rockbox/apps/voice_thread.c

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/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2007 Michael Sevakis
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
* as published by the Free Software Foundation; either version 2
* of the License, or (at your option) any later version.
*
* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
* KIND, either express or implied.
*
****************************************************************************/
#include "config.h"
#include "system.h"
#include "kernel.h"
#include "core_alloc.h"
#include "thread.h"
#include "appevents.h"
#include "voice_thread.h"
#include "talk.h"
#include "dsp_core.h"
#include "pcm.h"
#include "pcm_mixer.h"
#include "codecs/libspeex/speex/speex.h"
/* Default number of PCM frames to queue - adjust as necessary per-target */
#define VOICE_FRAMES 4
/* Define any of these as "1" and uncomment the LOGF_ENABLE line to log
regular and/or timeout messages */
#define VOICE_LOGQUEUES 0
#define VOICE_LOGQUEUES_SYS_TIMEOUT 0
/*#define LOGF_ENABLE*/
#include "logf.h"
#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
/* Minimum priority needs to be a bit elevated since voice has fairly low
latency */
#define PRIORITY_VOICE (PRIORITY_PLAYBACK-4)
/* A speex frame generally consists of 20ms of audio
* (http://www.speex.org/docs/manual/speex-manual/node10.html)
* for wideband mode this results in 320 samples of decoded PCM.
*/
#define VOICE_FRAME_COUNT 320 /* Samples / frame */
#define VOICE_SAMPLE_RATE 16000 /* Sample rate in HZ */
#define VOICE_SAMPLE_DEPTH 16 /* Sample depth in bits */
/* The max. wideband bitrate is 42.4 kbps
* (http://www.speex.org/docs/manual/speex-manual/node11.html). For 20ms
* this gives a maximum of 106 bytes for an encoded speex frame */
#define VOICE_MAX_ENCODED_FRAME_SIZE 106
/* Voice thread variables */
static unsigned int voice_thread_id = 0;
#if defined(CPU_MIPS)
/* MIPS is stack-hungry */
#define VOICE_STACK_EXTRA 0x500
#elif defined(CPU_COLDFIRE)
/* ISR uses any available stack - need a bit more room */
#define VOICE_STACK_EXTRA 0x400
#else
#define VOICE_STACK_EXTRA 0x3c0
#endif
static long voice_stack[(DEFAULT_STACK_SIZE + VOICE_STACK_EXTRA)/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 queue_sender_list voice_queue_sender_list SHAREDBSS_ATTR;
static int quiet_counter SHAREDDATA_ATTR = 0;
static bool voice_playing = false;
#define VOICE_PCM_FRAME_COUNT ((PLAY_SAMPR_MAX*VOICE_FRAME_COUNT + \
VOICE_SAMPLE_RATE) / VOICE_SAMPLE_RATE)
#define VOICE_PCM_FRAME_SIZE (VOICE_PCM_FRAME_COUNT*2*sizeof (int16_t))
/* Voice processing states */
enum voice_state
{
VOICE_STATE_MESSAGE = 0,
VOICE_STATE_DECODE,
VOICE_STATE_BUFFER_INSERT,
VOICE_STATE_QUIT,
};
/* A delay to not bring audio back to normal level too soon */
#define QUIET_COUNT 3
enum voice_thread_messages
{
Q_VOICE_PLAY = 0, /* Play a clip */
Q_VOICE_STOP, /* Stop current clip */
Q_VOICE_KILL, /* Kill voice thread till restart*/
};
/* Structure to store clip data callback info */
struct voice_info
{
/* Callback to get more clips */
voice_play_callback_t get_more;
/* Start of clip */
const void *start;
/* Size of clip */
size_t size;
};
/* Private thread data for its current state that must be passed to its
* internal functions */
struct voice_thread_data
{
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 */
int lookahead; /* Number of samples to drop at start of clip */
struct dsp_buffer src; /* Speex output buffer/input to DSP */
struct dsp_buffer *dst; /* Pointer to DSP output buffer for PCM */
};
/* Functions called in their repective state that return the next state to
state machine loop - compiler may inline them at its discretion */
static enum voice_state voice_message(struct voice_thread_data *td);
static enum voice_state voice_decode(struct voice_thread_data *td);
static enum voice_state voice_buffer_insert(struct voice_thread_data *td);
/* Might have lookahead and be skipping samples, so size is needed */
static struct voice_buf
{
/* Buffer for decoded samples */
spx_int16_t spx_outbuf[VOICE_FRAME_COUNT];
/* Queue frame indexes */
unsigned int volatile frame_in;
unsigned int volatile frame_out;
/* For PCM pointer adjustment */
struct voice_thread_data *td;
/* Buffers for mixing voice */
struct voice_pcm_frame
{
size_t size;
int16_t pcm[2*VOICE_PCM_FRAME_COUNT];
} frames[VOICE_FRAMES];
} *voice_buf = NULL;
static int voice_buf_hid = 0;
static int move_callback(int handle, void *current, void *new)
{
/* Have to adjust the pointers that point into things in voice_buf */
off_t diff = new - current;
struct voice_thread_data *td = voice_buf->td;
if (td != NULL)
{
td->src.p32[0] = SKIPBYTES(td->src.p32[0], diff);
td->src.p32[1] = SKIPBYTES(td->src.p32[1], diff);
if (td->dst != NULL) /* Only when calling dsp_process */
td->dst->p16out = SKIPBYTES(td->dst->p16out, diff);
mixer_adjust_channel_address(PCM_MIXER_CHAN_VOICE, diff);
}
voice_buf = new;
return BUFLIB_CB_OK;
(void)handle;
};
static void sync_callback(int handle, bool sync_on)
{
/* A move must not allow PCM to access the channel */
if (sync_on)
pcm_play_lock();
else
pcm_play_unlock();
(void)handle;
}
static struct buflib_callbacks ops =
{
.move_callback = move_callback,
.sync_callback = sync_callback,
};
/* Number of frames in queue */
static unsigned int voice_unplayed_frames(void)
{
return voice_buf->frame_in - voice_buf->frame_out;
}
/* Mixer channel callback */
static void voice_pcm_callback(const void **start, size_t *size)
{
unsigned int frame_out = ++voice_buf->frame_out;
if (voice_unplayed_frames() == 0)
return; /* Done! */
struct voice_pcm_frame *frame =
&voice_buf->frames[frame_out % VOICE_FRAMES];
*start = frame->pcm;
*size = frame->size;
}
/* Start playback of voice channel if not already playing */
static void voice_start_playback(void)
{
if (mixer_channel_status(PCM_MIXER_CHAN_VOICE) != CHANNEL_STOPPED ||
voice_unplayed_frames() == 0)
return;
struct voice_pcm_frame *frame =
&voice_buf->frames[voice_buf->frame_out % VOICE_FRAMES];
mixer_channel_play_data(PCM_MIXER_CHAN_VOICE, voice_pcm_callback,
frame->pcm, frame->size);
}
/* Stop the voice channel */
static void voice_stop_playback(void)
{
mixer_channel_stop(PCM_MIXER_CHAN_VOICE);
voice_buf->frame_in = voice_buf->frame_out = 0;
}
/* Grab a free PCM frame */
static int16_t * voice_buf_get(void)
{
if (voice_unplayed_frames() >= VOICE_FRAMES)
{
/* Full */
voice_start_playback();
return NULL;
}
return voice_buf->frames[voice_buf->frame_in % VOICE_FRAMES].pcm;
}
/* Commit a frame returned by voice_buf_get and set the actual size */
static void voice_buf_commit(int count)
{
if (count > 0)
{
unsigned int frame_in = voice_buf->frame_in;
voice_buf->frames[frame_in % VOICE_FRAMES].size =
count * 2 * sizeof (int16_t);
voice_buf->frame_in = frame_in + 1;
}
}
/* Stop any current clip and start playing a new one */
void voice_play_data(const void *start, size_t size,
voice_play_callback_t get_more)
{
if (voice_thread_id && start && size && get_more)
{
struct voice_info voice_clip =
{
.get_more = get_more,
.start = start,
.size = size,
};
LOGFQUEUE("mp3 >| voice Q_VOICE_PLAY");
queue_send(&voice_queue, Q_VOICE_PLAY, (intptr_t)&voice_clip);
}
}
/* Stop current voice clip from playing */
void voice_play_stop(void)
{
if (voice_thread_id != 0)
{
LOGFQUEUE("mp3 >| voice Q_VOICE_STOP");
queue_send(&voice_queue, Q_VOICE_STOP, 0);
}
}
/* This function is meant to be used by the buffer request functions to
ensure the codec is no longer active */
void voice_stop(void)
{
/* Unqueue all future clips */
talk_force_shutup();
}
/* 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. */
while (voice_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->dsp = dsp_get_config(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);
mixer_channel_set_amplitude(PCM_MIXER_CHAN_VOICE, MIX_AMP_UNITY);
voice_buf->td = td;
td->dst = NULL;
}
/* Voice thread message processing */
static enum voice_state voice_message(struct voice_thread_data *td)
{
queue_wait_w_tmo(&voice_queue, &td->ev,
quiet_counter > 0 ? HZ/10 : TIMEOUT_BLOCK);
switch (td->ev.id)
{
case Q_VOICE_PLAY:
LOGFQUEUE("voice < Q_VOICE_PLAY");
if (quiet_counter == 0)
{
/* Boost CPU now */
trigger_cpu_boost();
}
else
{
/* Stop any clip still playing */
voice_stop_playback();
dsp_configure(td->dsp, DSP_FLUSH, 0);
}
if (quiet_counter <= 0)
{
voice_playing = true;
dsp_configure(td->dsp, DSP_SET_OUT_FREQUENCY, mixer_get_frequency());
send_event(VOICE_EVENT_IS_PLAYING, &voice_playing);
}
quiet_counter = QUIET_COUNT;
/* Copy the clip info */
td->vi = *(struct voice_info *)td->ev.data;
/* We need nothing more from the sending thread - let it run */
queue_reply(&voice_queue, 1);
/* 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, (void *)td->vi.start,
td->vi.size);
speex_decoder_ctl(td->st, SPEEX_GET_LOOKAHEAD, &td->lookahead);
return VOICE_STATE_DECODE;
case Q_VOICE_KILL:
queue_delete(&voice_queue);
return VOICE_STATE_QUIT;
case SYS_TIMEOUT:
if (voice_unplayed_frames())
{
/* Waiting for PCM to finish */
break;
}
/* Drop through and stop the first time after clip runs out */
if (quiet_counter-- != QUIET_COUNT)
{
if (quiet_counter <= 0)
{
voice_playing = false;
send_event(VOICE_EVENT_IS_PLAYING, &voice_playing);
}
break;
}
/* Fall-through */
case Q_VOICE_STOP:
LOGFQUEUE("voice < Q_VOICE_STOP");
cancel_cpu_boost();
voice_stop_playback();
break;
/* No default: no other message ids are sent */
}
return VOICE_STATE_MESSAGE;
}
/* Decode frames or stop if all have completed */
static enum voice_state voice_decode(struct voice_thread_data *td)
{
if (!queue_empty(&voice_queue))
return VOICE_STATE_MESSAGE;
/* Decode the data */
if (speex_decode_int(td->st, &td->bits, voice_buf->spx_outbuf) < 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 && td->vi.size > 0)
{
/* Make bit buffer use our own buffer */
speex_bits_set_bit_buffer(&td->bits, (void *)td->vi.start,
td->vi.size);
/* Don't skip any samples when we're stringing clips together */
td->lookahead = 0;
}
else
{
/* If all clips are done and not playing, force pcm playback. */
if (voice_unplayed_frames() > 0)
voice_start_playback();
return VOICE_STATE_MESSAGE;
}
}
else
{
if (td->vi.size > VOICE_MAX_ENCODED_FRAME_SIZE
&& td->bits.charPtr > (int)(td->vi.size - VOICE_MAX_ENCODED_FRAME_SIZE)
&& td->vi.get_more != NULL)
{
/* request more data _before_ running out of data (requesting
* more after the fact prevents speex from successful decoding)
* place a hint telling the callback how much of the
* previous buffer we have consumed such that it can rewind
* as necessary */
int bitPtr = td->bits.bitPtr;
td->vi.size = td->bits.charPtr;
td->vi.get_more(&td->vi.start, &td->vi.size);
speex_bits_set_bit_buffer(&td->bits, (void *)td->vi.start,
td->vi.size);
td->bits.bitPtr = bitPtr;
}
yield();
/* Output the decoded frame */
td->src.remcount = VOICE_FRAME_COUNT - td->lookahead;
td->src.pin[0] = &voice_buf->spx_outbuf[td->lookahead];
td->src.pin[1] = NULL;
td->src.proc_mask = 0;
td->lookahead -= MIN(VOICE_FRAME_COUNT, td->lookahead);
if (td->src.remcount > 0)
return VOICE_STATE_BUFFER_INSERT;
}
return VOICE_STATE_DECODE;
}
/* Process the PCM samples in the DSP and send out for mixing */
static enum voice_state voice_buffer_insert(struct voice_thread_data *td)
{
if (!queue_empty(&voice_queue))
return VOICE_STATE_MESSAGE;
struct dsp_buffer dst;
if ((dst.p16out = voice_buf_get()) != NULL)
{
dst.remcount = 0;
dst.bufcount = VOICE_PCM_FRAME_COUNT;
td->dst = &dst;
dsp_process(td->dsp, &td->src, &dst);
td->dst = NULL;
voice_buf_commit(dst.remcount);
/* Unless other effects are introduced to voice that have delays,
all output should have been purged to dst in one call */
return td->src.remcount > 0 ?
VOICE_STATE_BUFFER_INSERT : VOICE_STATE_DECODE;
}
sleep(0);
return VOICE_STATE_BUFFER_INSERT;
}
/* Voice thread entrypoint */
static void voice_thread(void)
{
struct voice_thread_data td;
enum voice_state state = VOICE_STATE_MESSAGE;
voice_data_init(&td);
while (1)
{
switch (state)
{
case VOICE_STATE_MESSAGE:
state = voice_message(&td);
break;
case VOICE_STATE_DECODE:
state = voice_decode(&td);
break;
case VOICE_STATE_BUFFER_INSERT:
state = voice_buffer_insert(&td);
break;
case VOICE_STATE_QUIT:
logf("Exiting voice thread");
core_free(voice_buf_hid);
voice_buf_hid = 0;
return;
}
}
return;
}
/* kill voice thread and dont allow re-init*/
void voice_thread_kill(void)
{
queue_send(&voice_queue, Q_VOICE_KILL, 0);
}
/* Initialize buffers, all synchronization objects and create the thread */
void voice_thread_init(void)
{
if (voice_thread_id != 0)
return; /* Already did an init and succeeded at it */
voice_buf_hid = core_alloc_ex("voice buf", sizeof (*voice_buf), &ops);
if (voice_buf_hid <= 0)
{
logf("voice: core_alloc_ex failed");
return;
}
voice_buf = core_get_data(voice_buf_hid);
if (voice_buf == NULL)
{
logf("voice: core_get_data failed");
core_free(voice_buf_hid);
voice_buf_hid = 0;
return;
}
memset(voice_buf, 0, sizeof (*voice_buf));
logf("Starting voice thread");
queue_init(&voice_queue, false);
voice_thread_id = create_thread(voice_thread, voice_stack,
sizeof(voice_stack), 0, voice_thread_name
IF_PRIO(, PRIORITY_VOICE) IF_COP(, CPU));
queue_enable_queue_send(&voice_queue, &voice_queue_sender_list,
voice_thread_id);
}
#ifdef HAVE_PRIORITY_SCHEDULING
/* Set the voice thread priority */
void voice_thread_set_priority(int priority)
{
if (voice_thread_id == 0)
return;
if (priority > PRIORITY_VOICE)
priority = PRIORITY_VOICE;
thread_set_priority(voice_thread_id, priority);
}
#endif