rockbox/apps/voice_thread.c

451 lines
14 KiB
C

/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2007 Michael Sevakis
*
* 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.
*
****************************************************************************/
#include "system.h"
#include "thread.h"
#include "logf.h"
#include "voice_thread.h"
#include "talk.h"
#include "dsp.h"
#include "audio.h"
#include "playback.h"
#include "pcmbuf.h"
#include "codecs/libspeex/speex/speex.h"
/* Define any of these as "1" to log regular and/or timeout messages */
#define VOICE_LOGQUEUES 0
#define VOICE_LOGQUEUES_SYS_TIMEOUT 0
#if VOICE_LOGQUEUES
#define LOGFQUEUE logf
#else
#define LOGFQUEUE(...)
#endif
#if VOICE_LOGQUEUES_SYS_TIMEOUT
#define LOGFQUEUE_SYS_TIMEOUT logf
#else
#define LOGFQUEUE_SYS_TIMEOUT(...)
#endif
#ifndef IBSS_ATTR_VOICE_STACK
#define IBSS_ATTR_VOICE_STACK IBSS_ATTR
#endif
#define VOICE_FRAME_SIZE 320 /* Samples / frame */
#define VOICE_SAMPLE_RATE 16000 /* Sample rate in HZ */
#define VOICE_SAMPLE_DEPTH 16 /* Sample depth in bits */
/* Voice thread variables */
static struct thread_entry *voice_thread_p = NULL;
static long voice_stack[0x7c0/sizeof(long)] IBSS_ATTR_VOICE_STACK;
static const char voice_thread_name[] = "voice";
/* Voice thread synchronization objects */
static struct event_queue voice_queue SHAREDBSS_ATTR;
static struct mutex voice_mutex SHAREDBSS_ATTR;
static struct event voice_event SHAREDBSS_ATTR;
static struct queue_sender_list voice_queue_sender_list SHAREDBSS_ATTR;
/* Buffer for decoded samples */
static spx_int16_t voice_output_buf[VOICE_FRAME_SIZE] CACHEALIGN_ATTR;
enum voice_thread_states
{
TSTATE_STOPPED = 0, /* Voice thread is stopped and awaiting commands */
TSTATE_DECODE, /* Voice is decoding a clip */
TSTATE_BUFFER_INSERT, /* Voice is sending decoded audio to PCM */
};
enum voice_thread_messages
{
Q_VOICE_NULL = 0, /* A message for thread sync - no effect on state */
Q_VOICE_PLAY, /* Play a clip */
Q_VOICE_STOP, /* Stop current clip */
Q_VOICE_STATE, /* Query playing state */
};
/* Structure to store clip data callback info */
struct voice_info
{
pcm_more_callback_type get_more; /* Callback to get more clips */
unsigned char *start; /* Start of clip */
size_t size; /* Size of clip */
};
/* Private thread data for its current state that must be passed to its
* internal functions */
struct voice_thread_data
{
int state; /* Thread state (TSTATE_*) */
struct queue_event ev; /* Last queue event pulled from queue */
void *st; /* Decoder instance */
SpeexBits bits; /* Bit cursor */
struct dsp_config *dsp; /* DSP used for voice output */
struct voice_info vi; /* Copy of clip data */
const char *src[2]; /* Current output buffer pointers */
int lookahead; /* Number of samples to drop at start of clip */
int count; /* Count of samples remaining to send to PCM */
};
/* Audio playback is in a playing state? */
static inline bool playback_is_playing(void)
{
return (audio_status() & AUDIO_STATUS_PLAY) != 0;
}
/* Stop any current clip and start playing a new one */
void mp3_play_data(const unsigned char* start, int size,
pcm_more_callback_type get_more)
{
/* Shared struct to get data to the thread - once it replies, it has
* safely cached it in its own private data */
static struct voice_info voice_clip SHAREDBSS_ATTR;
if (get_more != NULL && start != NULL && (ssize_t)size > 0)
{
mutex_lock(&voice_mutex);
voice_clip.get_more = get_more;
voice_clip.start = (unsigned char *)start;
voice_clip.size = size;
LOGFQUEUE("mp3 >| voice Q_VOICE_PLAY");
queue_send(&voice_queue, Q_VOICE_PLAY, (intptr_t)&voice_clip);
mutex_unlock(&voice_mutex);
}
}
/* Stop current voice clip from playing */
void mp3_play_stop(void)
{
mutex_lock(&voice_mutex); /* Sync against voice_stop */
LOGFQUEUE("mp3 > voice Q_VOICE_STOP: 1");
queue_remove_from_head(&voice_queue, Q_VOICE_STOP);
queue_post(&voice_queue, Q_VOICE_STOP, 1);
mutex_unlock(&voice_mutex);
}
void mp3_play_pause(bool play)
{
/* a dummy */
(void)play;
}
/* Tell is voice is still in a playing state */
bool mp3_is_playing(void)
{
/* TODO: Implement a timeout or state query function for event objects */
LOGFQUEUE("mp3 >| voice Q_VOICE_STATE");
int state = queue_send(&voice_queue, Q_VOICE_STATE, 0);
return state != TSTATE_STOPPED;
}
/* This function is meant to be used by the buffer request functions to
ensure the codec is no longer active */
void voice_stop(void)
{
mutex_lock(&voice_mutex);
/* Stop the output and current clip */
LOGFQUEUE("mp3 >| voice Q_VOICE_STOP: 1");
queue_send(&voice_queue, Q_VOICE_STOP, 1);
/* Careful if using sync objects in talk.c - make sure locking order is
* observed with one or the other always granted first */
/* Unqueue all future clips */
talk_force_shutup();
/* Wait for any final queue_post to be processed */
LOGFQUEUE("mp3 >| voice Q_VOICE_NULL");
queue_send(&voice_queue, Q_VOICE_NULL, 0);
mutex_unlock(&voice_mutex);
} /* voice_stop */
/* Wait for voice to finish speaking. */
void voice_wait(void)
{
/* NOTE: One problem here is that we can't tell if another thread started a
* new clip by the time we wait. This should be resolvable if conditions
* ever require knowing the very clip you requested has finished. */
event_wait(&voice_event, STATE_SIGNALED);
/* Wait for PCM buffer to be exhausted. Works only if not playing. */
while(!playback_is_playing() && pcm_is_playing())
sleep(1);
}
/* Initialize voice thread data that must be valid upon starting and the
* setup the DSP parameters */
static void voice_data_init(struct voice_thread_data *td)
{
td->state = TSTATE_STOPPED;
td->dsp = (struct dsp_config *)dsp_configure(NULL, DSP_MYDSP,
CODEC_IDX_VOICE);
dsp_configure(td->dsp, DSP_RESET, 0);
dsp_configure(td->dsp, DSP_SET_FREQUENCY, VOICE_SAMPLE_RATE);
dsp_configure(td->dsp, DSP_SET_SAMPLE_DEPTH, VOICE_SAMPLE_DEPTH);
dsp_configure(td->dsp, DSP_SET_STEREO_MODE, STEREO_MONO);
}
/* Voice thread message processing */
static void voice_message(struct voice_thread_data *td)
{
while (1)
{
switch (td->ev.id)
{
case Q_VOICE_PLAY:
LOGFQUEUE("voice < Q_VOICE_PLAY");
/* Put up a block for completion signal */
event_set_state(&voice_event, STATE_NONSIGNALED);
/* Copy the clip info */
td->vi = *(struct voice_info *)td->ev.data;
/* Be sure audio buffer is initialized */
audio_restore_playback(AUDIO_WANT_VOICE);
/* We need nothing more from the sending thread - let it run */
queue_reply(&voice_queue, 1);
if (td->state == TSTATE_STOPPED)
{
/* Boost CPU now */
trigger_cpu_boost();
}
else if (!playback_is_playing())
{
/* Just voice, stop any clip still playing */
pcmbuf_play_stop();
}
/* Clean-start the decoder */
td->st = speex_decoder_init(&speex_wb_mode);
/* Make bit buffer use our own buffer */
speex_bits_set_bit_buffer(&td->bits, td->vi.start, td->vi.size);
speex_decoder_ctl(td->st, SPEEX_GET_LOOKAHEAD, &td->lookahead);
td->state = TSTATE_DECODE;
return;
case Q_VOICE_STOP:
LOGFQUEUE("voice < Q_VOICE_STOP: %d", ev.data);
if (td->ev.data != 0 && !playback_is_playing())
{
/* If not playing, it's just voice so stop pcm playback */
pcmbuf_play_stop();
}
/* Cancel boost */
cancel_cpu_boost();
td->state = TSTATE_STOPPED;
event_set_state(&voice_event, STATE_SIGNALED);
break;
case Q_VOICE_STATE:
LOGFQUEUE("voice < Q_VOICE_STATE");
queue_reply(&voice_queue, td->state);
if (td->state == TSTATE_STOPPED)
break; /* Not in a playback state */
return;
default:
/* Default messages get a reply and thread continues with no
* state transition */
LOGFQUEUE("voice < default");
if (td->state == TSTATE_STOPPED)
break; /* Not in playback state */
queue_reply(&voice_queue, 0);
return;
}
queue_wait(&voice_queue, &td->ev);
}
}
/* Voice thread entrypoint */
static void voice_thread(void)
{
struct voice_thread_data td;
voice_data_init(&td);
audio_wait_for_init();
goto message_wait;
while (1)
{
td.state = TSTATE_DECODE;
if (!queue_empty(&voice_queue))
{
message_wait:
queue_wait(&voice_queue, &td.ev);
message_process:
voice_message(&td);
/* Branch to initial start point or branch back to previous
* operation if interrupted by a message */
switch (td.state)
{
case TSTATE_DECODE: goto voice_decode;
case TSTATE_BUFFER_INSERT: goto buffer_insert;
default: goto message_wait;
}
}
voice_decode:
/* Decode the data */
if (speex_decode_int(td.st, &td.bits, voice_output_buf) < 0)
{
/* End of stream or error - get next clip */
td.vi.size = 0;
if (td.vi.get_more != NULL)
td.vi.get_more(&td.vi.start, &td.vi.size);
if (td.vi.start != NULL && (ssize_t)td.vi.size > 0)
{
/* Make bit buffer use our own buffer */
speex_bits_set_bit_buffer(&td.bits, td.vi.start, td.vi.size);
/* Don't skip any samples when we're stringing clips together */
td.lookahead = 0;
/* Paranoid check - be sure never to somehow get stuck in a
* loop without listening to the queue */
yield();
if (!queue_empty(&voice_queue))
goto message_wait;
else
goto voice_decode;
}
/* If all clips are done and not playing, force pcm playback. */
if (!pcm_is_playing())
pcmbuf_play_start();
/* Synthesize a stop request */
/* NOTE: We have no way to know when the pcm data placed in the
* buffer is actually consumed and playback has reached the end
* so until the info is available or inferred somehow, this will
* not be accurate and the stopped signal will come too soon.
* ie. You may not hear the "Shutting Down" splash even though
* it waits for voice to stop. */
td.ev.id = Q_VOICE_STOP;
td.ev.data = 0; /* Let PCM drain by itself */
yield();
goto message_process;
}
yield();
/* Output the decoded frame */
td.count = VOICE_FRAME_SIZE - td.lookahead;
td.src[0] = (const char *)&voice_output_buf[td.lookahead];
td.src[1] = NULL;
td.lookahead -= MIN(VOICE_FRAME_SIZE, td.lookahead);
buffer_insert:
/* Process the PCM samples in the DSP and send out for mixing */
td.state = TSTATE_BUFFER_INSERT;
while (td.count > 0)
{
int out_count = dsp_output_count(td.dsp, td.count);
int inp_count;
char *dest;
while (1)
{
if (!queue_empty(&voice_queue))
goto message_wait;
if ((dest = pcmbuf_request_voice_buffer(&out_count)) != NULL)
break;
yield();
}
/* Get the real input_size for output_size bytes, guarding
* against resampling buffer overflows. */
inp_count = dsp_input_count(td.dsp, out_count);
if (inp_count <= 0)
break;
/* Input size has grown, no error, just don't write more than
* length */
if (inp_count > td.count)
inp_count = td.count;
out_count = dsp_process(td.dsp, dest, td.src, inp_count);
if (out_count <= 0)
break;
pcmbuf_write_voice_complete(out_count);
td.count -= inp_count;
}
yield();
} /* end while */
} /* voice_thread */
/* Initialize all synchronization objects create the thread */
void voice_thread_init(void)
{
logf("Starting voice thread");
queue_init(&voice_queue, false);
mutex_init(&voice_mutex);
event_init(&voice_event, STATE_SIGNALED | EVENT_MANUAL);
voice_thread_p = create_thread(voice_thread, voice_stack,
sizeof(voice_stack), CREATE_THREAD_FROZEN,
voice_thread_name IF_PRIO(, PRIORITY_PLAYBACK) IF_COP(, CPU));
queue_enable_queue_send(&voice_queue, &voice_queue_sender_list,
voice_thread_p);
} /* voice_thread_init */
/* Unfreeze the voice thread */
void voice_thread_resume(void)
{
logf("Thawing voice thread");
thread_thaw(voice_thread_p);
}
#ifdef HAVE_PRIORITY_SCHEDULING
/* Set the voice thread priority */
void voice_thread_set_priority(int priority)
{
thread_set_priority(voice_thread_p, priority);
}
#endif