--- /dev/null
+/*
+ * This file is part of DisOrder.
+ * Copyright (C) 2007 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 clients/playrtp-alsa.c
+ * @brief RTP player - ALSA support
+ */
+
+#include <config.h>
+
+#if API_ALSA
+#include "types.h"
+
+#include <poll.h>
+#include <alsa/asoundlib.h>
+#include <assert.h>
+#include <pthread.h>
+
+#include "mem.h"
+#include "log.h"
+#include "vector.h"
+#include "heap.h"
+#include "playrtp.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 (%d): %d",
+ MAXSAMPLES * samplesize * 3, 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);
+ 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) {
+ /* Do the write */
+ const snd_pcm_sframes_t 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);
+ return -1;
+ default:
+ fatal(0, "error calling snd_pcm_writei: %ld",
+ (long)frames_written);
+ }
+ } else {
+ /* Success */
+ next_timestamp += frames_written * 2;
+ return 0;
+ }
+}
+
+/** @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);
+ } else
+ if((err = snd_pcm_drain(pcm)))
+ fatal(0, "error calling snd_pcm_drain: %d", err);
+ if((err = snd_pcm_nonblock(pcm, 1)))
+ fatal(0, "error calling snd_pcm_nonblock: %d", err);
+ playrtp_alsa_prepared = 0;
+}
+
+void playrtp_alsa(void) {
+ int escape;
+ const struct packet *p;
+
+ playrtp_alsa_init();
+ pthread_mutex_lock(&lock);
+ for(;;) {
+ /* Wait for the buffer to fill up a bit */
+ playrtp_fill_buffer();
+ playrtp_alsa_enable();
+ escape = 0;
+ 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);
+ }
+ 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);
+ }
+}
+
+#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) 2007 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 clients/playrtp-coreaudio.c
+ * @brief RTP player - Core Audio support
+ */
+
+#include <config.h>
+
+#if HAVE_COREAUDIO_AUDIOHARDWARE_H
+#include "types.h"
+
+#include <assert.h>
+#include <pthread.h>
+#include <CoreAudio/AudioHardware.h>
+
+#include "mem.h"
+#include "log.h"
+#include "vector.h"
+#include "heap.h"
+#include "playrtp.h"
+
+/** @brief Callback from Core Audio */
+static OSStatus adioproc
+ (AudioDeviceID attribute((unused)) inDevice,
+ const AudioTimeStamp attribute((unused)) *inNow,
+ const AudioBufferList attribute((unused)) *inInputData,
+ const AudioTimeStamp attribute((unused)) *inInputTime,
+ AudioBufferList *outOutputData,
+ const AudioTimeStamp attribute((unused)) *inOutputTime,
+ void attribute((unused)) *inClientData) {
+ UInt32 nbuffers = outOutputData->mNumberBuffers;
+ AudioBuffer *ab = outOutputData->mBuffers;
+ uint32_t samples_available;
+
+ pthread_mutex_lock(&lock);
+ while(nbuffers > 0) {
+ float *samplesOut = ab->mData;
+ size_t samplesOutLeft = ab->mDataByteSize / sizeof (float);
+
+ while(samplesOutLeft > 0) {
+ 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 *ptr = (void *)(p->samples_raw + offset);
+
+ samples_available = packet_end - next_timestamp;
+ if(samples_available > samplesOutLeft)
+ samples_available = samplesOutLeft;
+ next_timestamp += samples_available;
+ samplesOutLeft -= samples_available;
+ while(samples_available-- > 0)
+ *samplesOut++ = (int16_t)ntohs(*ptr++) * (0.5 / 32767);
+ /* We don't bother junking the packet - that'll be dealt with next time
+ * round */
+ } else {
+ /* No packet is ready to play (and there might be no packet at all) */
+ samples_available = p ? p->timestamp - next_timestamp
+ : samplesOutLeft;
+ if(samples_available > samplesOutLeft)
+ samples_available = samplesOutLeft;
+ //info("infill by %"PRIu32, samples_available);
+ /* Conveniently the buffer is 0 to start with */
+ next_timestamp += samples_available;
+ samplesOut += samples_available;
+ samplesOutLeft -= samples_available;
+ }
+ }
+ ++ab;
+ --nbuffers;
+ }
+ pthread_mutex_unlock(&lock);
+ return 0;
+}
+
+void playrtp_coreaudio(void) {
+ OSStatus status;
+ UInt32 propertySize;
+ AudioDeviceID adid;
+ AudioStreamBasicDescription asbd;
+
+ /* If this looks suspiciously like libao's macosx driver there's an
+ * excellent reason for that... */
+
+ /* TODO report errors as strings not numbers */
+ propertySize = sizeof adid;
+ status = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultOutputDevice,
+ &propertySize, &adid);
+ if(status)
+ fatal(0, "AudioHardwareGetProperty: %d", (int)status);
+ if(adid == kAudioDeviceUnknown)
+ fatal(0, "no output device");
+ propertySize = sizeof asbd;
+ status = AudioDeviceGetProperty(adid, 0, false,
+ kAudioDevicePropertyStreamFormat,
+ &propertySize, &asbd);
+ if(status)
+ fatal(0, "AudioHardwareGetProperty: %d", (int)status);
+ D(("mSampleRate %f", asbd.mSampleRate));
+ D(("mFormatID %08lx", asbd.mFormatID));
+ D(("mFormatFlags %08lx", asbd.mFormatFlags));
+ D(("mBytesPerPacket %08lx", asbd.mBytesPerPacket));
+ D(("mFramesPerPacket %08lx", asbd.mFramesPerPacket));
+ D(("mBytesPerFrame %08lx", asbd.mBytesPerFrame));
+ D(("mChannelsPerFrame %08lx", asbd.mChannelsPerFrame));
+ D(("mBitsPerChannel %08lx", asbd.mBitsPerChannel));
+ D(("mReserved %08lx", asbd.mReserved));
+ if(asbd.mFormatID != kAudioFormatLinearPCM)
+ fatal(0, "audio device does not support kAudioFormatLinearPCM");
+ status = AudioDeviceAddIOProc(adid, adioproc, 0);
+ if(status)
+ fatal(0, "AudioDeviceAddIOProc: %d", (int)status);
+ pthread_mutex_lock(&lock);
+ for(;;) {
+ /* Wait for the buffer to fill up a bit */
+ playrtp_fill_buffer();
+ /* Start playing now */
+ info("Playing...");
+ next_timestamp = pheap_first(&packets)->timestamp;
+ active = 1;
+ status = AudioDeviceStart(adid, adioproc);
+ if(status)
+ fatal(0, "AudioDeviceStart: %d", (int)status);
+ /* 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 */
+ status = AudioDeviceStop(adid, adioproc);
+ if(status)
+ fatal(0, "AudioDeviceStop: %d", (int)status);
+ active = 0;
+ /* Go back round */
+ }
+}
+
+#endif
+
+/*
+Local Variables:
+c-basic-offset:2
+comment-column:40
+fill-column:79
+indent-tabs-mode:nil
+End:
+*/
*
* The main thread is responsible for actually playing audio. In ALSA this
* means it waits until ALSA says it's ready for more audio which it then
- * plays.
+ * plays. See @ref clients/playrtp-alsa.c.
*
* In Core Audio the main thread is only responsible for starting and stopping
* play: the system does the actual playback in its own private thread, and
- * calls adioproc() to fetch the audio data.
+ * calls adioproc() to fetch the audio data. See @ref
+ * clients/playrtp-coreaudio.c.
*
* Sometimes it happens that there is no audio available to play. This may
* because the server went away, or a packet was dropped, or the server
#include <sys/uio.h>
#include <string.h>
#include <assert.h>
+#include <errno.h>
#include "log.h"
#include "mem.h"
#include "timeval.h"
#include "playrtp.h"
-#if HAVE_COREAUDIO_AUDIOHARDWARE_H
-# include <CoreAudio/AudioHardware.h>
-#endif
-#if API_ALSA
-#include <alsa/asoundlib.h>
-#endif
-
#define readahead linux_headers_are_borked
/** @brief RTP socket */
/** @brief Minimum low watermark
*
* We'll stop playing if there's only this many samples in the buffer. */
-static unsigned minbuffer = 2 * 44100 / 10; /* 0.2 seconds */
+unsigned minbuffer = 2 * 44100 / 10; /* 0.2 seconds */
/** @brief Buffer high watermark
*
/** @brief Condition variable signalled whenever @ref packets is changed */
pthread_cond_t cond = PTHREAD_COND_INITIALIZER;
+#if API_ALSA
+# define DEFAULT_BACKEND playrtp_alsa
+#elif HAVE_SYS_SOUNDCARD_H
+# define DEFAULT_BACKEND playrtp_oss
+#elif HAVE_COREAUDIO_AUDIOHARDWARE_H
+# define DEFAULT_BACKEND playrtp_coreaudio
+#else
+# error No known backend
+#endif
+
+/** @brief Backend to play with */
+static void (*backend)(void) = &DEFAULT_BACKEND;
+
HEAP_DEFINE(pheap, struct packet *, lt_packet);
static const struct option options[] = {
{ "buffer", required_argument, 0, 'b' },
{ "rcvbuf", required_argument, 0, 'R' },
{ "multicast", required_argument, 0, 'M' },
+#if HAVE_SYS_SOUNDCARD_H
+ { "oss", no_argument, 0, 'o' },
+#endif
+#if API_ALSA
+ { "alsa", no_argument, 0, 'a' },
+#endif
+#if HAVE_COREAUDIO_AUDIOHARDWARE_H
+ { "core-audio", no_argument, 0, 'c' },
+#endif
{ 0, 0, 0, 0 }
};
if(pheap_count(&packets)) {
struct packet *const p = pheap_remove(&packets);
nsamples -= p->nsamples;
- free_packet(p);
+ playrtp_free_packet(p);
pthread_cond_broadcast(&cond);
}
}
* thread which reads packets off the list and adds them to the heap.
*
* We keep memory allocation (mostly) very fast by keeping pre-allocated
- * packets around; see @ref new_packet().
+ * packets around; see @ref playrtp_new_packet().
*/
static void *listen_thread(void attribute((unused)) *arg) {
struct packet *p = 0;
for(;;) {
if(!p)
- p = new_packet();
+ p = playrtp_new_packet();
iov[0].iov_base = &header;
iov[0].iov_len = sizeof header;
iov[1].iov_base = p->samples_raw;
}
}
-/** @brief Return true if @p p contains @p timestamp
- *
- * Containment implies that a sample @p timestamp exists within the packet.
- */
-static inline int contains(const struct packet *p, uint32_t timestamp) {
- const uint32_t packet_start = p->timestamp;
- const uint32_t packet_end = p->timestamp + p->nsamples;
-
- return (ge(timestamp, packet_start)
- && lt(timestamp, packet_end));
-}
-
/** @brief Wait until the buffer is adequately full
*
* Must be called with @ref lock held.
*/
-static void fill_buffer(void) {
+void playrtp_fill_buffer(void) {
while(nsamples)
drop_first_packet();
info("Buffering...");
*
* Must be called with @ref lock held.
*/
-static struct packet *next_packet(void) {
+struct packet *playrtp_next_packet(void) {
while(pheap_count(&packets)) {
struct packet *const p = pheap_first(&packets);
if(le(p->timestamp + p->nsamples, next_timestamp)) {
return 0;
}
-#if HAVE_COREAUDIO_AUDIOHARDWARE_H
-/** @brief Callback from Core Audio */
-static OSStatus adioproc
- (AudioDeviceID attribute((unused)) inDevice,
- const AudioTimeStamp attribute((unused)) *inNow,
- const AudioBufferList attribute((unused)) *inInputData,
- const AudioTimeStamp attribute((unused)) *inInputTime,
- AudioBufferList *outOutputData,
- const AudioTimeStamp attribute((unused)) *inOutputTime,
- void attribute((unused)) *inClientData) {
- UInt32 nbuffers = outOutputData->mNumberBuffers;
- AudioBuffer *ab = outOutputData->mBuffers;
- uint32_t samples_available;
-
- pthread_mutex_lock(&lock);
- while(nbuffers > 0) {
- float *samplesOut = ab->mData;
- size_t samplesOutLeft = ab->mDataByteSize / sizeof (float);
-
- while(samplesOutLeft > 0) {
- const struct packet *p = 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 *ptr = (void *)(p->samples_raw + offset);
-
- samples_available = packet_end - next_timestamp;
- if(samples_available > samplesOutLeft)
- samples_available = samplesOutLeft;
- next_timestamp += samples_available;
- samplesOutLeft -= samples_available;
- while(samples_available-- > 0)
- *samplesOut++ = (int16_t)ntohs(*ptr++) * (0.5 / 32767);
- /* We don't bother junking the packet - that'll be dealt with next time
- * round */
- } else {
- /* No packet is ready to play (and there might be no packet at all) */
- samples_available = p ? p->timestamp - next_timestamp
- : samplesOutLeft;
- if(samples_available > samplesOutLeft)
- samples_available = samplesOutLeft;
- //info("infill by %"PRIu32, samples_available);
- /* Conveniently the buffer is 0 to start with */
- next_timestamp += samples_available;
- samplesOut += samples_available;
- samplesOutLeft -= samples_available;
- }
- }
- ++ab;
- --nbuffers;
- }
- pthread_mutex_unlock(&lock);
- return 0;
-}
-#endif
-
-
-#if API_ALSA
-/** @brief PCM handle */
-static snd_pcm_t *pcm;
-
-/** @brief True when @ref pcm is up and running */
-static int alsa_prepared = 1;
-
-/** @brief Initialize @ref pcm */
-static void setup_alsa(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 (%d): %d",
- MAXSAMPLES * samplesize * 3, 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);
- 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 alsa_writei(const void *s, size_t n) {
- /* Do the write */
- const snd_pcm_sframes_t 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);
- return -1;
- default:
- fatal(0, "error calling snd_pcm_writei: %ld",
- (long)frames_written);
- }
- } else {
- /* Success */
- next_timestamp += frames_written * 2;
- return 0;
- }
-}
-
-/** @brief Play the relevant part of a packet
- * @param p Packet to play
- * @return 0 on success, -1 on non-fatal error
- */
-static int alsa_play(const struct packet *p) {
- return 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 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 alsa_writei(zeros, samples_available);
-}
-
-/** @brief Reset ALSA state after we lost synchronization */
-static void alsa_reset(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);
- } else
- if((err = snd_pcm_drain(pcm)))
- fatal(0, "error calling snd_pcm_drain: %d", err);
- if((err = snd_pcm_nonblock(pcm, 1)))
- fatal(0, "error calling snd_pcm_nonblock: %d", err);
- alsa_prepared = 0;
-}
-#endif
-
/** @brief Play an RTP stream
*
* This is the guts of the program. It is responsible for:
pthread_create(<id, 0, listen_thread, 0);
/* We have a second thread to add received packets to the queue */
pthread_create(<id, 0, queue_thread, 0);
-#if API_ALSA
- {
- struct packet *p;
- int escape, err;
-
- /* Open the sound device */
- setup_alsa();
- pthread_mutex_lock(&lock);
- for(;;) {
- /* Wait for the buffer to fill up a bit */
- fill_buffer();
- if(!alsa_prepared) {
- if((err = snd_pcm_prepare(pcm)))
- fatal(0, "error calling snd_pcm_prepare: %d", err);
- alsa_prepared = 1;
- }
- escape = 0;
- 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 = next_packet();
- /* Play it or play some silence */
- if(contains(p, next_timestamp))
- escape = alsa_play(p);
- else
- escape = alsa_infill(p);
- }
- active = 0;
- /* We stop playing for a bit until the buffer re-fills */
- pthread_mutex_unlock(&lock);
- alsa_reset(escape);
- pthread_mutex_lock(&lock);
- }
-
- }
-#elif HAVE_COREAUDIO_AUDIOHARDWARE_H
- {
- OSStatus status;
- UInt32 propertySize;
- AudioDeviceID adid;
- AudioStreamBasicDescription asbd;
-
- /* If this looks suspiciously like libao's macosx driver there's an
- * excellent reason for that... */
-
- /* TODO report errors as strings not numbers */
- propertySize = sizeof adid;
- status = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultOutputDevice,
- &propertySize, &adid);
- if(status)
- fatal(0, "AudioHardwareGetProperty: %d", (int)status);
- if(adid == kAudioDeviceUnknown)
- fatal(0, "no output device");
- propertySize = sizeof asbd;
- status = AudioDeviceGetProperty(adid, 0, false,
- kAudioDevicePropertyStreamFormat,
- &propertySize, &asbd);
- if(status)
- fatal(0, "AudioHardwareGetProperty: %d", (int)status);
- D(("mSampleRate %f", asbd.mSampleRate));
- D(("mFormatID %08lx", asbd.mFormatID));
- D(("mFormatFlags %08lx", asbd.mFormatFlags));
- D(("mBytesPerPacket %08lx", asbd.mBytesPerPacket));
- D(("mFramesPerPacket %08lx", asbd.mFramesPerPacket));
- D(("mBytesPerFrame %08lx", asbd.mBytesPerFrame));
- D(("mChannelsPerFrame %08lx", asbd.mChannelsPerFrame));
- D(("mBitsPerChannel %08lx", asbd.mBitsPerChannel));
- D(("mReserved %08lx", asbd.mReserved));
- if(asbd.mFormatID != kAudioFormatLinearPCM)
- fatal(0, "audio device does not support kAudioFormatLinearPCM");
- status = AudioDeviceAddIOProc(adid, adioproc, 0);
- if(status)
- fatal(0, "AudioDeviceAddIOProc: %d", (int)status);
- pthread_mutex_lock(&lock);
- for(;;) {
- /* Wait for the buffer to fill up a bit */
- fill_buffer();
- /* Start playing now */
- info("Playing...");
- next_timestamp = pheap_first(&packets)->timestamp;
- active = 1;
- status = AudioDeviceStart(adid, adioproc);
- if(status)
- fatal(0, "AudioDeviceStart: %d", (int)status);
- /* 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 */
- status = AudioDeviceStop(adid, adioproc);
- if(status)
- fatal(0, "AudioDeviceStop: %d", (int)status);
- active = 0;
- /* Go back round */
- }
- }
-#else
-# error No known audio API
-#endif
+ /* The rest of the work is backend-specific */
+ backend();
}
/* display usage message and terminate */
" --max, -x FRAMES Buffer maximum size\n"
" --rcvbuf, -R BYTES Socket receive buffer size\n"
" --multicast, -M GROUP Join multicast group\n"
+#if API_ALSA
+ " --alsa, -a Use ALSA to play audio\n"
+#endif
+#if HAVE_SYS_SOUNDCARD_H
+ " --oss, -o Use OSS to play audio\n"
+#endif
+#if HAVE_COREAUDIO_AUDIOHARDWARE_H
+ " --core-audio, -c Use Core Audio to play audio\n"
+#endif
" --help, -h Display usage message\n"
" --version, -V Display version number\n"
);
mem_init();
if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale");
- while((n = getopt_long(argc, argv, "hVdD:m:b:x:L:R:M:", options, 0)) >= 0) {
+ while((n = getopt_long(argc, argv, "hVdD:m:b:x:L:R:M:aoc", options, 0)) >= 0) {
switch(n) {
case 'h': help();
case 'V': version();
case 'L': logfp = fopen(optarg, "w"); break;
case 'R': target_rcvbuf = atoi(optarg); break;
case 'M': multicast_group = optarg; break;
+#if API_ALSA
+ case 'a': backend = playrtp_alsa; break;
+#endif
+#if 0
+#if HAVE_SYS_SOUNDCARD_H
+ case 'o': backend = playrtp_oss; break;
+#endif
+#if HAVE_COREAUDIO_AUDIOHARDWARE_H
+ case 'c': backend = playrtp_coreaudio; break;
+#endif
+#endif
default: fatal(0, "invalid option");
}
}
~mdw / disorder / commitdiff