/*
* This file is part of DisOrder.
- * Copyright (C) 2007 Richard Kettlewell
+ * Copyright (C) 2008 Richard Kettlewell
*
* 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
*/
/** @file clients/playrtp-alsa.c
* @brief RTP player - ALSA support
+ *
+ * This has been rewritten to use the @ref alsabg.h interface and is therefore
+ * now closely modelled on @ref playrtp-coreaudio.c. Given a similar interface
+ * wrapping OSS the whole of playrtp could probably be greatly simplified.
*/
#include <config.h>
#include "vector.h"
#include "heap.h"
#include "playrtp.h"
+#include "alsabg.h"
-/** @brief PCM handle */
-static snd_pcm_t *pcm;
-
-/** @brief True when @ref pcm is up and running */
-static int playrtp_alsa_prepared = 1;
-
-static void playrtp_alsa_init(void) {
- snd_pcm_hw_params_t *hwparams;
- snd_pcm_sw_params_t *swparams;
- /* Only support one format for now */
- const int sample_format = SND_PCM_FORMAT_S16_BE;
- unsigned rate = 44100;
- const int channels = 2;
- const int samplesize = channels * sizeof(uint16_t);
- snd_pcm_uframes_t pcm_bufsize = MAXSAMPLES * samplesize * 3;
- /* If we can write more than this many samples we'll get a wakeup */
- const int avail_min = 256;
- int err;
-
- /* Open ALSA */
- if((err = snd_pcm_open(&pcm,
- device ? device : "default",
- SND_PCM_STREAM_PLAYBACK,
- SND_PCM_NONBLOCK)))
- fatal(0, "error from snd_pcm_open: %d", err);
- /* Set up 'hardware' parameters */
- snd_pcm_hw_params_alloca(&hwparams);
- if((err = snd_pcm_hw_params_any(pcm, hwparams)) < 0)
- fatal(0, "error from snd_pcm_hw_params_any: %d", err);
- if((err = snd_pcm_hw_params_set_access(pcm, hwparams,
- SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
- fatal(0, "error from snd_pcm_hw_params_set_access: %d", err);
- if((err = snd_pcm_hw_params_set_format(pcm, hwparams,
- sample_format)) < 0)
-
- fatal(0, "error from snd_pcm_hw_params_set_format (%d): %d",
- sample_format, err);
- if((err = snd_pcm_hw_params_set_rate_near(pcm, hwparams, &rate, 0)) < 0)
- fatal(0, "error from snd_pcm_hw_params_set_rate (%d): %d",
- rate, err);
- if((err = snd_pcm_hw_params_set_channels(pcm, hwparams,
- channels)) < 0)
- fatal(0, "error from snd_pcm_hw_params_set_channels (%d): %d",
- channels, err);
- if((err = snd_pcm_hw_params_set_buffer_size_near(pcm, hwparams,
- &pcm_bufsize)) < 0)
- fatal(0, "error from snd_pcm_hw_params_set_buffer_size (%ld): %d",
- (long)pcm_bufsize, err);
- if((err = snd_pcm_hw_params(pcm, hwparams)) < 0)
- fatal(0, "error calling snd_pcm_hw_params: %d", err);
- /* Set up 'software' parameters */
- snd_pcm_sw_params_alloca(&swparams);
- if((err = snd_pcm_sw_params_current(pcm, swparams)) < 0)
- fatal(0, "error calling snd_pcm_sw_params_current: %d", err);
- if((err = snd_pcm_sw_params_set_avail_min(pcm, swparams, avail_min)) < 0)
- fatal(0, "error calling snd_pcm_sw_params_set_avail_min %d: %d",
- avail_min, err);
- /* Default start threshold is 1, which means that PCM starts as soon as we've
- * written anything. Setting it to pcm_bufsize (around 15000) produces
- * -EINVAL. 1024 is a guess... */
- if((err = snd_pcm_sw_params_set_start_threshold(pcm, swparams, 1024)) < 0)
- fatal(0, "error calling snd_pcm_sw_params_set_start_threshold %d: %d",
- 1024, err);
- if((err = snd_pcm_sw_params(pcm, swparams)) < 0)
- fatal(0, "error calling snd_pcm_sw_params: %d", err);
-}
-
-/** @brief Wait until ALSA wants some audio */
-static void wait_alsa(void) {
- struct pollfd fds[64];
- int nfds, err;
- unsigned short events;
-
- for(;;) {
- do {
- if((nfds = snd_pcm_poll_descriptors(pcm,
- fds, sizeof fds / sizeof *fds)) < 0)
- fatal(0, "error calling snd_pcm_poll_descriptors: %d", nfds);
- } while(poll(fds, nfds, -1) < 0 && errno == EINTR);
- if((err = snd_pcm_poll_descriptors_revents(pcm, fds, nfds, &events)))
- fatal(0, "error calling snd_pcm_poll_descriptors_revents: %d", err);
- if(events & POLLOUT)
- return;
- }
-}
-
-/** @brief Play some sound via ALSA
- * @param s Pointer to sample data
- * @param n Number of samples
- * @return 0 on success, -1 on non-fatal error
- */
-static int playrtp_alsa_writei(const void *s, size_t n) {
- int err;
- snd_pcm_sframes_t frames_written;
-
- /* Do the write */
- frames_written = snd_pcm_writei(pcm, s, n / 2);
- if(frames_written < 0) {
- /* Something went wrong */
- switch(frames_written) {
- case -EAGAIN:
- return 0;
- case -EPIPE:
- error(0, "error calling snd_pcm_writei: %ld",
- (long)frames_written);
- if((err = snd_pcm_recover(pcm, -EPIPE, 0)) < 0) {
- error(0, "error calling snd_pcm_recover: %d", err);
- return -1;
- }
- frames_written = snd_pcm_writei(pcm, s, n / 2);
- if(frames_written == -EAGAIN)
- return 0;
- else if(frames_written < 0) {
- error(0, "error retrying snd_pcm_writei: %ld",
- (long)frames_written);
- return -1;
- }
- break;
- default:
- fatal(0, "error calling snd_pcm_writei: %ld",
- (long)frames_written);
- }
- }
- /* Success */
- next_timestamp += frames_written * 2;
- if(dump_buffer) {
- snd_pcm_sframes_t count;
- const int16_t *sp = s;
-
- for(count = 0; count < frames_written * 2; ++count) {
- dump_buffer[dump_index++] = (int16_t)ntohs(*sp++);
- dump_index %= dump_size;
- }
- }
- return 0;
-}
+/** @brief Callback from alsa_bg_collect() */
+static int playrtp_alsa_supply(void *dst,
+ unsigned supply_nsamples) {
+ unsigned samples_available;
-/** @brief Play the relevant part of a packet
- * @param p Packet to play
- * @return 0 on success, -1 on non-fatal error
- */
-static int playrtp_alsa_play(const struct packet *p) {
- return playrtp_alsa_writei(p->samples_raw + next_timestamp - p->timestamp,
- (p->timestamp + p->nsamples) - next_timestamp);
-}
-
-/** @brief Play some silence
- * @param p Next packet or NULL
- * @return 0 on success, -1 on non-fatal error
- */
-static int playrtp_alsa_infill(const struct packet *p) {
- static const uint16_t zeros[INFILL_SAMPLES];
- size_t samples_available = INFILL_SAMPLES;
-
- if(p && samples_available > p->timestamp - next_timestamp)
- samples_available = p->timestamp - next_timestamp;
- return playrtp_alsa_writei(zeros, samples_available);
-}
-
-static void playrtp_alsa_enable(void){
- int err;
-
- if(!playrtp_alsa_prepared) {
- if((err = snd_pcm_prepare(pcm)))
- fatal(0, "error calling snd_pcm_prepare: %d", err);
- playrtp_alsa_prepared = 1;
- }
-}
-
-/** @brief Reset ALSA state after we lost synchronization */
-static void playrtp_alsa_disable(int hard_reset) {
- int err;
-
- if((err = snd_pcm_nonblock(pcm, 0)))
- fatal(0, "error calling snd_pcm_nonblock: %d", err);
- if(hard_reset) {
- if((err = snd_pcm_drop(pcm)))
- fatal(0, "error calling snd_pcm_drop: %d", err);
+ pthread_mutex_lock(&lock);
+ const struct packet *p = playrtp_next_packet();
+ if(p && contains(p, next_timestamp)) {
+ /* This packet is ready to play */
+ const uint32_t packet_end = p->timestamp + p->nsamples;
+ const uint32_t offset = next_timestamp - p->timestamp;
+ const uint16_t *src = (void *)(p->samples_raw + offset);
+ samples_available = packet_end - next_timestamp;
+ if(samples_available > supply_nsamples)
+ samples_available = supply_nsamples;
+ next_timestamp += samples_available;
+ memcpy(dst, src, samples_available * sizeof (int16_t));
+ /* We don't bother junking the packet - that'll be dealt with next time
+ * round */
} else {
- if((err = snd_pcm_drain(pcm))) {
- error(0, "error calling snd_pcm_drain: %d", err);
- if((err = snd_pcm_drop(pcm)))
- fatal(0, "error calling snd_pcm_drop: %d", err);
- }
+ /* No packet is ready to play (and there might be no packet at all) */
+ samples_available = p ? p->timestamp - next_timestamp : supply_nsamples;
+ if(samples_available > supply_nsamples)
+ samples_available = supply_nsamples;
+ /*info("infill %d", samples_available);*/
+ next_timestamp += samples_available;
+ /* Unlike Core Audio the buffer is not guaranteed to be 0-filled */
+ memset(dst, 0, samples_available * sizeof (int16_t));
}
- if((err = snd_pcm_nonblock(pcm, 1)))
- fatal(0, "error calling snd_pcm_nonblock: %d", err);
- playrtp_alsa_prepared = 0;
+ pthread_mutex_unlock(&lock);
+ return samples_available;
}
void playrtp_alsa(void) {
- int escape;
- const struct packet *p;
-
- playrtp_alsa_init();
+ alsa_bg_init(device ? device : "default",
+ playrtp_alsa_supply);
pthread_mutex_lock(&lock);
for(;;) {
/* Wait for the buffer to fill up a bit */
playrtp_fill_buffer();
- playrtp_alsa_enable();
- escape = 0;
+ /* Start playing now */
info("Playing...");
- /* Keep playing until the buffer empties out, or ALSA tells us to get
- * lost */
- while((nsamples >= minbuffer
- || (nsamples > 0
- && contains(pheap_first(&packets), next_timestamp)))
- && !escape) {
- /* Wait for ALSA to ask us for more data */
- pthread_mutex_unlock(&lock);
- wait_alsa();
- pthread_mutex_lock(&lock);
- /* ALSA is ready for more data, find something to play */
- p = playrtp_next_packet();
- /* Play it or play some silence */
- if(contains(p, next_timestamp))
- escape = playrtp_alsa_play(p);
- else
- escape = playrtp_alsa_infill(p);
+ next_timestamp = pheap_first(&packets)->timestamp;
+ active = 1;
+ alsa_bg_enable();
+ /* Wait until the buffer empties out */
+ while(nsamples >= minbuffer
+ || (nsamples > 0
+ && contains(pheap_first(&packets), next_timestamp))) {
+ pthread_cond_wait(&cond, &lock);
}
+ /* Stop playing for a bit until the buffer re-fills */
+ alsa_bg_disable();
active = 0;
- /* We stop playing for a bit until the buffer re-fills */
- pthread_mutex_unlock(&lock);
- playrtp_alsa_disable(escape);
- pthread_mutex_lock(&lock);
+ /* Go back round */
}
}
for(;;) {
/* Get the next packet */
pthread_mutex_lock(&receive_lock);
- while(!received_packets)
+ while(!received_packets) {
pthread_cond_wait(&receive_cond, &receive_lock);
+ }
p = received_packets;
received_packets = p->next;
if(!received_packets)
* out of order then we guarantee dropouts. But for now... */
if(nsamples >= maxbuffer) {
pthread_mutex_lock(&lock);
- while(nsamples >= maxbuffer)
+ while(nsamples >= maxbuffer) {
pthread_cond_wait(&cond, &lock);
+ }
pthread_mutex_unlock(&lock);
}
/* Add the packet to the receive queue */
while(nsamples)
drop_first_packet();
info("Buffering...");
- while(nsamples < readahead)
+ while(nsamples < readahead) {
pthread_cond_wait(&cond, &lock);
+ }
next_timestamp = pheap_first(&packets)->timestamp;
active = 1;
}
#
# This file is part of DisOrder.
-# Copyright (C) 2004, 2005, 2006, 2007 Richard Kettlewell
+# Copyright (C) 2004-2008 Richard Kettlewell
#
# 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
libdisorder_a_SOURCES=charset.c charset.h \
addr.c addr.h \
+ alsabg.c alsabg.h \
authhash.c authhash.h \
basen.c basen.h \
base64.c base64.h \
--- /dev/null
+/*
+ * This file is part of DisOrder.
+ * Copyright (C) 2008 Richard Kettlewell
+ *
+ * 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 program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
+ * USA
+ */
+/** @file alsabg.c
+ * @brief Background-thread interface to ALSA
+ */
+
+#include <config.h>
+
+#if HAVE_ALSA_ASOUNDLIB_H
+#include "types.h"
+
+#include <alsa/asoundlib.h>
+#include <pthread.h>
+
+#include "alsabg.h"
+#include "log.h"
+
+/** @brief Output handle */
+static snd_pcm_t *pcm;
+
+/** @brief Called to get audio data */
+static alsa_bg_supply *supplyfn;
+
+static pthread_t alsa_bg_collect_tid, alsa_bg_play_tid;
+
+/** @brief Set to shut down the background threads */
+static int alsa_bg_shutdown = 0;
+
+/** @brief Number of channels (samples per frame) */
+#define CHANNELS 2
+
+/** @brief Number of bytes per samples */
+#define BYTES_PER_SAMPLE 2
+
+/** @brief Number of bytes per frame */
+#define BYTES_PER_FRAME (CHANNELS * BYTES_PER_SAMPLE)
+
+/** @brief Buffer size in bytes */
+#define BUFFER_BYTES 65536
+
+/** @brief Buffer size in frames */
+#define BUFFER_FRAMES (BUFFER_BYTES / BYTES_PER_FRAME)
+
+/** @brief Buffer size in samples */
+#define BUFFER_SAMPLES (BUFFER_BYTES / BYTES_PER_SAMPLE)
+
+/** @brief Audio buffer */
+static uint8_t alsa_bg_buffer[BUFFER_BYTES];
+
+/** @brief First playable byte in audio buffer */
+static unsigned alsa_bg_start;
+
+/** @brief Number of playable bytes in audio buffer */
+static unsigned alsa_bg_count;
+
+/** @brief Current enable status */
+static int alsa_bg_enabled;
+
+/** @brief Lock protecting audio buffer pointers */
+static pthread_mutex_t alsa_bg_lock = PTHREAD_MUTEX_INITIALIZER;
+
+/** @brief Signaled when buffer contents changes */
+static pthread_cond_t alsa_bg_cond = PTHREAD_COND_INITIALIZER;
+
+/** @brief Call a pthread_ function and fatal() on exit */
+#define ep(x) do { \
+ int prc; \
+ \
+ if((prc = (x))) \
+ fatal(prc, "%s", #x); \
+} while(0)
+
+/** @brief Data collection thread
+ *
+ * This thread collects audio data to play and stores it in the ring
+ * buffer.
+ */
+static void *alsa_bg_collect(void attribute((unused)) *arg) {
+ unsigned avail_start, avail_count;
+ int count;
+
+ ep(pthread_mutex_lock(&alsa_bg_lock));
+ for(;;) {
+ /* If we're shutting down, quit straight away */
+ if(alsa_bg_shutdown)
+ break;
+ /* While we're disabled or the buffer is full, just wait */
+ if(!alsa_bg_enabled || alsa_bg_count == BUFFER_BYTES) {
+ ep(pthread_cond_wait(&alsa_bg_cond, &alsa_bg_lock));
+ continue;
+ }
+ /* Figure out where and how big the gap we can write into is */
+ avail_start = alsa_bg_start + alsa_bg_count;
+ if(avail_start < BUFFER_BYTES)
+ avail_count = BUFFER_BYTES - avail_start;
+ else {
+ avail_start %= BUFFER_BYTES;
+ avail_count = alsa_bg_start - avail_start;
+ }
+ assert(avail_start < BUFFER_BYTES);
+ assert(avail_count <= BUFFER_BYTES);
+ assert(avail_count + alsa_bg_count <= BUFFER_BYTES);
+ ep(pthread_mutex_unlock(&alsa_bg_lock));
+ count = supplyfn(alsa_bg_buffer + avail_start,
+ avail_count / BYTES_PER_SAMPLE);
+ ep(pthread_mutex_lock(&alsa_bg_lock));
+ alsa_bg_count += count * BYTES_PER_SAMPLE;
+ assert(alsa_bg_start < BUFFER_BYTES);
+ assert(alsa_bg_count <= BUFFER_BYTES);
+ ep(pthread_cond_signal(&alsa_bg_cond));
+ }
+ ep(pthread_mutex_unlock(&alsa_bg_lock));
+ return 0;
+}
+
+/** @brief Playback thread
+ *
+ * This thread reads audio data out of the ring buffer and plays it back
+ */
+static void *alsa_bg_play(void attribute((unused)) *arg) {
+ int prepared = 1, err;
+ int start, nbytes, nframes, rframes;
+
+ ep(pthread_mutex_lock(&alsa_bg_lock));
+ for(;;) {
+ /* If we're shutting down, quit straight away */
+ if(alsa_bg_shutdown)
+ break;
+ /* Wait for some data to be available. (If we are marked disabled
+ * we keep on playing what we've got.) */
+ if(alsa_bg_count == 0) {
+ if(prepared) {
+ if((err = snd_pcm_drain(pcm)))
+ fatal(0, "snd_pcm_drain: %d", err);
+ prepared = 0;
+ }
+ ep(pthread_cond_wait(&alsa_bg_cond, &alsa_bg_lock));
+ continue;
+ }
+ /* Calculate how much we can play */
+ start = alsa_bg_start;
+ if(start + alsa_bg_count <= BUFFER_BYTES)
+ nbytes = alsa_bg_count;
+ else
+ nbytes = BUFFER_BYTES - start;
+ /* Limit how much of the buffer we play. The effect is that we return from
+ * _writei earlier, and therefore free up more buffer space to read fresh
+ * data into. /2 works fine, /4 is just conservative. /1 (i.e. abolishing
+ * the heuristic) produces noticably noisy output. */
+ if(nbytes > BUFFER_BYTES / 4)
+ nbytes = BUFFER_BYTES / 4;
+ assert((unsigned)nbytes <= alsa_bg_count);
+ nframes = nbytes / BYTES_PER_FRAME;
+ ep(pthread_mutex_unlock(&alsa_bg_lock));
+ /* Make sure the PCM is prepared */
+ if(!prepared) {
+ if((err = snd_pcm_prepare(pcm)))
+ fatal(0, "snd_pcm_prepare: %d", err);
+ prepared = 1;
+ }
+ /* Play what we can */
+ rframes = snd_pcm_writei(pcm, alsa_bg_buffer + start, nframes);
+ ep(pthread_mutex_lock(&alsa_bg_lock));
+ if(rframes < 0) {
+ error(0, "snd_pcm_writei: %d", rframes);
+ switch(rframes) {
+ case -EPIPE:
+ if((err = snd_pcm_recover(pcm, -EPIPE, 0)))
+ fatal(0, "snd_pcm_recover: %d", err);
+ break;
+ }
+ } else {
+ const int rbytes = rframes * BYTES_PER_FRAME;
+ /*fprintf(stderr, "%5d -> %5d\n", nbytes, rbytes);*/
+ /* Update the buffer pointers */
+ alsa_bg_count -= rbytes;
+ alsa_bg_start += rbytes;
+ if(alsa_bg_start >= BUFFER_BYTES)
+ alsa_bg_start -= BUFFER_BYTES;
+ assert(alsa_bg_start < BUFFER_BYTES);
+ assert(alsa_bg_count <= BUFFER_BYTES);
+ /* Let the collector know we've opened up some space */
+ ep(pthread_cond_signal(&alsa_bg_cond));
+ }
+ }
+ ep(pthread_mutex_unlock(&alsa_bg_lock));
+ return 0;
+}
+
+/** @brief Enable ALSA play */
+void alsa_bg_enable(void) {
+ ep(pthread_mutex_lock(&alsa_bg_lock));
+ alsa_bg_enabled = 1;
+ ep(pthread_cond_broadcast(&alsa_bg_cond));
+ ep(pthread_mutex_unlock(&alsa_bg_lock));
+}
+
+/** @brief Disable ALSA play */
+void alsa_bg_disable(void) {
+ ep(pthread_mutex_lock(&alsa_bg_lock));
+ alsa_bg_enabled = 0;
+ ep(pthread_cond_broadcast(&alsa_bg_cond));
+ ep(pthread_mutex_unlock(&alsa_bg_lock));
+}
+
+/** @brief Initialize background ALSA playback
+ * @param device Target device or NULL to use default
+ * @param supply Function to call to get audio data to play
+ *
+ * Playback is not initially enabled; see alsa_bg_enable(). When playback is
+ * enabled, @p supply will be called in a background thread to request audio
+ * data. It should return in a timely manner, but playback happens from a
+ * further thread and delays in @p supply will not delay transfer of data to
+ * the sound device (provided it doesn't actually run out).
+ */
+void alsa_bg_init(const char *device,
+ alsa_bg_supply *supply) {
+ snd_pcm_hw_params_t *hwparams;
+ /* Only support one format for now */
+ const int sample_format = SND_PCM_FORMAT_S16_BE;
+ unsigned rate = 44100;
+ int err;
+
+ if((err = snd_pcm_open(&pcm,
+ device ? device : "default",
+ SND_PCM_STREAM_PLAYBACK,
+ 0)))
+ fatal(0, "error from snd_pcm_open: %d", err);
+ /* Set up 'hardware' parameters */
+ snd_pcm_hw_params_alloca(&hwparams);
+ if((err = snd_pcm_hw_params_any(pcm, hwparams)) < 0)
+ fatal(0, "error from snd_pcm_hw_params_any: %d", err);
+ if((err = snd_pcm_hw_params_set_access(pcm, hwparams,
+ SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
+ fatal(0, "error from snd_pcm_hw_params_set_access: %d", err);
+ if((err = snd_pcm_hw_params_set_format(pcm, hwparams,
+ sample_format)) < 0)
+
+ fatal(0, "error from snd_pcm_hw_params_set_format (%d): %d",
+ sample_format, err);
+ if((err = snd_pcm_hw_params_set_rate_near(pcm, hwparams, &rate, 0)) < 0)
+ fatal(0, "error from snd_pcm_hw_params_set_rate (%d): %d",
+ rate, err);
+ if((err = snd_pcm_hw_params_set_channels(pcm, hwparams,
+ CHANNELS)) < 0)
+ fatal(0, "error from snd_pcm_hw_params_set_channels (%d): %d",
+ CHANNELS, err);
+ if((err = snd_pcm_hw_params(pcm, hwparams)) < 0)
+ fatal(0, "error calling snd_pcm_hw_params: %d", err);
+
+ /* Record the audio supply function */
+ supplyfn = supply;
+
+ /* Create the audio output thread */
+ alsa_bg_shutdown = 0;
+ alsa_bg_enabled = 0;
+ ep(pthread_create(&alsa_bg_collect_tid, 0, alsa_bg_collect, 0));
+ ep(pthread_create(&alsa_bg_play_tid, 0, alsa_bg_play, 0));
+}
+
+void alsa_bg_close(void) {
+ void *r;
+
+ /* Notify background threads that we're shutting down */
+ ep(pthread_mutex_lock(&alsa_bg_lock));
+ alsa_bg_enabled = 0;
+ alsa_bg_shutdown = 1;
+ ep(pthread_cond_signal(&alsa_bg_cond));
+ ep(pthread_mutex_unlock(&alsa_bg_lock));
+ /* Join background threads when they're done */
+ ep(pthread_join(alsa_bg_collect_tid, &r));
+ ep(pthread_join(alsa_bg_play_tid, &r));
+ /* Close audio device */
+ snd_pcm_close(pcm);
+ pcm = 0;
+}
+
+#endif
+
+/*
+Local Variables:
+c-basic-offset:2
+comment-column:40
+fill-column:79
+indent-tabs-mode:nil
+End:
+*/
--- /dev/null
+/*
+ * This file is part of DisOrder.
+ * Copyright (C) 2008 Richard Kettlewell
+ *
+ * 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 program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
+ * USA
+ */
+/** @file alsabg.h
+ * @brief Background-thread interface to ALSA
+ *
+ * This wraps ALSA with an interface which calls back to the client from a
+ * thread. It's not intended for completely general use, just what DisOrder
+ * needs.
+ */
+
+#ifndef ALSABG_H
+#define ALSABG_H
+
+/** @brief Supply audio callback
+ * @param dst Where to write audio data
+ * @param nsamples Number of samples to write
+ *
+ * This function should write up to @p *nsamples samples of data at
+ * @p dst, and return the number of samples written, or -1 if some error
+ * occurred. It will be called in a background thread.
+ */
+typedef int alsa_bg_supply(void *dst,
+ unsigned nsamples);
+
+void alsa_bg_init(const char *device,
+ alsa_bg_supply *supply);
+
+void alsa_bg_enable(void);
+
+void alsa_bg_disable(void);
+
+void alsa_bg_close(void);
+
+#endif /* ALSABG_H */
+
+/*
+Local Variables:
+c-basic-offset:2
+comment-column:40
+fill-column:79
+indent-tabs-mode:nil
+End:
+*/
~mdw / disorder / commitdiff