codec = true;
|
|
return ;
|
|
}
|
|
|
|
buf_ridx += amount;
|
|
if (buf_ridx >= filebuflen)
|
|
buf_ridx -= filebuflen;
|
|
cur_ti->available -= amount;
|
|
filebufused -= amount;
|
|
ci.curpos += amount;
|
|
codec_set_offset_callback(ci.curpos);
|
|
}
|
|
|
|
void codec_advance_buffer_loc_callback(void *ptr)
|
|
{
|
|
long amount;
|
|
|
|
if (current_codec == CODEC_IDX_VOICE)
|
|
amount = (int)ptr - (int)voicebuf;
|
|
else
|
|
amount = (int)ptr - (int)&filebuf[buf_ridx];
|
|
codec_advance_buffer_callback(amount);
|
|
}
|
|
|
|
off_t codec_mp3_get_filepos_callback(int newtime)
|
|
{
|
|
off_t newpos;
|
|
|
|
cur_ti->id3.elapsed = newtime;
|
|
newpos = mp3_get_file_pos();
|
|
|
|
return newpos;
|
|
}
|
|
|
|
void codec_seek_complete_callback(void)
|
|
{
|
|
/* assume we're called from non-voice codec, as they shouldn't seek */
|
|
ci.seek_time = 0;
|
|
pcmbuf_flush_audio();
|
|
}
|
|
|
|
bool codec_seek_buffer_callback(off_t newpos)
|
|
{
|
|
int difference;
|
|
|
|
if (current_codec == CODEC_IDX_VOICE)
|
|
return false;
|
|
|
|
if (newpos < 0)
|
|
newpos = 0;
|
|
|
|
if (newpos >= cur_ti->filesize)
|
|
newpos = cur_ti->filesize - 1;
|
|
|
|
difference = newpos - ci.curpos;
|
|
/* Seeking forward */
|
|
if (difference >= 0) {
|
|
logf("seek: +%d", difference);
|
|
codec_advance_buffer_callback(difference);
|
|
return true;
|
|
}
|
|
|
|
/* Seeking backward */
|
|
difference = -difference;
|
|
if (ci.curpos - difference < 0)
|
|
difference = ci.curpos;
|
|
|
|
/* We need to reload the song. */
|
|
if (newpos < cur_ti->start_pos)
|
|
return rebuffer_and_seek(newpos);
|
|
|
|
/* Seeking inside buffer space. */
|
|
logf("seek: -%d", difference);
|
|
filebufused += difference;
|
|
cur_ti->available += difference;
|
|
buf_ridx -= difference;
|
|
if (buf_ridx < 0)
|
|
buf_ridx = filebuflen + buf_ridx;
|
|
ci.curpos -= difference;
|
|
|
|
return true;
|
|
}
|
|
|
|
static void set_filebuf_watermark(int seconds)
|
|
{
|
|
long bytes;
|
|
|
|
if (current_codec == CODEC_IDX_VOICE)
|
|
return ;
|
|
|
|
if (!filebuf)
|
|
return; /* Audio buffers not yet set up */
|
|
|
|
bytes = MAX((int)cur_ti->id3.bitrate * seconds * (1000/8), conf_watermark);
|
|
bytes = MIN(bytes, filebuflen / 2);
|
|
conf_watermark = bytes;
|
|
}
|
|
|
|
void codec_configure_callback(int setting, void *value)
|
|
{
|
|
switch (setting) {
|
|
case CODEC_SET_FILEBUF_WATERMARK:
|
|
conf_watermark = (unsigned int)value;
|
|
set_filebuf_watermark(buffer_margin);
|
|
break;
|
|
|
|
case CODEC_SET_FILEBUF_CHUNKSIZE:
|
|
conf_filechunk = (unsigned int)value;
|
|
break;
|
|
|
|
case CODEC_SET_FILEBUF_LIMIT:
|
|
conf_bufferlimit = (unsigned int)value;
|
|
break;
|
|
|
|
case CODEC_DSP_ENABLE:
|
|
if ((bool)value)
|
|
ci.pcmbuf_insert = codec_pcmbuf_insert_callback;
|
|
else
|
|
ci.pcmbuf_insert = pcmbuf_insert_buffer;
|
|
break ;
|
|
|
|
default:
|
|
if (!dsp_configure(setting, value)) {
|
|
logf("Illegal key: %d", setting);
|
|
}
|
|
}
|
|
}
|
|
|
|
void audio_set_track_buffer_event(void (*handler)(struct mp3entry *id3,
|
|
bool last_track))
|
|
{
|
|
track_buffer_callback = handler;
|
|
}
|
|
|
|
void audio_set_track_unbuffer_event(void (*handler)(struct mp3entry *id3,
|
|
bool last_track))
|
|
{
|
|
track_unbuffer_callback = handler;
|
|
}
|
|
|
|
void audio_set_track_changed_event(void (*handler)(struct mp3entry *id3))
|
|
{
|
|
track_changed_callback = handler;
|
|
}
|
|
|
|
void codec_track_changed(void)
|
|
{
|
|
track_changed = true;
|
|
queue_post(&audio_queue, AUDIO_TRACK_CHANGED, 0);
|
|
}
|
|
|
|
/* Give codecs or file buffering the right amount of processing time
|
|
to prevent pcm audio buffer from going empty. */
|
|
void yield_codecs(void)
|
|
{
|
|
yield();
|
|
if (!pcm_is_playing() && !paused)
|
|
sleep(5);
|
|
while ((pcmbuf_is_crossfade_active() || pcmbuf_is_lowdata())
|
|
&& !ci.stop_codec && playing && queue_empty(&audio_queue)
|
|
&& filebufused > (128*1024))
|
|
sleep(1);
|
|
}
|
|
|
|
/* FIXME: This code should be made more generic and move to metadata.c */
|
|
void strip_id3v1_tag(void)
|
|
{
|
|
int i;
|
|
static const unsigned char tag[] = "TAG";
|
|
int tagptr;
|
|
bool found = true;
|
|
|
|
if (filebufused >= 128)
|
|
{
|
|
tagptr = buf_widx - 128;
|
|
if (tagptr < 0)
|
|
tagptr += filebuflen;
|
|
|
|
for(i = 0;i < 3;i++)
|
|
{
|
|
if(tagptr >= filebuflen)
|
|
tagptr -= filebuflen;
|
|
|
|
if(filebuf[tagptr] != tag[i])
|
|
{
|
|
found = false;
|
|
break;
|
|
}
|
|
|
|
tagptr++;
|
|
}
|
|
|
|
if(found)
|
|
{
|
|
/* Skip id3v1 tag */
|
|
logf("Skipping ID3v1 tag\n");
|
|
buf_widx -= 128;
|
|
tracks[track_widx].available -= 128;
|
|
filebufused -= 128;
|
|
}
|
|
}
|
|
}
|
|
|
|
void audio_fill_file_buffer(void)
|
|
{
|
|
long i, size;
|
|
int rc;
|
|
|
|
if (current_fd < 0)
|
|
return ;
|
|
|
|
/* Throw away buffered codec. */
|
|
if (tracks[track_widx].start_pos != 0)
|
|
tracks[track_widx].codecsize = 0;
|
|
|
|
mutex_lock(&mutex_bufferfill);
|
|
i = 0;
|
|
size = MIN(tracks[track_widx].filerem, AUDIO_FILL_CYCLE);
|
|
while (i < size) {
|
|
/* Give codecs some processing time. */
|
|
yield_codecs();
|
|
|
|
if (fill_bytesleft == 0)
|
|
break ;
|
|
rc = MIN(conf_filechunk, filebuflen - buf_widx);
|
|
rc = MIN(rc, fill_bytesleft);
|
|
rc = read(current_fd, &filebuf[buf_widx], rc);
|
|
if (rc <= 0) {
|
|
tracks[track_widx].filerem = 0;
|
|
strip_id3v1_tag();
|
|
break ;
|
|
}
|
|
|
|
buf_widx += rc;
|
|
if (buf_widx >= filebuflen)
|
|
buf_widx -= filebuflen;
|
|
i += rc;
|
|
tracks[track_widx].available += rc;
|
|
tracks[track_widx].filerem -= rc;
|
|
tracks[track_widx].filepos += rc;
|
|
filebufused += rc;
|
|
fill_bytesleft -= rc;
|
|
}
|
|
mutex_unlock(&mutex_bufferfill);
|
|
|
|
/*logf("Filled:%d/%d", tracks[track_widx].available,
|
|
tracks[track_widx].filerem);*/
|
|
}
|
|
|
|
bool loadcodec(const char *trackname, bool start_play)
|
|
{
|
|
char msgbuf[80];
|
|
off_t size;
|
|
unsigned int filetype;
|
|
int fd;
|
|
int i, rc;
|
|
const char *codec_path;
|
|
int copy_n;
|
|
int prev_track;
|
|
|
|
filetype = probe_file_format(trackname);
|
|
switch (filetype) {
|
|
case AFMT_OGG_VORBIS:
|
|
logf("Codec: Vorbis");
|
|
codec_path = CODEC_VORBIS;
|
|
break;
|
|
case AFMT_MPA_L1:
|
|
case AFMT_MPA_L2:
|
|
case AFMT_MPA_L3:
|
|
logf("Codec: MPA L1/L2/L3");
|
|
codec_path = CODEC_MPA_L3;
|
|
break;
|
|
case AFMT_PCM_WAV:
|
|
logf("Codec: PCM WAV");
|
|
codec_path = CODEC_WAV;
|
|
break;
|
|
case AFMT_FLAC:
|
|
logf("Codec: FLAC");
|
|
codec_path = CODEC_FLAC;
|
|
break;
|
|
case AFMT_A52:
|
|
logf("Codec: A52");
|
|
codec_path = CODEC_A52;
|
|
break;
|
|
case AFMT_MPC:
|
|
logf("Codec: Musepack");
|
|
codec_path = CODEC_MPC;
|
|
break;
|
|
case AFMT_WAVPACK:
|
|
logf("Codec: WAVPACK");
|
|
codec_path = CODEC_WAVPACK;
|
|
break;
|
|
case AFMT_ALAC:
|
|
logf("Codec: ALAC");
|
|
codec_path = CODEC_ALAC;
|
|
break;
|
|
case AFMT_AAC:
|
|
logf("Codec: AAC");
|
|
codec_path = CODEC_AAC;
|
|
break;
|
|
default:
|
|
logf("Codec: Unsupported");
|
|
snprintf(msgbuf, sizeof(msgbuf)-1, "No codec for: %s", trackname);
|
|
splash(HZ*2, true, msgbuf);
|
|
codec_path = NULL;
|
|
}
|
|
|
|
tracks[track_widx].id3.codectype = filetype;
|
|
tracks[track_widx].codecsize = 0;
|
|
if (codec_path == NULL)
|
|
return false;
|
|
|
|
if (!start_play) {
|
|
prev_track = track_widx - 1;
|
|
if (prev_track < 0)
|
|
prev_track = MAX_TRACK-1;
|
|
if (track_count > 0 && filetype == tracks[prev_track].id3.codectype) {
|
|
logf("Reusing prev. codec");
|
|
return true;
|
|
}
|
|
} else {
|
|
/* Load the codec directly from disk and save some memory. */
|
|
cur_ti = &tracks[track_widx];
|
|
ci.filesize = cur_ti->filesize;
|
|
ci.id3 = (struct mp3entry *)&cur_ti->id3;
|
|
ci.taginfo_ready = (bool *)&cur_ti->taginfo_ready;
|
|
ci.curpos = 0;
|
|
playing = true;
|
|
logf("Starting codec");
|
|
queue_post(&codec_queue, CODEC_LOAD_DISK, (void *)codec_path);
|
|
return true;
|
|
}
|
|
|
|
fd = open(codec_path, O_RDONLY);
|
|
if (fd < 0) {
|
|
logf("Codec doesn't exist!");
|
|
snprintf(msgbuf, sizeof(msgbuf)-1, "Couldn't load codec: %s", codec_path);
|
|
splash(HZ*2, true, msgbuf);
|
|
return false;
|
|
}
|
|
|
|
size = filesize(fd);
|
|
if ((off_t)fill_bytesleft < size + conf_watermark) {
|
|
logf("Not enough space");
|
|
/* Set codectype back to zero to indicate no codec was loaded. */
|
|
tracks[track_widx].id3.codectype = 0;
|
|
fill_bytesleft = 0;
|
|
close(fd);
|
|
return false;
|
|
}
|
|
|
|
i = 0;
|
|
while (i < size) {
|
|
yield_codecs();
|
|
|
|
copy_n = MIN(conf_filechunk, filebuflen - buf_widx);
|
|
rc = read(fd, &filebuf[buf_widx], copy_n);
|
|
if (rc < 0)
|
|
return false;
|
|
buf_widx += rc;
|
|
filebufused += rc;
|
|
fill_bytesleft -= rc;
|
|
if (buf_widx >= filebuflen)
|
|
buf_widx -= filebuflen;
|
|
i += rc;
|
|
}
|
|
close(fd);
|
|
logf("Done: %dB", i);
|
|
|
|
tracks[track_widx].codecsize = size;
|
|
|
|
return true;
|
|
}
|
|
|
|
bool read_next_metadata(void)
|
|
{
|
|
int fd;
|
|
char *trackname;
|
|
int next_track;
|
|
int status;
|
|
|
|
next_track = track_widx;
|
|
if (tracks[track_widx].taginfo_ready)
|
|
next_track++;
|
|
|
|
if (next_track >= MAX_TRACK)
|
|
next_track -= MAX_TRACK;
|
|
|
|
if (tracks[next_track].taginfo_ready)
|
|
return true;
|
|
|
|
trackname = playlist_peek(last_peek_offset + 1);
|
|
if (!trackname)
|
|
return false;
|
|
|
|
fd = open(trackname, O_RDONLY);
|
|
if (fd < 0)
|
|
return false;
|
|
|
|
/* Start buffer refilling also because we need to spin-up the disk. */
|
|
filling = true;
|
|
tracks[next_track].id3.codectype = probe_file_format(trackname);
|
|
status = get_metadata(&tracks[next_track],fd,trackname,v1first);
|
|
tracks[next_track].id3.codectype = 0;
|
|
track_changed = true;
|
|
close(fd);
|
|
|
|
return status;
|
|
}
|
|
|
|
bool audio_load_track(int offset, bool start_play, int peek_offset)
|
|
{
|
|
char *trackname;
|
|
int fd = -1;
|
|
off_t size;
|
|
int rc, i;
|
|
int copy_n;
|
|
|
|
/* Stop buffer filling if there is no free track entries.
|
|
Don't fill up the last track entry (we wan't to store next track
|
|
metadata there). */
|
|
if (track_count >= MAX_TRACK - 1) {
|
|
fill_bytesleft = 0;
|
|
return false;
|
|
}
|
|
|
|
/* Don't start loading track if the current write position already
|
|
contains a BUFFERED track. The entry may contain the metadata
|
|
which is ok. */
|
|
if (tracks[track_widx].filesize != 0)
|
|
return false;
|
|
|
|
last_index = playlist_get_display_index();
|
|
|
|
peek_again:
|
|
/* Get track name from current playlist read position. */
|
|
logf("Buffering track:%d/%d", track_widx, track_ridx);
|
|
/* Handle broken playlists. */
|
|
while ( (trackname = playlist_peek(peek_offset)) != NULL) {
|
|
fd = open(trackname, O_RDONLY);
|
|
if (fd < 0) {
|
|
logf("Open failed");
|
|
/* Skip invalid entry from playlist. */
|
|
playlist_skip_entry(NULL, peek_offset);
|
|
continue ;
|
|
}
|
|
break ;
|
|
}
|
|
|
|
if (!trackname) {
|
|
logf("End-of-playlist");
|
|
conf_watermark = 0;
|
|
return false;
|
|
}
|
|
|
|
/* Initialize track entry. */
|
|
size = filesize(fd);
|
|
tracks[track_widx].filerem = size;
|
|
tracks[track_widx].filesize = size;
|
|
tracks[track_widx].filepos = 0;
|
|
tracks[track_widx].available = 0;
|
|
//tracks[track_widx].taginfo_ready = false;
|
|
tracks[track_widx].playlist_offset = peek_offset;
|
|
last_peek_offset = peek_offset;
|
|
|
|
if (buf_widx >= filebuflen)
|
|
buf_widx -= filebuflen;
|
|
|
|
/* Set default values */
|
|
if (start_play) {
|
|
int last_codec = current_codec;
|
|
current_codec = CODEC_IDX_AUDIO;
|
|
conf_bufferlimit = 0;
|
|
conf_watermark = AUDIO_DEFAULT_WATERMARK;
|
|
conf_filechunk = AUDIO_DEFAULT_FILECHUNK;
|
|
dsp_configure(DSP_RESET, 0);
|
|
ci.configure(CODEC_DSP_ENABLE, false);
|
|
current_codec = last_codec;
|
|
}
|
|
|
|
/* Load the codec. */
|
|
tracks[track_widx].codecbuf = &filebuf[buf_widx];
|
|
if (!loadcodec(trackname, start_play)) {
|
|
close(fd);
|
|
/* Stop buffer filling if codec load failed. */
|
|
fill_bytesleft = 0;
|
|
/* Set filesize to zero to indicate no file was loaded. */
|
|
tracks[track_widx].filesize = 0;
|
|
tracks[track_widx].filerem = 0;
|
|
|
|
/* Try skipping to next track. */
|
|
if (fill_bytesleft > 0) {
|
|
/* Skip invalid entry from playlist. */
|
|
playlist_skip_entry(NULL, peek_offset);
|
|
goto peek_again;
|
|
}
|
|
return false;
|
|
}
|
|
// tracks[track_widx].filebuf = &filebuf[buf_widx];
|
|
tracks[track_widx].start_pos = 0;
|
|
|
|
/* Get track metadata if we don't already have it. */
|
|
if (!tracks[track_widx].taginfo_ready) {
|
|
if (!get_metadata(&tracks[track_widx],fd,trackname,v1first)) {
|
|
logf("Metadata error!");
|
|
tracks[track_widx].filesize = 0;
|
|
tracks[track_widx].filerem = 0;
|
|
close(fd);
|
|
/* Skip invalid entry from playlist. */
|
|
playlist_skip_entry(NULL, peek_offset);
|
|
goto peek_again;
|
|
}
|
|
}
|
|
set_filebuf_watermark(buffer_margin);
|
|
tracks[track_widx].id3.elapsed = 0;
|
|
|
|
/* Starting playback from an offset is only support in MPA at the moment */
|
|
if (offset > 0) {
|
|
switch (tracks[track_widx].id3.codectype) {
|
|
case AFMT_MPA_L2:
|
|
case AFMT_MPA_L3:
|
|
lseek(fd, offset, SEEK_SET);
|
|
tracks[track_widx].id3.offset = offset;
|
|
mp3_set_elapsed(&tracks[track_widx].id3);
|
|
tracks[track_widx].filepos = offset;
|
|
tracks[track_widx].filerem = tracks[track_widx].filesize - offset;
|
|
ci.curpos = offset;
|
|
tracks[track_widx].start_pos = offset;
|
|
break;
|
|
|
|
case AFMT_WAVPACK:
|
|
lseek(fd, offset, SEEK_SET);
|
|
tracks[track_widx].id3.offset = offset;
|
|
tracks[track_widx].id3.elapsed = tracks[track_widx].id3.length / 2;
|
|
tracks[track_widx].filepos = offset;
|
|
tracks[track_widx].filerem = tracks[track_widx].filesize - offset;
|
|
ci.curpos = offset;
|
|
tracks[track_widx].start_pos = offset;
|
|
break;
|
|
case AFMT_OGG_VORBIS:
|
|
case AFMT_FLAC:
|
|
tracks[track_widx].id3.offset = offset;
|
|
break;
|
|
}
|
|
}
|
|
|
|
if (start_play) {
|
|
track_count++;
|
|
codec_track_changed();
|
|
}
|
|
|
|
/* Do some initial file buffering. */
|
|
mutex_lock(&mutex_bufferfill);
|
|
i = tracks[track_widx].start_pos;
|
|
size = MIN(size, AUDIO_FILL_CYCLE);
|
|
while (i < size) {
|
|
/* Give codecs some processing time to prevent glitches. */
|
|
yield_codecs();
|
|
|
|
if (fill_bytesleft == 0)
|
|
break ;
|
|
|
|
copy_n = MIN(conf_filechunk, filebuflen - buf_widx);
|
|
copy_n = MIN(size - i, copy_n);
|
|
copy_n = MIN((int)fill_bytesleft, copy_n);
|
|
rc = read(fd, &filebuf[buf_widx], copy_n);
|
|
if (rc < copy_n) {
|
|
logf("File error!");
|
|
tracks[track_widx].filesize = 0;
|
|
tracks[track_widx].filerem = 0;
|
|
close(fd);
|
|
mutex_unlock(&mutex_bufferfill);
|
|
return false;
|
|
}
|
|
buf_widx += rc;
|
|
if (buf_widx >= filebuflen)
|
|
buf_widx -= filebuflen;
|
|
i += rc;
|
|
tracks[track_widx].available += rc;
|
|
tracks[track_widx].filerem -= rc;
|
|
filebufused += rc;
|
|
fill_bytesleft -= rc;
|
|
}
|
|
mutex_unlock(&mutex_bufferfill);
|
|
|
|
if (!start_play)
|
|
track_count++;
|
|
|
|
tracks[track_widx].filepos = i;
|
|
|
|
if (current_fd >= 0) {
|
|
close(current_fd);
|
|
current_fd = -1;
|
|
}
|
|
|
|
/* Leave the file handle open for faster buffer refill. */
|
|
if (tracks[track_widx].filerem != 0) {
|
|
current_fd = fd;
|
|
logf("Partially buf:%d", tracks[track_widx].available);
|
|
} else {
|
|
logf("Completely buf.");
|
|
close(fd);
|
|
|
|
strip_id3v1_tag();
|
|
|
|
if (++track_widx >= MAX_TRACK) {
|
|
track_widx = 0;
|
|
}
|
|
tracks[track_widx].filerem = 0;
|
|
}
|
|
|
|
return true;
|
|
}
|
|
|
|
void audio_play_start(int offset)
|
|
{
|
|
if (current_fd >= 0) {
|
|
close(current_fd);
|
|
current_fd = -1;
|
|
}
|
|
|
|
memset(&tracks, 0, sizeof(struct track_info) * MAX_TRACK);
|
|
sound_set(SOUND_VOLUME, global_settings.volume);
|
|
track_count = 0;
|
|
track_widx = 0;
|
|
track_ridx = 0;
|
|
buf_ridx = 0;
|
|
buf_widx = 0;
|
|
filebufused = 0;
|
|
pcmbuf_set_boost_mode(true);
|
|
|
|
fill_bytesleft = filebuflen;
|
|
filling = true;
|
|
last_peek_offset = -1;
|
|
|
|
if (audio_load_track(offset, true, 0)) {
|
|
if (track_buffer_callback) {
|
|
cur_ti->event_sent = true;
|
|
track_buffer_callback(&cur_ti->id3, true);
|
|
}
|
|
} else {
|
|
logf("Failure");
|
|
}
|
|
|
|
pcmbuf_set_boost_mode(false);
|
|
}
|
|
|
|
void audio_clear_track_entries(bool buffered_only)
|
|
{
|
|
int cur_idx, event_count;
|
|
int i;
|
|
|
|
cur_idx = track_widx;
|
|
event_count = 0;
|
|
for (i = 0; i < MAX_TRACK - track_count; i++) {
|
|
if (++cur_idx >= MAX_TRACK)
|
|
cur_idx = 0;
|
|
|
|
if (tracks[cur_idx].event_sent)
|
|
event_count++;
|
|
|
|
if (!track_unbuffer_callback)
|
|
memset(&tracks[cur_idx], 0, sizeof(struct track_info));
|
|
}
|
|
|
|
if (!track_unbuffer_callback)
|
|
return ;
|
|
|
|
cur_idx = track_widx;
|
|
for (i = 0; i < MAX_TRACK - track_count; i++) {
|
|
if (++cur_idx >= MAX_TRACK)
|
|
cur_idx = 0;
|
|
|
|
/* Send an event to notify that track has finished. */
|
|
if (tracks[cur_idx].event_sent) {
|
|
event_count--;
|
|
track_unbuffer_callback(&tracks[cur_idx].id3, event_count == 0);
|
|
}
|
|
|
|
if (tracks[cur_idx].event_sent || !buffered_only)
|
|
memset(&tracks[cur_idx], 0, sizeof(struct track_info));
|
|
}
|
|
}
|
|
|
|
/* Send callback events to notify about new tracks. */
|
|
static void generate_postbuffer_events(void)
|
|
{
|
|
int i;
|
|
int cur_ridx, event_count;
|
|
|
|
/* At first determine how many unsent events we have. */
|
|
cur_ridx = track_ridx;
|
|
event_count = 0;
|
|
for (i = 0; i < track_count; i++) {
|
|
if (!tracks[cur_ridx].event_sent)
|
|
event_count++;
|
|
if (++cur_ridx >= MAX_TRACK)
|
|
cur_ridx -= MAX_TRACK;
|
|
}
|
|
|
|
/* Now sent these events. */
|
|
cur_ridx = track_ridx;
|
|
for (i = 0; i < track_count; i++) {
|
|
if (!tracks[cur_ridx].event_sent) {
|
|
tracks[cur_ridx].event_sent = true;
|
|
event_count--;
|
|
/* We still want to set event_sent flags even if not using
|
|
event callbacks. */
|
|
if (track_buffer_callback)
|
|
track_buffer_callback(&tracks[cur_ridx].id3, event_count == 0);
|
|
}
|
|
if (++cur_ridx >= MAX_TRACK)
|
|
cur_ridx -= MAX_TRACK;
|
|
}
|
|
}
|
|
|
|
void initialize_buffer_fill(void)
|
|
{
|
|
int cur_idx, i;
|
|
|
|
|
|
fill_bytesleft = filebuflen - filebufused;
|
|
cur_ti->start_pos = ci.curpos;
|
|
|
|
if (filling)
|
|
return ;
|
|
|
|
pcmbuf_set_boost_mode(true);
|
|
|
|
filling = true;
|
|
|
|
/* Calculate real track count after throwing away old tracks. */
|
|
cur_idx = track_ridx;
|
|
for (i = 0; i < track_count; i++) {
|
|
if (cur_idx == track_widx)
|
|
break ;
|
|
|
|
if (++cur_idx >= MAX_TRACK)
|
|
cur_idx = 0;
|
|
}
|
|
|
|
track_count = i;
|
|
if (tracks[track_widx].filesize == 0) {
|
|
if (--track_widx < 0)
|
|
track_widx = MAX_TRACK - 1;
|
|
} else {
|
|
track_count++;
|
|
}
|
|
|
|
/* Mark all buffered entries null (not metadata for next track). */
|
|
audio_clear_track_entries(true);
|
|
}
|
|
|
|
void audio_check_buffer(void)
|
|
{
|
|
/* Start buffer filling as necessary. */
|
|
if ((!conf_watermark || filebufused > conf_watermark
|
|
|| !queue_empty(&audio_queue) || !playing || ci.stop_codec
|
|
|| ci.reload_codec) && !filling)
|
|
return ;
|
|
|
|
mutex_lock(&mutex_bufferfill);
|
|
initialize_buffer_fill();
|
|
mutex_unlock(&mutex_bufferfill);
|
|
|
|
/* Limit buffering size at first run. */
|
|
if (conf_bufferlimit && fill_bytesleft > conf_bufferlimit
|
|
- filebufused) {
|
|
fill_bytesleft = MAX(0, conf_bufferlimit - filebufused);
|
|
}
|
|
|
|
/* Try to load remainings of the file. */
|
|
if (tracks[track_widx].filerem > 0)
|
|
audio_fill_file_buffer();
|
|
|
|
/* Increase track write index as necessary. */
|
|
if (tracks[track_widx].filerem == 0 && tracks[track_widx].filesize != 0) {
|
|
if (++track_widx == MAX_TRACK)
|
|
track_widx = 0;
|
|
}
|
|
|
|
/* Load new files to fill the entire buffer. */
|
|
if (audio_load_track(0, false, last_peek_offset + 1)) {
|
|
|
|
} else if (tracks[track_widx].filerem == 0 || fill_bytesleft == 0) {
|
|
/* Read next unbuffered track's metadata as necessary. */
|
|
read_next_metadata();
|
|
|
|
generate_postbuffer_events();
|
|
filling = false;
|
|
conf_bufferlimit = 0;
|
|
pcmbuf_set_boost_mode(false);
|
|
|
|
#ifndef SIMULATOR
|
|
if (playing)
|
|
ata_sleep();
|
|
#endif
|
|
}
|
|
}
|
|
|
|
void audio_update_trackinfo(void)
|
|
{
|
|
if (new_track >= 0) {
|
|
buf_ridx += cur_ti->available;
|
|
filebufused -= cur_ti->available;
|
|
|
|
cur_ti = &tracks[track_ridx];
|
|
buf_ridx += cur_ti->codecsize;
|
|
filebufused -= cur_ti->codecsize;
|
|
if (buf_ridx >= filebuflen)
|
|
buf_ridx -= filebuflen;
|
|
|
|
if (!filling)
|
|
pcmbuf_set_boost_mode(false);
|
|
} else {
|
|
buf_ridx -= ci.curpos + cur_ti->codecsize;
|
|
filebufused += ci.curpos + cur_ti->codecsize;
|
|
cur_ti->available = cur_ti->filesize - cur_ti->filerem;
|
|
|
|
cur_ti = &tracks[track_ridx];
|
|
buf_ridx -= cur_ti->filesize;
|
|
filebufused += cur_ti->filesize;
|
|
cur_ti->available = cur_ti->filesize;
|
|
if (buf_ridx < 0)
|
|
buf_ridx = filebuflen + buf_ridx;
|
|
}
|
|
|
|
ci.filesize = cur_ti->filesize;
|
|
cur_ti->id3.elapsed = 0;
|
|
cur_ti->id3.offset = 0;
|
|
ci.id3 = (struct mp3entry *)&cur_ti->id3;
|
|
ci.curpos = 0;
|
|
cur_ti->start_pos = 0;
|
|
ci.taginfo_ready = (bool *)&cur_ti->taginfo_ready;
|
|
if (pcmbuf_is_crossfade_enabled() && !pcmbuf_is_crossfade_active()) {
|
|
pcmbuf_crossfade_init(new_track ? CROSSFADE_MODE_CROSSFADE
|
|
: global_settings.crossfade);
|
|
codec_track_changed();
|
|
} else {
|
|
pcmbuf_add_event(codec_track_changed);
|
|
}
|
|
last_index = playlist_get_display_index();
|
|
}
|
|
|
|
static void audio_stop_playback(void)
|
|
{
|
|
paused = false;
|
|
playing = false;
|
|
filling = false;
|
|
ci.stop_codec = true;
|
|
if (current_fd >= 0) {
|
|
close(current_fd);
|
|
current_fd = -1;
|
|
}
|
|
pcmbuf_play_stop();
|
|
while (audio_codec_loaded)
|
|
yield();
|
|
|
|
track_count = 0;
|
|
/* Mark all entries null. */
|
|
audio_clear_track_entries(false);
|
|
}
|
|
|
|
/* Request the next track with new codec. */
|
|
void audio_change_track(void)
|
|
{
|
|
logf("change track");
|
|
|
|
/* Wait for new track data. */
|
|
while (track_count <= 1 && filling)
|
|
yield();
|
|
|
|
/* If we are not filling, then it must be end-of-playlist. */
|
|
if (track_count <= 1) {
|
|
logf("No more tracks");
|
|
while (pcm_is_playing())
|
|
yield();
|
|
audio_stop_playback();
|
|
return ;
|
|
}
|
|
|
|
if (++track_ridx >= MAX_TRACK)
|
|
track_ridx = 0;
|
|
|
|
audio_update_trackinfo();
|
|
queue_post(&codec_queue, CODEC_LOAD, 0);
|
|
}
|
|
|
|
static int get_codec_base_type(int type)
|
|
{
|
|
switch (type) {
|
|
case AFMT_MPA_L1:
|
|
case AFMT_MPA_L2:
|
|
case AFMT_MPA_L3:
|
|
return AFMT_MPA_L3;
|
|
}
|
|
|
|
return type;
|
|
}
|
|
|
|
bool codec_request_next_track_callback(void)
|
|
{
|
|
if (current_codec == CODEC_IDX_VOICE) {
|
|
voice_remaining = 0;
|
|
/* Terminate the codec if they are messages waiting on the queue or
|
|
core has been requested the codec to be terminated. */
|
|
return !ci_voice.stop_codec && queue_empty(&voice_codec_queue);
|
|
}
|
|
|
|
if (ci.stop_codec || !playing)
|
|
return false;
|
|
|
|
logf("Request new track");
|
|
|
|
/* Advance to next track. */
|
|
if (ci.reload_codec && new_track > 0) {
|
|
if (!playlist_check(new_track)) {
|
|
ci.reload_codec = false;
|
|
return false;
|
|
}
|
|
last_peek_offset--;
|
|
playlist_next(new_track);
|
|
if (++track_ridx == MAX_TRACK)
|
|
track_ridx = 0;
|
|
|
|
/* Wait for new track data (codectype 0 is invalid). When a correct
|
|
codectype is set, we can assume that the filesize is correct. */
|
|
while (tracks[track_ridx].id3.codectype == 0 && filling
|
|
&& !ci.stop_codec)
|
|
yield();
|
|
|
|
if (tracks[track_ridx].filesize == 0) {
|
|
logf("Loading from disk...");
|
|
new_track = 0;
|
|
last_index = -1;
|
|
queue_post(&audio_queue, AUDIO_PLAY, 0);
|
|
return false;
|
|
}
|
|
}
|
|
|
|
/* Advance to previous track. */
|
|
else if (ci.reload_codec && new_track < 0) {
|
|
if (!playlist_check(new_track)) {
|
|
ci.reload_codec = false;
|
|
return false;
|
|
}
|
|
last_peek_offset++;
|
|
playlist_next(new_track);
|
|
if (--track_ridx < 0)
|
|
track_ridx = MAX_TRACK-1;
|
|
if (tracks[track_ridx].filesize == 0 ||
|
|
filebufused+ci.curpos+tracks[track_ridx].filesize
|
|
/*+ (off_t)tracks[track_ridx].codecsize*/ > filebuflen) {
|
|
logf("Loading from disk...");
|
|
new_track = 0;
|
|
last_index = -1;
|
|
queue_post(&audio_queue, AUDIO_PLAY, 0);
|
|
return false;
|
|
}
|
|
}
|
|
|
|
/* Codec requested track change (next track). */
|
|
else {
|
|
if (!playlist_check(1)) {
|
|
ci.reload_codec = false;
|
|
return false;
|
|
}
|
|
last_peek_offset--;
|
|
playlist_next(1);
|
|
if (++track_ridx >= MAX_TRACK)
|
|
track_ridx = 0;
|
|
|
|
/* Wait for new track data (codectype 0 is invalid). When a correct
|
|
codectype is set, we can assume that the filesize is correct. */
|
|
while (tracks[track_ridx].id3.codectype == 0 && filling
|
|
&& !ci.stop_codec)
|
|
yield();
|
|
|
|
if (tracks[track_ridx].filesize == 0) {
|
|
logf("No more tracks [2]");
|
|
ci.stop_codec = true;
|
|
new_track = 0;
|
|
last_index = -1;
|
|
queue_post(&audio_queue, AUDIO_PLAY, 0);
|
|
return false;
|
|
}
|
|
}
|
|
|
|
ci.reload_codec = false;
|
|
|
|
/* Check if the next codec is the same file. */
|
|
if (get_codec_base_type(cur_ti->id3.codectype) !=
|
|
get_codec_base_type(tracks[track_ridx].id3.codectype)) {
|
|
logf("New codec:%d/%d", cur_ti->id3.codectype,
|
|
tracks[track_ridx].id3.codectype);
|
|
if (--track_ridx < 0)
|
|
track_ridx = MAX_TRACK-1;
|
|
new_track = 0;
|
|
return false;
|
|
}
|
|
|
|
logf("On-the-fly change");
|
|
audio_update_trackinfo();
|
|
new_track = 0;
|
|
|
|
return true;
|
|
}
|
|
|
|
/* Invalidates all but currently playing track. */
|
|
void audio_invalidate_tracks(void)
|
|
{
|
|
if (track_count == 0) {
|
|
/* This call doesn't seem necessary anymore. Uncomment it
|
|
if things break */
|
|
/* queue_post(&audio_queue, AUDIO_PLAY, 0); */
|
|
return ;
|
|
}
|
|
|
|
track_count = 1;
|
|
last_peek_offset = 0;
|
|
track_widx = track_ridx;
|
|
/* Mark all other entries null (also buffered wrong metadata). */
|
|
audio_clear_track_entries(false);
|
|
filebufused = cur_ti->available;
|
|
buf_widx = buf_ridx + cur_ti->available;
|
|
if (buf_widx >= filebuflen)
|
|
buf_widx -= filebuflen;
|
|
read_next_metadata();
|
|
}
|
|
|
|
static void initiate_track_change(int peek_index)
|
|
{
|
|
if (!playlist_check(peek_index))
|
|
return ;
|
|
|
|
/* Detect if disk is spinning.. */
|
|
if (filling) {
|
|
playlist_next(peek_index);
|
|
queue_post(&audio_queue, AUDIO_PLAY, 0);
|
|
} else {
|
|
new_track = peek_index;
|
|
ci.reload_codec = true;
|
|
if (!pcmbuf_is_crossfade_enabled())
|
|
pcmbuf_flush_audio();
|
|
}
|
|
|
|
codec_track_changed();
|
|
}
|
|
|
|
void audio_thread(void)
|
|
{
|
|
struct event ev;
|
|
|
|
while (1) {
|
|
yield_codecs();
|
|
|
|
audio_check_buffer();
|
|
|
|
queue_wait_w_tmo(&audio_queue, &ev, 0);
|
|
switch (ev.id) {
|
|
case AUDIO_PLAY:
|
|
/* Refuse to start playback if we are already playing
|
|
the requested track. This is needed because when skipping
|
|
tracks fast, AUDIO_PLAY commands will get queued with the
|
|
the same track and playback will stutter. */
|
|
if (last_index == playlist_get_display_index() && playing
|
|
&& pcm_is_playing()) {
|
|
logf("already playing req. track");
|
|
break ;
|
|
}
|
|
|
|
/* Do not start crossfading if audio is paused. */
|
|
if (paused) {
|
|
audio_stop_playback();
|
|
paused = false;
|
|
}
|
|
|
|
logf("starting...");
|
|
playing = true;
|
|
ci.stop_codec = true;
|
|
ci.reload_codec = false;
|
|
ci.seek_time = 0;
|
|
pcmbuf_crossfade_init(CROSSFADE_MODE_CROSSFADE);
|
|
while (audio_codec_loaded)
|
|
yield();
|
|
audio_play_start((int)ev.data);
|
|
|
|
playlist_update_resume_info(audio_current_track());
|
|
|
|
/* If there are no tracks in the playlist, then the playlist
|
|
was empty or none of the filenames were valid. No point
|
|
in playing an empty playlist. */
|
|
if (playlist_amount() == 0) {
|
|
audio_stop_playback();
|
|
}
|
|
break ;
|
|
|
|
case AUDIO_STOP:
|
|
if (playing)
|
|
playlist_update_resume_info(audio_current_track());
|
|
audio_stop_playback();
|
|
break ;
|
|
|
|
case AUDIO_PAUSE:
|
|
logf("audio_pause");
|
|
/* We will pause the pcm playback in audiobuffer insert function
|
|
to prevent a loop inside the pcm buffer. */
|
|
// pcm_play_pause(false);
|
|
paused = true;
|
|
break ;
|
|
|
|
case AUDIO_RESUME:
|
|
logf("audio_resume");
|
|
pcm_play_pause(true);
|
|
paused = false;
|
|
break ;
|
|
|
|
case AUDIO_NEXT:
|
|
logf("audio_next");
|
|
if (global_settings.beep)
|
|
pcmbuf_beep(5000, 100, 2500*global_settings.beep);
|
|
initiate_track_change(1);
|
|
break ;
|
|
|
|
case AUDIO_PREV:
|
|
logf("audio_prev");
|
|
if (global_settings.beep)
|
|
pcmbuf_beep(5000, 100, 2500*global_settings.beep);
|
|
initiate_track_change(-1);
|
|
break;
|
|
|
|
case AUDIO_FLUSH:
|
|
audio_invalidate_tracks();
|
|
break ;
|
|
|
|
case AUDIO_TRACK_CHANGED:
|
|
if (track_changed_callback)
|
|
track_changed_callback(&cur_ti->id3);
|
|
playlist_update_resume_info(audio_current_track());
|
|
break ;
|
|
|
|
case AUDIO_CODEC_DONE:
|
|
//if (playing)
|
|
// audio_change_track();
|
|
break ;
|
|
|
|
#ifndef SIMULATOR
|
|
case SYS_USB_CONNECTED:
|
|
logf("USB: Audio core");
|
|
audio_stop_playback();
|
|
usb_acknowledge(SYS_USB_CONNECTED_ACK);
|
|
usb_wait_for_disconnect(&audio_queue);
|
|
break ;
|
|
#endif
|
|
case SYS_TIMEOUT:
|
|
if (playing)
|
|
playlist_update_resume_info(audio_current_track());
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
|
|
void codec_thread(void)
|
|
{
|
|
struct event ev;
|
|
long codecsize;
|
|
int status;
|
|
int wrap;
|
|
|
|
while (1) {
|
|
status = 0;
|
|
queue_wait(&codec_queue, &ev);
|
|
switch (ev.id) {
|
|
case CODEC_LOAD_DISK:
|
|
ci.stop_codec = false;
|
|
audio_codec_loaded = true;
|
|
mutex_lock(&mutex_codecthread);
|
|
current_codec = CODEC_IDX_AUDIO;
|
|
status = codec_load_file((char *)ev.data, &ci);
|
|
mutex_unlock(&mutex_codecthread);
|
|
break ;
|
|
|
|
case CODEC_LOAD:
|
|
logf("Codec start");
|
|
codecsize = cur_ti->codecsize;
|
|
if (codecsize == 0) {
|
|
logf("Codec slot is empty!");
|
|
/* Wait for the pcm buffer to go empty */
|
|
while (pcm_is_playing())
|
|
yield();
|
|
audio_stop_playback();
|
|
break ;
|
|
}
|
|
|
|
ci.stop_codec = false;
|
|
wrap = (int)&filebuf[filebuflen] - (int)cur_ti->codecbuf;
|
|
audio_codec_loaded = true;
|
|
mutex_lock(&mutex_codecthread);
|
|
current_codec = CODEC_IDX_AUDIO;
|
|
status = codec_load_ram(cur_ti->codecbuf, codecsize,
|
|
&filebuf[0], wrap, &ci);
|
|
mutex_unlock(&mutex_codecthread);
|
|
break ;
|
|
|
|
#ifndef SIMULATOR
|
|
case SYS_USB_CONNECTED:
|
|
while (voice_codec_loaded) {
|
|
if (current_codec != CODEC_IDX_VOICE)
|
|
swap_codec();
|
|
sleep(1);
|
|
}
|
|
logf("USB: Audio codec");
|
|
usb_acknowledge(SYS_USB_CONNECTED_ACK);
|
|
usb_wait_for_disconnect(&codec_queue);
|
|
break ;
|
|
#endif
|
|
}
|
|
|
|
audio_codec_loaded = false;
|
|
|
|
switch (ev.id) {
|
|
case CODEC_LOAD_DISK:
|
|
case CODEC_LOAD:
|
|
if (status != CODEC_OK) {
|
|
logf("Codec failure");
|
|
audio_stop_playback();
|
|
splash(HZ*2, true, "Codec failure");
|
|
} else {
|
|
logf("Codec finished");
|
|
}
|
|
|
|
if (playing && !ci.stop_codec && !ci.reload_codec) {
|
|
audio_change_track();
|
|
continue ;
|
|
} else if (ci.stop_codec) {
|
|
//playing = false;
|
|
}
|
|
//queue_post(&audio_queue, AUDIO_CODEC_DONE, (void *)status);
|
|
}
|
|
}
|
|
}
|
|
|
|
static void reset_buffer(void)
|
|
{
|
|
filebuf = &audiobuf[MALLOC_BUFSIZE];
|
|
filebuflen = audiobufend - audiobuf - pcmbuf_get_bufsize()
|
|
- PCMBUF_GUARD - MALLOC_BUFSIZE - GUARD_BUFSIZE;
|
|
|
|
if (talk_get_bufsize() && voice_codec_loaded)
|
|
{
|
|
filebuf = &filebuf[talk_get_bufsize()];
|
|
filebuflen -= 2*CODEC_IRAM_SIZE + 2*CODEC_SIZE + talk_get_bufsize();
|
|
}
|
|
}
|
|
|
|
void voice_codec_thread(void)
|
|
{
|
|
struct event ev;
|
|
int status;
|
|
|
|
current_codec = CODEC_IDX_AUDIO;
|
|
voice_codec_loaded = false;
|
|
while (1) {
|
|
status = 0;
|
|
voice_is_playing = false;
|
|
queue_wait(&voice_codec_queue, &ev);
|
|
switch (ev.id) {
|
|
case CODEC_LOAD_DISK:
|
|
logf("Loading voice codec");
|
|
audio_stop_playback();
|
|
mutex_lock(&mutex_codecthread);
|
|
current_codec = CODEC_IDX_VOICE;
|
|
dsp_configure(DSP_RESET, 0);
|
|
ci.configure(CODEC_DSP_ENABLE, (bool *)true);
|
|
voice_remaining = 0;
|
|
voice_getmore = NULL;
|
|
voice_codec_loaded = true;
|
|
reset_buffer();
|
|
ci_voice.stop_codec = false;
|
|
|
|
status = codec_load_file((char *)ev.data, &ci_voice);
|
|
|
|
logf("Voice codec finished");
|
|
audio_stop_playback();
|
|
mutex_unlock(&mutex_codecthread);
|
|
current_codec = CODEC_IDX_AUDIO;
|
|
voice_codec_loaded = false;
|
|
reset_buffer();
|
|
break ;
|
|
|
|
#ifndef SIMULATOR
|
|
case SYS_USB_CONNECTED:
|
|
logf("USB: Voice codec");
|
|
usb_acknowledge(SYS_USB_CONNECTED_ACK);
|
|
usb_wait_for_disconnect(&voice_codec_queue);
|
|
break ;
|
|
#endif
|
|
}
|
|
}
|
|
}
|
|
|
|
void voice_init(void)
|
|
{
|
|
if (!filebuf)
|
|
return; /* Audio buffers not yet set up */
|
|
|
|
while (voice_codec_loaded)
|
|
{
|
|
logf("Terminating voice codec");
|
|
ci_voice.stop_codec = true;
|
|
if (current_codec != CODEC_IDX_VOICE)
|
|
swap_codec();
|
|
sleep(1);
|
|
}
|
|
|
|
if (!talk_get_bufsize())
|
|
return ;
|
|
|
|
logf("Starting voice codec");
|
|
queue_post(&voice_codec_queue, CODEC_LOAD_DISK, (void *)CODEC_MPA_L3);
|
|
while (!voice_codec_loaded)
|
|
sleep(1);
|
|
}
|
|
|
|
struct mp3entry* audio_current_track(void)
|
|
{
|
|
// logf("audio_current_track");
|
|
|
|
if (track_count > 0 && cur_ti->taginfo_ready)
|
|
return (struct mp3entry *)&cur_ti->id3;
|
|
else
|
|
return NULL;
|
|
}
|
|
|
|
struct mp3entry* audio_next_track(void)
|
|
{
|
|
int next_idx = track_ridx + 1;
|
|
|
|
if (track_count == 0)
|
|
return NULL;
|
|
|
|
if (next_idx >= MAX_TRACK)
|
|
next_idx = 0;
|
|
|
|
if (!tracks[next_idx].taginfo_ready)
|
|
return NULL;
|
|
|
|
//logf("audio_next_track");
|
|
|
|
return &tracks[next_idx].id3;
|
|
}
|
|
|
|
bool audio_has_changed_track(void)
|
|
{
|
|
if (track_changed && track_count > 0 && playing) {
|
|
if (!cur_ti->taginfo_ready)
|
|
return false;
|
|
track_changed = false;
|
|
return true;
|
|
}
|
|
|
|
return false;
|
|
}
|
|
|
|
void audio_play(int offset)
|
|
{
|
|
logf("audio_play");
|
|
last_index = -1;
|
|
queue_post(&audio_queue, AUDIO_PLAY, (void *)offset);
|
|
}
|
|
|
|
void audio_stop(void)
|
|
{
|
|
logf("audio_stop");
|
|
queue_post(&audio_queue, AUDIO_STOP, 0);
|
|
while (playing || audio_codec_loaded)
|
|
yield();
|
|
}
|
|
|
|
bool mp3_pause_done(void)
|
|
{
|
|
return paused;
|
|
}
|
|
|
|
void audio_pause(void)
|
|
{
|
|
queue_post(&audio_queue, AUDIO_PAUSE, 0);
|
|
}
|
|
|
|
void audio_resume(void)
|
|
{
|
|
queue_post(&audio_queue, AUDIO_RESUME, 0);
|
|
}
|
|
|
|
void audio_next(void)
|
|
{
|
|
queue_post(&audio_queue, AUDIO_NEXT, 0);
|
|
}
|
|
|
|
void audio_prev(void)
|
|
{
|
|
queue_post(&audio_queue, AUDIO_PREV, 0);
|
|
}
|
|
|
|
void audio_ff_rewind(int newpos)
|
|
{
|
|
logf("rewind: %d", newpos);
|
|
if (playing) {
|
|
pcmbuf_play_stop();
|
|
ci.seek_time = newpos+1;
|
|
}
|
|
}
|
|
|
|
void audio_flush_and_reload_tracks(void)
|
|
{
|
|
logf("flush & reload");
|
|
queue_post(&audio_queue, AUDIO_FLUSH, 0);
|
|
}
|
|
|
|
void audio_error_clear(void)
|
|
{
|
|
}
|
|
|
|
int audio_status(void)
|
|
{
|
|
int ret = 0;
|
|
|
|
if (playing)
|
|
ret |= AUDIO_STATUS_PLAY;
|
|
|
|
if (paused)
|
|
ret |= AUDIO_STATUS_PAUSE;
|
|
|
|
return ret;
|
|
}
|
|
|
|
int audio_get_file_pos(void)
|
|
{
|
|
return 0;
|
|
}
|
|
|
|
|
|
/* Copied from mpeg.c. Should be moved somewhere else. */
|
|
static void mp3_set_elapsed(struct mp3entry* id3)
|
|
{
|
|
if ( id3->vbr ) {
|
|
if ( id3->has_toc ) {
|
|
/* calculate elapsed time using TOC */
|
|
int i;
|
|
unsigned int remainder, plen, relpos, nextpos;
|
|
|
|
/* find wich percent we're at */
|
|
for (i=0; i<100; i++ )
|
|
{
|
|
if ( id3->offset < id3->toc[i] * (id3->filesize / 256) )
|
|
{
|
|
break;
|
|
}
|
|
}
|
|
|
|
i--;
|
|
if (i < 0)
|
|
i = 0;
|
|
|
|
relpos = id3->toc[i];
|
|
|
|
if (i < 99)
|
|
{
|
|
nextpos = id3->toc[i+1];
|
|
}
|
|
else
|
|
{
|
|
nextpos = 256;
|
|
}
|
|
|
|
remainder = id3->offset - (relpos * (id3->filesize / 256));
|
|
|
|
/* set time for this percent (divide before multiply to prevent
|
|
overflow on long files. loss of precision is negligible on
|
|
short files) */
|
|
id3->elapsed = i * (id3->length / 100);
|
|
|
|
/* calculate remainder time */
|
|
plen = (nextpos - relpos) * (id3->filesize / 256);
|
|
id3->elapsed += (((remainder * 100) / plen) *
|
|
(id3->length / 10000));
|
|
}
|
|
else {
|
|
/* no TOC exists. set a rough estimate using average bitrate */
|
|
int tpk = id3->length / (id3->filesize / 1024);
|
|
id3->elapsed = id3->offset / 1024 * tpk;
|
|
}
|
|
}
|
|
else
|
|
/* constant bitrate, use exact calculation */
|
|
id3->elapsed = id3->offset / (id3->bitrate / 8);
|
|
}
|
|
|
|
/* Copied from mpeg.c. Should be moved somewhere else. */
|
|
int mp3_get_file_pos(void)
|
|
{
|
|
int pos = -1;
|
|
struct mp3entry *id3 = audio_current_track();
|
|
|
|
if (id3->vbr)
|
|
{
|
|
if (id3->has_toc)
|
|
{
|
|
/* Use the TOC to find the new position */
|
|
unsigned int percent, remainder;
|
|
int curtoc, nexttoc, plen;
|
|
|
|
percent = (id3->elapsed*100)/id3->length;
|
|
if (percent > 99)
|
|
percent = 99;
|
|
|
|
curtoc = id3->toc[percent];
|
|
|
|
if (percent < 99)
|
|
nexttoc = id3->toc[percent+1];
|
|
else
|
|
nexttoc = 256;
|
|
|
|
pos = (id3->filesize/256)*curtoc;
|
|
|
|
/* Use the remainder to get a more accurate position */
|
|
remainder = (id3->elapsed*100)%id3->length;
|
|
remainder = (remainder*100)/id3->length;
|
|
plen = (nexttoc - curtoc)*(id3->filesize/256);
|
|
pos += (plen/100)*remainder;
|
|
}
|
|
else
|
|
{
|
|
/* No TOC exists, estimate the new position */
|
|
pos = (id3->filesize / (id3->length / 1000)) *
|
|
(id3->elapsed / 1000);
|
|
}
|
|
}
|
|
else if (id3->bitrate)
|
|
pos = id3->elapsed * (id3->bitrate / 8);
|
|
else
|
|
{
|
|
return -1;
|
|
}
|
|
|
|
if (pos >= (int)(id3->filesize - id3->id3v1len))
|
|
{
|
|
/* Don't seek right to the end of the file so that we can
|
|
transition properly to the next song */
|
|
pos = id3->filesize - id3->id3v1len - 1;
|
|
}
|
|
else if (pos < (int)id3->first_frame_offset)
|
|
{
|
|
/* skip past id3v2 tag and other leading garbage */
|
|
pos = id3->first_frame_offset;
|
|
}
|
|
return pos;
|
|
}
|
|
|
|
void mp3_play_data(const unsigned char* start, int size,
|
|
void (*get_more)(unsigned char** start, int* size))
|
|
{
|
|
voice_getmore = get_more;
|
|
voicebuf = (unsigned char *)start;
|
|
voice_remaining = size;
|
|
voice_is_playing = true;
|
|
pcmbuf_reset_mixpos();
|
|
}
|
|
|
|
void audio_set_buffer_margin(int setting)
|
|
{
|
|
static const int lookup[] = {5, 15, 30, 60, 120, 180, 300, 600};
|
|
buffer_margin = lookup[setting];
|
|
logf("buffer margin: %ds", buffer_margin);
|
|
set_filebuf_watermark(buffer_margin);
|
|
}
|
|
|
|
/* Set crossfade & PCM buffer length. */
|
|
void audio_set_crossfade(int type)
|
|
{
|
|
long size;
|
|
bool was_playing = playing;
|
|
int offset = 0;
|
|
static const int lookup[] = {1, 2, 4, 6, 8, 10, 12, 14};
|
|
int seconds = lookup[global_settings.crossfade_duration];
|
|
|
|
if (!filebuf)
|
|
return; /* Audio buffers not yet set up */
|
|
|
|
/* Store the track resume position */
|
|
if (playing)
|
|
offset = cur_ti->id3.offset;
|
|
|
|
if (type == CROSSFADE_MODE_OFF)
|
|
seconds = 1;
|
|
|
|
/* Buffer has to be at least 2s long. */
|
|
seconds += 2;
|
|
logf("buf len: %d", seconds);
|
|
size = seconds * (NATIVE_FREQUENCY*4);
|
|
if (pcmbuf_get_bufsize() == size)
|
|
return ;
|
|
|
|
/* Playback has to be stopped before changing the buffer size. */
|
|
audio_stop_playback();
|
|
|
|
/* Re-initialize audio system. */
|
|
if (was_playing)
|
|
splash(0, true, str(LANG_RESTARTING_PLAYBACK));
|
|
pcmbuf_init(size);
|
|
pcmbuf_crossfade_enable(type != CROSSFADE_MODE_OFF);
|
|
reset_buffer();
|
|
logf("abuf:%dB", pcmbuf_get_bufsize());
|
|
logf("fbuf:%dB", filebuflen);
|
|
|
|
voice_init();
|
|
|
|
/* Restart playback. */
|
|
if (was_playing) {
|
|
audio_play(offset);
|
|
|
|
/* Wait for the playback to start again (and display the splash
|
|
screen during that period. */
|
|
playing = true;
|
|
while (playing && !audio_codec_loaded)
|
|
yield();
|
|
}
|
|
}
|
|
|
|
void mpeg_id3_options(bool _v1first)
|
|
{
|
|
v1first = _v1first;
|
|
}
|
|
|
|
void test_buffer_event(struct mp3entry *id3, bool last_track)
|
|
{
|
|
(void)id3;
|
|
(void)last_track;
|
|
|
|
logf("be:%d%s", last_track, id3->path);
|
|
}
|
|
|
|
void test_unbuffer_event(struct mp3entry *id3, bool last_track)
|
|
{
|
|
(void)id3;
|
|
(void)last_track;
|
|
|
|
logf("ube:%d%s", last_track, id3->path);
|
|
}
|
|
|
|
void audio_init(void)
|
|
{
|
|
static bool voicetagtrue = true;
|
|
|
|
logf("audio api init");
|
|
pcm_init();
|
|
filebufused = 0;
|
|
filling = false;
|
|
current_codec = CODEC_IDX_AUDIO;
|
|
filebuf = &audiobuf[MALLOC_BUFSIZE];
|
|
playing = false;
|
|
audio_codec_loaded = false;
|
|
voice_is_playing = false;
|
|
paused = false;
|
|
track_changed = false;
|
|
current_fd = -1;
|
|
track_buffer_callback = NULL;
|
|
track_unbuffer_callback = NULL;
|
|
track_changed_callback = NULL;
|
|
/* Just to prevent cur_ti never be anything random. */
|
|
cur_ti = &tracks[0];
|
|
|
|
audio_set_track_buffer_event(test_buffer_event);
|
|
audio_set_track_unbuffer_event(test_unbuffer_event);
|
|
|
|
/* Initialize codec api. */
|
|
ci.read_filebuf = codec_filebuf_callback;
|
|
ci.pcmbuf_insert = pcmbuf_insert_buffer;
|
|
ci.pcmbuf_insert_split = codec_pcmbuf_insert_split_callback;
|
|
ci.get_codec_memory = get_codec_memory_callback;
|
|
ci.request_buffer = codec_request_buffer_callback;
|
|
ci.advance_buffer = codec_advance_buffer_callback;
|
|
ci.advance_buffer_loc = codec_advance_buffer_loc_callback;
|
|
ci.request_next_track = codec_request_next_track_callback;
|
|
ci.mp3_get_filepos = codec_mp3_get_filepos_callback;
|
|
ci.seek_buffer = codec_seek_buffer_callback;
|
|
ci.seek_complete = codec_seek_complete_callback;
|
|
ci.set_elapsed = codec_set_elapsed_callback;
|
|
ci.set_offset = codec_set_offset_callback;
|
|
ci.configure = codec_configure_callback;
|
|
|
|
memcpy(&ci_voice, &ci, sizeof(struct codec_api));
|
|
memset(&id3_voice, 0, sizeof(struct mp3entry));
|
|
ci_voice.taginfo_ready = &voicetagtrue;
|
|
ci_voice.id3 = &id3_voice;
|
|
ci_voice.pcmbuf_insert = codec_pcmbuf_insert_callback;
|
|
id3_voice.frequency = 11200;
|
|
id3_voice.length = 1000000L;
|
|
|
|
mutex_init(&mutex_bufferfill);
|
|
mutex_init(&mutex_codecthread);
|
|
|
|
queue_init(&audio_queue);
|
|
queue_init(&codec_queue);
|
|
queue_init(&voice_codec_queue);
|
|
|
|
create_thread(codec_thread, codec_stack, sizeof(codec_stack),
|
|
codec_thread_name);
|
|
create_thread(voice_codec_thread, voice_codec_stack,
|
|
sizeof(voice_codec_stack), voice_codec_thread_name);
|
|
create_thread(audio_thread, audio_stack, sizeof(audio_stack),
|
|
audio_thread_name);
|
|
|
|
/* Apply relevant settings */
|
|
audio_set_buffer_margin(global_settings.buffer_margin);
|
|
audio_set_crossfade(global_settings.crossfade);
|
|
}
|
|
|
|
|