diff options
Diffstat (limited to 'apps/plugins/mpegplayer/audio_thread.c')
| -rw-r--r-- | apps/plugins/mpegplayer/audio_thread.c | 721 |
1 files changed, 0 insertions, 721 deletions
diff --git a/apps/plugins/mpegplayer/audio_thread.c b/apps/plugins/mpegplayer/audio_thread.c deleted file mode 100644 index 764ad111f2..0000000000 --- a/apps/plugins/mpegplayer/audio_thread.c +++ /dev/null @@ -1,721 +0,0 @@ -/*************************************************************************** - * __________ __ ___. - * 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 */ - struct dsp_buffer src; /* Current audio data for DSP processing */ -}; - -/* 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 head 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: - 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: - /* 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(rb->thread_self(), - 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 = rb->dsp_get_config(CODEC_IDX_AUDIO); - rb->dsp_configure(td.dsp, DSP_SET_OUT_FREQUENCY, CLOCK_RATE); -#ifdef HAVE_PITCHCONTROL - rb->sound_set_pitch(PITCH_SPEED_100); - rb->dsp_set_timestretch(PITCH_SPEED_100); -#endif - rb->dsp_configure(td.dsp, DSP_RESET, 0); - rb->dsp_configure(td.dsp, DSP_FLUSH, 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; - /* Anything else is interpreted as an exit */ - default: - { -#ifdef HAVE_PRIORITY_SCHEDULING - rb->thread_set_priority(rb->thread_self(), 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; /* Still have frames to decode */ - - /* Used up remainder of compressed audio buffer. Wait for - * samples on PCM buffer to finish playing. */ - audio_queue_reset(); - - while (1) - { - if (pcm_output_empty()) - { - td.state = TSTATE_EOS; - stream_generate_event(&audio_str, STREAM_EV_COMPLETE, 0); - break; - } - - pcm_output_drain(); - str_get_msg_w_tmo(&audio_str, &td.ev, 1); - - if (td.ev.id != SYS_TIMEOUT) - break; - } - - 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; - td.status = STREAM_ERROR; - stream_generate_event(&audio_str, STREAM_EV_COMPLETE, 0); - - 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_SET_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.src.remcount = synth.pcm.length; - td.src.pin[0] = synth.pcm.samples[0]; - td.src.pin[1] = synth.pcm.samples[1]; - td.src.proc_mask = 0; - - td.state = TSTATE_RENDER_WAIT; - - /* Add a frame of audio to the pcm buffer. Maximum is 1152 samples. */ - render_wait: - rb->yield(); - - while (1) - { - struct dsp_buffer dst; - dst.remcount = 0; - dst.bufcount = MAX(td.src.remcount, 1024); - - ssize_t size = dst.bufcount * 2 * sizeof(int16_t); - - /* Wait for required amount of free buffer space */ - while ((dst.p16out = pcm_output_get_buffer(&size)) == NULL) - { - /* Wait one frame */ - int timeout = dst.bufcount*HZ / td.samplerate; - str_get_msg_w_tmo(&audio_str, &td.ev, MAX(timeout, 1)); - if (td.ev.id != SYS_TIMEOUT) - goto message_process; - } - - dst.bufcount = size / (2 * sizeof (int16_t)); - rb->dsp_process(td.dsp, &td.src, &dst); - - if (dst.remcount > 0) - { - /* Make this data available to DMA */ - pcm_output_commit_data(dst.remcount * 2 * sizeof(int16_t), - 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 += dst.remcount; - } - else if (td.src.remcount <= 0) - { - break; - } - } - } /* 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; - } -} |