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rockbox/apps/plugins/mpegplayer/audio_thread.c
Rafaël Carré 5574af8334 mpegplayer mad: apply r27655 fix 
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@27659 a1c6a512-1295-4272-9138-f99709370657
2010-08-01 16:38:58 +00:00

716 lines
22 KiB
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/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* mpegplayer audio thread implementation
*
* 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 "plugin.h"
#include "mpegplayer.h"
#include "codecs/libmad/bit.h"
#include "codecs/libmad/mad.h"
/** Audio stream and thread **/
struct pts_queue_slot;
struct audio_thread_data
{
struct queue_event ev; /* Our event queue to receive commands */
int state; /* Thread state */
int status; /* Media status (STREAM_PLAYING, etc.) */
int mad_errors; /* A count of the errors in each frame */
unsigned samplerate; /* Current stream sample rate */
int nchannels; /* Number of audio channels */
struct dsp_config *dsp; /* The DSP we're using */
};
/* The audio thread is stolen from the core codec thread */
static struct event_queue audio_str_queue SHAREDBSS_ATTR;
static struct queue_sender_list audio_str_queue_send SHAREDBSS_ATTR;
struct stream audio_str IBSS_ATTR;
/* libmad related definitions */
static struct mad_stream stream IBSS_ATTR;
static struct mad_frame frame IBSS_ATTR;
static struct mad_synth synth IBSS_ATTR;
/*sbsample buffer for mad_frame*/
mad_fixed_t sbsample[2][36][32];
/* 2567 bytes */
static unsigned char mad_main_data[MAD_BUFFER_MDLEN];
/* There isn't enough room for this in IRAM on PortalPlayer, but there
is for Coldfire. */
/* 4608 bytes */
#if defined(CPU_COLDFIRE) || defined(CPU_S5L870X)
static mad_fixed_t mad_frame_overlap[2][32][18] IBSS_ATTR;
#else
static mad_fixed_t mad_frame_overlap[2][32][18];
#endif
/** A queue for saving needed information about MPEG audio packets **/
#define AUDIODESC_QUEUE_LEN (1 << 5) /* 32 should be way more than sufficient -
if not, the case is handled */
#define AUDIODESC_QUEUE_MASK (AUDIODESC_QUEUE_LEN-1)
struct audio_frame_desc
{
uint32_t time; /* Time stamp for packet in audio ticks */
ssize_t size; /* Number of unprocessed bytes left in packet */
};
/* This starts out wr == rd but will never be emptied to zero during
streaming again in order to support initializing the first packet's
timestamp without a special case */
struct
{
/* Compressed audio data */
uint8_t *start; /* Start of encoded audio buffer */
uint8_t *ptr; /* Pointer to next encoded audio data */
ssize_t used; /* Number of bytes in MPEG audio buffer */
/* Compressed audio data descriptors */
unsigned read, write;
struct audio_frame_desc *curr; /* Current slot */
struct audio_frame_desc descs[AUDIODESC_QUEUE_LEN];
} audio_queue;
static inline int audiodesc_queue_count(void)
{
return audio_queue.write - audio_queue.read;
}
static inline bool audiodesc_queue_full(void)
{
return audio_queue.used >= MPA_MAX_FRAME_SIZE + MAD_BUFFER_GUARD ||
audiodesc_queue_count() >= AUDIODESC_QUEUE_LEN;
}
/* Increments the queue tail postion - should be used to preincrement */
static inline void audiodesc_queue_add_tail(void)
{
if (audiodesc_queue_full())
{
DEBUGF("audiodesc_queue_add_tail: audiodesc queue full!\n");
return;
}
audio_queue.write++;
}
/* Increments the queue tail position - leaves one slot as current */
static inline bool audiodesc_queue_remove_head(void)
{
if (audio_queue.write == audio_queue.read)
return false;
audio_queue.read++;
return true;
}
/* Returns the "tail" at the index just behind the write index */
static inline struct audio_frame_desc * audiodesc_queue_tail(void)
{
return &audio_queue.descs[(audio_queue.write - 1) & AUDIODESC_QUEUE_MASK];
}
/* Returns a pointer to the current head */
static inline struct audio_frame_desc * audiodesc_queue_head(void)
{
return &audio_queue.descs[audio_queue.read & AUDIODESC_QUEUE_MASK];
}
/* Resets the pts queue - call when starting and seeking */
static void audio_queue_reset(void)
{
audio_queue.ptr = audio_queue.start;
audio_queue.used = 0;
audio_queue.read = 0;
audio_queue.write = 0;
rb->memset(audio_queue.descs, 0, sizeof (audio_queue.descs));
audio_queue.curr = audiodesc_queue_head();
}
static void audio_queue_advance_pos(ssize_t len)
{
audio_queue.ptr += len;
audio_queue.used -= len;
audio_queue.curr->size -= len;
}
static int audio_buffer(struct stream *str, enum stream_parse_mode type)
{
int ret = STREAM_OK;
/* Carry any overshoot to the next size since we're technically
-size bytes into it already. If size is negative an audio
frame was split across packets. Old has to be saved before
moving the head. */
if (audio_queue.curr->size <= 0 && audiodesc_queue_remove_head())
{
struct audio_frame_desc *old = audio_queue.curr;
audio_queue.curr = audiodesc_queue_head();
audio_queue.curr->size += old->size;
old->size = 0;
}
/* Add packets to compressed audio buffer until it's full or the
* timestamp queue is full - whichever happens first */
while (!audiodesc_queue_full())
{
ret = parser_get_next_data(str, type);
struct audio_frame_desc *curr;
ssize_t len;
if (ret != STREAM_OK)
break;
/* Get data from next audio packet */
len = str->curr_packet_end - str->curr_packet;
if (str->pkt_flags & PKT_HAS_TS)
{
audiodesc_queue_add_tail();
curr = audiodesc_queue_tail();
curr->time = TS_TO_TICKS(str->pts);
/* pts->size should have been zeroed when slot was
freed */
}
else
{
/* Add to the one just behind the tail - this may be
* the head or the previouly added tail - whether or
* not we'll ever reach this is quite in question
* since audio always seems to have every packet
* timestamped */
curr = audiodesc_queue_tail();
}
curr->size += len;
/* Slide any remainder over to beginning */
if (audio_queue.ptr > audio_queue.start && audio_queue.used > 0)
{
rb->memmove(audio_queue.start, audio_queue.ptr,
audio_queue.used);
}
/* Splice this packet onto any remainder */
rb->memcpy(audio_queue.start + audio_queue.used,
str->curr_packet, len);
audio_queue.used += len;
audio_queue.ptr = audio_queue.start;
rb->yield();
}
return ret;
}
/* Initialise libmad */
static void init_mad(void)
{
/* init the sbsample buffer */
frame.sbsample_prev = &sbsample;
frame.sbsample = &sbsample;
/* We do this so libmad doesn't try to call codec_calloc(). This needs to
* be called before mad_stream_init(), mad_frame_inti() and
* mad_synth_init(). */
frame.overlap = &mad_frame_overlap;
stream.main_data = &mad_main_data;
/* Call mad initialization. Those will zero the arrays frame.overlap,
* frame.sbsample and frame.sbsample_prev. Therefore there is no need to
* zero them here. */
mad_stream_init(&stream);
mad_frame_init(&frame);
mad_synth_init(&synth);
}
/* Sync audio stream to a particular frame - see main decoder loop for
* detailed remarks */
static int audio_sync(struct audio_thread_data *td,
struct str_sync_data *sd)
{
int retval = STREAM_MATCH;
uint32_t sdtime = TS_TO_TICKS(clip_time(&audio_str, sd->time));
uint32_t time;
uint32_t duration = 0;
struct stream *str;
struct stream tmp_str;
struct mad_header header;
struct mad_stream stream;
if (td->ev.id == STREAM_SYNC)
{
/* Actually syncing for playback - use real stream */
time = 0;
str = &audio_str;
}
else
{
/* Probing - use temp stream */
time = INVALID_TIMESTAMP;
str = &tmp_str;
str->id = audio_str.id;
}
str->hdr.pos = sd->sk.pos;
str->hdr.limit = sd->sk.pos + sd->sk.len;
mad_stream_init(&stream);
mad_header_init(&header);
while (1)
{
if (audio_buffer(str, STREAM_PM_RANDOM_ACCESS) == STREAM_DATA_END)
{
DEBUGF("audio_sync:STR_DATA_END\n aqu:%ld swl:%ld swr:%ld\n",
(long)audio_queue.used, str->hdr.win_left, str->hdr.win_right);
if (audio_queue.used <= MAD_BUFFER_GUARD)
goto sync_data_end;
}
stream.error = 0;
mad_stream_buffer(&stream, audio_queue.ptr, audio_queue.used);
if (stream.sync && mad_stream_sync(&stream) < 0)
{
DEBUGF(" audio: mad_stream_sync failed\n");
audio_queue_advance_pos(MAX(audio_queue.curr->size - 1, 1));
continue;
}
stream.sync = 0;
if (mad_header_decode(&header, &stream) < 0)
{
DEBUGF(" audio: mad_header_decode failed:%s\n",
mad_stream_errorstr(&stream));
audio_queue_advance_pos(1);
continue;
}
duration = 32*MAD_NSBSAMPLES(&header);
time = audio_queue.curr->time;
DEBUGF(" audio: ft:%u t:%u fe:%u nsamp:%u sampr:%u\n",
(unsigned)TICKS_TO_TS(time), (unsigned)sd->time,
(unsigned)TICKS_TO_TS(time + duration),
(unsigned)duration, header.samplerate);
audio_queue_advance_pos(stream.this_frame - audio_queue.ptr);
if (time <= sdtime && sdtime < time + duration)
{
DEBUGF(" audio: ft<=t<fe\n");
retval = STREAM_PERFECT_MATCH;
break;
}
else if (time > sdtime)
{
DEBUGF(" audio: ft>t\n");
break;
}
audio_queue_advance_pos(stream.next_frame - audio_queue.ptr);
audio_queue.curr->time += duration;
rb->yield();
}
sync_data_end:
if (td->ev.id == STREAM_FIND_END_TIME)
{
if (time != INVALID_TIMESTAMP)
{
time = TICKS_TO_TS(time);
duration = TICKS_TO_TS(duration);
sd->time = time + duration;
retval = STREAM_PERFECT_MATCH;
}
else
{
retval = STREAM_NOT_FOUND;
}
}
DEBUGF(" audio header: 0x%02X%02X%02X%02X\n",
(unsigned)audio_queue.ptr[0], (unsigned)audio_queue.ptr[1],
(unsigned)audio_queue.ptr[2], (unsigned)audio_queue.ptr[3]);
return retval;
(void)td;
}
static void audio_thread_msg(struct audio_thread_data *td)
{
while (1)
{
intptr_t reply = 0;
switch (td->ev.id)
{
case STREAM_PLAY:
td->status = STREAM_PLAYING;
switch (td->state)
{
case TSTATE_INIT:
td->state = TSTATE_DECODE;
case TSTATE_DECODE:
case TSTATE_RENDER_WAIT:
case TSTATE_RENDER_WAIT_END:
break;
case TSTATE_EOS:
/* At end of stream - no playback possible so fire the
* completion event */
stream_generate_event(&audio_str, STREAM_EV_COMPLETE, 0);
break;
}
break;
case STREAM_PAUSE:
td->status = STREAM_PAUSED;
reply = td->state != TSTATE_EOS;
break;
case STREAM_STOP:
if (td->state == TSTATE_DATA)
stream_clear_notify(&audio_str, DISK_BUF_DATA_NOTIFY);
td->status = STREAM_STOPPED;
td->state = TSTATE_EOS;
reply = true;
break;
case STREAM_RESET:
if (td->state == TSTATE_DATA)
stream_clear_notify(&audio_str, DISK_BUF_DATA_NOTIFY);
td->status = STREAM_STOPPED;
td->state = TSTATE_INIT;
td->samplerate = 0;
td->nchannels = 0;
init_mad();
td->mad_errors = 0;
audio_queue_reset();
reply = true;
break;
case STREAM_NEEDS_SYNC:
reply = true; /* Audio always needs to */
break;
case STREAM_SYNC:
case STREAM_FIND_END_TIME:
if (td->state != TSTATE_INIT)
break;
reply = audio_sync(td, (struct str_sync_data *)td->ev.data);
break;
case DISK_BUF_DATA_NOTIFY:
/* Our bun is done */
if (td->state != TSTATE_DATA)
break;
td->state = TSTATE_DECODE;
str_data_notify_received(&audio_str);
break;
case STREAM_QUIT:
/* Time to go - make thread exit */
td->state = TSTATE_EOS;
return;
}
str_reply_msg(&audio_str, reply);
if (td->status == STREAM_PLAYING)
{
switch (td->state)
{
case TSTATE_DECODE:
case TSTATE_RENDER_WAIT:
case TSTATE_RENDER_WAIT_END:
/* These return when in playing state */
return;
}
}
str_get_msg(&audio_str, &td->ev);
}
}
static void audio_thread(void)
{
struct audio_thread_data td;
#ifdef HAVE_PRIORITY_SCHEDULING
/* Up the priority since the core DSP over-yields internally */
int old_priority = rb->thread_set_priority(THREAD_ID_CURRENT,
PRIORITY_PLAYBACK-4);
#endif
rb->memset(&td, 0, sizeof (td));
td.status = STREAM_STOPPED;
td.state = TSTATE_EOS;
/* We need this here to init the EMAC for Coldfire targets */
init_mad();
td.dsp = (struct dsp_config *)rb->dsp_configure(NULL, DSP_MYDSP,
CODEC_IDX_AUDIO);
rb->sound_set_pitch(PITCH_SPEED_100);
rb->dsp_configure(td.dsp, DSP_RESET, 0);
rb->dsp_configure(td.dsp, DSP_SET_SAMPLE_DEPTH, MAD_F_FRACBITS);
goto message_wait;
/* This is the decoding loop. */
while (1)
{
td.state = TSTATE_DECODE;
/* Check for any pending messages and process them */
if (str_have_msg(&audio_str))
{
message_wait:
/* Wait for a message to be queued */
str_get_msg(&audio_str, &td.ev);
message_process:
/* Process a message already dequeued */
audio_thread_msg(&td);
switch (td.state)
{
/* These states are the only ones that should return */
case TSTATE_DECODE: goto audio_decode;
case TSTATE_RENDER_WAIT: goto render_wait;
case TSTATE_RENDER_WAIT_END: goto render_wait_end;
/* Anything else is interpreted as an exit */
default:
{
#ifdef HAVE_PRIORITY_SCHEDULING
rb->thread_set_priority(THREAD_ID_CURRENT, old_priority);
#endif
return;
}
}
}
audio_decode:
/** Buffering **/
switch (audio_buffer(&audio_str, STREAM_PM_STREAMING))
{
case STREAM_DATA_NOT_READY:
{
td.state = TSTATE_DATA;
goto message_wait;
} /* STREAM_DATA_NOT_READY: */
case STREAM_DATA_END:
{
if (audio_queue.used > MAD_BUFFER_GUARD)
break;
/* Used up remainder of compressed audio buffer.
* Force any residue to play if audio ended before
* reaching the threshold */
td.state = TSTATE_RENDER_WAIT_END;
audio_queue_reset();
render_wait_end:
pcm_output_drain();
while (pcm_output_used() > (ssize_t)PCMOUT_LOW_WM)
{
str_get_msg_w_tmo(&audio_str, &td.ev, 1);
if (td.ev.id != SYS_TIMEOUT)
goto message_process;
}
td.state = TSTATE_EOS;
if (td.status == STREAM_PLAYING)
stream_generate_event(&audio_str, STREAM_EV_COMPLETE, 0);
rb->yield();
goto message_wait;
} /* STREAM_DATA_END: */
}
/** Decoding **/
mad_stream_buffer(&stream, audio_queue.ptr, audio_queue.used);
int mad_stat = mad_frame_decode(&frame, &stream);
ssize_t len = stream.next_frame - audio_queue.ptr;
if (mad_stat != 0)
{
DEBUGF("audio: Stream error: %s\n",
mad_stream_errorstr(&stream));
/* If something's goofed - try to perform resync by moving
* at least one byte at a time */
audio_queue_advance_pos(MAX(len, 1));
if (stream.error == MAD_ERROR_BUFLEN)
{
/* This makes the codec support partially corrupted files */
if (++td.mad_errors <= MPA_MAX_FRAME_SIZE)
{
stream.error = 0;
rb->yield();
continue;
}
DEBUGF("audio: Too many errors\n");
}
else if (MAD_RECOVERABLE(stream.error))
{
/* libmad says it can recover - just keep on decoding */
rb->yield();
continue;
}
else
{
/* Some other unrecoverable error */
DEBUGF("audio: Unrecoverable error\n");
}
/* This is too hard - bail out */
td.state = TSTATE_EOS;
if (td.status == STREAM_PLAYING)
stream_generate_event(&audio_str, STREAM_EV_COMPLETE, 0);
td.status = STREAM_ERROR;
goto message_wait;
}
/* Adjust sizes by the frame size */
audio_queue_advance_pos(len);
td.mad_errors = 0; /* Clear errors */
/* Generate the pcm samples */
mad_synth_frame(&synth, &frame);
/** Output **/
if (frame.header.samplerate != td.samplerate)
{
td.samplerate = frame.header.samplerate;
rb->dsp_configure(td.dsp, DSP_SWITCH_FREQUENCY,
td.samplerate);
}
if (MAD_NCHANNELS(&frame.header) != td.nchannels)
{
td.nchannels = MAD_NCHANNELS(&frame.header);
rb->dsp_configure(td.dsp, DSP_SET_STEREO_MODE,
td.nchannels == 1 ?
STEREO_MONO : STEREO_NONINTERLEAVED);
}
td.state = TSTATE_RENDER_WAIT;
/* Add a frame of audio to the pcm buffer. Maximum is 1152 samples. */
render_wait:
if (synth.pcm.length > 0)
{
struct pcm_frame_header *dst_hdr = pcm_output_get_buffer();
const char *src[2] =
{ (char *)synth.pcm.samples[0], (char *)synth.pcm.samples[1] };
int out_count = (synth.pcm.length * CLOCK_RATE
+ (td.samplerate - 1)) / td.samplerate;
ssize_t size = sizeof(*dst_hdr) + out_count*4;
/* Wait for required amount of free buffer space */
while (pcm_output_free() < size)
{
/* Wait one frame */
int timeout = out_count*HZ / td.samplerate;
str_get_msg_w_tmo(&audio_str, &td.ev, MAX(timeout, 1));
if (td.ev.id != SYS_TIMEOUT)
goto message_process;
}
out_count = rb->dsp_process(td.dsp, dst_hdr->data, src,
synth.pcm.length);
if (out_count <= 0)
break;
dst_hdr->size = sizeof(*dst_hdr) + out_count*4;
dst_hdr->time = audio_queue.curr->time;
/* As long as we're on this timestamp, the time is just
incremented by the number of samples */
audio_queue.curr->time += out_count;
/* Make this data available to DMA */
pcm_output_add_data();
}
rb->yield();
} /* end decoding loop */
}
/* Initializes the audio thread resources and starts the thread */
bool audio_thread_init(void)
{
/* Initialise the encoded audio buffer and its descriptors */
audio_queue.start = mpeg_malloc(AUDIOBUF_ALLOC_SIZE,
MPEG_ALLOC_AUDIOBUF);
if (audio_queue.start == NULL)
return false;
/* Start the audio thread */
audio_str.hdr.q = &audio_str_queue;
rb->queue_init(audio_str.hdr.q, false);
/* We steal the codec thread for audio */
rb->codec_thread_do_callback(audio_thread, &audio_str.thread);
rb->queue_enable_queue_send(audio_str.hdr.q, &audio_str_queue_send,
audio_str.thread);
/* Wait for thread to initialize */
str_send_msg(&audio_str, STREAM_NULL, 0);
return true;
}
/* Stops the audio thread */
void audio_thread_exit(void)
{
if (audio_str.thread != 0)
{
str_post_msg(&audio_str, STREAM_QUIT, 0);
rb->codec_thread_do_callback(NULL, NULL);
audio_str.thread = 0;
}
}
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