forked from len0rd/rockbox
Make speex the new voice format for SWCODEC targets (non-Archos). Remove codec swapping and build speex voice decoding directly into the core binary.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@15668 a1c6a512-1295-4272-9138-f99709370657
This commit is contained in:
Michael Sevakis
parent
75432619e8
commit
99617d71ba
29 changed files with 753 additions and 1153 deletions
444
apps/voice_thread.c
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444
apps/voice_thread.c
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/***************************************************************************
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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) 2007 Michael Sevakis
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*
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* All files in this archive are subject to the GNU General Public License.
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* See the file COPYING in the source tree root for full license agreement.
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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 "system.h"
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#include "thread.h"
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#include "logf.h"
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#include "voice_thread.h"
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#include "talk.h"
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#include "dsp.h"
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#include "audio.h"
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#include "pcmbuf.h"
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#include "codecs/libspeex/speex/speex.h"
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/* Define any of these as "1" to log regular and/or timeout messages */
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#define VOICE_LOGQUEUES 0
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#define VOICE_LOGQUEUES_SYS_TIMEOUT 0
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#if VOICE_LOGQUEUES
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#define LOGFQUEUE logf
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#else
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#define LOGFQUEUE(...)
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#endif
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#if VOICE_LOGQUEUES_SYS_TIMEOUT
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#define LOGFQUEUE_SYS_TIMEOUT logf
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#else
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#define LOGFQUEUE_SYS_TIMEOUT(...)
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#endif
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#ifndef IBSS_ATTR_VOICE_STACK
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#define IBSS_ATTR_VOICE_STACK IBSS_ATTR
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#endif
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#define VOICE_FRAME_SIZE 320 /* Samples / frame */
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#define VOICE_SAMPLE_RATE 16000 /* Sample rate in HZ */
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#define VOICE_SAMPLE_DEPTH 16 /* Sample depth in bits */
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/* Voice thread variables */
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static struct thread_entry *voice_thread_p = NULL;
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static long voice_stack[0x740/sizeof(long)] IBSS_ATTR_VOICE_STACK;
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static const char voice_thread_name[] = "voice";
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/* Voice thread synchronization objects */
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static struct event_queue voice_queue NOCACHEBSS_ATTR;
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static struct mutex voice_mutex NOCACHEBSS_ATTR;
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static struct event voice_event NOCACHEBSS_ATTR;
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static struct queue_sender_list voice_queue_sender_list NOCACHEBSS_ATTR;
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/* Buffer for decoded samples */
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static spx_int16_t voice_output_buf[VOICE_FRAME_SIZE] CACHEALIGN_ATTR;
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enum voice_thread_states
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{
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TSTATE_STOPPED = 0, /* Voice thread is stopped and awaiting commands */
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TSTATE_DECODE, /* Voice is decoding a clip */
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TSTATE_BUFFER_INSERT, /* Voice is sending decoded audio to PCM */
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};
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enum voice_thread_messages
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{
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Q_VOICE_NULL = 0, /* A message for thread sync - no effect on state */
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Q_VOICE_PLAY, /* Play a clip */
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Q_VOICE_STOP, /* Stop current clip */
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Q_VOICE_STATE, /* Query playing state */
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};
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/* Structure to store clip data callback info */
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struct voice_info
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{
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pcm_more_callback_type get_more; /* Callback to get more clips */
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unsigned char *start; /* Start of clip */
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ssize_t size; /* Size of clip */
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};
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/* Private thread data for its current state that must be passed to its
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* internal functions */
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struct voice_thread_data
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{
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int state; /* Thread state (TSTATE_*) */
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struct queue_event ev; /* Last queue event pulled from queue */
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void *st; /* Decoder instance */
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SpeexBits bits; /* Bit cursor */
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struct dsp_config *dsp; /* DSP used for voice output */
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struct voice_info vi; /* Copy of clip data */
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const char *src[2]; /* Current output buffer pointers */
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int lookahead; /* Number of samples to drop at start of clip */
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int count; /* Count of samples remaining to send to PCM */
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};
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/* Audio playback is in a playing state? */
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static inline bool playback_is_playing(void)
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{
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return (audio_status() & AUDIO_STATUS_PLAY) != 0;
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}
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/* Stop any current clip and start playing a new one */
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void mp3_play_data(const unsigned char* start, int size,
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pcm_more_callback_type get_more)
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{
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/* Shared struct to get data to the thread - once it replies, it has
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* safely cached it in its own private data */
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static struct voice_info voice_clip NOCACHEBSS_ATTR;
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if (get_more != NULL && start != NULL && size > 0)
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{
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mutex_lock(&voice_mutex);
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voice_clip.get_more = get_more;
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voice_clip.start = (unsigned char *)start;
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voice_clip.size = size;
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LOGFQUEUE("mp3 >| voice Q_VOICE_PLAY");
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queue_send(&voice_queue, Q_VOICE_PLAY, (intptr_t)&voice_clip);
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mutex_unlock(&voice_mutex);
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}
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}
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/* Stop current voice clip from playing */
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void mp3_play_stop(void)
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{
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mutex_lock(&voice_mutex); /* Sync against voice_stop */
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LOGFQUEUE("mp3 >| voice Q_VOICE_STOP: 1");
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queue_send(&voice_queue, Q_VOICE_STOP, 1);
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mutex_unlock(&voice_mutex);
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}
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void mp3_play_pause(bool play)
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{
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/* a dummy */
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(void)play;
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}
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/* Tell is voice is still in a playing state */
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bool mp3_is_playing(void)
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{
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/* TODO: Implement a timeout or state query function for event objects */
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LOGFQUEUE("mp3 >| voice Q_VOICE_STATE");
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int state = queue_send(&voice_queue, Q_VOICE_STATE, 0);
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return state != TSTATE_STOPPED;
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}
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/* This function is meant to be used by the buffer request functions to
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ensure the codec is no longer active */
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void voice_stop(void)
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{
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mutex_lock(&voice_mutex);
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/* Stop the output and current clip */
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mp3_play_stop();
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/* Careful if using sync objects in talk.c - make sure locking order is
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* observed with one or the other always granted first */
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/* Unqueue all future clips */
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talk_force_shutup();
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mutex_unlock(&voice_mutex);
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} /* voice_stop */
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/* Wait for voice to finish speaking. */
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void voice_wait(void)
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{
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/* NOTE: One problem here is that we can't tell if another thread started a
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* new clip by the time we wait. This should be resolvable if conditions
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* ever require knowing the very clip you requested has finished. */
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event_wait(&voice_event, STATE_SIGNALED);
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}
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/* Initialize voice thread data that must be valid upon starting and the
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* setup the DSP parameters */
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static void voice_data_init(struct voice_thread_data *td)
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{
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td->state = TSTATE_STOPPED;
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td->dsp = (struct dsp_config *)dsp_configure(NULL, DSP_MYDSP,
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CODEC_IDX_VOICE);
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dsp_configure(td->dsp, DSP_RESET, 0);
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dsp_configure(td->dsp, DSP_SET_FREQUENCY, VOICE_SAMPLE_RATE);
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dsp_configure(td->dsp, DSP_SET_SAMPLE_DEPTH, VOICE_SAMPLE_DEPTH);
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dsp_configure(td->dsp, DSP_SET_STEREO_MODE, STEREO_MONO);
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}
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/* Voice thread message processing */
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static void voice_message(struct voice_thread_data *td)
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{
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while (1)
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{
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switch (td->ev.id)
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{
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case Q_VOICE_PLAY:
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LOGFQUEUE("voice < Q_VOICE_PLAY");
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/* Put up a block for completion signal */
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event_set_state(&voice_event, STATE_NONSIGNALED);
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/* Copy the clip info */
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td->vi = *(struct voice_info *)td->ev.data;
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/* Be sure audio buffer is initialized */
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audio_restore_playback(AUDIO_WANT_VOICE);
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/* We need nothing more from the sending thread - let it run */
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queue_reply(&voice_queue, 1);
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if (td->state == TSTATE_STOPPED)
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{
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/* Boost CPU now */
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trigger_cpu_boost();
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}
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else if (!playback_is_playing())
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{
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/* Just voice, stop any clip still playing */
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pcmbuf_play_stop();
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}
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/* Clean-start the decoder */
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td->st = speex_decoder_init(&speex_wb_mode);
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/* Make bit buffer use our own buffer */
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speex_bits_set_bit_buffer(&td->bits, td->vi.start, td->vi.size);
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speex_decoder_ctl(td->st, SPEEX_GET_LOOKAHEAD, &td->lookahead);
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td->state = TSTATE_DECODE;
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return;
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case Q_VOICE_STOP:
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LOGFQUEUE("voice < Q_VOICE_STOP: %d", ev.data);
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if (td->ev.data != 0 && !playback_is_playing())
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{
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/* If not playing, it's just voice so stop pcm playback */
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pcmbuf_play_stop();
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}
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/* Cancel boost */
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sleep(0);
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td->state = TSTATE_STOPPED;
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event_set_state(&voice_event, STATE_SIGNALED);
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break;
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case Q_VOICE_STATE:
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LOGFQUEUE("voice < Q_VOICE_STATE");
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queue_reply(&voice_queue, td->state);
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if (td->state == TSTATE_STOPPED)
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break; /* Not in a playback state */
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return;
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default:
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/* Default messages get a reply and thread continues with no
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* state transition */
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LOGFQUEUE("voice < default");
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if (td->state == TSTATE_STOPPED)
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break; /* Not in playback state */
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queue_reply(&voice_queue, 0);
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return;
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}
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queue_wait(&voice_queue, &td->ev);
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}
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}
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/* Voice thread entrypoint */
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static void voice_thread(void)
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{
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struct voice_thread_data td;
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voice_data_init(&td);
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goto message_wait;
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while (1)
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{
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td.state = TSTATE_DECODE;
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if (!queue_empty(&voice_queue))
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{
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message_wait:
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queue_wait(&voice_queue, &td.ev);
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message_process:
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voice_message(&td);
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/* Branch to initial start point or branch back to previous
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* operation if interrupted by a message */
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switch (td.state)
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{
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case TSTATE_DECODE: goto voice_decode;
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case TSTATE_BUFFER_INSERT: goto buffer_insert;
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default: goto message_wait;
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}
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}
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voice_decode:
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/* Check if all data was exhausted for this clip */
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if (speex_bits_remaining(&td.bits) < 8)
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{
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voice_error:
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/* Get next clip */
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td.vi.size = 0;
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if (td.vi.get_more != NULL)
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td.vi.get_more(&td.vi.start, &td.vi.size);
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if (td.vi.start != NULL && td.vi.size > 0)
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{
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/* Make bit buffer use our own buffer */
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speex_bits_set_bit_buffer(&td.bits, td.vi.start, td.vi.size);
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speex_decoder_ctl(td.st, SPEEX_GET_LOOKAHEAD, &td.lookahead);
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yield();
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if (!queue_empty(&voice_queue))
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goto message_wait;
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else
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goto voice_decode;
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}
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/* If all clips are done and not playing, force pcm playback. */
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if (!pcm_is_playing())
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pcmbuf_play_start();
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/* Synthesize a stop request */
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/* NOTE: We have no way to know when the pcm data placed in the
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* buffer is actually consumed and playback has reached the end
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* so until the info is available or inferred somehow, this will
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* not be accurate and the stopped signal will come too soon.
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* ie. You may not hear the "Shutting Down" splash even though
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* it waits for voice to stop. */
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td.ev.id = Q_VOICE_STOP;
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td.ev.data = 0; /* Let PCM drain by itself */
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yield();
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goto message_process;
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}
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/* Decode the data */
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int status = speex_decode_int(td.st, &td.bits, voice_output_buf);
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yield();
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if (status == -2)
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goto voice_error; /* error - try some more */
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/* Output the decoded frame */
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td.count = VOICE_FRAME_SIZE - td.lookahead;
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td.src[0] = (const char *)&voice_output_buf[td.lookahead];
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td.src[1] = NULL;
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td.lookahead -= MIN(VOICE_FRAME_SIZE, td.lookahead);
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buffer_insert:
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/* Process the PCM samples in the DSP and send out for mixing */
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td.state = TSTATE_BUFFER_INSERT;
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while (td.count > 0)
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{
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int out_count = dsp_output_count(td.dsp, td.count);
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int inp_count;
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char *dest;
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while (1)
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{
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if (!queue_empty(&voice_queue))
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goto message_wait;
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if ((dest = pcmbuf_request_voice_buffer(&out_count)) != NULL)
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break;
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yield();
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}
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/* Get the real input_size for output_size bytes, guarding
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* against resampling buffer overflows. */
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inp_count = dsp_input_count(td.dsp, out_count);
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if (inp_count <= 0)
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break;
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/* Input size has grown, no error, just don't write more than
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* length */
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if (inp_count > td.count)
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inp_count = td.count;
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out_count = dsp_process(td.dsp, dest, td.src, inp_count);
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if (out_count <= 0)
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break;
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pcmbuf_write_voice_complete(out_count);
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td.count -= inp_count;
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}
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yield();
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} /* end while */
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} /* voice_thread */
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/* Initialize all synchronization objects create the thread */
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void voice_thread_init(void)
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{
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logf("Starting voice thread");
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queue_init(&voice_queue, false);
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queue_enable_queue_send(&voice_queue, &voice_queue_sender_list);
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mutex_init(&voice_mutex);
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event_init(&voice_event, STATE_SIGNALED | EVENT_MANUAL);
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voice_thread_p = create_thread(voice_thread, voice_stack,
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sizeof(voice_stack), CREATE_THREAD_FROZEN,
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voice_thread_name IF_PRIO(, PRIORITY_PLAYBACK) IF_COP(, CPU));
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} /* voice_thread_init */
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/* Unfreeze the voice thread */
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void voice_thread_resume(void)
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{
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logf("Thawing voice thread");
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thread_thaw(voice_thread_p);
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/* Wait for initialization to complete (a very short wait until the
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* voice thread is available to process messages) */
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queue_send(&voice_queue, Q_VOICE_NULL, 0);
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}
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#ifdef HAVE_PRIORITY_SCHEDULING
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/* Set the voice thread priority */
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void voice_thread_set_priority(int priority)
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{
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thread_set_priority(voice_thread_p, priority);
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}
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#endif
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