03f373c9cd
This unifies the talk.c for all targets. The only separation is left is TALK_PROGRESSIVE_LOAD: When this is defined the talk buffer will not be initially prefilled. This is useful for super slow storage or when the buffer is not large enough to prefill it with useful clips (the prefill code could be made smarter too). The buffer size can be adjusted. By default lowmem uses 100k while other targets load the entire file. The bigger the more clips can be cached but with diminishing returns. Change-Id: Ife38fb101c53093117e6638c40c65f7d177a31d4
1612 lines
48 KiB
C
1612 lines
48 KiB
C
/***************************************************************************
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* __________ __ ___.
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* Open \______ \ ____ ____ | | _\_ |__ _______ ___
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* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
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* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
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* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
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* \/ \/ \/ \/ \/
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* $Id$
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*
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* Copyright (C) 2004 Jörg Hohensohn
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*
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* This module collects the Talkbox and voice UI functions.
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* (Talkbox reads directory names from mp3 clips called thumbnails,
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* the voice UI lets menus and screens "talk" from a voicefile in memory.
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*
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* This program is free software; you can redistribute it and/or
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* modify it under the terms of the GNU General Public License
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* as published by the Free Software Foundation; either version 2
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* of the License, or (at your option) any later version.
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*
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* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
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* KIND, either express or implied.
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*
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****************************************************************************/
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#include <stdio.h>
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#include <stddef.h>
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#include "string-extra.h"
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#include "file.h"
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#include "system.h"
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#include "kernel.h"
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#include "settings.h"
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#include "settings_list.h"
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#if CONFIG_CODEC == SWCODEC
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#include "voice_thread.h"
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#else
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#include "mp3_playback.h"
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#endif
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#include "audio.h"
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#include "lang.h"
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#include "talk.h"
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#include "metadata.h"
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/*#define LOGF_ENABLE*/
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#include "logf.h"
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#include "bitswap.h"
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#include "structec.h"
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#include "plugin.h" /* plugin_get_buffer() */
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#include "debug.h"
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#include "panic.h"
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/* Memory layout varies between targets because the
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Archos (MASCODEC) devices cannot mix voice and audio playback
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MASCODEC | MASCODEC | SWCODEC
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(playing) | (stopped) |
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voicebuf-----------+-----------+------------
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audio | voice | voice
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|-----------|------------
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| thumbnail | thumbnail
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| |------------
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| | filebuf
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| |------------
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| | audio
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voicebufend----------+-----------+------------
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SWCODEC allocates dedicated buffers (except voice and thumbnail are together
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in the talkbuf), MASCODEC reuses audiobuf. */
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/***************** Constants *****************/
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#define QUEUE_SIZE 64 /* must be a power of two */
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#define QUEUE_MASK (QUEUE_SIZE-1)
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const char* const dir_thumbnail_name = "_dirname.talk";
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const char* const file_thumbnail_ext = ".talk";
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/***************** Functional Macros *****************/
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#define QUEUE_LEVEL ((queue_write - queue_read) & QUEUE_MASK)
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#define LOADED_MASK 0x80000000 /* MSB */
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/* swcodec: cap thumbnail buffer to MAX_THUMNAIL_BUFSIZE since audio keeps
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* playing while voice
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* hwcodec: just use whatever is left in the audiobuffer, music
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* playback is impossible => no cap */
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#if CONFIG_CODEC == SWCODEC
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#define MAX_THUMBNAIL_BUFSIZE 0x10000
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#endif
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#define DEFAULT_VOICE_LANG "english"
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/***************** Data types *****************/
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struct clip_entry /* one entry of the index table */
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{
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int offset; /* offset from start of voicefile file */
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int size; /* size of the clip */
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};
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struct voicefile_header /* file format of our voice file */
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{
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int version; /* version of the voicefile */
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int target_id; /* the rockbox target the file was made for */
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int table; /* offset to index table, (=header size) */
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int id1_max; /* number of "normal" clips contained in above index */
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int id2_max; /* number of "voice only" clips contained in above index */
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/* The header is folled by the index tables (n*struct clip_entry),
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* which is followed by the mp3/speex encoded clip data */
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};
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/***************** Globals *****************/
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#if (CONFIG_CODEC == SWCODEC && MEMORYSIZE <= 2) || defined(ONDIO_SERIES)
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/* On low memory swcodec targets the entire voice file wouldn't fit in memory
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* together with codecs, so we load clips each time they are accessed.
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* The Ondios have slow storage access and loading the entire voice file would
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* take several seconds, so we use the same mechanism. */
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#define TALK_PROGRESSIVE_LOAD
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#if !defined(ONDIO_SERIES)
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/* 70+ clips should fit into 100k */
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#define MAX_CLIP_BUFFER_SIZE (100000)
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#endif
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#endif
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#ifndef MAX_CLIP_BUFFER_SIZE
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#define MAX_CLIP_BUFFER_SIZE (1<<30) /* 1GB should be enough for everybody */
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#endif
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/* Multiple thumbnails can be loaded back-to-back in this buffer. */
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static volatile int thumbnail_buf_used SHAREDBSS_ATTR; /* length of data in
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thumbnail buffer */
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static long size_for_thumbnail; /* total thumbnail buffer size */
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static struct voicefile_header voicefile; /* loaded voicefile */
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static bool has_voicefile; /* a voicefile file is present */
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static bool need_shutup; /* is there possibly any voice playing to be shutup */
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static bool force_enqueue_next; /* enqueue next utterance even if enqueue is false */
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static int queue_write; /* write index of queue, by application */
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static int queue_read; /* read index of queue, by ISR context */
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#if CONFIG_CODEC == SWCODEC
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/* protects queue_read, queue_write and thumbnail_buf_used */
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static struct mutex queue_mutex SHAREDBSS_ATTR;
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#define talk_queue_lock() ({ mutex_lock(&queue_mutex); })
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#define talk_queue_unlock() ({ mutex_unlock(&queue_mutex); })
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#else
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#define talk_queue_lock() ({ })
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#define talk_queue_unlock() ({ })
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#endif /* CONFIG_CODEC */
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static int sent; /* how many bytes handed over to playback, owned by ISR */
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static unsigned char curr_hd[3]; /* current frame header, for re-sync */
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static unsigned char last_lang[MAX_FILENAME+1]; /* name of last used lang file (in talk_init) */
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static bool talk_initialized; /* true if talk_init has been called */
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static bool give_buffer_away; /* true if we should give the buffers away in shrink_callback if requested */
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static int talk_temp_disable_count; /* if positive, temporarily disable voice UI (not saved) */
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/* size of the voice data in the voice file and the actually allocated buffer
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* for it. voicebuf_size is always smaller or equal to voicefile_size */
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static unsigned long voicefile_size, voicebuf_size;
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struct queue_entry /* one entry of the internal queue */
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{
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int offset; /* actually a buflib handle if type == HANDLE */
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int length; /* total length of the clip */
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int remaining; /* bytes that still need to be deoded */
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/* for large QUEUE_SIZE values it might be worthwhile to merge the type
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* into the bits of the above members (as a space saver). For small values
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* the required extra code outweights this so it's not done here */
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enum offset_type {
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TALK_HANDLE,
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THUMB_OFFSET,
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} type;
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};
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static struct buflib_context clip_ctx;
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struct clip_cache_metadata {
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long tick;
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int handle, voice_id;
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};
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static int metadata_table_handle;
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static unsigned max_clips;
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static int cache_hits, cache_misses;
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static struct queue_entry queue[QUEUE_SIZE]; /* queue of scheduled clips */
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static struct queue_entry silence, *last_clip;
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/***************** Private implementation *****************/
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static int index_handle, talk_handle, thumb_handle;
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static int move_callback(int handle, void *current, void *new)
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{
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(void)handle; (void)current; (void)new;
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if (handle == talk_handle && !buflib_context_relocate(&clip_ctx, new))
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return BUFLIB_CB_CANNOT_MOVE;
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return BUFLIB_CB_OK;
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}
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static struct mutex read_buffer_mutex;
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/* on HWCODEC only voice xor audio can be active at a time */
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static bool check_audio_status(void)
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{
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#if CONFIG_CODEC != SWCODEC
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if (audio_status()) /* busy, buffer in use */
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return false;
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/* ensure playback is given up on the buffer */
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audio_hard_stop();
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#endif
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return true;
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}
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/* ISR (mp3_callback()) must not run during moving of the clip buffer,
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* because the MAS may get out-of-sync */
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static void sync_callback(int handle, bool sync_on)
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{
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(void) handle;
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if (sync_on)
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mutex_lock(&read_buffer_mutex);
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else
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mutex_unlock(&read_buffer_mutex);
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}
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static ssize_t read_to_handle_ex(int fd, struct buflib_context *ctx, int handle,
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int handle_offset, size_t count)
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{
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unsigned char *buf;
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ssize_t ret;
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mutex_lock(&read_buffer_mutex);
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if (!ctx)
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buf = core_get_data(handle);
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else
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buf = buflib_get_data(ctx, handle);
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buf += handle_offset;
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ret = read(fd, buf, count);
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mutex_unlock(&read_buffer_mutex);
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return ret;
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}
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static ssize_t read_to_handle(int fd, int handle, int handle_offset, size_t count)
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{
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return read_to_handle_ex(fd, NULL, handle, handle_offset, count);
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}
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static int shrink_callback(int handle, unsigned hints, void *start, size_t old_size)
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{
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(void)start;(void)old_size;(void)hints;
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int *h;
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#if (MAX_CLIP_BUFFER_SIZE < (MEMORYSIZE<<20) || (MEMORYSIZE > 2))
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/* on low-mem and when the voice buffer size is not limited (i.e.
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* on 2MB HWCODEC) we effectively own the entire buffer because
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* the voicefile takes up all RAM. This blocks other Rockbox parts
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* from allocating, especially during bootup. Therefore always give
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* up the buffer and reload when clips are played back. On high-mem
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* or when the clip buffer is limited to a few 100K this provision is
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* not necessary. */
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if (give_buffer_away
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&& (hints & BUFLIB_SHRINK_POS_MASK) == BUFLIB_SHRINK_POS_MASK)
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#endif
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{
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if (handle == talk_handle)
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h = &talk_handle;
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else //if (handle == index_handle)
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h = &index_handle;
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mutex_lock(&read_buffer_mutex);
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/* the clip buffer isn't usable without index table */
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if (handle == index_handle && talk_handle > 0)
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talk_handle = core_free(talk_handle);
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*h = core_free(handle);
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mutex_unlock(&read_buffer_mutex);
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return BUFLIB_CB_OK;
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}
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return BUFLIB_CB_CANNOT_SHRINK;
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}
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static int thumb_shrink_callback(int handle, unsigned hints, void *start, size_t old_size)
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{
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(void)start;(void)old_size;(void)hints;
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if (handle == thumb_handle && thumbnail_buf_used == 0)
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{
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mutex_lock(&read_buffer_mutex);
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thumb_handle = core_free(handle);
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mutex_unlock(&read_buffer_mutex);
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return BUFLIB_CB_OK;
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}
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return BUFLIB_CB_CANNOT_SHRINK;
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}
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static struct buflib_callbacks talk_ops = {
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.move_callback = move_callback,
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.sync_callback = sync_callback,
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.shrink_callback = shrink_callback,
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};
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static struct buflib_callbacks thumb_ops = {
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.move_callback = move_callback,
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.sync_callback = sync_callback,
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.shrink_callback = thumb_shrink_callback,
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};
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static int open_voicefile(void)
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{
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char buf[64];
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char* p_lang = DEFAULT_VOICE_LANG; /* default */
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if ( global_settings.lang_file[0] &&
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global_settings.lang_file[0] != 0xff )
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{ /* try to open the voice file of the selected language */
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p_lang = (char *)global_settings.lang_file;
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}
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snprintf(buf, sizeof(buf), LANG_DIR "/%s.voice", p_lang);
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return open(buf, O_RDONLY);
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}
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static int id2index(int id)
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{
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int index = id;
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if (id > VOICEONLY_DELIMITER)
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{ /* voice-only entries use the second part of the table.
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The first string comes after VOICEONLY_DELIMITER so we need to
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substract VOICEONLY_DELIMITER + 1 */
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index -= VOICEONLY_DELIMITER + 1;
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if (index >= voicefile.id2_max)
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return -1; /* must be newer than we have */
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index += voicefile.id1_max; /* table 2 is behind table 1 */
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}
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else
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{ /* normal use of the first table */
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if (id >= voicefile.id1_max)
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return -1; /* must be newer than we have */
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}
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return index;
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}
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static int index2id(int index)
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{
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int id = index;
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if (index >= voicefile.id2_max + voicefile.id1_max)
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return -1;
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if (index >= voicefile.id1_max)
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{ /* must be voice-only if it exceeds table 1 */
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id -= voicefile.id1_max;
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/* The first string comes after VOICEONLY_DELIMITER so we need to
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add VOICEONLY_DELIMITER + 1 */
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id += VOICEONLY_DELIMITER + 1;
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}
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return id;
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}
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static int free_oldest_clip(void)
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{
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unsigned i;
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int oldest = 0;
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long age, now;
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struct clip_entry* clipbuf;
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struct clip_cache_metadata *cc = buflib_get_data(&clip_ctx, metadata_table_handle);
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for(age = i = 0, now = current_tick; i < max_clips; i++)
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{
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if (cc[i].handle && (now - cc[i].tick) > age
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&& cc[i].voice_id != VOICE_PAUSE) /* never consider silence */
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{
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age = now - cc[i].tick;
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oldest = i;
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}
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}
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cc = &cc[oldest];
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cc->handle = buflib_free(&clip_ctx, cc->handle);
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/* need to clear the LOADED bit too */
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clipbuf = core_get_data(index_handle);
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clipbuf[id2index(cc->voice_id)].size &= ~LOADED_MASK;
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return oldest;
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}
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/* common code for load_initial_clips() and get_clip() */
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static void add_cache_entry(int clip_handle, int table_index, int id)
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{
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unsigned i;
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struct clip_cache_metadata *cc = buflib_get_data(&clip_ctx, metadata_table_handle);
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if (table_index != -1)
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{
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/* explicit slot; use that */
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cc = &cc[table_index];
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if (cc->handle > 0) panicf("%s(): Slot already used", __func__);
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}
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else
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{ /* find an empty slot */
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for(i = 0; cc[i].handle && i < max_clips; i++) ;
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if (i == max_clips) /* no free slot in the cache table? */
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i = free_oldest_clip();
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cc = &cc[i];
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}
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cc->handle = clip_handle;
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cc->tick = current_tick;
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cc->voice_id = id;
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}
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static ssize_t read_clip_data(int fd, int index, int clip_handle)
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{
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struct clip_entry* clipbuf;
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size_t clipsize;
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ssize_t ret;
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if (fd < 0)
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{
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buflib_free(&clip_ctx, clip_handle);
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return -1; /* open error */
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}
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clipbuf = core_get_data(index_handle);
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/* this must not be called with LOADED_MASK set in clipsize */
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clipsize = clipbuf[index].size;
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lseek(fd, clipbuf[index].offset, SEEK_SET);
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ret = read_to_handle_ex(fd, &clip_ctx, clip_handle, 0, clipsize);
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if (ret < 0 || clipsize != (size_t)ret)
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{
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buflib_free(&clip_ctx, clip_handle);
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return -2; /* read error */
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}
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clipbuf = core_get_data(index_handle);
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clipbuf[index].size |= LOADED_MASK; /* mark as loaded */
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return ret;
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}
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static void load_initial_clips(int fd)
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{
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(void) fd;
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#ifndef TALK_PROGRESSIVE_LOAD
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unsigned index, i;
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unsigned num_clips = voicefile.id1_max + voicefile.id2_max;
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for(index = i = 0; index < num_clips && i < max_clips; index++)
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{
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int handle;
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struct clip_entry* clipbuf = core_get_data(index_handle);
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size_t clipsize = clipbuf[index].size;
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if (clipsize == 0) /* clip not included in voicefile */
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continue;
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handle = buflib_alloc(&clip_ctx, clipsize);
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if (handle < 0)
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break;
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if (read_clip_data(fd, index, handle) < 0)
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break;
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add_cache_entry(handle, i++, index2id(index));
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}
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#endif
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}
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/* fetch a clip from the voice file */
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static int get_clip(long id, struct queue_entry *q)
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{
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int index;
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int retval = -1;
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struct clip_entry* clipbuf;
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size_t clipsize;
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enum offset_type type;
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index = id2index(id);
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if (index == -1)
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return -1;
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clipbuf = core_get_data(index_handle);
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clipsize = clipbuf[index].size;
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if (clipsize == 0) /* clip not included in voicefile */
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return -1;
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if (!(clipsize & LOADED_MASK))
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{ /* clip needs loading */
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int fd, handle, oldest = -1;
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ssize_t ret;
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cache_misses++;
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/* free clips from cache until this one succeeds to allocate */
|
|
while ((handle = buflib_alloc(&clip_ctx, clipsize)) < 0)
|
|
oldest = free_oldest_clip();
|
|
/* handle should now hold a valid alloc. Load from disk
|
|
* and insert into cache */
|
|
fd = open_voicefile();
|
|
ret = read_clip_data(fd, index, handle);
|
|
close(fd);
|
|
if (ret < 0)
|
|
return ret;
|
|
/* finally insert into metadata table */
|
|
add_cache_entry(handle, oldest, id);
|
|
retval = handle;
|
|
}
|
|
else
|
|
{ /* clip is in memory already; find where it was loaded */
|
|
cache_hits++;
|
|
struct clip_cache_metadata *cc;
|
|
static int i;
|
|
cc = buflib_get_data(&clip_ctx, metadata_table_handle);
|
|
for (i = 0; cc[i].voice_id != id || !cc[i].handle; i++) ;
|
|
cc[i].tick = current_tick; /* reset age */
|
|
clipsize &= ~LOADED_MASK; /* without the extra bit gives true size */
|
|
retval = cc[i].handle;
|
|
}
|
|
type = TALK_HANDLE;
|
|
|
|
q->offset = retval;
|
|
q->length = clipsize;
|
|
q->remaining = clipsize;
|
|
q->type = type;
|
|
return 0;
|
|
}
|
|
|
|
static bool load_index_table(int fd, const struct voicefile_header *hdr)
|
|
{
|
|
ssize_t ret;
|
|
struct clip_entry *buf;
|
|
|
|
if (index_handle > 0) /* nothing to do? */
|
|
return true;
|
|
|
|
ssize_t alloc_size = (hdr->id1_max + hdr->id2_max) * sizeof(struct clip_entry);
|
|
index_handle = core_alloc_ex("voice index", alloc_size, &talk_ops);
|
|
if (index_handle < 0)
|
|
return false;
|
|
|
|
ret = read_to_handle(fd, index_handle, 0, alloc_size);
|
|
|
|
if (ret == alloc_size)
|
|
{
|
|
buf = core_get_data(index_handle);
|
|
for (int i = 0; i < hdr->id1_max + hdr->id2_max; i++)
|
|
{
|
|
#ifdef ROCKBOX_LITTLE_ENDIAN
|
|
/* doesn't yield() */
|
|
structec_convert(&buf[i], "ll", 1, true);
|
|
#endif
|
|
}
|
|
}
|
|
else
|
|
index_handle = core_free(index_handle);
|
|
|
|
return ret == alloc_size;
|
|
}
|
|
|
|
static bool load_header(int fd, struct voicefile_header *hdr)
|
|
{
|
|
ssize_t got_size = read(fd, hdr, sizeof(*hdr));
|
|
if (got_size != sizeof(*hdr))
|
|
return false;
|
|
|
|
#ifdef ROCKBOX_LITTLE_ENDIAN
|
|
logf("Byte swapping voice file");
|
|
structec_convert(&voicefile, "lllll", 1, true);
|
|
#endif
|
|
return true;
|
|
}
|
|
|
|
static bool create_clip_buffer(size_t max_size)
|
|
{
|
|
size_t alloc_size;
|
|
/* just allocate, populate on an as-needed basis later */
|
|
talk_handle = core_alloc_ex("voice data", max_size, &talk_ops);
|
|
if (talk_handle < 0)
|
|
goto alloc_err;
|
|
|
|
buflib_init(&clip_ctx, core_get_data(talk_handle), max_size);
|
|
|
|
/* the first alloc is the clip metadata table */
|
|
alloc_size = max_clips * sizeof(struct clip_cache_metadata);
|
|
metadata_table_handle = buflib_alloc(&clip_ctx, alloc_size);
|
|
memset(buflib_get_data(&clip_ctx, metadata_table_handle), 0, alloc_size);
|
|
|
|
return true;
|
|
|
|
alloc_err:
|
|
index_handle = core_free(index_handle);
|
|
return false;
|
|
}
|
|
|
|
static bool alloc_thumbnail_buf(void)
|
|
{
|
|
int handle;
|
|
size_t size;
|
|
if (thumb_handle > 0)
|
|
return true; /* nothing to do? */
|
|
#if CONFIG_CODEC == SWCODEC
|
|
/* try to allocate the max. first, and take whatever we can get if that
|
|
* fails */
|
|
size = MAX_THUMBNAIL_BUFSIZE;
|
|
handle = core_alloc_ex("voice thumb", MAX_THUMBNAIL_BUFSIZE, &thumb_ops);
|
|
if (handle < 0)
|
|
{
|
|
size = core_allocatable();
|
|
handle = core_alloc_ex("voice thumb", size, &thumb_ops);
|
|
}
|
|
#else
|
|
/* on HWCODEC, just use the rest of the remaining buffer,
|
|
* normal playback cannot happen anyway */
|
|
size = audio_buffer_available();
|
|
handle = core_alloc_ex("voice thumb", size, &thumb_ops);
|
|
#endif
|
|
thumb_handle = handle;
|
|
size_for_thumbnail = (handle > 0) ? size : 0;
|
|
return handle > 0;
|
|
}
|
|
|
|
/* load the voice file into the mp3 buffer */
|
|
static bool load_voicefile_index(int fd)
|
|
{
|
|
if (fd < 0) /* failed to open */
|
|
return false;
|
|
|
|
/* load the header first */
|
|
if (!load_header(fd, &voicefile))
|
|
return false;
|
|
|
|
/* format check */
|
|
if (voicefile.table == sizeof(struct voicefile_header))
|
|
{
|
|
if (voicefile.version == VOICE_VERSION &&
|
|
voicefile.target_id == TARGET_ID)
|
|
{
|
|
if (load_index_table(fd, &voicefile))
|
|
return true;
|
|
}
|
|
}
|
|
|
|
logf("Incompatible voice file");
|
|
return false;
|
|
}
|
|
|
|
/* this function caps the voicefile buffer and allocates it. It can
|
|
* be called after talk_init(), e.g. when the voice was temporarily disabled.
|
|
* The buffer size has to be capped again each time because the available
|
|
* audio buffer changes over time */
|
|
static bool load_voicefile_data(int fd)
|
|
{
|
|
voicebuf_size = voicefile_size;
|
|
/* cap to the max. number of clips or the size of the available audio
|
|
* buffer which we grab. We leave some to the rest of the system.
|
|
* While that reduces our buffer size it improves the chance that
|
|
* other allocs succeed without disabling voice which would require
|
|
* reloading the voice from disk (as we do not shrink our buffer when
|
|
* other code attempts new allocs these would fail) */
|
|
ssize_t cap = MIN(MAX_CLIP_BUFFER_SIZE, audio_buffer_available() - (64<<10));
|
|
if (UNLIKELY(cap < 0))
|
|
{
|
|
logf("Not enough memory for voice. Disabling...\n");
|
|
return false;
|
|
}
|
|
else if (voicebuf_size > (size_t)cap)
|
|
voicebuf_size = cap;
|
|
|
|
/* just allocate, populate on an as-needed basis later
|
|
* re-create the clip buffer to ensure clip_ctx is up-to-date */
|
|
if (talk_handle > 0)
|
|
talk_handle = core_free(talk_handle);
|
|
if (!create_clip_buffer(voicebuf_size))
|
|
return false;
|
|
|
|
load_initial_clips(fd);
|
|
/* make sure to have the silence clip, if available return value can
|
|
* be cached globally even for TALK_PROGRESSIVE_LOAD because the
|
|
* VOICE_PAUSE clip is specially handled */
|
|
get_clip(VOICE_PAUSE, &silence);
|
|
|
|
/* not an error if this fails here, might try again when the
|
|
* actual thumbnails are attempted to be played back */
|
|
alloc_thumbnail_buf();
|
|
|
|
return true;
|
|
}
|
|
|
|
/* most, if not all, clips should be well below 32k (largest in english.lang is
|
|
* 4.5K). Currently there is a problem with voice decoding such that clips
|
|
* cannot be decoded in chunks. Once that is resolved this buffer could be
|
|
* smaller and clips be decoded in multiple chunks */
|
|
static unsigned char commit_buffer[32<<10];
|
|
|
|
static void* commit_transfer(struct queue_entry *qe, size_t *size)
|
|
{
|
|
void *buf = NULL; /* shut up gcc */
|
|
static unsigned char *bufpos = commit_buffer;
|
|
int offset = qe->offset;
|
|
#if CONFIG_CODEC != SWCODEC
|
|
sent = MIN(qe->remaining, 0xFFFF);
|
|
#else
|
|
sent = qe->remaining;
|
|
#endif
|
|
sent = MIN((size_t)sent, sizeof(commit_buffer));
|
|
switch (qe->type)
|
|
{
|
|
case TALK_HANDLE: buf = buflib_get_data(&clip_ctx, offset); break;
|
|
case THUMB_OFFSET: buf = core_get_data(thumb_handle) + offset; break;
|
|
}
|
|
/* adjust buffer position to what has been played already */
|
|
buf += (qe->length - qe->remaining);
|
|
memcpy(bufpos, buf, sent);
|
|
*size = sent;
|
|
|
|
|
|
return commit_buffer;
|
|
}
|
|
|
|
static inline bool is_silence(struct queue_entry *qe)
|
|
{
|
|
if (silence.length > 0) /* silence clip available? */
|
|
return (qe->offset == silence.offset && qe->type == silence.type);
|
|
else
|
|
return false;
|
|
}
|
|
|
|
/* called in ISR context (on HWCODEC) if mp3 data got consumed */
|
|
static void mp3_callback(const void** start, size_t* size)
|
|
{
|
|
struct queue_entry *qe = &queue[queue_read];
|
|
qe->remaining -= sent; /* we completed this */
|
|
|
|
if (qe->remaining > 0) /* current clip not finished? */
|
|
{ /* feed the next 64K-1 chunk */
|
|
*start = commit_transfer(qe, size);
|
|
return;
|
|
}
|
|
|
|
talk_queue_lock();
|
|
/* check if thumbnails have been played */
|
|
if (qe->type == THUMB_OFFSET)
|
|
{
|
|
if (qe->remaining == 0 && (qe->length + qe->offset) == thumbnail_buf_used)
|
|
thumbnail_buf_used = 0;
|
|
}
|
|
|
|
/* increment read position for the just played clip */
|
|
queue_read = (queue_read + 1) & QUEUE_MASK;
|
|
|
|
if (QUEUE_LEVEL == 0)
|
|
{
|
|
if (!is_silence(last_clip) && last_clip->type != THUMB_OFFSET)
|
|
{ /* add silence clip when queue runs empty playing a voice clip,
|
|
* only if the previous clip wasn't silence or thumbnail */
|
|
queue[queue_write] = silence;
|
|
queue_write = (queue_write + 1) & QUEUE_MASK;
|
|
}
|
|
else
|
|
{
|
|
*size = 0; /* end of data */
|
|
}
|
|
}
|
|
|
|
if (QUEUE_LEVEL != 0) /* queue is not empty? */
|
|
{ /* start next clip */
|
|
last_clip = &queue[queue_read];
|
|
*start = commit_transfer(last_clip, size);
|
|
curr_hd[0] = commit_buffer[1];
|
|
curr_hd[1] = commit_buffer[2];
|
|
curr_hd[2] = commit_buffer[3];
|
|
}
|
|
|
|
talk_queue_unlock();
|
|
}
|
|
|
|
/***************** Public routines *****************/
|
|
|
|
/* stop the playback and the pending clips */
|
|
void talk_force_shutup(void)
|
|
{
|
|
/* Most of this is MAS only */
|
|
#if CONFIG_CODEC != SWCODEC
|
|
#ifdef SIMULATOR
|
|
return;
|
|
#endif
|
|
unsigned char* pos;
|
|
unsigned char* search;
|
|
unsigned char* end;
|
|
int len;
|
|
unsigned offset;
|
|
if (QUEUE_LEVEL == 0) /* has ended anyway */
|
|
return;
|
|
|
|
#if CONFIG_CPU == SH7034
|
|
CHCR3 &= ~0x0001; /* disable the DMA (and therefore the interrupt also) */
|
|
#endif /* CONFIG_CPU == SH7034 */
|
|
/* search next frame boundary and continue up to there */
|
|
pos = search = mp3_get_pos();
|
|
offset = queue[queue_read].offset;
|
|
switch (queue[queue_read].type)
|
|
{
|
|
case THUMB_OFFSET: end = core_get_data(thumb_handle) + offset; break;
|
|
case TALK_HANDLE: end = buflib_get_data(&clip_ctx, offset); break;
|
|
default: end = NULL; /* shut up gcc */
|
|
}
|
|
len = queue[queue_read].length;
|
|
|
|
if (pos >= end && pos <= (end+len)) /* really our clip? */
|
|
{ /* (for strange reasons this isn't nesessarily the case) */
|
|
/* find the next frame boundary */
|
|
while (search < (end+len)) /* search the remaining data */
|
|
{
|
|
if (*search++ != 0xFF) /* quick search for frame sync byte */
|
|
continue; /* (this does the majority of the job) */
|
|
|
|
/* look at the (bitswapped) rest of header candidate */
|
|
if (search[0] == curr_hd[0] /* do the quicker checks first */
|
|
&& search[2] == curr_hd[2]
|
|
&& (search[1] & 0x30) == (curr_hd[1] & 0x30)) /* sample rate */
|
|
{
|
|
search--; /* back to the sync byte */
|
|
break; /* From looking at it, this is our header. */
|
|
}
|
|
}
|
|
|
|
if (search-pos)
|
|
{ /* play old data until the frame end, to keep the MAS in sync */
|
|
sent = search-pos;
|
|
|
|
queue_write = (queue_read + 1) & QUEUE_MASK; /* will be empty after next callback */
|
|
queue[queue_read].length = sent; /* current one ends after this */
|
|
|
|
#if CONFIG_CPU == SH7034
|
|
DTCR3 = sent; /* let the DMA finish this frame */
|
|
CHCR3 |= 0x0001; /* re-enable DMA */
|
|
#endif /* CONFIG_CPU == SH7034 */
|
|
thumbnail_buf_used = 0;
|
|
return;
|
|
}
|
|
}
|
|
#endif /* CONFIG_CODEC != SWCODEC */
|
|
|
|
/* Either SWCODEC, or MAS had nothing to do (was frame boundary or not our clip) */
|
|
mp3_play_stop();
|
|
talk_queue_lock();
|
|
queue_write = queue_read = 0; /* reset the queue */
|
|
thumbnail_buf_used = 0;
|
|
talk_queue_unlock();
|
|
need_shutup = false;
|
|
}
|
|
|
|
/* Shutup the voice, except if force_enqueue_next is set. */
|
|
void talk_shutup(void)
|
|
{
|
|
if (need_shutup && !force_enqueue_next)
|
|
talk_force_shutup();
|
|
}
|
|
|
|
/* schedule a clip, at the end or discard the existing queue */
|
|
static void queue_clip(struct queue_entry *clip, bool enqueue)
|
|
{
|
|
struct queue_entry *qe;
|
|
int queue_level;
|
|
|
|
if (!enqueue)
|
|
talk_shutup(); /* cut off all the pending stuff */
|
|
/* Something is being enqueued, force_enqueue_next override is no
|
|
longer in effect. */
|
|
force_enqueue_next = false;
|
|
|
|
if (!clip->length)
|
|
return; /* safety check */
|
|
#if CONFIG_CPU == SH7034
|
|
/* disable the DMA temporarily, to be safe of race condition */
|
|
CHCR3 &= ~0x0001;
|
|
#endif
|
|
talk_queue_lock();
|
|
queue_level = QUEUE_LEVEL; /* check old level */
|
|
qe = &queue[queue_write];
|
|
|
|
if (queue_level < QUEUE_SIZE - 1) /* space left? */
|
|
{
|
|
queue[queue_write] = *clip;
|
|
queue_write = (queue_write + 1) & QUEUE_MASK;
|
|
}
|
|
talk_queue_unlock();
|
|
|
|
if (queue_level == 0)
|
|
{ /* queue was empty, we have to do the initial start */
|
|
size_t size;
|
|
void *buf = commit_transfer(qe, &size);
|
|
last_clip = qe;
|
|
mp3_play_data(buf, size, mp3_callback);
|
|
curr_hd[0] = commit_buffer[1];
|
|
curr_hd[1] = commit_buffer[2];
|
|
curr_hd[2] = commit_buffer[3];
|
|
mp3_play_pause(true); /* kickoff audio */
|
|
}
|
|
else
|
|
{
|
|
#if CONFIG_CPU == SH7034
|
|
CHCR3 |= 0x0001; /* re-enable DMA */
|
|
#endif
|
|
}
|
|
|
|
need_shutup = true;
|
|
|
|
return;
|
|
}
|
|
|
|
#if CONFIG_CODEC == SWCODEC
|
|
/* return if a voice codec is required or not */
|
|
static bool talk_voice_required(void)
|
|
{
|
|
return (has_voicefile) /* Voice file is available */
|
|
|| (global_settings.talk_dir_clip) /* Thumbnail clips are required */
|
|
|| (global_settings.talk_file_clip);
|
|
}
|
|
#endif
|
|
|
|
/***************** Public implementation *****************/
|
|
|
|
void talk_init(void)
|
|
{
|
|
int filehandle;
|
|
talk_temp_disable_count = 0;
|
|
if (talk_initialized && !strcasecmp(last_lang, global_settings.lang_file))
|
|
{
|
|
/* not a new file, nothing to do */
|
|
return;
|
|
}
|
|
|
|
if(!talk_initialized)
|
|
{
|
|
#if CONFIG_CODEC == SWCODEC
|
|
mutex_init(&queue_mutex);
|
|
#endif /* CONFIG_CODEC == SWCODEC */
|
|
mutex_init(&read_buffer_mutex);
|
|
}
|
|
talk_initialized = true;
|
|
strlcpy((char *)last_lang, (char *)global_settings.lang_file,
|
|
MAX_FILENAME);
|
|
|
|
/* reset some states */
|
|
queue_write = queue_read = 0; /* reset the queue */
|
|
memset(&voicefile, 0, sizeof(voicefile));
|
|
|
|
silence.offset = -1; /* pause clip not accessible */
|
|
voicefile_size = has_voicefile = 0;
|
|
/* need to free these as their size depends on the voice file, and
|
|
* this function is called when the talk voice file changes */
|
|
if (index_handle > 0) index_handle = core_free(index_handle);
|
|
if (talk_handle > 0) talk_handle = core_free(talk_handle);
|
|
/* don't free thumb handle, it doesn't depend on the actual voice file
|
|
* and so we can re-use it if it's already allocated in any event */
|
|
|
|
filehandle = open_voicefile();
|
|
if (filehandle < 0)
|
|
return;
|
|
|
|
if (!load_voicefile_index(filehandle))
|
|
goto out;
|
|
|
|
/* Now determine the maximum buffer size needed for the voicefile.
|
|
* The below pretends the entire voicefile would be loaded. The buffer
|
|
* size is eventually capped later on in load_voicefile_data() */
|
|
int num_clips = voicefile.id1_max + voicefile.id2_max;
|
|
int non_empty = num_clips;
|
|
int total_size = 0, avg_size;
|
|
struct clip_entry *clips = core_get_data(index_handle);
|
|
/* check for the average clip size to estimate the maximum number of
|
|
* clips the buffer can hold */
|
|
for (int i = 0; i<num_clips; i++)
|
|
{
|
|
if (clips[i].size)
|
|
total_size += ALIGN_UP(clips[i].size, sizeof(void *));
|
|
else
|
|
non_empty -= 1;
|
|
}
|
|
avg_size = total_size / non_empty;
|
|
max_clips = MIN((int)(MAX_CLIP_BUFFER_SIZE/avg_size) + 1, non_empty);
|
|
voicefile_size = total_size;
|
|
/* additionally to the clips we need a table to record the age of the clips
|
|
* so that, when memory is tight, only the most recently used ones are kept */
|
|
voicefile_size += sizeof(struct clip_cache_metadata) * max_clips;
|
|
/* compensate a bit for buflib alloc overhead. */
|
|
voicefile_size += BUFLIB_ALLOC_OVERHEAD * max_clips + 64;
|
|
has_voicefile = true;
|
|
|
|
load_voicefile_data(filehandle);
|
|
|
|
#if CONFIG_CODEC == SWCODEC
|
|
/* Initialize the actual voice clip playback engine as well */
|
|
if (talk_voice_required())
|
|
voice_thread_init();
|
|
#endif
|
|
|
|
out:
|
|
close(filehandle); /* close again, this was just to detect presence */
|
|
filehandle = -1;
|
|
}
|
|
|
|
/* somebody else claims the mp3 buffer, e.g. for regular play/record */
|
|
void talk_buffer_set_policy(int policy)
|
|
{
|
|
switch(policy)
|
|
{
|
|
case TALK_BUFFER_DEFAULT:
|
|
case TALK_BUFFER_HOLD: give_buffer_away = false; break;
|
|
case TALK_BUFFER_LOOSE: give_buffer_away = true; break;
|
|
default: DEBUGF("Ignoring unknown policy\n"); break;
|
|
}
|
|
}
|
|
|
|
/* play a voice ID from voicefile */
|
|
int talk_id(int32_t id, bool enqueue)
|
|
{
|
|
int32_t unit;
|
|
int decimals;
|
|
struct queue_entry clip;
|
|
|
|
if (!has_voicefile)
|
|
return 0; /* no voicefile loaded, not an error -> pretent success */
|
|
if (talk_temp_disable_count > 0)
|
|
return -1; /* talking has been disabled */
|
|
if (!check_audio_status())
|
|
return -1;
|
|
|
|
if (talk_handle <= 0 || index_handle <= 0) /* reload needed? */
|
|
{
|
|
int fd = open_voicefile();
|
|
if (fd < 0
|
|
|| !load_voicefile_index(fd)
|
|
|| !load_voicefile_data(fd))
|
|
return -1;
|
|
}
|
|
|
|
if (id == -1) /* -1 is an indication for silence */
|
|
return -1;
|
|
|
|
decimals = (((uint32_t)id) >> DECIMAL_SHIFT) & 0x7;
|
|
|
|
/* check if this is a special ID, with a value */
|
|
unit = ((uint32_t)id) >> UNIT_SHIFT;
|
|
if (unit || decimals)
|
|
{ /* sign-extend the value */
|
|
id = (uint32_t)id << (32-DECIMAL_SHIFT);
|
|
id >>= (32-DECIMAL_SHIFT);
|
|
|
|
talk_value_decimal(id, unit, decimals, enqueue); /* speak it */
|
|
return 0; /* and stop, end of special case */
|
|
}
|
|
|
|
if (get_clip(id, &clip) < 0)
|
|
return -1; /* not present */
|
|
|
|
#ifdef LOGF_ENABLE
|
|
if (id > VOICEONLY_DELIMITER)
|
|
logf("\ntalk_id: Say voice clip 0x%x\n", id);
|
|
else
|
|
logf("\ntalk_id: Say '%s'\n", str(id));
|
|
#endif
|
|
|
|
queue_clip(&clip, enqueue);
|
|
|
|
return 0;
|
|
}
|
|
/* Speaks zero or more IDs (from an array). */
|
|
int talk_idarray(const long *ids, bool enqueue)
|
|
{
|
|
int r;
|
|
if(!ids)
|
|
return 0;
|
|
while(*ids != TALK_FINAL_ID)
|
|
{
|
|
if((r = talk_id(*ids++, enqueue)) <0)
|
|
return r;
|
|
enqueue = true;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
/* Make sure the current utterance is not interrupted by the next one. */
|
|
void talk_force_enqueue_next(void)
|
|
{
|
|
force_enqueue_next = true;
|
|
}
|
|
|
|
/* play a thumbnail from file */
|
|
/* Returns size of spoken thumbnail, so >0 means something is spoken,
|
|
<=0 means something went wrong. */
|
|
static int _talk_file(const char* filename,
|
|
const long *prefix_ids, bool enqueue)
|
|
{
|
|
int fd;
|
|
int size;
|
|
int thumb_used;
|
|
#if CONFIG_CODEC != SWCODEC
|
|
struct mp3entry info;
|
|
#endif
|
|
|
|
/* reload needed? */
|
|
if (talk_temp_disable_count > 0)
|
|
return -1; /* talking has been disabled */
|
|
if (!check_audio_status())
|
|
return -1;
|
|
|
|
if (!alloc_thumbnail_buf())
|
|
return -1;
|
|
|
|
#if CONFIG_CODEC != SWCODEC
|
|
if(mp3info(&info, filename)) /* use this to find real start */
|
|
{
|
|
return 0; /* failed to open, or invalid */
|
|
}
|
|
#endif
|
|
|
|
if (!enqueue)
|
|
/* shutup now to free the thumbnail buffer */
|
|
talk_shutup();
|
|
|
|
fd = open(filename, O_RDONLY);
|
|
if (fd < 0) /* failed to open */
|
|
{
|
|
return 0;
|
|
}
|
|
|
|
thumb_used = thumbnail_buf_used;
|
|
if(filesize(fd) > size_for_thumbnail -thumb_used)
|
|
{ /* Don't play truncated clips */
|
|
close(fd);
|
|
return 0;
|
|
}
|
|
|
|
#if CONFIG_CODEC != SWCODEC
|
|
lseek(fd, info.first_frame_offset, SEEK_SET); /* behind ID data */
|
|
#endif
|
|
|
|
size = read_to_handle(fd, thumb_handle, thumb_used, size_for_thumbnail - thumb_used);
|
|
close(fd);
|
|
|
|
/* ToDo: find audio, skip ID headers and trailers */
|
|
|
|
if (size > 0) /* Don't play missing clips */
|
|
{
|
|
struct queue_entry clip;
|
|
#if CONFIG_CODEC != SWCODEC && !defined(SIMULATOR)
|
|
/* bitswap doesnt yield() */
|
|
bitswap(core_get_data(thumb_handle), size);
|
|
#endif
|
|
if(prefix_ids)
|
|
/* prefix thumbnail by speaking these ids, but only now
|
|
that we know there's actually a thumbnail to be
|
|
spoken. */
|
|
talk_idarray(prefix_ids, true);
|
|
talk_queue_lock();
|
|
thumbnail_buf_used = thumb_used + size;
|
|
talk_queue_unlock();
|
|
clip = (struct queue_entry){ .offset = thumb_used, .length = size, .remaining = size, .type = THUMB_OFFSET };
|
|
queue_clip(&clip, true);
|
|
}
|
|
|
|
return size;
|
|
}
|
|
|
|
int talk_file(const char *root, const char *dir, const char *file,
|
|
const char *ext, const long *prefix_ids, bool enqueue)
|
|
/* Play a thumbnail file */
|
|
{
|
|
char buf[MAX_PATH];
|
|
/* Does root end with a slash */
|
|
char *slash = (root && root[0]
|
|
&& root[strlen(root)-1] != '/') ? "/" : "";
|
|
snprintf(buf, MAX_PATH, "%s%s%s%s%s%s",
|
|
root ? root : "", slash,
|
|
dir ? dir : "", dir ? "/" : "",
|
|
file ? file : "",
|
|
ext ? ext : "");
|
|
return _talk_file(buf, prefix_ids, enqueue);
|
|
}
|
|
|
|
static int talk_spell_basename(const char *path,
|
|
const long *prefix_ids, bool enqueue)
|
|
{
|
|
if(prefix_ids)
|
|
{
|
|
talk_idarray(prefix_ids, enqueue);
|
|
enqueue = true;
|
|
}
|
|
char buf[MAX_PATH];
|
|
/* Spell only the path component after the last slash */
|
|
strlcpy(buf, path, sizeof(buf));
|
|
if(strlen(buf) >1 && buf[strlen(buf)-1] == '/')
|
|
/* strip trailing slash */
|
|
buf[strlen(buf)-1] = '\0';
|
|
char *ptr = strrchr(buf, '/');
|
|
if(ptr && strlen(buf) >1)
|
|
++ptr;
|
|
else ptr = buf;
|
|
return talk_spell(ptr, enqueue);
|
|
}
|
|
|
|
/* Play a file's .talk thumbnail, fallback to spelling the filename, or
|
|
go straight to spelling depending on settings. */
|
|
int talk_file_or_spell(const char *dirname, const char *filename,
|
|
const long *prefix_ids, bool enqueue)
|
|
{
|
|
if (global_settings.talk_file_clip)
|
|
{ /* .talk clips enabled */
|
|
if(talk_file(dirname, NULL, filename, file_thumbnail_ext,
|
|
prefix_ids, enqueue) >0)
|
|
return 0;
|
|
}
|
|
if (global_settings.talk_file == 2)
|
|
/* Either .talk clips are disabled, or as a fallback */
|
|
return talk_spell_basename(filename, prefix_ids, enqueue);
|
|
return 0;
|
|
}
|
|
|
|
#if CONFIG_CODEC == SWCODEC
|
|
/* Play a directory's .talk thumbnail, fallback to spelling the filename, or
|
|
go straight to spelling depending on settings. */
|
|
int talk_dir_or_spell(const char* dirname,
|
|
const long *prefix_ids, bool enqueue)
|
|
{
|
|
if (global_settings.talk_dir_clip)
|
|
{ /* .talk clips enabled */
|
|
if(talk_file(dirname, NULL, dir_thumbnail_name, NULL,
|
|
prefix_ids, enqueue) >0)
|
|
return 0;
|
|
}
|
|
if (global_settings.talk_dir == 2)
|
|
/* Either .talk clips disabled or as a fallback */
|
|
return talk_spell_basename(dirname, prefix_ids, enqueue);
|
|
return 0;
|
|
}
|
|
#endif
|
|
|
|
/* say a numeric value, this word ordering works for english,
|
|
but not necessarily for other languages (e.g. german) */
|
|
int talk_number(long n, bool enqueue)
|
|
{
|
|
int level = 2; /* mille count */
|
|
long mil = 1000000000; /* highest possible "-illion" */
|
|
|
|
if (talk_temp_disable_count > 0)
|
|
return -1; /* talking has been disabled */
|
|
if (!check_audio_status())
|
|
return -1;
|
|
|
|
if (!enqueue)
|
|
talk_shutup(); /* cut off all the pending stuff */
|
|
|
|
if (n==0)
|
|
{ /* special case */
|
|
talk_id(VOICE_ZERO, true);
|
|
return 0;
|
|
}
|
|
|
|
if (n<0)
|
|
{
|
|
talk_id(VOICE_MINUS, true);
|
|
n = -n;
|
|
}
|
|
|
|
while (n)
|
|
{
|
|
int segment = n / mil; /* extract in groups of 3 digits */
|
|
n -= segment * mil; /* remove the used digits from number */
|
|
mil /= 1000; /* digit place for next round */
|
|
|
|
if (segment)
|
|
{
|
|
int hundreds = segment / 100;
|
|
int ones = segment % 100;
|
|
|
|
if (hundreds)
|
|
{
|
|
talk_id(VOICE_ZERO + hundreds, true);
|
|
talk_id(VOICE_HUNDRED, true);
|
|
}
|
|
|
|
/* combination indexing */
|
|
if (ones > 20)
|
|
{
|
|
int tens = ones/10 + 18;
|
|
talk_id(VOICE_ZERO + tens, true);
|
|
ones %= 10;
|
|
}
|
|
|
|
/* direct indexing */
|
|
if (ones)
|
|
talk_id(VOICE_ZERO + ones, true);
|
|
|
|
/* add billion, million, thousand */
|
|
if (mil)
|
|
talk_id(VOICE_THOUSAND + level, true);
|
|
}
|
|
level--;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
/* Say time duration/interval. Input is time in seconds,
|
|
say hours,minutes,seconds. */
|
|
static int talk_time_unit(long secs, bool enqueue)
|
|
{
|
|
int hours, mins;
|
|
if (!enqueue)
|
|
talk_shutup();
|
|
if((hours = secs/3600)) {
|
|
secs %= 3600;
|
|
talk_value(hours, UNIT_HOUR, true);
|
|
}
|
|
if((mins = secs/60)) {
|
|
secs %= 60;
|
|
talk_value(mins, UNIT_MIN, true);
|
|
}
|
|
if((secs) || (!hours && !mins))
|
|
talk_value(secs, UNIT_SEC, true);
|
|
else if(!hours && secs)
|
|
talk_number(secs, true);
|
|
return 0;
|
|
}
|
|
|
|
void talk_fractional(char *tbuf, int value, int unit)
|
|
{
|
|
int i;
|
|
/* strip trailing zeros from the fraction */
|
|
for (i = strlen(tbuf) - 1; (i >= 0) && (tbuf[i] == '0'); i--)
|
|
tbuf[i] = '\0';
|
|
|
|
talk_number(value, true);
|
|
if (tbuf[0] != 0)
|
|
{
|
|
talk_id(LANG_POINT, true);
|
|
talk_spell(tbuf, true);
|
|
}
|
|
talk_id(unit, true);
|
|
}
|
|
|
|
int talk_value(long n, int unit, bool enqueue)
|
|
{
|
|
return talk_value_decimal(n, unit, 0, enqueue);
|
|
}
|
|
|
|
/* singular/plural aware saying of a value */
|
|
int talk_value_decimal(long n, int unit, int decimals, bool enqueue)
|
|
{
|
|
int unit_id;
|
|
static const int unit_voiced[] =
|
|
{ /* lookup table for the voice ID of the units */
|
|
[0 ... UNIT_LAST-1] = -1, /* regular ID, int, signed */
|
|
[UNIT_MS]
|
|
= VOICE_MILLISECONDS, /* here come the "real" units */
|
|
[UNIT_SEC]
|
|
= VOICE_SECONDS,
|
|
[UNIT_MIN]
|
|
= VOICE_MINUTES,
|
|
[UNIT_HOUR]
|
|
= VOICE_HOURS,
|
|
[UNIT_KHZ]
|
|
= VOICE_KHZ,
|
|
[UNIT_DB]
|
|
= VOICE_DB,
|
|
[UNIT_PERCENT]
|
|
= VOICE_PERCENT,
|
|
[UNIT_MAH]
|
|
= VOICE_MILLIAMPHOURS,
|
|
[UNIT_PIXEL]
|
|
= VOICE_PIXEL,
|
|
[UNIT_PER_SEC]
|
|
= VOICE_PER_SEC,
|
|
[UNIT_HERTZ]
|
|
= VOICE_HERTZ,
|
|
[UNIT_MB]
|
|
= LANG_MEGABYTE,
|
|
[UNIT_KBIT]
|
|
= VOICE_KBIT_PER_SEC,
|
|
[UNIT_PM_TICK]
|
|
= VOICE_PM_UNITS_PER_TICK,
|
|
};
|
|
|
|
static const int pow10[] = { /* 10^0 - 10^7 */
|
|
1, 10, 100, 1000, 10000, 100000, 1000000, 10000000
|
|
};
|
|
|
|
char tbuf[8];
|
|
char fmt[] = "%0nd";
|
|
|
|
if (talk_temp_disable_count > 0)
|
|
return -1; /* talking has been disabled */
|
|
if (!check_audio_status())
|
|
return -1;
|
|
|
|
/* special case for time duration */
|
|
if (unit == UNIT_TIME)
|
|
return talk_time_unit(n, enqueue);
|
|
|
|
if (unit < 0 || unit >= UNIT_LAST)
|
|
unit_id = -1;
|
|
else
|
|
unit_id = unit_voiced[unit];
|
|
|
|
if ((n==1 || n==-1) /* singular? */
|
|
&& unit_id >= VOICE_SECONDS && unit_id <= VOICE_HOURS)
|
|
{
|
|
unit_id--; /* use the singular for those units which have */
|
|
}
|
|
|
|
/* special case with a "plus" before */
|
|
if (n > 0 && (unit == UNIT_SIGNED || unit == UNIT_DB))
|
|
{
|
|
talk_id(VOICE_PLUS, enqueue);
|
|
enqueue = true;
|
|
}
|
|
|
|
if (decimals)
|
|
{
|
|
/* needed for the "-0.5" corner case */
|
|
if (n < 0)
|
|
{
|
|
talk_id(VOICE_MINUS, enqueue);
|
|
n = -n;
|
|
}
|
|
|
|
fmt[2] = '0' + decimals;
|
|
|
|
snprintf(tbuf, sizeof(tbuf), fmt, n % pow10[decimals]);
|
|
talk_fractional(tbuf, n / pow10[decimals], unit_id);
|
|
|
|
return 0;
|
|
}
|
|
|
|
talk_number(n, enqueue); /* say the number */
|
|
talk_id(unit_id, true); /* say the unit, if any */
|
|
|
|
return 0;
|
|
}
|
|
|
|
/* spell a string */
|
|
int talk_spell(const char* spell, bool enqueue)
|
|
{
|
|
char c; /* currently processed char */
|
|
|
|
if (talk_temp_disable_count > 0)
|
|
return -1; /* talking has been disabled */
|
|
if (!check_audio_status())
|
|
return -1;
|
|
|
|
if (!enqueue)
|
|
talk_shutup(); /* cut off all the pending stuff */
|
|
|
|
while ((c = *spell++) != '\0')
|
|
{
|
|
/* if this grows into too many cases, I should use a table */
|
|
if (c >= 'A' && c <= 'Z')
|
|
talk_id(VOICE_CHAR_A + c - 'A', true);
|
|
else if (c >= 'a' && c <= 'z')
|
|
talk_id(VOICE_CHAR_A + c - 'a', true);
|
|
else if (c >= '0' && c <= '9')
|
|
talk_id(VOICE_ZERO + c - '0', true);
|
|
else if (c == '-')
|
|
talk_id(VOICE_MINUS, true);
|
|
else if (c == '+')
|
|
talk_id(VOICE_PLUS, true);
|
|
else if (c == '.')
|
|
talk_id(VOICE_DOT, true);
|
|
else if (c == ' ')
|
|
talk_id(VOICE_PAUSE, true);
|
|
else if (c == '/')
|
|
talk_id(VOICE_CHAR_SLASH, true);
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
void talk_disable(bool disable)
|
|
{
|
|
if (disable)
|
|
talk_temp_disable_count++;
|
|
else
|
|
talk_temp_disable_count--;
|
|
}
|
|
|
|
void talk_setting(const void *global_settings_variable)
|
|
{
|
|
const struct settings_list *setting;
|
|
if (!global_settings.talk_menu)
|
|
return;
|
|
setting = find_setting(global_settings_variable, NULL);
|
|
if (setting == NULL)
|
|
return;
|
|
if (setting->lang_id)
|
|
talk_id(setting->lang_id,false);
|
|
}
|
|
|
|
|
|
#if CONFIG_RTC
|
|
void talk_date(const struct tm *tm, bool enqueue)
|
|
{
|
|
talk_id(LANG_MONTH_JANUARY + tm->tm_mon, enqueue);
|
|
talk_number(tm->tm_mday, true);
|
|
talk_number(1900 + tm->tm_year, true);
|
|
}
|
|
|
|
void talk_time(const struct tm *tm, bool enqueue)
|
|
{
|
|
if (global_settings.timeformat == 1)
|
|
{
|
|
/* Voice the hour */
|
|
long am_pm_id = VOICE_AM;
|
|
int hour = tm->tm_hour;
|
|
if (hour >= 12)
|
|
{
|
|
am_pm_id = VOICE_PM;
|
|
hour -= 12;
|
|
}
|
|
if (hour == 0)
|
|
hour = 12;
|
|
talk_number(hour, enqueue);
|
|
|
|
/* Voice the minutes */
|
|
if (tm->tm_min == 0)
|
|
{
|
|
/* Say o'clock if the minute is 0. */
|
|
talk_id(VOICE_OCLOCK, true);
|
|
}
|
|
else
|
|
{
|
|
/* Pronounce the leading 0 */
|
|
if(tm->tm_min < 10)
|
|
talk_id(VOICE_OH, true);
|
|
talk_number(tm->tm_min, true);
|
|
}
|
|
talk_id(am_pm_id, true);
|
|
}
|
|
else
|
|
{
|
|
/* Voice the time in 24 hour format */
|
|
talk_number(tm->tm_hour, enqueue);
|
|
if (tm->tm_min == 0)
|
|
{
|
|
talk_id(VOICE_HUNDRED, true);
|
|
talk_id(VOICE_HOUR, true);
|
|
}
|
|
else
|
|
{
|
|
/* Pronounce the leading 0 */
|
|
if(tm->tm_min < 10)
|
|
talk_id(VOICE_OH, true);
|
|
talk_number(tm->tm_min, true);
|
|
}
|
|
}
|
|
}
|
|
|
|
#endif /* CONFIG_RTC */
|
|
|
|
bool talk_get_debug_data(struct talk_debug_data *data)
|
|
{
|
|
char* p_lang = DEFAULT_VOICE_LANG; /* default */
|
|
|
|
memset(data, 0, sizeof(*data));
|
|
|
|
if (!has_voicefile || index_handle <= 0)
|
|
return false;
|
|
|
|
if (global_settings.lang_file[0] && global_settings.lang_file[0] != 0xff)
|
|
p_lang = (char *)global_settings.lang_file;
|
|
|
|
struct clip_entry *clips = core_get_data(index_handle);
|
|
int cached = 0;
|
|
int real_clips = 0;
|
|
|
|
strlcpy(data->voicefile, p_lang, sizeof(data->voicefile));
|
|
data->num_clips = voicefile.id1_max + voicefile.id2_max;
|
|
data->avg_clipsize = data->max_clipsize = 0;
|
|
data->min_clipsize = INT_MAX;
|
|
for(int i = 0; i < data->num_clips; i++)
|
|
{
|
|
int size = clips[i].size & (~LOADED_MASK);
|
|
if (!size) continue;
|
|
real_clips += 1;
|
|
if (size < data->min_clipsize)
|
|
data->min_clipsize = size;
|
|
if (size > data->max_clipsize)
|
|
data->max_clipsize = size;
|
|
data->avg_clipsize += size;
|
|
if (clips[i].size & LOADED_MASK)
|
|
cached++;
|
|
}
|
|
data->avg_clipsize /= real_clips;
|
|
data->num_empty_clips = data->num_clips - real_clips;
|
|
data->memory_allocated = sizeof(commit_buffer) + sizeof(voicefile)
|
|
+ data->num_clips * sizeof(struct clip_entry)
|
|
+ voicebuf_size + size_for_thumbnail;
|
|
data->memory_used = data->memory_allocated - size_for_thumbnail + thumbnail_buf_used;
|
|
if (talk_handle > 0)
|
|
data->memory_used -= buflib_available(&clip_ctx);
|
|
data->cached_clips = cached;
|
|
data->cache_hits = cache_hits;
|
|
data->cache_misses = cache_misses;
|
|
|
|
return true;
|
|
}
|