/*************************************************************************** * __________ __ ___. * 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: Check for a possibly broken codepath on a rapid skip, stop event */ /* TODO: same in reverse ^^ */ /* TODO: Also play, stop ^^ */ /* 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. */ /* This is also necesary to prevent the problem with buffer overwriting on * automatic track changes */ #include #include #include #include #include "system.h" #include "thread.h" #include "file.h" #include "lcd.h" #include "font.h" #include "backlight.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 "pcm_playback.h" #include "pcm_record.h" #include "buffer.h" #include "dsp.h" #include "abrepeat.h" #include "tagcache.h" #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 "talk.h" #ifdef CONFIG_TUNER #include "radio.h" #endif #include "splash.h" static volatile bool audio_codec_loaded; static volatile bool voice_codec_loaded; static volatile bool playing; static volatile bool paused; #define CODEC_VORBIS "/.rockbox/codecs/vorbis.codec" #define CODEC_MPA_L3 "/.rockbox/codecs/mpa.codec" #define CODEC_FLAC "/.rockbox/codecs/flac.codec" #define CODEC_WAV "/.rockbox/codecs/wav.codec" #define CODEC_A52 "/.rockbox/codecs/a52.codec" #define CODEC_MPC "/.rockbox/codecs/mpc.codec" #define CODEC_WAVPACK "/.rockbox/codecs/wavpack.codec" #define CODEC_ALAC "/.rockbox/codecs/alac.codec" #define CODEC_AAC "/.rockbox/codecs/aac.codec" #define CODEC_SHN "/.rockbox/codecs/shorten.codec" #define CODEC_AIFF "/.rockbox/codecs/aiff.codec" #define CODEC_SID "/.rockbox/codecs/sid.codec" /* 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) 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, 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, }; /* As defined in plugins/lib/xxx2wav.h */ #define MALLOC_BUFSIZE (512*1024) #define GUARD_BUFSIZE (32*1024) /* As defined in plugin.lds */ #if CONFIG_CPU == PP5020 || CONFIG_CPU == PP5002 #define CODEC_IRAM_ORIGIN 0x4000c000 #else #define CODEC_IRAM_ORIGIN 0x1000c000 #endif #define CODEC_IRAM_SIZE 0xc000 #ifndef SIMULATOR extern bool audio_is_initialized; #else static bool audio_is_initialized = false; #endif /* Buffer control thread. */ static struct event_queue audio_queue; static long audio_stack[(DEFAULT_STACK_SIZE + 0x1000)/sizeof(long)]; static const char audio_thread_name[] = "audio"; /* Codec thread. */ static struct event_queue codec_queue; static long codec_stack[(DEFAULT_STACK_SIZE + 0x2000)/sizeof(long)] IBSS_ATTR; static const char codec_thread_name[] = "codec"; /* Voice codec thread. */ static struct event_queue voice_codec_queue; static long voice_codec_stack[(DEFAULT_STACK_SIZE + 0x2000)/sizeof(long)] IBSS_ATTR; static const char voice_codec_thread_name[] = "voice codec"; struct voice_info { void (*callback)(unsigned char **start, int *size); int size; char *buf; }; static struct mutex mutex_codecthread; static struct event_queue codec_callback_queue; static struct mp3entry id3_voice; static char *voicebuf; static size_t voice_remaining; static bool voice_is_playing; static void (*voice_getmore)(unsigned char** start, int* size); static int voice_thread_num = -1; /* Is file buffer currently being refilled? */ static volatile bool filling IDATA_ATTR; volatile int current_codec IDATA_ATTR; extern unsigned char codecbuf[]; /* Ring buffer where tracks and codecs are loaded. */ static char *filebuf; /* Total size of the ring buffer. */ size_t filebuflen; /* Bytes available in the buffer. */ size_t filebufused; /* Ring buffer read and write indexes. */ static volatile size_t buf_ridx IDATA_ATTR; static volatile size_t buf_widx IDATA_ATTR; #ifndef SIMULATOR static unsigned char *iram_buf[2]; #endif static unsigned char *dram_buf[2]; /* Step count to the next unbuffered track. */ static int last_peek_offset; /* Track information (count in file buffer, read/write indexes for track ring structure. */ static int track_ridx; static int track_widx; static bool track_changed; /* Partially loaded song's file handle to continue buffering later. */ static int current_fd; /* Information about how many bytes left on the buffer re-fill run. */ static size_t fill_bytesleft; /* Track info structure about songs in the file buffer. */ static struct track_info tracks[MAX_TRACK]; /* Pointer to track info structure about current song playing. */ static struct track_info *cur_ti; static struct track_info *prev_ti; /* Have we reached end of the current playlist. */ static bool playlist_end = false; /* Codec API including function callbacks. */ extern struct codec_api ci; extern struct codec_api ci_voice; /* Was the skip being executed manual or automatic? */ static bool automatic_skip; static bool dir_skip = false; static bool new_playlist = false; /* Callback function to call when current track has really changed. */ void (*track_changed_callback)(struct mp3entry *id3); void (*track_buffer_callback)(struct mp3entry *id3, bool last_track); void (*track_unbuffer_callback)(struct mp3entry *id3, bool last_track); static void playback_init(void); /* Configuration */ static size_t conf_watermark; static size_t conf_filechunk; static size_t buffer_margin; static bool v1first = false; static void mp3_set_elapsed(struct mp3entry* id3); static int mp3_get_file_pos(void); static void audio_clear_track_entries( bool clear_buffered, bool clear_unbuffered, bool may_yield); static bool initialize_buffer_fill(bool clear_tracks); static void audio_fill_file_buffer( bool start_play, bool rebuffer, size_t offset); static void swap_codec(void) { int my_codec = current_codec; logf("swapping out codec:%d", my_codec); /* Save our current IRAM and DRAM */ #ifndef SIMULATOR memcpy(iram_buf[my_codec], (unsigned char *)CODEC_IRAM_ORIGIN, CODEC_IRAM_SIZE); #endif memcpy(dram_buf[my_codec], codecbuf, CODEC_SIZE); do { /* Release my semaphore and force a task switch. */ mutex_unlock(&mutex_codecthread); yield(); mutex_lock(&mutex_codecthread); /* Loop until the other codec has locked and run */ } while (my_codec == current_codec); current_codec = my_codec; /* Reload our IRAM and DRAM */ #ifndef SIMULATOR memcpy((unsigned char *)CODEC_IRAM_ORIGIN, iram_buf[my_codec], CODEC_IRAM_SIZE); #endif invalidate_icache(); memcpy(codecbuf, dram_buf[my_codec], CODEC_SIZE); logf("resuming codec:%d", my_codec); } #ifdef HAVE_ADJUSTABLE_CPU_FREQ static void voice_boost_cpu(bool state) { static bool voice_cpu_boosted = false; if (state != voice_cpu_boosted) { cpu_boost(state); voice_cpu_boosted = state; } } #else #define voice_boost_cpu(state) do { } while(0) #endif 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) 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) swap_codec(); } else pcmbuf_write_complete(output_size); length -= input_size; } return true; } 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); if (voice_is_playing && pcm_is_playing() && pcmbuf_usage() > 30 && pcmbuf_mix_free() > 80) { swap_codec(); } length -= input_size; } return true; } 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 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* get_voice_memory_callback(size_t *size) { *size = 0; return NULL; } static void* get_codec_memory_callback(size_t *size) { *size = MALLOC_BUFSIZE; if (voice_codec_loaded) return &audiobuf[talk_get_bufsize()]; else return audiobuf; } static void pcmbuf_position_callback(size_t size) ICODE_ATTR; static void pcmbuf_position_callback(size_t size) { 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 voice_set_elapsed_callback(unsigned int value) { (void)value; } 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 voice_set_offset_callback(size_t value) { (void)value; } 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 bool filebuf_is_lowdata(void) { return filebufused < AUDIO_FILEBUF_CRITICAL; } static bool have_tracks(void) { return track_ridx != track_widx || cur_ti->filesize; } static bool 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 (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; } static void advance_buffer_counters(size_t amount) { buf_ridx += amount; if (buf_ridx >= filebuflen) buf_ridx -= filebuflen; ci.curpos += amount; cur_ti->available -= amount; filebufused -= amount; /* Start buffer filling as necessary. */ if (!pcmbuf_is_lowdata() && !filling) { if (conf_watermark && filebufused <= conf_watermark && playing) queue_post(&audio_queue, Q_AUDIO_FILL_BUFFER, 0); } } static size_t voice_filebuf_callback(void *ptr, size_t size) { (void)ptr; (void)size; return 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) 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 */ advance_buffer_counters(copy_n); /* Return the actual amount of data copied to the buffer */ return copy_n; } 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) queue_wait_w_tmo(&voice_codec_queue, &ev, 0); else if (playing) { queue_wait_w_tmo(&voice_codec_queue, &ev, 0); if (ev.id == SYS_TIMEOUT) ev.id = Q_AUDIO_PLAY; } else queue_wait(&voice_codec_queue, &ev); switch (ev.id) { case Q_AUDIO_PLAY: if (playing) swap_codec(); break; case Q_VOICE_STOP: if (voice_is_playing) { /* Clear the current buffer */ voice_is_playing = false; voice_getmore = NULL; voice_remaining = 0; voicebuf = NULL; voice_boost_cpu(false); ci_voice.new_track = 1; /* Force the codec to think it's changing tracks */ *realsize = 0; return NULL; } else break; case SYS_USB_CONNECTED: logf("USB: Voice codec"); usb_acknowledge(SYS_USB_CONNECTED_ACK); if (audio_codec_loaded) swap_codec(); usb_wait_for_disconnect(&voice_codec_queue); break; case Q_VOICE_PLAY: { struct voice_info *voice_data; voice_is_playing = true; voice_boost_cpu(true); voice_data = ev.data; voice_remaining = voice_data->size; voicebuf = voice_data->buf; voice_getmore = voice_data->callback; } case SYS_TIMEOUT: goto voice_play_clip; } } 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) { queue_post(&voice_codec_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; } 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) 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]; } 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; } /* Count the data BETWEEN the selected tracks */ static size_t 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 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 += buffer_count_tracks(old_track_ridx, new_track_ridx); if (amount > filebufused) return false; logf("bwf:%ldB",amount); /* Wind the buffer to the beginning of the target track or its codec */ buf_ridx += amount; filebufused -= amount; /* Check and handle buffer wrapping */ if (buf_ridx >= filebuflen) buf_ridx -= filebuflen; return true; } static bool 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; buf_back = buf_ridx; amount = ci.curpos; if (buf_ridx < buf_widx) buf_back += filebuflen; buf_back -= 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 += 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); /* Check and handle buffer wrapping */ if (amount > buf_ridx) buf_ridx += filebuflen; /* Rewind the buffer to the beginning of the target track or its codec */ buf_ridx -= amount; filebufused += 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; } 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 */ buf_ridx = buf_widx = 0; track_widx = track_ridx; cur_ti = &tracks[track_ridx]; audio_clear_track_entries(true, true, false); filebufused = 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 { 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) { 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) { 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) { queue_post(&codec_callback_queue, Q_CODEC_REQUEST_FAILED, 0); return; } if (new_playlist) { ci.new_track = 1; new_playlist = false; } track_ridx += ci.new_track; track_ridx &= MAX_TRACK_MASK; /* Save the old track */ prev_ti = cur_ti; /* Move to the new track */ cur_ti = &tracks[track_ridx]; 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 (!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 (!buffer_wind_backward(track_ridx, old_track_ridx)) audio_rebuffer(); } else { cur_ti->taginfo_ready = false; audio_rebuffer(); } } skip_done: audio_update_trackinfo(); queue_post(&codec_callback_queue, Q_CODEC_REQUEST_COMPLETE, 0); } static void rebuffer_and_seek(size_t newpos) { int fd; char *trackname; trackname = playlist_peek(0); /* (Re-)open current track's file handle. */ fd = open(trackname, O_RDONLY); if (fd < 0) { logf("Open 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; filebufused = 0; buf_widx = buf_ridx = 0; last_peek_offset = 0; filling = false; initialize_buffer_fill(true); if (newpos > AUDIO_REBUFFER_GUESS_SIZE) { buf_ridx += AUDIO_REBUFFER_GUESS_SIZE; cur_ti->start_pos = newpos - AUDIO_REBUFFER_GUESS_SIZE; } else { buf_ridx += newpos; cur_ti->start_pos = 0; } cur_ti->filerem = cur_ti->filesize - cur_ti->start_pos; cur_ti->available = 0; lseek(current_fd, cur_ti->start_pos, SEEK_SET); queue_post(&codec_callback_queue, Q_CODEC_REQUEST_COMPLETE, 0); } static void voice_advance_buffer_callback(size_t amount) { amount = MIN(amount, voice_remaining); voicebuf += amount; voice_remaining -= amount; } 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; 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: ci.stop_codec = true; case Q_CODEC_REQUEST_COMPLETE: return; default: logf("Bad event on ccq"); ci.stop_codec = true; return; } } advance_buffer_counters(amount); codec_set_offset_callback(ci.curpos); } static void voice_advance_buffer_loc_callback(void *ptr) { size_t amount = (size_t)ptr - (size_t)voicebuf; voice_advance_buffer_callback(amount); } 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); } static off_t voice_mp3_get_filepos_callback(int newtime) { (void)newtime; return 0; } static off_t codec_mp3_get_filepos_callback(int newtime) { off_t newpos; cur_ti->id3.elapsed = newtime; newpos = mp3_get_file_pos(); return newpos; } static void voice_do_nothing(void) { return; } 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 voice_seek_buffer_callback(size_t newpos) { (void)newpos; return false; } static bool codec_seek_buffer_callback(size_t newpos) { int difference; 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; queue_post(&audio_queue, Q_AUDIO_REBUFFER_SEEK, (void *)newpos); queue_wait(&codec_callback_queue, &ev); switch (ev.id) { case Q_CODEC_REQUEST_COMPLETE: return true; case Q_CODEC_REQUEST_FAILED: ci.stop_codec = true; return false; default: logf("Bad event on ccq"); return false; } } /* Seeking inside buffer space. */ logf("seek: -%d", difference); filebufused += difference; cur_ti->available += difference; if (buf_ridx < (unsigned)difference) buf_ridx += filebuflen; buf_ridx -= difference; ci.curpos -= difference; return true; } 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 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; 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; } static void codec_track_changed(void) { automatic_skip = false; track_changed = true; queue_post(&audio_queue, Q_AUDIO_TRACK_CHANGED, 0); } static void pcmbuf_track_changed_callback(void) { pcmbuf_set_position_callback(NULL); codec_track_changed(); } /* 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 yield_codecs(void) { yield(); if (!queue_empty(&audio_queue)) return true; while ((pcmbuf_is_crossfade_active() || pcmbuf_is_lowdata()) && !ci.stop_codec && playing && !filebuf_is_lowdata()) { sleep(1); if (!queue_empty(&audio_queue)) return true; } return false; } /* FIXME: This code should be made more generic and move to metadata.c */ static void strip_id3v1_tag(void) { int i; static const unsigned char tag[] = "TAG"; size_t tag_idx; size_t cur_idx; tag_idx = buf_widx; if (tag_idx < 128) tag_idx += filebuflen; tag_idx -= 128; if (filebufused > 128 && tag_idx > buf_ridx) { cur_idx = tag_idx; for(i = 0;i < 3;i++) { if(filebuf[cur_idx] != tag[i]) return; if(++cur_idx >= filebuflen) cur_idx -= filebuflen; } /* Skip id3v1 tag */ logf("Skipping ID3v1 tag"); buf_widx = tag_idx; tracks[track_widx].available -= 128; tracks[track_widx].filesize -= 128; filebufused -= 128; } } static void audio_read_file(bool quick) { size_t copy_n; int rc; /* If we're called and no file is open, this is an error */ if (current_fd < 0) { logf("Bad fd in arf"); /* Stop this buffer cycle immediately */ fill_bytesleft = 0; /* Give some hope of miraculous recovery by forcing a track reload */ tracks[track_widx].filesize = 0; return ; } while (tracks[track_widx].filerem > 0) { if (fill_bytesleft == 0) break ; /* copy_n is the largest chunk that is safe to read */ copy_n = MIN(conf_filechunk, filebuflen - buf_widx); copy_n = MIN(copy_n, fill_bytesleft); /* rc is the actual amount read */ rc = read(current_fd, &filebuf[buf_widx], copy_n); if (rc <= 0) { /* Reached the end of the file */ tracks[track_widx].filerem = 0; break ; } buf_widx += rc; if (buf_widx >= filebuflen) buf_widx -= filebuflen; tracks[track_widx].available += rc; tracks[track_widx].filerem -= rc; filebufused += rc; if (fill_bytesleft > (unsigned)rc) fill_bytesleft -= rc; else fill_bytesleft = 0; /* Let the codec process until it is out of the danger zone, or there * is an event to handle. In the latter case, break this fill cycle * immediately */ if (quick || yield_codecs()) break; } if (tracks[track_widx].filerem == 0) { logf("Finished buf:%dB", tracks[track_widx].filesize); close(current_fd); current_fd = -1; strip_id3v1_tag(); track_widx++; track_widx &= MAX_TRACK_MASK; tracks[track_widx].filesize = 0; } else { logf("Partially buf:%dB", tracks[track_widx].filesize - tracks[track_widx].filerem); } } static void codec_discard_codec_callback(void) { if (cur_ti->has_codec) { cur_ti->has_codec = false; filebufused -= cur_ti->codecsize; buf_ridx += cur_ti->codecsize; if (buf_ridx >= filebuflen) buf_ridx -= filebuflen; } #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 const char *get_codec_path(int codectype) { switch (codectype) { case AFMT_OGG_VORBIS: logf("Codec: Vorbis"); return CODEC_VORBIS; case AFMT_MPA_L1: case AFMT_MPA_L2: case AFMT_MPA_L3: logf("Codec: MPA L1/L2/L3"); return CODEC_MPA_L3; case AFMT_PCM_WAV: logf("Codec: PCM WAV"); return CODEC_WAV; case AFMT_FLAC: logf("Codec: FLAC"); return CODEC_FLAC; case AFMT_A52: logf("Codec: A52"); return CODEC_A52; case AFMT_MPC: logf("Codec: Musepack"); return CODEC_MPC; case AFMT_WAVPACK: logf("Codec: WAVPACK"); return CODEC_WAVPACK; case AFMT_ALAC: logf("Codec: ALAC"); return CODEC_ALAC; case AFMT_AAC: logf("Codec: AAC"); return CODEC_AAC; case AFMT_SHN: logf("Codec: SHN"); return CODEC_SHN; case AFMT_AIFF: logf("Codec: PCM AIFF"); return CODEC_AIFF; case AFMT_SID: logf("Codec: SID"); return CODEC_SID; default: logf("Codec: Unsupported"); return NULL; } } static bool loadcodec(bool start_play) { size_t size; int fd; int rc; size_t copy_n; int prev_track; const char *codec_path = get_codec_path(tracks[track_widx].id3.codectype); if (codec_path == NULL) return false; tracks[track_widx].has_codec = false; tracks[track_widx].codecsize = 0; if (start_play) { /* Load the codec directly from disk and save some memory. */ track_ridx = track_widx; cur_ti = &tracks[track_ridx]; ci.filesize = cur_ti->filesize; ci.id3 = &cur_ti->id3; ci.taginfo_ready = &cur_ti->taginfo_ready; ci.curpos = 0; playing = true; queue_post(&codec_queue, Q_CODEC_LOAD_DISK, (void *)codec_path); 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; } } } fd = open(codec_path, O_RDONLY); if (fd < 0) { logf("Codec doesn't exist!"); return false; } size = filesize(fd); /* Never load a partial codec */ if (fill_bytesleft < size) { logf("Not enough space"); fill_bytesleft = 0; close(fd); return false; } while (tracks[track_widx].codecsize < size) { copy_n = MIN(conf_filechunk, filebuflen - buf_widx); rc = read(fd, &filebuf[buf_widx], copy_n); if (rc < 0) return false; filebufused += rc; if (fill_bytesleft > (unsigned)rc) fill_bytesleft -= rc; else fill_bytesleft = 0; buf_widx += rc; if (buf_widx >= filebuflen) buf_widx -= filebuflen; tracks[track_widx].codecsize += rc; yield_codecs(); } tracks[track_widx].has_codec = true; close(fd); logf("Done: %dB", size); return true; } static bool 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; } 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 (!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; 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 (!loadcodec(start_play)) { if (tracks[track_widx].codecsize) { /* Must undo the buffer write of the partial codec */ logf("Partial codec loaded"); fill_bytesleft += tracks[track_widx].codecsize; filebufused -= tracks[track_widx].codecsize; if (buf_widx < tracks[track_widx].codecsize) buf_widx += filebuflen; buf_widx -= tracks[track_widx].codecsize; tracks[track_widx].codecsize = 0; } /* 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; /* Try skipping to next track if there is space. */ if (fill_bytesleft > 0) { /* This is an error condition unless the fill_bytesleft is 0 */ 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; } return false; } 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; mp3_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: tracks[track_widx].id3.offset = offset; break; } } logf("alt:%s", trackname); // tracks[track_widx].buf_idx = buf_widx; audio_read_file(rebuffer); return true; } /* Note that this function might yield(). */ static void audio_clear_track_entries( bool clear_buffered, bool clear_unbuffered, bool may_yield) { 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 (clear_buffered) { if (last_idx >= 0) { /* If there is an unbuffer callback, call it, otherwise, * just clear the track */ if (track_unbuffer_callback) { if (may_yield) yield_codecs(); 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)); } } static void 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()); } if (voice_is_playing) { while (voice_is_playing && !queue_empty(&voice_codec_queue)) yield(); } filebufused = 0; playing = false; filling = false; paused = false; stop_codec_flush(); if (current_fd >= 0) { close(current_fd); current_fd = -1; } /* Mark all entries null. */ audio_clear_track_entries(true, false, false); memset(tracks, 0, sizeof(struct track_info) * MAX_TRACK); } static void audio_play_start(size_t offset) { #ifdef CONFIG_TUNER /* check if radio is playing */ if (get_radio_status() != FMRADIO_OFF) radio_stop(); #endif /* Wait for any previously playing audio to flush - TODO: Not necessary? */ while (audio_codec_loaded) stop_codec_flush(); track_changed = true; playlist_end = false; playing = true; ci.new_track = 0; ci.seek_time = 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; filebufused = 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); } /* Send callback events to notify about new tracks. */ static void generate_postbuffer_events(void) { int cur_idx; int last_idx = -1; logf("Postbuffer:%d/%d",track_ridx,track_widx); if (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); } } } static bool initialize_buffer_fill(bool clear_tracks) { /* Don't initialize if we're already initialized */ if (filling) return true; /* Don't start buffer fill if buffer is already full. */ if (filebufused > conf_watermark && !filling) return false; logf("Starting buffer fill"); fill_bytesleft = filebuflen - filebufused; /* TODO: This doesn't look right, and might explain some problems with * seeking in large files (to offsets larger than filebuflen). * And what about buffer wraps? * * This really doesn't look right, so don't use it. */ // if (buf_ridx > cur_ti->buf_idx) // cur_ti->start_pos = buf_ridx - cur_ti->buf_idx; /* 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(true, false, true); /* 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; if (!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) audio_read_file(false); else if (!audio_load_track(offset, start_play, rebuffer)) fill_bytesleft = 0; if (!had_next_track && audio_next_track()) track_changed = true; /* If we're done buffering */ if (fill_bytesleft == 0) { read_next_metadata(); generate_postbuffer_events(); filling = false; #ifndef SIMULATOR if (playing) ata_sleep(); #endif } } static void track_skip_done(bool was_manual) { /* Manual track change (always crossfade or flush audio). */ if (was_manual) { pcmbuf_crossfade_init(true); 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(pcmbuf_position_callback); pcmbuf_set_event_handler(pcmbuf_track_changed_callback); } } static bool load_next_track(void) { struct event ev; 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; } cpu_boost(true); 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; } cpu_boost(false); switch (ev.id) { case Q_CODEC_REQUEST_COMPLETE: track_skip_done(!automatic_skip); return true; case Q_CODEC_REQUEST_FAILED: ci.new_track = 0; ci.stop_codec = true; return false; default: logf("Bad event on ccq"); ci.stop_codec = true; return false; } } static bool voice_request_next_track_callback(void) { ci_voice.new_track = 0; return true; } 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 (!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; } } /* Invalidates all but currently playing track. */ void audio_invalidate_tracks(void) { if (have_tracks()) { last_peek_offset = 0; playlist_end = false; track_widx = track_ridx; audio_clear_track_entries(true, true, 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; /* Mark all other entries null (also buffered wrong metadata). */ filebufused = cur_ti->available; buf_widx = buf_ridx + cur_ti->available; if (buf_widx >= filebuflen) buf_widx -= filebuflen; 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 (have_tracks()) { playlist_end = false; track_widx = track_ridx; audio_clear_track_entries(true, true, 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 */ filebufused = cur_ti->available; buf_widx = buf_ridx + cur_ti->available; if (buf_widx >= filebuflen) buf_widx -= filebuflen; } /* 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 initiate_track_change(long direction) { if (playlist_check(direction)) { playlist_end = false; /* Flag track changed immediately so wps can update instantly. * No need to wait for disk to spin up or message to travel * through the deep queues as this info is only for the wps. */ track_changed = true; ci.new_track += direction; } } static void initiate_dir_change(long direction) { playlist_end = false; dir_skip = true; ci.new_track = direction; } void audio_thread(void) { struct event ev; /* At first initialize audio system in background. */ playback_init(); while (1) { if (filling) { queue_wait_w_tmo(&audio_queue, &ev, 0); if (ev.id == SYS_TIMEOUT) ev.id = Q_AUDIO_FILL_BUFFER; } else queue_wait_w_tmo(&audio_queue, &ev, HZ); switch (ev.id) { case 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: logf("starting..."); audio_clear_track_entries(true, false, true); audio_play_start((size_t)ev.data); break ; case Q_AUDIO_STOP: logf("audio_stop"); audio_stop_playback(); break ; case Q_AUDIO_PAUSE: logf("audio_%s",ev.data?"pause":"resume"); pcmbuf_pause((bool)ev.data); paused = (bool)ev.data; break ; case Q_AUDIO_SKIP: logf("audio_skip"); initiate_track_change((long)ev.data); break; case Q_AUDIO_PRE_FF_REWIND: if (!playing) break; logf("pre_ff_rewind"); pcmbuf_pause(true); break; case Q_AUDIO_FF_REWIND: if (!playing) break ; logf("ff_rewind"); ci.seek_time = (long)ev.data+1; break ; case Q_AUDIO_REBUFFER_SEEK: logf("Re-buffering song w/seek"); rebuffer_and_seek((size_t)ev.data); break; case Q_AUDIO_CHECK_NEW_TRACK: logf("Check new track buffer"); audio_check_new_track(); break; case Q_AUDIO_DIR_SKIP: logf("audio_dir_skip"); playlist_end = false; if (global_settings.beep) pcmbuf_beep(5000, 100, 2500*global_settings.beep); initiate_dir_change((long)ev.data); break; case Q_AUDIO_NEW_PLAYLIST: logf("new_playlist"); audio_new_playlist(); break; case Q_AUDIO_FLUSH: logf("flush & reload"); audio_invalidate_tracks(); break ; case 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 ; #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: break; } } } 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: logf("Codec load disk"); audio_codec_loaded = true; if (voice_codec_loaded) queue_post(&voice_codec_queue, Q_AUDIO_PLAY, 0); mutex_lock(&mutex_codecthread); current_codec = CODEC_IDX_AUDIO; ci.stop_codec = false; status = codec_load_file((const char *)ev.data, &ci); mutex_unlock(&mutex_codecthread); break ; case Q_CODEC_LOAD: logf("Codec load ram"); 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; queue_post(&codec_queue, Q_AUDIO_PLAY, 0); break ; } audio_codec_loaded = true; if (voice_codec_loaded) queue_post(&voice_codec_queue, Q_AUDIO_PLAY, 0); mutex_lock(&mutex_codecthread); 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); mutex_unlock(&mutex_codecthread); break ; #ifndef SIMULATOR case SYS_USB_CONNECTED: queue_clear(&codec_queue); logf("USB: Audio codec"); usb_acknowledge(SYS_USB_CONNECTED_ACK); if (voice_codec_loaded) swap_codec(); usb_wait_for_disconnect(&codec_queue); break ; #endif } 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: if (playing) { const char *codec_path; if (ci.new_track || status != CODEC_OK) { if (!ci.new_track) { logf("Codec failure"); gui_syncsplash(HZ*2, true, "Codec failure"); } if (!load_next_track()) { queue_post(&codec_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()) sleep(1); queue_post(&audio_queue, Q_AUDIO_STOP, 0); break; } } if (cur_ti->has_codec) queue_post(&codec_queue, Q_CODEC_LOAD, 0); else { codec_path = get_codec_path(cur_ti->id3.codectype); queue_post(&codec_queue, Q_CODEC_LOAD_DISK, (void *)codec_path); } } } } } static void reset_buffer(void) { size_t offset; filebuf = (char *)&audiobuf[MALLOC_BUFSIZE]; filebuflen = audiobufend - audiobuf - MALLOC_BUFSIZE - GUARD_BUFSIZE - (pcmbuf_get_bufsize() + get_pcmbuf_descsize() + PCMBUF_MIX_CHUNK * 2); if (talk_get_bufsize()) { filebuf = &filebuf[talk_get_bufsize()]; filebuflen -= 2*CODEC_IRAM_SIZE + 2*CODEC_SIZE + talk_get_bufsize(); #ifndef SIMULATOR iram_buf[0] = &filebuf[filebuflen]; iram_buf[1] = &filebuf[filebuflen+CODEC_IRAM_SIZE]; #endif dram_buf[0] = (unsigned char *)&filebuf[filebuflen+CODEC_IRAM_SIZE*2]; dram_buf[1] = (unsigned char *)&filebuf[filebuflen+CODEC_IRAM_SIZE*2+CODEC_SIZE]; } /* Ensure that everything is aligned */ offset = (-(size_t)filebuf) & 3; filebuf += offset; filebuflen -= offset; filebuflen &= ~3; } static void voice_codec_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(CODEC_MPA_L3, &ci_voice); logf("Voice codec finished"); mutex_unlock(&mutex_codecthread); voice_codec_loaded = false; } } void voice_init(void) { if (!filebuf) return; /* Audio buffers not yet set up */ if (voice_thread_num >= 0) { logf("Terminating voice codec"); remove_thread(voice_thread_num); if (current_codec == CODEC_IDX_VOICE) mutex_unlock(&mutex_codecthread); queue_delete(&voice_codec_queue); voice_thread_num = -1; voice_codec_loaded = false; } if (!talk_get_bufsize()) return ; logf("Starting voice codec"); queue_init(&voice_codec_queue); voice_thread_num = create_thread(voice_codec_thread, voice_codec_stack, sizeof(voice_codec_stack), voice_codec_thread_name); while (!voice_codec_loaded) yield(); } struct mp3entry* audio_current_track(void) { const char *filename; const char *p; static struct mp3entry temp_id3; int cur_idx; cur_idx = track_ridx + ci.new_track; 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(ci.new_track); 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 (!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"); if (playing && offset <= 0) queue_post(&audio_queue, Q_AUDIO_NEW_PLAYLIST, 0); else { if (playing) audio_stop(); playing = true; queue_post(&audio_queue, Q_AUDIO_PLAY, (void *)offset); } } void audio_stop(void) { queue_post(&audio_queue, Q_AUDIO_STOP, 0); while (playing || audio_codec_loaded) yield(); } bool mp3_pause_done(void) { return pcm_is_paused(); } void audio_pause(void) { queue_post(&audio_queue, Q_AUDIO_PAUSE, (void *)true); } void audio_resume(void) { queue_post(&audio_queue, Q_AUDIO_PAUSE, (void *)false); } void audio_next(void) { if (global_settings.beep) pcmbuf_beep(5000, 100, 2500*global_settings.beep); /* Should be safe to do outside of thread, that way we get * the instant wps response at least. */ initiate_track_change(1); // queue_post(&audio_queue, Q_AUDIO_SKIP, (void *)1); } void audio_prev(void) { if (global_settings.beep) pcmbuf_beep(5000, 100, 2500*global_settings.beep); initiate_track_change(-1); // queue_post(&audio_queue, Q_AUDIO_SKIP, (void *)-1); } void audio_next_dir(void) { queue_post(&audio_queue, Q_AUDIO_DIR_SKIP, (void *)1); } void audio_prev_dir(void) { queue_post(&audio_queue, Q_AUDIO_DIR_SKIP, (void *)-1); } void audio_pre_ff_rewind(void) { queue_post(&audio_queue, Q_AUDIO_PRE_FF_REWIND, 0); } void audio_ff_rewind(long newpos) { queue_post(&audio_queue, Q_AUDIO_FF_REWIND, (int *)newpos); } void audio_flush_and_reload_tracks(void) { queue_post(&audio_queue, Q_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; } /* TODO: 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 */ if (id3->bitrate != 0) id3->elapsed = id3->offset / (id3->bitrate / 8); } } /* Copied from mpeg.c. Should be moved somewhere else. */ static 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; /* 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; /* skip past id3v2 tag and other leading garbage */ else if (pos < (int)id3->first_frame_offset) 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)) { static struct voice_info voice_clip; voice_clip.callback = get_more; voice_clip.buf = (char *)start; voice_clip.size = size; queue_post(&voice_codec_queue, Q_VOICE_STOP, 0); queue_post(&voice_codec_queue, Q_VOICE_PLAY, &voice_clip); voice_is_playing = true; voice_boost_cpu(true); } void mp3_play_stop(void) { queue_post(&voice_codec_queue, Q_VOICE_STOP, 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; int seconds = 1; if (!filebuf) return; /* Audio buffers not yet set up */ 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); if (pcmbuf_get_bufsize() == 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. */ queue_post(&audio_queue, Q_AUDIO_STOP, 0); while (audio_codec_loaded) yield(); gui_syncsplash(0, true, (char *)str(LANG_RESTARTING_PLAYBACK)); } /* Re-initialize audio system. */ pcmbuf_init(size); pcmbuf_crossfade_enable(enable); reset_buffer(); logf("abuf:%dB", pcmbuf_get_bufsize()); logf("fbuf:%dB", filebuflen); voice_init(); /* Restart playback. */ if (was_playing) { playing = true; queue_post(&audio_queue, Q_AUDIO_PLAY, (void *)offset); /* Wait for the playback to start again (and display the splash screen during that period. */ while (playing && !audio_codec_loaded) yield(); } } void mpeg_id3_options(bool _v1first) { v1first = _v1first; } #ifdef ROCKBOX_HAS_LOGF void test_track_changed_event(struct mp3entry *id3) { (void)id3; logf("tce:%s", id3->path); } #endif static void playback_init(void) { static bool voicetagtrue = true; struct event ev; logf("playback api init"); pcm_init(); #if defined(HAVE_RECORDING) && !defined(SIMULATOR) /* Set the input multiplexer to Line In */ pcm_rec_mux(0); #endif #ifdef ROCKBOX_HAS_LOGF audio_set_track_changed_event(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 = 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; ci.discard_codec = codec_discard_codec_callback; /* Initialize voice codec api. */ 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 = get_voice_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; create_thread(codec_thread, codec_stack, sizeof(codec_stack), codec_thread_name); 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 } filebuf = (char *)&audiobuf[MALLOC_BUFSIZE]; audio_set_crossfade(global_settings.crossfade); audio_is_initialized = true; sound_settings_apply(); } void audio_preinit(void) { logf("playback system pre-init"); filebufused = 0; filling = false; current_codec = CODEC_IDX_AUDIO; playing = false; paused = false; audio_codec_loaded = false; voice_is_playing = 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]; mutex_init(&mutex_codecthread); queue_init(&audio_queue); queue_init(&codec_queue); /* clear, not init to create a private queue */ queue_clear(&codec_callback_queue); create_thread(audio_thread, audio_stack, sizeof(audio_stack), audio_thread_name); } void audio_init(void) { logf("playback system post-init"); queue_post(&audio_queue, Q_AUDIO_POSTINIT, 0); }