/*************************************************************************** * __________ __ ___. * Open \______ \ ____ ____ | | _\_ |__ _______ ___ * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ / * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < < * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \ * \/ \/ \/ \/ \/ * $Id$ * * Copyright (C) 2005 Miika Pekkarinen * * All files in this archive are subject to the GNU General Public License. * See the file COPYING in the source tree root for full license agreement. * * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY * KIND, either express or implied. * ****************************************************************************/ /* TODO: Can use the track changed callback to detect end of track and seek * in the previous track until this happens */ /* Design: we have prev_ti already, have a conditional for what type of seek * to do on a seek request, if it is a previous track seek, skip previous, * and in the request_next_track callback set the offset up the same way that * starting from an offset works. */ /* TODO: Pause should be handled in here, rather than PCMBUF so that voice can * play whilst audio is paused */ #include #include #include #include #include "system.h" #include "thread.h" #include "file.h" #include "lcd.h" #include "font.h" #include "button.h" #include "kernel.h" #include "tree.h" #include "debug.h" #include "sprintf.h" #include "settings.h" #include "codecs.h" #include "audio.h" #include "logf.h" #include "mp3_playback.h" #include "usb.h" #include "status.h" #include "main_menu.h" #include "ata.h" #include "screens.h" #include "playlist.h" #include "playback.h" #include "pcmbuf.h" #include "buffer.h" #include "dsp.h" #include "abrepeat.h" #ifdef HAVE_TAGCACHE #include "tagcache.h" #endif #ifdef HAVE_LCD_BITMAP #include "icons.h" #include "peakmeter.h" #include "action.h" #endif #include "lang.h" #include "bookmark.h" #include "misc.h" #include "sound.h" #include "metadata.h" #include "splash.h" #include "talk.h" #include "ata_idle_notify.h" #ifdef HAVE_RECORDING #include "recording.h" #include "talk.h" #endif #define PLAYBACK_VOICE /* default point to start buffer refill */ #define AUDIO_DEFAULT_WATERMARK (1024*512) /* amount of data to read in one read() call */ #define AUDIO_DEFAULT_FILECHUNK (1024*32) /* point at which the file buffer will fight for CPU time */ #define AUDIO_FILEBUF_CRITICAL (1024*128) /* amount of guess-space to allow for codecs that must hunt and peck * for their correct seeek target, 32k seems a good size */ #define AUDIO_REBUFFER_GUESS_SIZE (1024*32) /* macros to enable logf for queues logging on SYS_TIMEOUT can be disabled */ #ifdef SIMULATOR /* Define this for logf output of all queuing except SYS_TIMEOUT */ #define PLAYBACK_LOGQUEUES /* Define this to logf SYS_TIMEOUT messages */ #define PLAYBACK_LOGQUEUES_SYS_TIMEOUT #endif #ifdef PLAYBACK_LOGQUEUES #define LOGFQUEUE(s) logf("%s", s) #else #define LOGFQUEUE(s) #endif #ifdef PLAYBACK_LOGQUEUES_SYS_TIMEOUT #define LOGFQUEUE_SYS_TIMEOUT(s) logf("%s", s) #else #define LOGFQUEUE_SYS_TIMEOUT(s) #endif /* Define one constant that includes recording related functionality */ #if defined(HAVE_RECORDING) && !defined(SIMULATOR) #define AUDIO_HAVE_RECORDING #endif enum { Q_AUDIO_PLAY = 1, Q_AUDIO_STOP, Q_AUDIO_PAUSE, Q_AUDIO_SKIP, Q_AUDIO_PRE_FF_REWIND, Q_AUDIO_FF_REWIND, Q_AUDIO_REBUFFER_SEEK, Q_AUDIO_CHECK_NEW_TRACK, Q_AUDIO_FLUSH, Q_AUDIO_TRACK_CHANGED, Q_AUDIO_DIR_SKIP, Q_AUDIO_NEW_PLAYLIST, Q_AUDIO_POSTINIT, Q_AUDIO_FILL_BUFFER, #if MEM > 8 Q_AUDIO_FILL_BUFFER_IF_ACTIVE_ATA, #endif #ifdef AUDIO_HAVE_RECORDING Q_AUDIO_LOAD_ENCODER, #endif Q_CODEC_REQUEST_PENDING, Q_CODEC_REQUEST_COMPLETE, Q_CODEC_REQUEST_FAILED, Q_VOICE_PLAY, Q_VOICE_STOP, Q_CODEC_LOAD, Q_CODEC_LOAD_DISK, #ifdef AUDIO_HAVE_RECORDING Q_ENCODER_LOAD_DISK, Q_ENCODER_RECORD, #endif }; /* As defined in plugins/lib/xxx2wav.h */ #if MEM > 1 #define MALLOC_BUFSIZE (512*1024) #define GUARD_BUFSIZE (32*1024) #else #define MALLOC_BUFSIZE (100*1024) #define GUARD_BUFSIZE (8*1024) #endif /* As defined in plugin.lds */ #if CONFIG_CPU == PP5020 || CONFIG_CPU == PP5002 #define CODEC_IRAM_ORIGIN 0x4000c000 #define CODEC_IRAM_SIZE 0xc000 #elif defined(IAUDIO_X5) #define CODEC_IRAM_ORIGIN 0x10010000 #define CODEC_IRAM_SIZE 0x10000 #else #define CODEC_IRAM_ORIGIN 0x1000c000 #define CODEC_IRAM_SIZE 0xc000 #endif #ifndef IBSS_ATTR_VOICE_STACK #define IBSS_ATTR_VOICE_STACK IBSS_ATTR #endif #ifndef SIMULATOR extern bool audio_is_initialized; #else static bool audio_is_initialized = false; #endif /* Variables are commented with the threads that use them: * * A=audio, C=codec, V=voice. A suffix of - indicates that * * the variable is read but not updated on that thread. */ /* TBD: Split out "audio" and "playback" (ie. calling) threads */ /* Main state control */ static struct event_queue codec_callback_queue; /* Queue for codec callback responses */ static volatile bool audio_codec_loaded; /* Is codec loaded? (C/A-) */ static volatile bool playing; /* Is audio playing? (A) */ static volatile bool paused; /* Is audio paused? (A/C-) */ static volatile bool filling IDATA_ATTR; /* Is file buffer currently being refilled? (A/C-) */ /* Ring buffer where tracks and codecs are loaded */ static unsigned char *filebuf; /* Pointer to start of ring buffer (A/C-) */ size_t filebuflen; /* Total size of the ring buffer FIXME: make static (A/C-)*/ static volatile size_t buf_ridx IDATA_ATTR; /* Ring buffer read position (A/C) FIXME? should be (C/A-) */ static volatile size_t buf_widx IDATA_ATTR; /* Ring buffer read position (A/C-) */ #define BUFFER_STATE_TRASHED -1 /* Buffer is in a trashed state and must be reset */ #define BUFFER_STATE_NORMAL 0 /* Buffer is arranged for voice and audio */ #define BUFFER_STATE_VOICED_ONLY 1 /* Buffer is arranged for voice-only use */ static int buffer_state = BUFFER_STATE_TRASHED; /* Buffer state */ #define RINGBUF_ADD(p,v) ((p+v)=v) ? p-v : p+filebuflen-v) #define RINGBUF_ADD_CROSS(p1,v,p2) ((p1 voice Q_VOICE_STOP"); queue_post(&voice_queue, Q_VOICE_STOP, 0); LOGFQUEUE("mp3 > voice Q_VOICE_PLAY"); queue_post(&voice_queue, Q_VOICE_PLAY, &voice_clip); voice_thread_start = true; trigger_cpu_boost(); #else (void) start; (void) size; (void) get_more; #endif } void mp3_play_stop(void) { #ifdef PLAYBACK_VOICE queue_remove_from_head(&voice_queue, Q_VOICE_STOP); LOGFQUEUE("mp3 > voice Q_VOICE_STOP"); queue_post(&voice_queue, Q_VOICE_STOP, (void *)1); #endif } bool mp3_pause_done(void) { return pcm_is_paused(); } void mpeg_id3_options(bool _v1first) { v1first = _v1first; } unsigned char *audio_get_buffer(bool talk_buf, size_t *buffer_size) { unsigned char *buf = audiobuf; unsigned char *end = audiobufend; audio_stop(); if (talk_buf || !talk_voice_required() || buffer_state == BUFFER_STATE_TRASHED) { logf("get buffer: talk_buf"); /* ok to use everything from audiobuf to audiobufend */ if (buffer_state != BUFFER_STATE_TRASHED) talk_buffer_steal(); buffer_state = BUFFER_STATE_TRASHED; } else { /* skip talk buffer and move pcm buffer to end */ logf("get buffer: voice"); mp3_play_stop(); buf += talk_get_bufsize(); end -= pcmbuf_init(pcmbuf_get_bufsize(), audiobufend); buffer_state = BUFFER_STATE_VOICED_ONLY; } *buffer_size = end - buf; return buf; } #ifdef HAVE_RECORDING unsigned char *audio_get_recording_buffer(size_t *buffer_size) { /* don't allow overwrite of voice swap area or we'll trash the swapped-out voice codec but can use whole thing if none */ unsigned char *end = iram_buf[CODEC_IDX_VOICE] ? iram_buf[CODEC_IDX_VOICE] : audiobufend; audio_stop(); talk_buffer_steal(); buffer_state = BUFFER_STATE_TRASHED; *buffer_size = end - audiobuf; return (unsigned char *)audiobuf; } bool audio_load_encoder(int afmt) { #ifndef SIMULATOR const char *enc_fn = get_codec_filename(afmt | CODEC_TYPE_ENCODER); if (!enc_fn) return false; audio_remove_encoder(); ci.enc_codec_loaded = 0; /* clear any previous error condition */ LOGFQUEUE("audio > Q_AUDIO_LOAD_ENCODER"); queue_post(&audio_queue, Q_AUDIO_LOAD_ENCODER, (void *)enc_fn); while (ci.enc_codec_loaded == 0) yield(); logf("codec loaded: %d", ci.enc_codec_loaded); return ci.enc_codec_loaded > 0; #else (void)afmt; return true; #endif } /* audio_load_encoder */ void audio_remove_encoder(void) { #ifndef SIMULATOR /* force encoder codec unload (if currently loaded) */ if (ci.enc_codec_loaded <= 0) return; ci.stop_codec = true; while (ci.enc_codec_loaded > 0) yield(); #endif } /* audio_remove_encoder */ #endif /* HAVE_RECORDING */ struct mp3entry* audio_current_track(void) { const char *filename; const char *p; static struct mp3entry temp_id3; int cur_idx; int offset = ci.new_track + wps_offset; cur_idx = track_ridx + offset; cur_idx &= MAX_TRACK_MASK; if (tracks[cur_idx].taginfo_ready) return &tracks[cur_idx].id3; memset(&temp_id3, 0, sizeof(struct mp3entry)); filename = playlist_peek(offset); if (!filename) filename = "No file!"; #ifdef HAVE_TC_RAMCACHE if (tagcache_fill_tags(&temp_id3, filename)) return &temp_id3; #endif p = strrchr(filename, '/'); if (!p) p = filename; else p++; strncpy(temp_id3.path, p, sizeof(temp_id3.path)-1); temp_id3.title = &temp_id3.path[0]; return &temp_id3; } struct mp3entry* audio_next_track(void) { int next_idx = track_ridx; if (!audio_have_tracks()) return NULL; next_idx++; next_idx &= MAX_TRACK_MASK; if (!tracks[next_idx].taginfo_ready) return NULL; return &tracks[next_idx].id3; } bool audio_has_changed_track(void) { if (track_changed) { track_changed = false; return true; } return false; } void audio_play(long offset) { logf("audio_play"); #ifdef PLAYBACK_VOICE /* Truncate any existing voice output so we don't have spelling * etc. over the first part of the played track */ LOGFQUEUE("mp3 > voice Q_VOICE_STOP"); queue_post(&voice_queue, Q_VOICE_STOP, (void *)1); #endif /* Start playback */ if (playing && offset <= 0) { LOGFQUEUE("audio > audio Q_AUDIO_NEW_PLAYLIST"); queue_post(&audio_queue, Q_AUDIO_NEW_PLAYLIST, 0); } else { LOGFQUEUE("audio > audio Q_AUDIO_STOP"); queue_post(&audio_queue, Q_AUDIO_STOP, 0); LOGFQUEUE("audio > audio Q_AUDIO_PLAY"); queue_post(&audio_queue, Q_AUDIO_PLAY, (void *)offset); } /* Don't return until playback has actually started */ while (!playing) yield(); } void audio_stop(void) { /* Stop playback */ LOGFQUEUE("audio > audio Q_AUDIO_STOP"); queue_post(&audio_queue, Q_AUDIO_STOP, 0); /* Don't return until playback has actually stopped */ while(playing || !queue_empty(&audio_queue)) yield(); } void audio_pause(void) { LOGFQUEUE("audio > audio Q_AUDIO_PAUSE"); queue_post(&audio_queue, Q_AUDIO_PAUSE, (void *)true); } void audio_resume(void) { LOGFQUEUE("audio > audio Q_AUDIO_PAUSE resume"); queue_post(&audio_queue, Q_AUDIO_PAUSE, (void *)false); } void audio_next(void) { if (playlist_check(ci.new_track + wps_offset + 1)) { if (global_settings.beep) pcmbuf_beep(5000, 100, 2500*global_settings.beep); LOGFQUEUE("audio > audio Q_AUDIO_SKIP 1"); queue_post(&audio_queue, Q_AUDIO_SKIP, (void *)1); /* Keep wps fast while our message travels inside deep playback queues. */ wps_offset++; track_changed = true; } else { /* No more tracks. */ if (global_settings.beep) pcmbuf_beep(1000, 100, 1000*global_settings.beep); } } void audio_prev(void) { if (playlist_check(ci.new_track + wps_offset - 1)) { if (global_settings.beep) pcmbuf_beep(5000, 100, 2500*global_settings.beep); LOGFQUEUE("audio > audio Q_AUDIO_SKIP -1"); queue_post(&audio_queue, Q_AUDIO_SKIP, (void *)-1); /* Keep wps fast while our message travels inside deep playback queues. */ wps_offset--; track_changed = true; } else { /* No more tracks. */ if (global_settings.beep) pcmbuf_beep(1000, 100, 1000*global_settings.beep); } } void audio_next_dir(void) { LOGFQUEUE("audio > audio Q_AUDIO_DIR_SKIP 1"); queue_post(&audio_queue, Q_AUDIO_DIR_SKIP, (void *)1); } void audio_prev_dir(void) { LOGFQUEUE("audio > audio Q_AUDIO_DIR_SKIP -1"); queue_post(&audio_queue, Q_AUDIO_DIR_SKIP, (void *)-1); } void audio_pre_ff_rewind(void) { LOGFQUEUE("audio > audio Q_AUDIO_PRE_FF_REWIND"); queue_post(&audio_queue, Q_AUDIO_PRE_FF_REWIND, 0); } void audio_ff_rewind(long newpos) { LOGFQUEUE("audio > audio Q_AUDIO_FF_REWIND"); queue_post(&audio_queue, Q_AUDIO_FF_REWIND, (int *)newpos); } void audio_flush_and_reload_tracks(void) { LOGFQUEUE("audio > audio Q_AUDIO_FLUSH"); queue_post(&audio_queue, Q_AUDIO_FLUSH, 0); } void audio_error_clear(void) { #ifdef AUDIO_HAVE_RECORDING extern void pcm_rec_error_clear(void); pcm_rec_error_clear(); #endif } int audio_status(void) { int ret = 0; if (playing) ret |= AUDIO_STATUS_PLAY; if (paused) ret |= AUDIO_STATUS_PAUSE; #ifdef HAVE_RECORDING /* Do this here for constitency with mpeg.c version */ extern unsigned long pcm_rec_status(void); ret |= pcm_rec_status(); #endif return ret; } int audio_get_file_pos(void) { return 0; } 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 enable) { size_t size; bool was_playing = (playing && audio_is_initialized); size_t offset = 0; #if MEM > 1 int seconds = 1; #endif if (!filebuf) return; /* Audio buffers not yet set up */ #if MEM > 1 if (enable) seconds = global_settings.crossfade_fade_out_delay + global_settings.crossfade_fade_out_duration; /* Buffer has to be at least 2s long. */ seconds += 2; logf("buf len: %d", seconds); size = seconds * (NATIVE_FREQUENCY*4); #else enable = 0; size = NATIVE_FREQUENCY*2; #endif if (buffer_state == BUFFER_STATE_NORMAL && pcmbuf_is_same_size(size)) return ; if (was_playing) { /* Store the track resume position */ offset = CUR_TI->id3.offset; /* Playback has to be stopped before changing the buffer size. */ gui_syncsplash(0, true, (char *)str(LANG_RESTARTING_PLAYBACK)); audio_stop(); } voice_stop(); /* Re-initialize audio system. */ audio_reset_buffer(size); pcmbuf_crossfade_enable(enable); logf("abuf:%dB", pcmbuf_get_bufsize()); logf("fbuf:%dB", filebuflen); voice_init(); /* Restart playback. */ if (was_playing) audio_play(offset); } void audio_preinit(void) { logf("playback system pre-init"); filling = false; current_codec = CODEC_IDX_AUDIO; playing = false; paused = false; audio_codec_loaded = false; #ifdef PLAYBACK_VOICE voice_is_playing = false; voice_thread_start = false; voice_codec_loaded = false; #endif track_changed = false; current_fd = -1; track_buffer_callback = NULL; track_unbuffer_callback = NULL; track_changed_callback = NULL; track_ridx = 0; /* Just to prevent CUR_TI from being anything random. */ prev_ti = &tracks[MAX_TRACK-1]; /* And prevent prev_ti being random too */ #ifdef PLAYBACK_VOICE mutex_init(&mutex_codecthread); #endif queue_init(&audio_queue, true); queue_init(&codec_queue, true); /* create a private queue */ queue_init(&codec_callback_queue, false); create_thread(audio_thread, audio_stack, sizeof(audio_stack), audio_thread_name IF_PRIO(, PRIORITY_BUFFERING)); } void audio_init(void) { LOGFQUEUE("audio > audio Q_AUDIO_POSTINIT"); queue_post(&audio_queue, Q_AUDIO_POSTINIT, 0); } void voice_init(void) { #ifdef PLAYBACK_VOICE if (!filebuf) return; /* Audio buffers not yet set up */ if (voice_thread_p) return; if (!talk_voice_required()) return; logf("Starting voice codec"); queue_init(&voice_queue, true); voice_thread_p = create_thread(voice_thread, voice_stack, sizeof(voice_stack), voice_thread_name IF_PRIO(, PRIORITY_PLAYBACK)); while (!voice_codec_loaded) yield(); #endif } /* voice_init */ void voice_stop(void) { #ifdef PLAYBACK_VOICE /* Messages should not be posted to voice codec queue unless it is the current codec or deadlocks happen. */ if (current_codec != CODEC_IDX_VOICE) return; LOGFQUEUE("mp3 > voice Q_VOICE_STOP"); queue_post(&voice_queue, Q_VOICE_STOP, 0); while (voice_is_playing || !queue_empty(&voice_queue)) yield(); if (!playing) pcmbuf_play_stop(); #endif } /* voice_stop */ /* --- Routines called from multiple threads --- */ #ifdef PLAYBACK_VOICE static void swap_codec(void) { int my_codec = current_codec; logf("swapping out codec:%d", my_codec); /* Save our current IRAM and DRAM */ memcpy(iram_buf[my_codec], (unsigned char *)CODEC_IRAM_ORIGIN, CODEC_IRAM_SIZE); memcpy(dram_buf[my_codec], codecbuf, CODEC_SIZE); /* Release my semaphore */ mutex_unlock(&mutex_codecthread); /* Loop until the other codec has locked and run */ do { /* Release my semaphore and force a task switch. */ yield(); } while (my_codec == current_codec); /* Wait for other codec to unlock */ mutex_lock(&mutex_codecthread); /* Take control */ current_codec = my_codec; /* Reload our IRAM and DRAM */ memcpy((unsigned char *)CODEC_IRAM_ORIGIN, iram_buf[my_codec], CODEC_IRAM_SIZE); invalidate_icache(); memcpy(codecbuf, dram_buf[my_codec], CODEC_SIZE); logf("resuming codec:%d", my_codec); } #endif static void set_filebuf_watermark(int seconds) { size_t bytes; if (current_codec == CODEC_IDX_VOICE) return; if (!filebuf) return; /* Audio buffers not yet set up */ bytes = MAX(CUR_TI->id3.bitrate * seconds * (1000/8), conf_watermark); bytes = MIN(bytes, filebuflen / 2); conf_watermark = bytes; } static const char * get_codec_filename(int cod_spec) { const char *fname; #ifdef HAVE_RECORDING /* Can choose decoder or encoder if one available */ int type = cod_spec & CODEC_TYPE_MASK; int afmt = cod_spec & CODEC_AFMT_MASK; if ((unsigned)afmt >= AFMT_NUM_CODECS) type = AFMT_UNKNOWN | (type & CODEC_TYPE_MASK); fname = (type == CODEC_TYPE_ENCODER) ? audio_formats[afmt].codec_enc_root_fn : audio_formats[afmt].codec_root_fn; logf("%s: %d - %s", (type == CODEC_TYPE_ENCODER) ? "Encoder" : "Decoder", afmt, fname ? fname : ""); #else /* !HAVE_RECORDING */ /* Always decoder */ if ((unsigned)cod_spec >= AFMT_NUM_CODECS) cod_spec = AFMT_UNKNOWN; fname = audio_formats[cod_spec].codec_root_fn; logf("Codec: %d - %s", cod_spec, fname ? fname : ""); #endif /* HAVE_RECORDING */ return fname; } /* get_codec_filename */ /* --- Voice thread --- */ #ifdef PLAYBACK_VOICE static bool voice_pcmbuf_insert_split_callback( const void *ch1, const void *ch2, size_t length) { const char* src[2]; char *dest; long input_size; size_t output_size; src[0] = ch1; src[1] = ch2; if (dsp_stereo_mode() == STEREO_NONINTERLEAVED) length *= 2; /* Length is per channel */ while (length) { long est_output_size = dsp_output_size(length); while ((dest = pcmbuf_request_voice_buffer(est_output_size, &output_size, playing)) == NULL) { if (playing && audio_codec_loaded) swap_codec(); else yield(); } /* Get the real input_size for output_size bytes, guarding * against resampling buffer overflows. */ input_size = dsp_input_size(output_size); if (input_size <= 0) { DEBUGF("Error: dsp_input_size(%ld=dsp_output_size(%ld))=%ld<=0\n", output_size, length, input_size); /* If this happens, there are samples of codec data that don't * become a number of pcm samples, and something is broken */ return false; } /* Input size has grown, no error, just don't write more than length */ if ((size_t)input_size > length) input_size = length; output_size = dsp_process(dest, src, input_size); if (playing) { pcmbuf_mix_voice(output_size); if ((pcmbuf_usage() < 10 || pcmbuf_mix_free() < 30) && audio_codec_loaded) swap_codec(); } else pcmbuf_write_complete(output_size); length -= input_size; } return true; } /* voice_pcmbuf_insert_split_callback */ static bool voice_pcmbuf_insert_callback(const char *buf, size_t length) { /* TODO: The audiobuffer API should probably be updated, and be based on * pcmbuf_insert_split(). */ long real_length = length; if (dsp_stereo_mode() == STEREO_NONINTERLEAVED) length /= 2; /* Length is per channel */ /* Second channel is only used for non-interleaved stereo. */ return voice_pcmbuf_insert_split_callback(buf, buf + (real_length / 2), length); } static void* voice_get_memory_callback(size_t *size) { *size = 0; return NULL; } static void voice_set_elapsed_callback(unsigned int value) { (void)value; } static void voice_set_offset_callback(size_t value) { (void)value; } static size_t voice_filebuf_callback(void *ptr, size_t size) { (void)ptr; (void)size; return 0; } static void* voice_request_buffer_callback(size_t *realsize, size_t reqsize) { struct event ev; if (ci_voice.new_track) { *realsize = 0; return NULL; } while (1) { if (voice_is_playing || playing) queue_wait_w_tmo(&voice_queue, &ev, 0); else queue_wait(&voice_queue, &ev); if (!voice_is_playing) { if (ev.id == SYS_TIMEOUT) ev.id = Q_AUDIO_PLAY; } switch (ev.id) { case Q_AUDIO_PLAY: LOGFQUEUE("voice < Q_AUDIO_PLAY"); if (playing) { if (audio_codec_loaded) swap_codec(); yield(); } break; #ifdef AUDIO_HAVE_RECORDING case Q_ENCODER_RECORD: LOGFQUEUE("voice < Q_ENCODER_RECORD"); swap_codec(); break; #endif case Q_VOICE_STOP: LOGFQUEUE("voice < Q_VOICE_STOP"); if (ev.data == (void *)1 && !playing && pcm_is_playing()) { /* Aborting: Slight hack - flush PCM buffer if only being used for voice */ pcmbuf_play_stop(); } if (voice_is_playing) { /* Clear the current buffer */ voice_is_playing = false; voice_getmore = NULL; voice_remaining = 0; voicebuf = NULL; /* Force the codec to think it's changing tracks */ ci_voice.new_track = 1; *realsize = 0; return NULL; } else break; case SYS_USB_CONNECTED: LOGFQUEUE("voice < SYS_USB_CONNECTED"); usb_acknowledge(SYS_USB_CONNECTED_ACK); if (audio_codec_loaded) swap_codec(); usb_wait_for_disconnect(&voice_queue); break; case Q_VOICE_PLAY: LOGFQUEUE("voice < Q_VOICE_PLAY"); if (!voice_is_playing) { /* Set up new voice data */ struct voice_info *voice_data; voice_is_playing = true; trigger_cpu_boost(); voice_data = ev.data; voice_remaining = voice_data->size; voicebuf = voice_data->buf; voice_getmore = voice_data->callback; } goto voice_play_clip; case SYS_TIMEOUT: LOGFQUEUE_SYS_TIMEOUT("voice < SYS_TIMEOUT"); goto voice_play_clip; default: LOGFQUEUE("voice < default"); } } voice_play_clip: if (voice_remaining == 0 || voicebuf == NULL) { if (voice_getmore) voice_getmore((unsigned char **)&voicebuf, (int *)&voice_remaining); /* If this clip is done */ if (voice_remaining == 0) { LOGFQUEUE("voice > voice Q_VOICE_STOP"); queue_post(&voice_queue, Q_VOICE_STOP, 0); /* Force pcm playback. */ if (!pcm_is_playing()) pcmbuf_play_start(); } } *realsize = MIN(voice_remaining, reqsize); if (*realsize == 0) return NULL; return voicebuf; } /* voice_request_buffer_callback */ static void voice_advance_buffer_callback(size_t amount) { amount = MIN(amount, voice_remaining); voicebuf += amount; voice_remaining -= amount; } static void voice_advance_buffer_loc_callback(void *ptr) { size_t amount = (size_t)ptr - (size_t)voicebuf; voice_advance_buffer_callback(amount); } static off_t voice_mp3_get_filepos_callback(int newtime) { (void)newtime; return 0; } static void voice_do_nothing(void) { return; } static bool voice_seek_buffer_callback(size_t newpos) { (void)newpos; return false; } static bool voice_request_next_track_callback(void) { ci_voice.new_track = 0; return true; } static void voice_thread(void) { while (1) { logf("Loading voice codec"); voice_codec_loaded = true; mutex_lock(&mutex_codecthread); current_codec = CODEC_IDX_VOICE; dsp_configure(DSP_RESET, 0); voice_remaining = 0; voice_getmore = NULL; codec_load_file(get_codec_filename(AFMT_MPA_L3), &ci_voice); logf("Voice codec finished"); voice_codec_loaded = false; mutex_unlock(&mutex_codecthread); } } /* voice_thread */ #endif /* PLAYBACK_VOICE */ /* --- Codec thread --- */ static bool codec_pcmbuf_insert_split_callback( const void *ch1, const void *ch2, size_t length) { const char* src[2]; char *dest; long input_size; size_t output_size; src[0] = ch1; src[1] = ch2; if (dsp_stereo_mode() == STEREO_NONINTERLEAVED) length *= 2; /* Length is per channel */ while (length) { long est_output_size = dsp_output_size(length); /* Prevent audio from a previous track from playing */ if (ci.new_track || ci.stop_codec) return true; while ((dest = pcmbuf_request_buffer(est_output_size, &output_size)) == NULL) { sleep(1); if (ci.seek_time || ci.new_track || ci.stop_codec) return true; } /* Get the real input_size for output_size bytes, guarding * against resampling buffer overflows. */ input_size = dsp_input_size(output_size); if (input_size <= 0) { DEBUGF("Error: dsp_input_size(%ld=dsp_output_size(%ld))=%ld<=0\n", output_size, length, input_size); /* If this happens, there are samples of codec data that don't * become a number of pcm samples, and something is broken */ return false; } /* Input size has grown, no error, just don't write more than length */ if ((size_t)input_size > length) input_size = length; output_size = dsp_process(dest, src, input_size); pcmbuf_write_complete(output_size); #ifdef PLAYBACK_VOICE if ((voice_is_playing || voice_thread_start) && pcm_is_playing() && voice_codec_loaded && pcmbuf_usage() > 30 && pcmbuf_mix_free() > 80) { voice_thread_start = false; swap_codec(); } #endif length -= input_size; } return true; } /* codec_pcmbuf_insert_split_callback */ static bool codec_pcmbuf_insert_callback(const char *buf, size_t length) { /* TODO: The audiobuffer API should probably be updated, and be based on * pcmbuf_insert_split(). */ long real_length = length; if (dsp_stereo_mode() == STEREO_NONINTERLEAVED) length /= 2; /* Length is per channel */ /* Second channel is only used for non-interleaved stereo. */ return codec_pcmbuf_insert_split_callback(buf, buf + (real_length / 2), length); } static void* codec_get_memory_callback(size_t *size) { *size = MALLOC_BUFSIZE; return &audiobuf[talk_get_bufsize()]; } static void codec_pcmbuf_position_callback(size_t size) ICODE_ATTR; static void codec_pcmbuf_position_callback(size_t size) { /* This is called from an ISR, so be quick */ unsigned int time = size * 1000 / 4 / NATIVE_FREQUENCY + prev_ti->id3.elapsed; if (time >= prev_ti->id3.length) { pcmbuf_set_position_callback(NULL); prev_ti->id3.elapsed = prev_ti->id3.length; } else prev_ti->id3.elapsed = time; } static void codec_set_elapsed_callback(unsigned int value) { unsigned int latency; if (ci.seek_time) return; #ifdef AB_REPEAT_ENABLE ab_position_report(value); #endif latency = pcmbuf_get_latency(); if (value < latency) CUR_TI->id3.elapsed = 0; else if (value - latency > CUR_TI->id3.elapsed || value - latency < CUR_TI->id3.elapsed - 2) { CUR_TI->id3.elapsed = value - latency; } } static void codec_set_offset_callback(size_t value) { unsigned int latency; if (ci.seek_time) return; latency = pcmbuf_get_latency() * CUR_TI->id3.bitrate / 8; if (value < latency) CUR_TI->id3.offset = 0; else CUR_TI->id3.offset = value - latency; } static void codec_advance_buffer_counters(size_t amount) { buf_ridx = RINGBUF_ADD(buf_ridx, amount); ci.curpos += amount; CUR_TI->available -= amount; /* Start buffer filling as necessary. */ if (!pcmbuf_is_lowdata() && !filling) { if (FILEBUFUSED < conf_watermark && playing && !playlist_end) { LOGFQUEUE("codec > audio Q_AUDIO_FILL_BUFFER"); queue_post(&audio_queue, Q_AUDIO_FILL_BUFFER, 0); } } } /* copy up-to size bytes into ptr and return the actual size copied */ static size_t codec_filebuf_callback(void *ptr, size_t size) { char *buf = (char *)ptr; size_t copy_n; size_t part_n; if (ci.stop_codec || !playing) return 0; /* The ammount to copy is the lesser of the requested amount and the * amount left of the current track (both on disk and already loaded) */ copy_n = MIN(size, CUR_TI->available + CUR_TI->filerem); /* Nothing requested OR nothing left */ if (copy_n == 0) return 0; /* Let the disk buffer catch fill until enough data is available */ while (copy_n > CUR_TI->available) { if (!filling) { LOGFQUEUE("codec > audio Q_AUDIO_FILL_BUFFER"); queue_post(&audio_queue, Q_AUDIO_FILL_BUFFER, 0); } sleep(1); if (ci.stop_codec || ci.new_track) return 0; } /* Copy as much as possible without wrapping */ part_n = MIN(copy_n, filebuflen - buf_ridx); memcpy(buf, &filebuf[buf_ridx], part_n); /* Copy the rest in the case of a wrap */ if (part_n < copy_n) { memcpy(&buf[part_n], &filebuf[0], copy_n - part_n); } /* Update read and other position pointers */ codec_advance_buffer_counters(copy_n); /* Return the actual amount of data copied to the buffer */ return copy_n; } /* codec_filebuf_callback */ static void* codec_request_buffer_callback(size_t *realsize, size_t reqsize) { size_t short_n, copy_n, buf_rem; if (!playing) { *realsize = 0; return NULL; } copy_n = MIN(reqsize, CUR_TI->available + CUR_TI->filerem); if (copy_n == 0) { *realsize = 0; return NULL; } while (copy_n > CUR_TI->available) { if (!filling) { LOGFQUEUE("codec > audio Q_AUDIO_FILL_BUFFER"); queue_post(&audio_queue, Q_AUDIO_FILL_BUFFER, 0); } sleep(1); if (ci.stop_codec || ci.new_track) { *realsize = 0; return NULL; } } /* How much is left at the end of the file buffer before wrap? */ buf_rem = filebuflen - buf_ridx; /* If we can't satisfy the request without wrapping */ if (buf_rem < copy_n) { /* How short are we? */ short_n = copy_n - buf_rem; /* If we can fudge it with the guardbuf */ if (short_n < GUARD_BUFSIZE) memcpy(&filebuf[filebuflen], &filebuf[0], short_n); else copy_n = buf_rem; } *realsize = copy_n; return (char *)&filebuf[buf_ridx]; } /* codec_request_buffer_callback */ 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; } static void codec_advance_buffer_callback(size_t amount) { if (amount > CUR_TI->available + CUR_TI->filerem) amount = CUR_TI->available + CUR_TI->filerem; while (amount > CUR_TI->available && filling) sleep(1); if (amount > CUR_TI->available) { struct event ev; LOGFQUEUE("codec > audio Q_AUDIO_REBUFFER_SEEK"); queue_post(&audio_queue, Q_AUDIO_REBUFFER_SEEK, (void *)(ci.curpos + amount)); queue_wait(&codec_callback_queue, &ev); switch (ev.id) { case Q_CODEC_REQUEST_FAILED: LOGFQUEUE("codec < Q_CODEC_REQUEST_FAILED"); ci.stop_codec = true; return; case Q_CODEC_REQUEST_COMPLETE: LOGFQUEUE("codec < Q_CODEC_REQUEST_COMPLETE"); return; default: LOGFQUEUE("codec < default"); ci.stop_codec = true; return; } } codec_advance_buffer_counters(amount); codec_set_offset_callback(ci.curpos); } static void codec_advance_buffer_loc_callback(void *ptr) { size_t amount = (size_t)ptr - (size_t)&filebuf[buf_ridx]; codec_advance_buffer_callback(amount); } /* Copied from mpeg.c. Should be moved somewhere else. */ static int codec_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; pos += id3->first_frame_offset; /* Don't seek right to the end of the file so that we can transition properly to the next song */ if (pos >= (int)(id3->filesize - id3->id3v1len)) pos = id3->filesize - id3->id3v1len - 1; return pos; } static off_t codec_mp3_get_filepos_callback(int newtime) { off_t newpos; CUR_TI->id3.elapsed = newtime; newpos = codec_get_file_pos(); return newpos; } static void codec_seek_complete_callback(void) { logf("seek_complete"); if (pcm_is_paused()) { /* If this is not a seamless seek, clear the buffer */ pcmbuf_play_stop(); /* If playback was not 'deliberately' paused, unpause now */ if (!paused) pcmbuf_pause(false); } ci.seek_time = 0; } static bool codec_seek_buffer_callback(size_t newpos) { int difference; logf("codec_seek_buffer_callback"); if (newpos >= CUR_TI->filesize) newpos = CUR_TI->filesize - 1; difference = newpos - ci.curpos; if (difference >= 0) { /* Seeking forward */ 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) { struct event ev; LOGFQUEUE("codec > audio Q_AUDIO_REBUFFER_SEEK"); queue_post(&audio_queue, Q_AUDIO_REBUFFER_SEEK, (void *)newpos); queue_wait(&codec_callback_queue, &ev); switch (ev.id) { case Q_CODEC_REQUEST_COMPLETE: LOGFQUEUE("codec < Q_CODEC_REQUEST_COMPLETE"); return true; case Q_CODEC_REQUEST_FAILED: LOGFQUEUE("codec < Q_CODEC_REQUEST_FAILED"); ci.stop_codec = true; return false; default: LOGFQUEUE("codec < default"); return false; } } /* Seeking inside buffer space. */ logf("seek: -%d", difference); CUR_TI->available += difference; buf_ridx = RINGBUF_SUB(buf_ridx, (unsigned)difference); ci.curpos -= difference; return true; } static void codec_configure_callback(int setting, void *value) { switch (setting) { case CODEC_SET_FILEBUF_WATERMARK: conf_watermark = (unsigned long)value; set_filebuf_watermark(buffer_margin); break; case CODEC_SET_FILEBUF_CHUNKSIZE: conf_filechunk = (unsigned long)value; break; case CODEC_SET_FILEBUF_PRESEEK: conf_preseek = (unsigned long)value; break; default: if (!dsp_configure(setting, value)) { logf("Illegal key:%d", setting); } } } static void codec_track_changed(void) { automatic_skip = false; LOGFQUEUE("codec > audio Q_AUDIO_TRACK_CHANGED"); queue_post(&audio_queue, Q_AUDIO_TRACK_CHANGED, 0); } static void codec_pcmbuf_track_changed_callback(void) { pcmbuf_set_position_callback(NULL); codec_track_changed(); } static void codec_discard_codec_callback(void) { if (CUR_TI->has_codec) { CUR_TI->has_codec = false; buf_ridx = RINGBUF_ADD(buf_ridx, CUR_TI->codecsize); } #if 0 /* Check if a buffer desync has happened, log it and stop playback. */ if (buf_ridx != CUR_TI->buf_idx) { int offset = CUR_TI->buf_idx - buf_ridx; size_t new_used = FILEBUFUSED - offset; logf("Buf off :%d=%d-%d", offset, CUR_TI->buf_idx, buf_ridx); logf("Used off:%d",FILEBUFUSED - new_used); /* This is a fatal internal error and it's not safe to * continue playback. */ ci.stop_codec = true; queue_post(&audio_queue, Q_AUDIO_STOP, 0); } #endif } static void codec_track_skip_done(bool was_manual) { /* Manual track change (always crossfade or flush audio). */ if (was_manual) { pcmbuf_crossfade_init(true); LOGFQUEUE("codec > audio Q_AUDIO_TRACK_CHANGED"); queue_post(&audio_queue, Q_AUDIO_TRACK_CHANGED, 0); } /* Automatic track change w/crossfade, if not in "Track Skip Only" mode. */ else if (pcmbuf_is_crossfade_enabled() && !pcmbuf_is_crossfade_active() && global_settings.crossfade != 2) { pcmbuf_crossfade_init(false); codec_track_changed(); } /* Gapless playback. */ else { pcmbuf_set_position_callback(codec_pcmbuf_position_callback); pcmbuf_set_event_handler(codec_pcmbuf_track_changed_callback); } } static bool codec_load_next_track(void) { struct event ev; prev_track_elapsed = CUR_TI->id3.elapsed; if (ci.seek_time) codec_seek_complete_callback(); #ifdef AB_REPEAT_ENABLE ab_end_of_track_report(); #endif logf("Request new track"); if (ci.new_track == 0) { ci.new_track++; automatic_skip = true; } trigger_cpu_boost(); LOGFQUEUE("codec > audio Q_AUDIO_CHECK_NEW_TRACK"); queue_post(&audio_queue, Q_AUDIO_CHECK_NEW_TRACK, 0); while (1) { queue_wait(&codec_callback_queue, &ev); if (ev.id == Q_CODEC_REQUEST_PENDING) { if (!automatic_skip) pcmbuf_play_stop(); } else break; } switch (ev.id) { case Q_CODEC_REQUEST_COMPLETE: LOGFQUEUE("codec < Q_CODEC_REQUEST_COMPLETE"); codec_track_skip_done(!automatic_skip); return true; case Q_CODEC_REQUEST_FAILED: LOGFQUEUE("codec < Q_CODEC_REQUEST_FAILED"); ci.new_track = 0; ci.stop_codec = true; return false; default: LOGFQUEUE("codec < default"); ci.stop_codec = true; return false; } } static bool codec_request_next_track_callback(void) { int prev_codectype; if (ci.stop_codec || !playing) return false; prev_codectype = get_codec_base_type(CUR_TI->id3.codectype); if (!codec_load_next_track()) return false; /* Check if the next codec is the same file. */ if (prev_codectype == get_codec_base_type(CUR_TI->id3.codectype)) { logf("New track loaded"); codec_discard_codec_callback(); return true; } else { logf("New codec:%d/%d", CUR_TI->id3.codectype, prev_codectype); return false; } } static void codec_thread(void) { struct event ev; int status; size_t wrap; while (1) { status = 0; queue_wait(&codec_queue, &ev); switch (ev.id) { case Q_CODEC_LOAD_DISK: LOGFQUEUE("codec < Q_CODEC_LOAD_DISK"); audio_codec_loaded = true; #ifdef PLAYBACK_VOICE /* Don't sent messages to voice codec if it's not current */ if (voice_codec_loaded && current_codec == CODEC_IDX_VOICE) { LOGFQUEUE("codec > voice Q_AUDIO_PLAY"); queue_post(&voice_queue, Q_AUDIO_PLAY, 0); } mutex_lock(&mutex_codecthread); #endif current_codec = CODEC_IDX_AUDIO; ci.stop_codec = false; status = codec_load_file((const char *)ev.data, &ci); #ifdef PLAYBACK_VOICE mutex_unlock(&mutex_codecthread); #endif break ; case Q_CODEC_LOAD: LOGFQUEUE("codec < Q_CODEC_LOAD"); if (!CUR_TI->has_codec) { logf("Codec slot is empty!"); /* Wait for the pcm buffer to go empty */ while (pcm_is_playing()) yield(); /* This must be set to prevent an infinite loop */ ci.stop_codec = true; LOGFQUEUE("codec > codec Q_AUDIO_PLAY"); queue_post(&codec_queue, Q_AUDIO_PLAY, 0); break ; } audio_codec_loaded = true; #ifdef PLAYBACK_VOICE if (voice_codec_loaded && current_codec == CODEC_IDX_VOICE) { LOGFQUEUE("codec > voice Q_AUDIO_PLAY"); queue_post(&voice_queue, Q_AUDIO_PLAY, 0); } mutex_lock(&mutex_codecthread); #endif current_codec = CODEC_IDX_AUDIO; ci.stop_codec = false; wrap = (size_t)&filebuf[filebuflen] - (size_t)CUR_TI->codecbuf; status = codec_load_ram(CUR_TI->codecbuf, CUR_TI->codecsize, &filebuf[0], wrap, &ci); #ifdef PLAYBACK_VOICE mutex_unlock(&mutex_codecthread); #endif break ; #ifdef AUDIO_HAVE_RECORDING case Q_ENCODER_LOAD_DISK: LOGFQUEUE("codec < Q_ENCODER_LOAD_DISK"); audio_codec_loaded = false; /* Not audio codec! */ #ifdef PLAYBACK_VOICE if (voice_codec_loaded && current_codec == CODEC_IDX_VOICE) { LOGFQUEUE("codec > voice Q_ENCODER_RECORD"); queue_post(&voice_queue, Q_ENCODER_RECORD, NULL); } mutex_lock(&mutex_codecthread); #endif logf("loading encoder"); current_codec = CODEC_IDX_AUDIO; ci.stop_codec = false; status = codec_load_file((const char *)ev.data, &ci); #ifdef PLAYBACK_VOICE mutex_unlock(&mutex_codecthread); #endif logf("encoder stopped"); break; #endif /* AUDIO_HAVE_RECORDING */ #ifndef SIMULATOR case SYS_USB_CONNECTED: LOGFQUEUE("codec < SYS_USB_CONNECTED"); queue_clear(&codec_queue); usb_acknowledge(SYS_USB_CONNECTED_ACK); usb_wait_for_disconnect(&codec_queue); break; #endif default: LOGFQUEUE("codec < default"); } if (audio_codec_loaded) { if (ci.stop_codec) { status = CODEC_OK; if (!playing) pcmbuf_play_stop(); } audio_codec_loaded = false; } switch (ev.id) { case Q_CODEC_LOAD_DISK: case Q_CODEC_LOAD: LOGFQUEUE("codec < Q_CODEC_LOAD"); if (playing) { if (ci.new_track || status != CODEC_OK) { if (!ci.new_track) { logf("Codec failure"); gui_syncsplash(HZ*2, true, "Codec failure"); } if (!codec_load_next_track()) { // queue_post(&codec_queue, Q_AUDIO_STOP, 0); LOGFQUEUE("codec > audio Q_AUDIO_STOP"); queue_post(&audio_queue, Q_AUDIO_STOP, 0); break; } } else { logf("Codec finished"); if (ci.stop_codec) { /* Wait for the audio to stop playing before * triggering the WPS exit */ while(pcm_is_playing()) { CUR_TI->id3.elapsed = CUR_TI->id3.length - pcmbuf_get_latency(); sleep(1); } LOGFQUEUE("codec > audio Q_AUDIO_STOP"); queue_post(&audio_queue, Q_AUDIO_STOP, 0); break; } } if (CUR_TI->has_codec) { LOGFQUEUE("codec > codec Q_CODEC_LOAD"); queue_post(&codec_queue, Q_CODEC_LOAD, 0); } else { const char *codec_fn = get_codec_filename(CUR_TI->id3.codectype); LOGFQUEUE("codec > codec Q_CODEC_LOAD_DISK"); queue_post(&codec_queue, Q_CODEC_LOAD_DISK, (void *)codec_fn); } } break; #ifdef AUDIO_HAVE_RECORDING case Q_ENCODER_LOAD_DISK: LOGFQUEUE("codec < Q_ENCODER_LOAD_DISK"); if (status == CODEC_OK) break; logf("Encoder failure"); gui_syncsplash(HZ*2, true, "Encoder failure"); if (ci.enc_codec_loaded < 0) break; logf("Encoder failed to load"); ci.enc_codec_loaded = -1; break; #endif /* AUDIO_HAVE_RECORDING */ default: LOGFQUEUE("codec < default"); } /* end switch */ } } /* --- Audio thread --- */ static bool audio_filebuf_is_lowdata(void) { return FILEBUFUSED < AUDIO_FILEBUF_CRITICAL; } static bool audio_have_tracks(void) { return track_ridx != track_widx || CUR_TI->filesize; } static bool audio_have_free_tracks(void) { if (track_widx < track_ridx) return track_widx + 1 < track_ridx; else if (track_ridx == 0) return track_widx < MAX_TRACK - 1; return true; } int audio_track_count(void) { if (audio_have_tracks()) { int relative_track_widx = track_widx; if (track_ridx > track_widx) relative_track_widx += MAX_TRACK; return relative_track_widx - track_ridx + 1; } return 0; } long audio_filebufused(void) { return (long) FILEBUFUSED; } /* Count the data BETWEEN the selected tracks */ static size_t audio_buffer_count_tracks(int from_track, int to_track) { size_t amount = 0; bool need_wrap = to_track < from_track; while (1) { if (++from_track >= MAX_TRACK) { from_track -= MAX_TRACK; need_wrap = false; } if (from_track >= to_track && !need_wrap) break; amount += tracks[from_track].codecsize + tracks[from_track].filesize; } return amount; } static bool audio_buffer_wind_forward(int new_track_ridx, int old_track_ridx) { size_t amount; /* Start with the remainder of the previously playing track */ amount = tracks[old_track_ridx].filesize - ci.curpos; /* Then collect all data from tracks in between them */ amount += audio_buffer_count_tracks(old_track_ridx, new_track_ridx); logf("bwf:%ldB", (long) amount); if (amount > FILEBUFUSED) return false; /* Wind the buffer to the beginning of the target track or its codec */ buf_ridx = RINGBUF_ADD(buf_ridx, amount); return true; } static bool audio_buffer_wind_backward(int new_track_ridx, int old_track_ridx) { /* Available buffer data */ size_t buf_back; /* Start with the previously playing track's data and our data */ size_t amount; amount = ci.curpos; buf_back = RINGBUF_SUB(buf_ridx, buf_widx); /* If we're not just resetting the current track */ if (new_track_ridx != old_track_ridx) { /* Need to wind to before the old track's codec and our filesize */ amount += tracks[old_track_ridx].codecsize; amount += tracks[new_track_ridx].filesize; /* Rewind the old track to its beginning */ tracks[old_track_ridx].available = tracks[old_track_ridx].filesize - tracks[old_track_ridx].filerem; } /* If the codec was ever buffered */ if (tracks[new_track_ridx].codecsize) { /* Add the codec to the needed size */ amount += tracks[new_track_ridx].codecsize; tracks[new_track_ridx].has_codec = true; } /* Then collect all data from tracks between new and old */ amount += audio_buffer_count_tracks(new_track_ridx, old_track_ridx); /* Do we have space to make this skip? */ if (amount > buf_back) return false; logf("bwb:%ldB",amount); /* Rewind the buffer to the beginning of the target track or its codec */ buf_ridx = RINGBUF_SUB(buf_ridx, amount); /* Reset to the beginning of the new track */ tracks[new_track_ridx].available = tracks[new_track_ridx].filesize; return true; } static void audio_update_trackinfo(void) { ci.filesize = CUR_TI->filesize; CUR_TI->id3.elapsed = 0; CUR_TI->id3.offset = 0; ci.id3 = &CUR_TI->id3; ci.curpos = 0; ci.taginfo_ready = &CUR_TI->taginfo_ready; } /* Yield to codecs for as long as possible if they are in need of data * return true if the caller should break to let the audio thread process * new events */ static bool audio_yield_codecs(void) { yield(); if (!queue_empty(&audio_queue)) return true; while ((pcmbuf_is_crossfade_active() || pcmbuf_is_lowdata()) && !ci.stop_codec && playing && !audio_filebuf_is_lowdata()) { if (filling) yield(); else sleep(2); if (!queue_empty(&audio_queue)) return true; } return false; } static void audio_clear_track_entries(bool clear_unbuffered) { int cur_idx = track_widx; int last_idx = -1; logf("Clearing tracks:%d/%d, %d", track_ridx, track_widx, clear_unbuffered); /* Loop over all tracks from write-to-read */ while (1) { cur_idx++; cur_idx &= MAX_TRACK_MASK; if (cur_idx == track_ridx) break; /* If the track is buffered, conditionally clear/notify, * otherwise clear the track if that option is selected */ if (tracks[cur_idx].event_sent) { if (last_idx >= 0) { /* If there is an unbuffer callback, call it, otherwise, * just clear the track */ if (track_unbuffer_callback) track_unbuffer_callback(&tracks[last_idx].id3, false); memset(&tracks[last_idx], 0, sizeof(struct track_info)); } last_idx = cur_idx; } else if (clear_unbuffered) memset(&tracks[cur_idx], 0, sizeof(struct track_info)); } /* We clear the previous instance of a buffered track throughout * the above loop to facilitate 'last' detection. Clear/notify * the last track here */ if (last_idx >= 0) { if (track_unbuffer_callback) track_unbuffer_callback(&tracks[last_idx].id3, true); memset(&tracks[last_idx], 0, sizeof(struct track_info)); } } /* FIXME: This code should be made more generic and move to metadata.c */ static void audio_strip_tags(void) { int i; static const unsigned char tag[] = "TAG"; static const unsigned char apetag[] = "APETAGEX"; size_t tag_idx; size_t cur_idx; size_t len, version; tag_idx = RINGBUF_SUB(buf_widx, 128); if (FILEBUFUSED > 128 && tag_idx > buf_ridx) { cur_idx = tag_idx; for(i = 0;i < 3;i++) { if(filebuf[cur_idx] != tag[i]) goto strip_ape_tag; cur_idx = RINGBUF_ADD(cur_idx, 1); } /* Skip id3v1 tag */ logf("Skipping ID3v1 tag"); buf_widx = tag_idx; tracks[track_widx].available -= 128; tracks[track_widx].filesize -= 128; } strip_ape_tag: /* Check for APE tag (look for the APE tag footer) */ tag_idx = RINGBUF_SUB(buf_widx, 32); if (FILEBUFUSED > 32 && tag_idx > buf_ridx) { cur_idx = tag_idx; for(i = 0;i < 8;i++) { if(filebuf[cur_idx] != apetag[i]) return; cur_idx = RINGBUF_ADD(cur_idx, 1); } /* Read the version and length from the footer */ version = letoh32(*(long *)(filebuf + tag_idx + 8)); len = letoh32(*(long *)(filebuf + tag_idx + 12)); if (version == 2000) len += 32; /* APEv2 has a 32 byte header */ /* Skip APE tag */ if (FILEBUFUSED > len) { logf("Skipping APE tag (%dB)", len); buf_widx = RINGBUF_SUB(buf_widx, len); tracks[track_widx].available -= len; tracks[track_widx].filesize -= len; } } } /* Returns true if a whole file is read, false otherwise */ static bool audio_read_file(size_t minimum) { bool ret_val = false; /* If we're called and no file is open, this is an error */ if (current_fd < 0) { logf("Bad fd in arf"); /* Give some hope of miraculous recovery by forcing a track reload */ tracks[track_widx].filesize = 0; /* Stop this buffering run */ return ret_val; } trigger_cpu_boost(); while (tracks[track_widx].filerem > 0) { size_t copy_n; int overlap; int rc; /* copy_n is the largest chunk that is safe to read */ copy_n = MIN(conf_filechunk, filebuflen - buf_widx); /* buf_widx == buf_ridx is defined as buffer empty, not buffer full */ if (RINGBUF_ADD_CROSS(buf_widx,copy_n,buf_ridx) >= 0) break; /* rc is the actual amount read */ rc = read(current_fd, &filebuf[buf_widx], copy_n); if (rc < 0) { logf("File ended %dB early", tracks[track_widx].filerem); tracks[track_widx].filesize -= tracks[track_widx].filerem; tracks[track_widx].filerem = 0; break; } /* How much of the playing track did we overwrite */ if (buf_widx == CUR_TI->buf_idx) { /* Special handling; zero or full overlap? */ if (track_widx == track_ridx && CUR_TI->available == 0) overlap = 0; else overlap = rc; } else overlap = RINGBUF_ADD_CROSS(buf_widx,rc,CUR_TI->buf_idx); if ((unsigned)rc > tracks[track_widx].filerem) { logf("Bad: rc-filerem=%d, fixing", rc-tracks[track_widx].filerem); tracks[track_widx].filesize += rc - tracks[track_widx].filerem; tracks[track_widx].filerem = rc; } /* Advance buffer */ buf_widx = RINGBUF_ADD(buf_widx, rc); tracks[track_widx].available += rc; tracks[track_widx].filerem -= rc; /* If we write into the playing track, adjust it's buffer info */ if (overlap > 0) { CUR_TI->buf_idx += overlap; CUR_TI->start_pos += overlap; } /* For a rebuffer, fill at least this minimum */ if (minimum > (unsigned)rc) minimum -= rc; /* Let the codec process up to the watermark */ /* Break immediately if this is a quick buffer, or there is an event */ else if (minimum || audio_yield_codecs()) { /* Exit quickly, but don't stop the overall buffering process */ ret_val = true; break; } } if (tracks[track_widx].filerem == 0) { logf("Finished buf:%dB", tracks[track_widx].filesize); close(current_fd); current_fd = -1; audio_strip_tags(); track_widx++; track_widx &= MAX_TRACK_MASK; tracks[track_widx].filesize = 0; return true; } else { logf("%s buf:%dB", ret_val?"Quick":"Partially", tracks[track_widx].filesize - tracks[track_widx].filerem); return ret_val; } } static bool audio_loadcodec(bool start_play) { size_t size = 0; int fd; int rc; size_t copy_n; int prev_track; char codec_path[MAX_PATH]; /* Full path to codec */ const char * codec_fn = get_codec_filename(tracks[track_widx].id3.codectype); if (codec_fn == NULL) return false; tracks[track_widx].has_codec = false; if (start_play) { /* Load the codec directly from disk and save some memory. */ track_ridx = track_widx; ci.filesize = CUR_TI->filesize; ci.id3 = &CUR_TI->id3; ci.taginfo_ready = &CUR_TI->taginfo_ready; ci.curpos = 0; LOGFQUEUE("codec > codec Q_CODEC_LOAD_DISK"); queue_post(&codec_queue, Q_CODEC_LOAD_DISK, (void *)codec_fn); return true; } else { /* If we already have another track than this one buffered */ if (track_widx != track_ridx) { prev_track = (track_widx - 1) & MAX_TRACK_MASK; /* If the previous codec is the same as this one, there is no need * to put another copy of it on the file buffer */ if (get_codec_base_type(tracks[track_widx].id3.codectype) == get_codec_base_type(tracks[prev_track].id3.codectype) && audio_codec_loaded) { logf("Reusing prev. codec"); return true; } } } codec_get_full_path(codec_path, codec_fn); fd = open(codec_path, O_RDONLY); if (fd < 0) { logf("Codec doesn't exist!"); return false; } tracks[track_widx].codecsize = filesize(fd); /* Never load a partial codec */ if (RINGBUF_ADD_CROSS(buf_widx,tracks[track_widx].codecsize,buf_ridx) >= 0) { logf("Not enough space"); close(fd); return false; } while (size < tracks[track_widx].codecsize) { copy_n = MIN(conf_filechunk, filebuflen - buf_widx); rc = read(fd, &filebuf[buf_widx], copy_n); if (rc < 0) { close(fd); /* This is an error condition, likely the codec file is corrupt */ logf("Partial codec loaded"); /* Must undo the buffer write of the partial codec */ buf_widx = RINGBUF_SUB(buf_widx, size); tracks[track_widx].codecsize = 0; return false; } buf_widx = RINGBUF_ADD(buf_widx, rc); size += rc; } tracks[track_widx].has_codec = true; close(fd); logf("Done: %dB", size); return true; } /* TODO: Copied from mpeg.c. Should be moved somewhere else. */ static void audio_set_elapsed(struct mp3entry* id3) { unsigned long offset = id3->offset > id3->first_frame_offset ? id3->offset - id3->first_frame_offset : 0; 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 ( 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 = 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 - id3->first_frame_offset - id3->id3v1len) / 1024); id3->elapsed = offset / 1024 * tpk; } } else { /* constant bitrate, use exact calculation */ if (id3->bitrate != 0) id3->elapsed = offset / (id3->bitrate / 8); } } static bool audio_load_track(int offset, bool start_play, bool rebuffer) { char *trackname; off_t size; char msgbuf[80]; /* 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 (!audio_have_free_tracks()) { logf("No free tracks"); return false; } if (current_fd >= 0) { logf("Nonzero fd in alt"); close(current_fd); current_fd = -1; } last_peek_offset++; peek_again: logf("Buffering track:%d/%d", track_widx, track_ridx); /* Get track name from current playlist read position. */ while ((trackname = playlist_peek(last_peek_offset)) != NULL) { /* Handle broken playlists. */ current_fd = open(trackname, O_RDONLY); if (current_fd < 0) { logf("Open failed"); /* Skip invalid entry from playlist. */ playlist_skip_entry(NULL, last_peek_offset); } else break; } if (!trackname) { logf("End-of-playlist"); playlist_end = true; return false; } /* Initialize track entry. */ size = filesize(current_fd); tracks[track_widx].filerem = size; tracks[track_widx].filesize = size; tracks[track_widx].available = 0; /* Set default values */ if (start_play) { int last_codec = current_codec; current_codec = CODEC_IDX_AUDIO; conf_watermark = AUDIO_DEFAULT_WATERMARK; conf_filechunk = AUDIO_DEFAULT_FILECHUNK; conf_preseek = AUDIO_REBUFFER_GUESS_SIZE; dsp_configure(DSP_RESET, 0); current_codec = last_codec; } /* Get track metadata if we don't already have it. */ if (!tracks[track_widx].taginfo_ready) { if (get_metadata(&tracks[track_widx],current_fd,trackname,v1first)) { if (start_play) { track_changed = true; playlist_update_resume_info(audio_current_track()); } } else { logf("mde:%s!",trackname); /* Set filesize to zero to indicate no file was loaded. */ tracks[track_widx].filesize = 0; tracks[track_widx].filerem = 0; close(current_fd); current_fd = -1; /* Skip invalid entry from playlist. */ playlist_skip_entry(NULL, last_peek_offset); tracks[track_widx].taginfo_ready = false; goto peek_again; } } /* Load the codec. */ tracks[track_widx].codecbuf = &filebuf[buf_widx]; if (!audio_loadcodec(start_play)) { /* Set filesize to zero to indicate no file was loaded. */ tracks[track_widx].filesize = 0; tracks[track_widx].filerem = 0; close(current_fd); current_fd = -1; if (tracks[track_widx].codecsize) { /* No space for codec on buffer, not an error */ tracks[track_widx].codecsize = 0; return false; } /* This is an error condition, either no codec was found, or reading * the codec file failed part way through, either way, skip the track */ snprintf(msgbuf, sizeof(msgbuf)-1, "No codec for: %s", trackname); /* We should not use gui_syncplash from audio thread! */ gui_syncsplash(HZ*2, true, msgbuf); /* Skip invalid entry from playlist. */ playlist_skip_entry(NULL, last_peek_offset); tracks[track_widx].taginfo_ready = false; goto peek_again; } tracks[track_widx].start_pos = 0; set_filebuf_watermark(buffer_margin); tracks[track_widx].id3.elapsed = 0; if (offset > 0) { switch (tracks[track_widx].id3.codectype) { case AFMT_MPA_L1: case AFMT_MPA_L2: case AFMT_MPA_L3: lseek(current_fd, offset, SEEK_SET); tracks[track_widx].id3.offset = offset; audio_set_elapsed(&tracks[track_widx].id3); tracks[track_widx].filerem = size - offset; ci.curpos = offset; tracks[track_widx].start_pos = offset; break; case AFMT_WAVPACK: lseek(current_fd, offset, SEEK_SET); tracks[track_widx].id3.offset = offset; tracks[track_widx].id3.elapsed = tracks[track_widx].id3.length / 2; tracks[track_widx].filerem = size - offset; ci.curpos = offset; tracks[track_widx].start_pos = offset; break; case AFMT_OGG_VORBIS: case AFMT_FLAC: case AFMT_PCM_WAV: case AFMT_A52: case AFMT_AAC: tracks[track_widx].id3.offset = offset; break; } } logf("alt:%s", trackname); tracks[track_widx].buf_idx = buf_widx; return audio_read_file(rebuffer); } static bool audio_read_next_metadata(void) { int fd; char *trackname; int next_idx; int status; next_idx = track_widx; if (tracks[next_idx].taginfo_ready) { next_idx++; next_idx &= MAX_TRACK_MASK; if (tracks[next_idx].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; status = get_metadata(&tracks[next_idx],fd,trackname,v1first); /* Preload the glyphs in the tags */ if (status) { if (tracks[next_idx].id3.title) lcd_getstringsize(tracks[next_idx].id3.title, NULL, NULL); if (tracks[next_idx].id3.artist) lcd_getstringsize(tracks[next_idx].id3.artist, NULL, NULL); if (tracks[next_idx].id3.album) lcd_getstringsize(tracks[next_idx].id3.album, NULL, NULL); } close(fd); return status; } /* Send callback events to notify about new tracks. */ static void audio_generate_postbuffer_events(void) { int cur_idx; int last_idx = -1; logf("Postbuffer:%d/%d",track_ridx,track_widx); if (audio_have_tracks()) { cur_idx = track_ridx; while (1) { if (!tracks[cur_idx].event_sent) { if (last_idx >= 0 && !tracks[last_idx].event_sent) { /* Mark the event 'sent' even if we don't really send one */ tracks[last_idx].event_sent = true; if (track_buffer_callback) track_buffer_callback(&tracks[last_idx].id3, false); } last_idx = cur_idx; } if (cur_idx == track_widx) break; cur_idx++; cur_idx &= MAX_TRACK_MASK; } if (last_idx >= 0 && !tracks[last_idx].event_sent) { tracks[last_idx].event_sent = true; if (track_buffer_callback) track_buffer_callback(&tracks[last_idx].id3, true); } /* Force WPS reload. */ track_changed = true; } } static bool audio_initialize_buffer_fill(bool clear_tracks) { /* Don't initialize if we're already initialized */ if (filling) return true; logf("Starting buffer fill"); /* Set the filling flag true before calling audio_clear_tracks as that * function can yield and we start looping. */ filling = true; if (clear_tracks) audio_clear_track_entries(false); /* Save the current resume position once. */ playlist_update_resume_info(audio_current_track()); return true; } static void audio_fill_file_buffer( bool start_play, bool rebuffer, size_t offset) { bool had_next_track = audio_next_track() != NULL; bool continue_buffering; if (!audio_initialize_buffer_fill(!start_play)) return ; /* If we have a partially buffered track, continue loading, * otherwise load a new track */ if (tracks[track_widx].filesize > 0) continue_buffering = audio_read_file(rebuffer); else continue_buffering = audio_load_track(offset, start_play, rebuffer); if (!had_next_track && audio_next_track()) track_changed = true; /* If we're done buffering */ if (!continue_buffering) { audio_read_next_metadata(); audio_generate_postbuffer_events(); filling = false; } #ifndef SIMULATOR ata_sleep(); #endif } static void audio_rebuffer(void) { logf("Forcing rebuffer"); /* Notify the codec that this will take a while */ /* Currently this can cause some problems (logf in reverse order): * Codec load error:-1 * Codec load disk * Codec: Unsupported * Codec finished * New codec:0/3 * Clearing tracks:7/7, 1 * Forcing rebuffer * Check new track buffer * Request new track * Clearing tracks:5/5, 0 * Starting buffer fill * Clearing tracks:5/5, 1 * Re-buffering song w/seek */ //if (!filling) // queue_post(&codec_callback_queue, Q_CODEC_REQUEST_PENDING, 0); /* Stop in progress fill, and clear open file descriptor */ if (current_fd >= 0) { close(current_fd); current_fd = -1; } filling = false; /* Reset buffer and track pointers */ CUR_TI->buf_idx = buf_ridx = buf_widx = 0; track_widx = track_ridx; audio_clear_track_entries(true); CUR_TI->available = 0; /* Fill the buffer */ last_peek_offset = -1; CUR_TI->filesize = 0; CUR_TI->start_pos = 0; ci.curpos = 0; if (!CUR_TI->taginfo_ready) memset(&CUR_TI->id3, 0, sizeof(struct mp3entry)); audio_fill_file_buffer(false, true, 0); } static void audio_check_new_track(void) { int track_count = audio_track_count(); int old_track_ridx = track_ridx; bool forward; if (dir_skip) { dir_skip = false; if (playlist_next_dir(ci.new_track)) { ci.new_track = 0; CUR_TI->taginfo_ready = false; audio_rebuffer(); goto skip_done; } else { LOGFQUEUE("audio > codec Q_CODEC_REQUEST_FAILED"); queue_post(&codec_callback_queue, Q_CODEC_REQUEST_FAILED, 0); return; } } if (new_playlist) ci.new_track = 0; /* If the playlist isn't that big */ if (!playlist_check(ci.new_track)) { if (ci.new_track >= 0) { LOGFQUEUE("audio > codec Q_CODEC_REQUEST_FAILED"); queue_post(&codec_callback_queue, Q_CODEC_REQUEST_FAILED, 0); return; } /* Find the beginning backward if the user over-skips it */ while (!playlist_check(++ci.new_track)) if (ci.new_track >= 0) { LOGFQUEUE("audio > codec Q_CODEC_REQUEST_FAILED"); queue_post(&codec_callback_queue, Q_CODEC_REQUEST_FAILED, 0); return; } } /* Update the playlist */ last_peek_offset -= ci.new_track; if (playlist_next(ci.new_track) < 0) { LOGFQUEUE("audio > codec Q_CODEC_REQUEST_FAILED"); queue_post(&codec_callback_queue, Q_CODEC_REQUEST_FAILED, 0); return; } if (new_playlist) { ci.new_track = 1; new_playlist = false; } /* Save the old track */ prev_ti = CUR_TI; /* Move to the new track */ track_ridx += ci.new_track; track_ridx &= MAX_TRACK_MASK; if (automatic_skip) playlist_end = false; track_changed = !automatic_skip; /* If it is not safe to even skip this many track entries */ if (ci.new_track >= track_count || ci.new_track <= track_count - MAX_TRACK) { ci.new_track = 0; CUR_TI->taginfo_ready = false; audio_rebuffer(); goto skip_done; } forward = ci.new_track > 0; ci.new_track = 0; /* If the target track is clearly not in memory */ if (CUR_TI->filesize == 0 || !CUR_TI->taginfo_ready) { audio_rebuffer(); goto skip_done; } /* The track may be in memory, see if it really is */ if (forward) { if (!audio_buffer_wind_forward(track_ridx, old_track_ridx)) audio_rebuffer(); } else { int cur_idx = track_ridx; bool taginfo_ready = true; bool wrap = track_ridx > old_track_ridx; while (1) { cur_idx++; cur_idx &= MAX_TRACK_MASK; if (!(wrap || cur_idx < old_track_ridx)) break; /* If we hit a track in between without valid tag info, bail */ if (!tracks[cur_idx].taginfo_ready) { taginfo_ready = false; break; } tracks[cur_idx].available = tracks[cur_idx].filesize; if (tracks[cur_idx].codecsize) tracks[cur_idx].has_codec = true; } if (taginfo_ready) { if (!audio_buffer_wind_backward(track_ridx, old_track_ridx)) audio_rebuffer(); } else { CUR_TI->taginfo_ready = false; audio_rebuffer(); } } skip_done: audio_update_trackinfo(); LOGFQUEUE("audio > codec Q_CODEC_REQUEST_COMPLETE"); queue_post(&codec_callback_queue, Q_CODEC_REQUEST_COMPLETE, 0); } static void audio_rebuffer_and_seek(size_t newpos) { size_t real_preseek; int fd; char *trackname; /* (Re-)open current track's file handle. */ trackname = playlist_peek(0); fd = open(trackname, O_RDONLY); if (fd < 0) { LOGFQUEUE("audio > codec Q_CODEC_REQUEST_FAILED"); queue_post(&codec_callback_queue, Q_CODEC_REQUEST_FAILED, 0); return; } if (current_fd >= 0) close(current_fd); current_fd = fd; playlist_end = false; ci.curpos = newpos; /* Clear codec buffer. */ track_widx = track_ridx; tracks[track_widx].buf_idx = buf_widx = buf_ridx = 0; last_peek_offset = 0; filling = false; audio_initialize_buffer_fill(true); /* This may have been tweaked by the id3v1 code */ CUR_TI->filesize=filesize(fd); if (newpos > conf_preseek) { CUR_TI->start_pos = newpos - conf_preseek; lseek(current_fd, CUR_TI->start_pos, SEEK_SET); CUR_TI->filerem = CUR_TI->filesize - CUR_TI->start_pos; real_preseek = conf_preseek; } else { CUR_TI->start_pos = 0; CUR_TI->filerem = CUR_TI->filesize; real_preseek = newpos; } CUR_TI->available = 0; audio_read_file(real_preseek); /* Account for the data we just read that is 'behind' us now */ CUR_TI->available -= real_preseek; buf_ridx = RINGBUF_ADD(buf_ridx, real_preseek); LOGFQUEUE("audio > codec Q_CODEC_REQUEST_COMPLETE"); queue_post(&codec_callback_queue, Q_CODEC_REQUEST_COMPLETE, 0); } 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; } unsigned long audio_prev_elapsed(void) { return prev_track_elapsed; } static void audio_stop_codec_flush(void) { ci.stop_codec = true; pcmbuf_pause(true); while (audio_codec_loaded) yield(); /* If the audio codec is not loaded any more, and the audio is still * playing, it is now and _only_ now safe to call this function from the * audio thread */ if (pcm_is_playing()) pcmbuf_play_stop(); pcmbuf_pause(paused); } static void audio_stop_playback(void) { /* If we were playing, save resume information */ if (playing) { /* Save the current playing spot, or NULL if the playlist has ended */ playlist_update_resume_info( (playlist_end && ci.stop_codec)?NULL:audio_current_track()); } filling = false; paused = false; audio_stop_codec_flush(); playing = false; if (current_fd >= 0) { close(current_fd); current_fd = -1; } /* Mark all entries null. */ audio_clear_track_entries(false); } static void audio_play_start(size_t offset) { #if defined(HAVE_RECORDING) || defined(CONFIG_TUNER) rec_set_source(AUDIO_SRC_PLAYBACK, SRCF_PLAYBACK); #endif /* Wait for any previously playing audio to flush - TODO: Not necessary? */ audio_stop_codec_flush(); /* must reset the buffer before any playback begins if needed */ if (buffer_state != BUFFER_STATE_NORMAL) audio_reset_buffer(pcmbuf_get_bufsize()); track_changed = true; playlist_end = false; playing = true; ci.new_track = 0; ci.seek_time = 0; wps_offset = 0; if (current_fd >= 0) { close(current_fd); current_fd = -1; } sound_set_volume(global_settings.volume); track_widx = track_ridx = 0; buf_ridx = buf_widx = 0; /* Mark all entries null. */ memset(tracks, 0, sizeof(struct track_info) * MAX_TRACK); last_peek_offset = -1; audio_fill_file_buffer(true, false, offset); } /* Invalidates all but currently playing track. */ static void audio_invalidate_tracks(void) { if (audio_have_tracks()) { last_peek_offset = 0; playlist_end = false; track_widx = track_ridx; /* Mark all other entries null (also buffered wrong metadata). */ audio_clear_track_entries(true); /* If the current track is fully buffered, advance the write pointer */ if (tracks[track_widx].filerem == 0) track_widx = (track_widx + 1) & MAX_TRACK_MASK; buf_widx = RINGBUF_ADD(buf_ridx, CUR_TI->available); audio_read_next_metadata(); } } static void audio_new_playlist(void) { /* Prepare to start a new fill from the beginning of the playlist */ last_peek_offset = -1; if (audio_have_tracks()) { playlist_end = false; track_widx = track_ridx; audio_clear_track_entries(true); track_widx++; track_widx &= MAX_TRACK_MASK; /* Stop reading the current track */ CUR_TI->filerem = 0; close(current_fd); current_fd = -1; /* Mark the current track as invalid to prevent skipping back to it */ CUR_TI->taginfo_ready = false; /* Invalidate the buffer other than the playing track */ buf_widx = RINGBUF_ADD(buf_ridx, CUR_TI->available); } /* Signal the codec to initiate a track change forward */ new_playlist = true; ci.new_track = 1; audio_fill_file_buffer(false, true, 0); } static void audio_initiate_track_change(long direction) { playlist_end = false; ci.new_track += direction; wps_offset -= direction; } static void audio_initiate_dir_change(long direction) { playlist_end = false; dir_skip = true; ci.new_track = direction; } /* * Layout audio buffer as follows: * [|TALK]|MALLOC|FILE|GUARD|PCM|AUDIOCODEC|[VOICECODEC|] */ static void audio_reset_buffer(size_t pcmbufsize) { /* see audio_get_recording_buffer if this is modified */ size_t offset; logf("audio_reset_buffer"); logf(" size:%08X", pcmbufsize); /* Initially set up file buffer as all space available */ filebuf = audiobuf + MALLOC_BUFSIZE + talk_get_bufsize(); filebuflen = audiobufend - filebuf; /* Allow for codec(s) at end of audio buffer */ if (talk_voice_required()) { #ifdef PLAYBACK_VOICE /* Allow 2 codecs at end of audio buffer */ filebuflen -= 2 * (CODEC_IRAM_SIZE + CODEC_SIZE); iram_buf[CODEC_IDX_AUDIO] = filebuf + filebuflen; dram_buf[CODEC_IDX_AUDIO] = iram_buf[CODEC_IDX_AUDIO] + CODEC_IRAM_SIZE; iram_buf[CODEC_IDX_VOICE] = dram_buf[CODEC_IDX_AUDIO] + CODEC_SIZE; dram_buf[CODEC_IDX_VOICE] = iram_buf[CODEC_IDX_VOICE] + CODEC_IRAM_SIZE; #endif } else { #ifdef PLAYBACK_VOICE /* Allow for 1 codec at end of audio buffer */ filebuflen -= CODEC_IRAM_SIZE + CODEC_SIZE; iram_buf[CODEC_IDX_AUDIO] = filebuf + filebuflen; dram_buf[CODEC_IDX_AUDIO] = iram_buf[CODEC_IDX_AUDIO] + CODEC_IRAM_SIZE; iram_buf[CODEC_IDX_VOICE] = NULL; dram_buf[CODEC_IDX_VOICE] = NULL; #endif } filebuflen -= pcmbuf_init(pcmbufsize, filebuf + filebuflen) + GUARD_BUFSIZE; /* Ensure that file buffer is aligned */ offset = -(size_t)filebuf & 3; filebuf += offset; filebuflen -= offset; filebuflen &= ~3; /* Clear any references to the file buffer */ buffer_state = BUFFER_STATE_NORMAL; } #ifdef ROCKBOX_HAS_LOGF static void audio_test_track_changed_event(struct mp3entry *id3) { (void)id3; logf("tce:%s", id3->path); } #endif static void audio_playback_init(void) { #ifdef PLAYBACK_VOICE static bool voicetagtrue = true; static struct mp3entry id3_voice; #endif struct event ev; logf("playback api init"); pcm_init(); #ifdef AUDIO_HAVE_RECORDING rec_set_source(AUDIO_SRC_PLAYBACK, SRCF_PLAYBACK); #endif #ifdef ROCKBOX_HAS_LOGF audio_set_track_changed_event(audio_test_track_changed_event); #endif /* Initialize codec api. */ ci.read_filebuf = codec_filebuf_callback; ci.pcmbuf_insert = codec_pcmbuf_insert_callback; ci.pcmbuf_insert_split = codec_pcmbuf_insert_split_callback; ci.get_codec_memory = codec_get_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; ci.discard_codec = codec_discard_codec_callback; /* Initialize voice codec api. */ #ifdef PLAYBACK_VOICE memcpy(&ci_voice, &ci, sizeof(struct codec_api)); memset(&id3_voice, 0, sizeof(struct mp3entry)); ci_voice.read_filebuf = voice_filebuf_callback; ci_voice.pcmbuf_insert = voice_pcmbuf_insert_callback; ci_voice.pcmbuf_insert_split = voice_pcmbuf_insert_split_callback; ci_voice.get_codec_memory = voice_get_memory_callback; ci_voice.request_buffer = voice_request_buffer_callback; ci_voice.advance_buffer = voice_advance_buffer_callback; ci_voice.advance_buffer_loc = voice_advance_buffer_loc_callback; ci_voice.request_next_track = voice_request_next_track_callback; ci_voice.mp3_get_filepos = voice_mp3_get_filepos_callback; ci_voice.seek_buffer = voice_seek_buffer_callback; ci_voice.seek_complete = voice_do_nothing; ci_voice.set_elapsed = voice_set_elapsed_callback; ci_voice.set_offset = voice_set_offset_callback; ci_voice.discard_codec = voice_do_nothing; ci_voice.taginfo_ready = &voicetagtrue; ci_voice.id3 = &id3_voice; id3_voice.frequency = 11200; id3_voice.length = 1000000L; #endif codec_thread_p = create_thread(codec_thread, codec_stack, sizeof(codec_stack), codec_thread_name IF_PRIO(, PRIORITY_PLAYBACK)); while (1) { queue_wait(&audio_queue, &ev); if (ev.id == Q_AUDIO_POSTINIT) break ; #ifndef SIMULATOR if (ev.id == SYS_USB_CONNECTED) { logf("USB: Audio preinit"); usb_acknowledge(SYS_USB_CONNECTED_ACK); usb_wait_for_disconnect(&audio_queue); } #endif } /* initialize the buffer */ filebuf = audiobuf; /* must be non-NULL for audio_set_crossfade */ buffer_state = BUFFER_STATE_TRASHED; /* force it */ audio_set_crossfade(global_settings.crossfade); audio_is_initialized = true; sound_settings_apply(); } #if MEM > 8 /* we dont want this rebuffering on targets with little ram because the disk may never spin down */ bool ata_fillbuffer_callback(void) { queue_post(&audio_queue, Q_AUDIO_FILL_BUFFER_IF_ACTIVE_ATA, 0); return true; } #endif static void audio_thread(void) { struct event ev; #if MEM > 8 size_t high_watermark; #endif /* At first initialize audio system in background. */ audio_playback_init(); #if MEM > 8 high_watermark = (3*filebuflen)/4; #endif while (1) { if (filling) { queue_wait_w_tmo(&audio_queue, &ev, 0); if (ev.id == SYS_TIMEOUT) ev.id = Q_AUDIO_FILL_BUFFER; } #if MEM > 8 else { queue_wait_w_tmo(&audio_queue, &ev, HZ/2); if ( (ev.id == SYS_TIMEOUT) && (FILEBUFUSED < high_watermark)) register_ata_idle_func(ata_fillbuffer_callback); } #else queue_wait_w_tmo(&audio_queue, &ev, HZ/2); #endif switch (ev.id) { #if MEM > 8 case Q_AUDIO_FILL_BUFFER_IF_ACTIVE_ATA: /* only fill if the disk is still spining */ #ifndef SIMULATOR if (!ata_disk_is_active()) break; #endif #endif /* MEM > 8 */ /* else fall through to Q_AUDIO_FILL_BUFFER */ case Q_AUDIO_FILL_BUFFER: LOGFQUEUE("audio < Q_AUDIO_FILL_BUFFER"); if (!filling) if (!playing || playlist_end || ci.stop_codec) break; audio_fill_file_buffer(false, false, 0); break; case Q_AUDIO_PLAY: LOGFQUEUE("audio < Q_AUDIO_PLAY"); audio_clear_track_entries(false); audio_play_start((size_t)ev.data); break ; case Q_AUDIO_STOP: LOGFQUEUE("audio < Q_AUDIO_STOP"); audio_stop_playback(); break ; case Q_AUDIO_PAUSE: LOGFQUEUE("audio < Q_AUDIO_PAUSE"); pcmbuf_pause((bool)ev.data); paused = (bool)ev.data; break ; case Q_AUDIO_SKIP: LOGFQUEUE("audio < Q_AUDIO_SKIP"); audio_initiate_track_change((long)ev.data); break; case Q_AUDIO_PRE_FF_REWIND: LOGFQUEUE("audio < Q_AUDIO_PRE_FF_REWIND"); if (!playing) break; pcmbuf_pause(true); break; case Q_AUDIO_FF_REWIND: LOGFQUEUE("audio < Q_AUDIO_FF_REWIND"); if (!playing) break ; ci.seek_time = (long)ev.data+1; break ; case Q_AUDIO_REBUFFER_SEEK: LOGFQUEUE("audio < Q_AUDIO_REBUFFER_SEEK"); audio_rebuffer_and_seek((size_t)ev.data); break; case Q_AUDIO_CHECK_NEW_TRACK: LOGFQUEUE("audio < Q_AUDIO_CHECK_NEW_TRACK"); audio_check_new_track(); break; case Q_AUDIO_DIR_SKIP: LOGFQUEUE("audio < Q_AUDIO_DIR_SKIP"); playlist_end = false; audio_initiate_dir_change((long)ev.data); break; case Q_AUDIO_NEW_PLAYLIST: LOGFQUEUE("audio < Q_AUDIO_NEW_PLAYLIST"); audio_new_playlist(); break; case Q_AUDIO_FLUSH: LOGFQUEUE("audio < Q_AUDIO_FLUSH"); audio_invalidate_tracks(); break ; case Q_AUDIO_TRACK_CHANGED: LOGFQUEUE("audio < Q_AUDIO_TRACK_CHANGED"); if (track_changed_callback) track_changed_callback(&CUR_TI->id3); track_changed = true; playlist_update_resume_info(audio_current_track()); break ; #ifdef AUDIO_HAVE_RECORDING case Q_AUDIO_LOAD_ENCODER: LOGFQUEUE("audio < Q_AUDIO_LOAD_ENCODER"); LOGFQUEUE("audio > codec Q_ENCODER_LOAD_DISK"); queue_post(&codec_queue, Q_ENCODER_LOAD_DISK, ev.data); break; #endif #ifndef SIMULATOR case SYS_USB_CONNECTED: LOGFQUEUE("audio < SYS_USB_CONNECTED"); audio_stop_playback(); usb_acknowledge(SYS_USB_CONNECTED_ACK); usb_wait_for_disconnect(&audio_queue); break ; #endif case SYS_TIMEOUT: LOGFQUEUE_SYS_TIMEOUT("audio < SYS_TIMEOUT"); break; default: LOGFQUEUE("audio < default"); } } }