and a placeholder RTP one lurking around too.
disorder-playrtp now uses the uniform audio interface (and is
considerably simplified as a result).
disorderfm_LDADD=$(LIBOBJS) ../lib/libdisorder.a $(LIBGC) $(LIBICONV)
disorderfm_DEPENDENCIES=$(LIBOBJS) ../lib/libdisorder.a
-disorder_playrtp_SOURCES=playrtp.c playrtp.h playrtp-mem.c \
- playrtp-alsa.c playrtp-coreaudio.c playrtp-oss.c
+disorder_playrtp_SOURCES=playrtp.c playrtp.h playrtp-mem.c
disorder_playrtp_LDADD=$(LIBOBJS) ../lib/libdisorder.a \
$(LIBASOUND) $(LIBPCRE) $(LIBICONV) $(LIBGCRYPT) $(COREAUDIO) \
$(LIBDB) $(LIBPTHREAD)
+++ /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 3 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, see <http://www.gnu.org/licenses/>.
- */
-/** @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 "common.h"
-
-#if HAVE_ALSA_ASOUNDLIB_H
-
-#include <poll.h>
-#include <alsa/asoundlib.h>
-#include <pthread.h>
-#include <arpa/inet.h>
-
-#include "mem.h"
-#include "log.h"
-#include "vector.h"
-#include "heap.h"
-#include "playrtp.h"
-#include "alsabg.h"
-
-/** @brief Callback from alsa_bg_collect() */
-static int playrtp_alsa_supply(void *dst,
- unsigned supply_nsamples) {
- unsigned samples_available;
- const struct packet *p;
-
- pthread_mutex_lock(&lock);
- 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 {
- /* 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));
- }
- pthread_mutex_unlock(&lock);
- return samples_available;
-}
-
-void playrtp_alsa(void) {
- 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();
- /* Start playing now */
- info("Playing...");
- 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;
- /* Go back round */
- }
-}
-
-#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 3 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, see <http://www.gnu.org/licenses/>.
- */
-/** @file clients/playrtp-coreaudio.c
- * @brief RTP player - Core Audio support
- */
-
-#include "common.h"
-
-#if HAVE_COREAUDIO_AUDIOHARDWARE_H
-#include <pthread.h>
-
-#include "mem.h"
-#include "log.h"
-#include "vector.h"
-#include "heap.h"
-#include "playrtp.h"
-#include "coreaudio.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;
- if(dump_buffer) {
- size_t n;
-
- for(n = 0; n < samples_available; ++n) {
- dump_buffer[dump_index++] = (int16_t)ntohs(ptr[n]);
- dump_index %= dump_size;
- }
- }
- 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;
- if(dump_buffer) {
- size_t n;
-
- for(n = 0; n < samples_available; ++n) {
- dump_buffer[dump_index++] = 0;
- dump_index %= dump_size;
- }
- }
- }
- }
- ++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 */
- /* Identify the device to use */
- adid = coreaudio_getdevice(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:
-*/
+++ /dev/null
-/*
- * This file is part of DisOrder.
- * Copyright (C) 2007 Richard Kettlewell
- * Portions copyright (C) 2007 Ross Younger
- *
- * 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 3 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, see <http://www.gnu.org/licenses/>.
- */
-/** @file clients/playrtp-oss.c
- * @brief RTP player - OSS and empeg support
- */
-
-#include "common.h"
-
-#if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
-
-#include <poll.h>
-#include <sys/ioctl.h>
-#if !EMPEG_HOST
-#include <sys/soundcard.h>
-#endif
-#include <pthread.h>
-#include <fcntl.h>
-#include <unistd.h>
-#include <errno.h>
-#include <arpa/inet.h>
-
-#include "mem.h"
-#include "log.h"
-#include "vector.h"
-#include "heap.h"
-#include "syscalls.h"
-#include "playrtp.h"
-
-/** @brief /dev/dsp (or whatever) */
-static int playrtp_oss_fd = -1;
-
-/** @brief Audio buffer */
-static char *playrtp_oss_buffer;
-
-/** @brief Size of @ref playrtp_oss_buffer in bytes */
-static int playrtp_oss_bufsize;
-
-/** @brief Number of bytes used in @ref playrtp_oss_buffer */
-static int playrtp_oss_bufused;
-
-/** @brief Open and configure the OSS audio device */
-static void playrtp_oss_enable(void) {
- if(playrtp_oss_fd == -1) {
-#if EMPEG_HOST
- /* empeg audio driver only knows /dev/audio, only supports the equivalent
- * of AFMT_S16_NE, has a fixed buffer size, and does not support the
- * SNDCTL_ ioctls. */
- if(!device)
- device = "/dev/audio";
- if((playrtp_oss_fd = open(device, O_WRONLY)) < 0)
- fatal(errno, "error opening %s", device);
- playrtp_oss_bufsize = 4608;
-#else
- int rate = 44100, stereo = 1, format = AFMT_S16_BE;
- if(!device) {
- if(access("/dev/dsp", W_OK) == 0)
- device = "/dev/dsp";
- else if(access("/dev/audio", W_OK) == 0)
- device = "/dev/audio";
- else
- fatal(0, "cannot determine default audio device");
- }
- if((playrtp_oss_fd = open(device, O_WRONLY)) < 0)
- fatal(errno, "error opening %s", device);
- if(ioctl(playrtp_oss_fd, SNDCTL_DSP_SETFMT, &format) < 0)
- fatal(errno, "ioctl SNDCTL_DSP_SETFMT");
- if(ioctl(playrtp_oss_fd, SNDCTL_DSP_STEREO, &stereo) < 0)
- fatal(errno, "ioctl SNDCTL_DSP_STEREO");
- if(ioctl(playrtp_oss_fd, SNDCTL_DSP_SPEED, &rate) < 0)
- fatal(errno, "ioctl SNDCTL_DSP_SPEED");
- if(rate != 44100)
- error(0, "asking for 44100Hz, got %dHz", rate);
- if(ioctl(playrtp_oss_fd, SNDCTL_DSP_GETBLKSIZE, &playrtp_oss_bufsize) < 0)
- fatal(errno, "ioctl SNDCTL_DSP_GETBLKSIZE");
- info("OSS buffer size %d", playrtp_oss_bufsize);
-#endif
- playrtp_oss_buffer = xmalloc(playrtp_oss_bufsize);
- playrtp_oss_bufused = 0;
- nonblock(playrtp_oss_fd);
- }
-}
-
-/** @brief Flush the OSS output buffer
- * @return 0 on success, non-0 on error
- */
-static int playrtp_oss_flush(void) {
- int nbyteswritten;
-
- if(!playrtp_oss_bufused)
- return 0; /* nothing to do */
- /* 0 out the unused portion of the buffer */
- memset(playrtp_oss_buffer + playrtp_oss_bufused, 0,
- playrtp_oss_bufsize - playrtp_oss_bufused);
-#if EMPEG_HOST
- /* empeg audio driver insists on native-endian samples */
- {
- uint16_t *ptr,
- *const limit = (uint16_t *)(playrtp_oss_buffer + playrtp_oss_bufused);
-
- for(ptr = (uint16_t *)playrtp_oss_buffer; ptr < limit; ++ptr)
- *ptr = ntohs(*ptr);
- }
-#endif
- for(;;) {
- nbyteswritten = write(playrtp_oss_fd,
- playrtp_oss_buffer, playrtp_oss_bufsize);
- if(nbyteswritten < 0) {
- switch(errno) {
- case EINTR:
- break; /* try again */
- case EAGAIN:
- return 0; /* try later */
- default:
- error(errno, "error writing to %s", device);
- return -1;
- }
- } else {
- if(nbyteswritten < playrtp_oss_bufsize)
- error(0, "%s: short write (%d/%d)",
- device, nbyteswritten, playrtp_oss_bufsize);
- if(dump_buffer) {
- int count;
- const int16_t *sp = (const int16_t *)playrtp_oss_buffer;
-
- for(count = 0; count < playrtp_oss_bufsize; count += sizeof(int16_t)) {
- dump_buffer[dump_index++] = (int16_t)ntohs(*sp++);
- dump_index %= dump_size;
- }
- }
- playrtp_oss_bufused = 0;
- return 0;
- }
- }
-}
-
-/** @brief Wait until the audio device can accept more data */
-static void playrtp_oss_wait(void) {
- struct pollfd fds[1];
- int n;
-
- do {
- fds[0].fd = playrtp_oss_fd;
- fds[0].events = POLLOUT;
- while((n = poll(fds, 1, -1)) < 0 && errno == EINTR)
- ;
- if(n < 0)
- fatal(errno, "calling poll");
- } while(!(fds[0].revents & (POLLOUT|POLLERR)));
-}
-
-/** @brief Close the OSS output device
- * @param hard If nonzero, drop pending data
- */
-static void playrtp_oss_disable(int hard) {
- if(hard) {
-#if !EMPEG_HOST
- /* No SNDCTL_DSP_ ioctls on empeg */
- if(ioctl(playrtp_oss_fd, SNDCTL_DSP_RESET, 0) < 0)
- error(errno, "ioctl SNDCTL_DSP_RESET");
-#endif
- } else
- playrtp_oss_flush();
- xclose(playrtp_oss_fd);
- playrtp_oss_fd = -1;
- free(playrtp_oss_buffer);
- playrtp_oss_buffer = 0;
-}
-
-/** @brief Write samples to OSS output device
- * @param data Pointer to sample data
- * @param samples Number of samples
- * @return 0 on success, non-0 on error
- */
-static int playrtp_oss_write(const char *data, size_t samples) {
- long bytes = samples * sizeof(int16_t);
- while(bytes > 0) {
- int n = playrtp_oss_bufsize - playrtp_oss_bufused;
-
- if(n > bytes)
- n = bytes;
- memcpy(playrtp_oss_buffer + playrtp_oss_bufused, data, n);
- bytes -= n;
- data += n;
- playrtp_oss_bufused += n;
- if(playrtp_oss_bufused == playrtp_oss_bufsize)
- if(playrtp_oss_flush())
- return -1;
- }
- next_timestamp += samples;
- return 0;
-}
-
-/** @brief Play some data from packet @p p
- *
- * @p p is assumed to contain @ref next_timestamp.
- */
-static int playrtp_oss_play(const struct packet *p) {
- return playrtp_oss_write
- ((const char *)(p->samples_raw + next_timestamp - p->timestamp),
- (p->timestamp + p->nsamples) - next_timestamp);
-}
-
-/** @brief Play some silence before packet @p p
- *
- * @p p is assumed to be entirely before @ref next_timestamp.
- */
-static int playrtp_oss_infill(const struct packet *p) {
- static const char zeros[INFILL_SAMPLES * sizeof(int16_t)];
- size_t samples_available = INFILL_SAMPLES;
-
- if(p && samples_available > p->timestamp - next_timestamp)
- samples_available = p->timestamp - next_timestamp;
- return playrtp_oss_write(zeros, samples_available);
-}
-
-/** @brief OSS backend for playrtp */
-void playrtp_oss(void) {
- int escape;
- const struct packet *p;
-
- pthread_mutex_lock(&lock);
- for(;;) {
- /* Wait for the buffer to fill up a bit */
- playrtp_fill_buffer();
- playrtp_oss_enable();
- escape = 0;
- info("Playing...");
- /* Keep playing until the buffer empties out, we get an error */
- while((nsamples >= minbuffer
- || (nsamples > 0
- && contains(pheap_first(&packets), next_timestamp)))
- && !escape) {
- /* Wait until we can play more */
- pthread_mutex_unlock(&lock);
- playrtp_oss_wait();
- pthread_mutex_lock(&lock);
- /* Device 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_oss_play(p);
- else
- escape = playrtp_oss_infill(p);
- }
- active = 0;
- /* We stop playing for a bit until the buffer re-fills */
- pthread_mutex_unlock(&lock);
- playrtp_oss_disable(escape);
- pthread_mutex_lock(&lock);
- }
-}
-
-#endif
-
-/*
-Local Variables:
-c-basic-offset:2
-comment-column:40
-fill-column:79
-indent-tabs-mode:nil
-End:
-*/
#include "playrtp.h"
#include "inputline.h"
#include "version.h"
+#include "uaudio.h"
#define readahead linux_headers_are_borked
static FILE *logfp;
/** @brief Output device */
-const char *device;
/** @brief Minimum low watermark
*
/** @brief Condition variable signalled whenever @ref packets is changed */
pthread_cond_t cond = PTHREAD_COND_INITIALIZER;
-#if DEFAULT_BACKEND == BACKEND_ALSA
-# define DEFAULT_PLAYRTP_BACKEND playrtp_alsa
-#elif DEFAULT_BACKEND == BACKEND_OSS
-# define DEFAULT_PLAYRTP_BACKEND playrtp_oss
-#elif DEFAULT_BACKEND == BACKEND_COREAUDIO
-# define DEFAULT_PLAYRTP_BACKEND playrtp_coreaudio
-#endif
-
/** @brief Backend to play with */
-static void (*backend)(void) = DEFAULT_PLAYRTP_BACKEND;
+static const struct uaudio *backend;
HEAP_DEFINE(pheap, struct packet *, lt_packet);
return 0;
}
-/** @brief Play an RTP stream
- *
- * This is the guts of the program. It is responsible for:
- * - starting the listening thread
- * - opening the audio device
- * - reading ahead to build up a buffer
- * - arranging for audio to be played
- * - detecting when the buffer has got too small and re-buffering
- */
-static void play_rtp(void) {
- pthread_t ltid;
- int err;
-
- /* We receive and convert audio data in a background thread */
- if((err = pthread_create(<id, 0, listen_thread, 0)))
- fatal(err, "pthread_create listen_thread");
- /* We have a second thread to add received packets to the queue */
- if((err = pthread_create(<id, 0, queue_thread, 0)))
- fatal(err, "pthread_create queue_thread");
- /* The rest of the work is backend-specific */
- backend();
-}
-
/* display usage message and terminate */
static void help(void) {
xprintf("Usage:\n"
exit(0);
}
+static size_t playrtp_callback(int16_t *buffer,
+ size_t max_samples,
+ void attribute((unused)) *userdata) {
+ size_t samples;
+
+ pthread_mutex_lock(&lock);
+ /* Get the next packet, junking any that are now in the past */
+ const struct packet *p = playrtp_next_packet();
+ if(p && contains(p, next_timestamp)) {
+ /* This packet is ready to play; the desired next timestamp points
+ * somewhere into it. */
+
+ /* Timestamp of end of packet */
+ const uint32_t packet_end = p->timestamp + p->nsamples;
+
+ /* Offset of desired next timestamp into current packet */
+ const uint32_t offset = next_timestamp - p->timestamp;
+
+ /* Pointer to audio data */
+ const uint16_t *ptr = (void *)(p->samples_raw + offset);
+
+ /* Compute number of samples left in packet, limited to output buffer
+ * size */
+ samples = packet_end - next_timestamp;
+ if(samples > max_samples)
+ samples = max_samples;
+
+ /* Copy into buffer, converting to native endianness */
+ size_t i = samples;
+ int16_t *bufptr = buffer;
+ while(i > 0) {
+ *bufptr++ = (int16_t)ntohs(*ptr++);
+ --i;
+ }
+ /* We don't junk the packet here; a subsequent call to
+ * playrtp_next_packet() will dispose of it (if it's actually done with). */
+ } else {
+ /* There is no suitable packet. We introduce 0s up to the next packet, or
+ * to fill the buffer if there's no next packet or that's too many. The
+ * comparison with max_samples deals with the otherwise troubling overflow
+ * case. */
+ samples = p ? p->timestamp - next_timestamp : max_samples;
+ if(samples > max_samples)
+ samples = max_samples;
+ //info("infill by %zu", samples);
+ memset(buffer, 0, samples * sizeof *buffer);
+ }
+ /* Debug dump */
+ if(dump_buffer) {
+ for(size_t i = 0; i < samples; ++i) {
+ dump_buffer[dump_index++] = buffer[i];
+ dump_index %= dump_size;
+ }
+ }
+ /* Advance timestamp */
+ next_timestamp += samples;
+ pthread_mutex_unlock(&lock);
+ return samples;
+}
+
int main(int argc, char **argv) {
int n, err;
struct addrinfo *res;
};
union any_sockaddr mgroup;
const char *dumpfile = 0;
+ const char *device = 0;
+ pthread_t ltid;
static const struct addrinfo prefs = {
.ai_flags = AI_PASSIVE,
mem_init();
if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale");
+ backend = uaudio_apis[0];
while((n = getopt_long(argc, argv, "hVdD:m:b:x:L:R:M:aocC:r", options, 0)) >= 0) {
switch(n) {
case 'h': help();
case 'L': logfp = fopen(optarg, "w"); break;
case 'R': target_rcvbuf = atoi(optarg); break;
#if HAVE_ALSA_ASOUNDLIB_H
- case 'a': backend = playrtp_alsa; break;
+ case 'a': backend = &uaudio_alsa; break;
#endif
#if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
- case 'o': backend = playrtp_oss; break;
+ case 'o': backend = &uaudio_oss; break;
#endif
#if HAVE_COREAUDIO_AUDIOHARDWARE_H
- case 'c': backend = playrtp_coreaudio; break;
+ case 'c': backend = &uaudio_coreaudio; break;
#endif
case 'C': configfile = optarg; break;
case 's': control_socket = optarg; break;
fatal(errno, "mapping %s", dumpfile);
info("dumping to %s", dumpfile);
}
- play_rtp();
+ /* Choose output device */
+ if(device)
+ uaudio_set("device", device);
+ /* Set up output */
+ backend->start(playrtp_callback, NULL);
+ /* We receive and convert audio data in a background thread */
+ if((err = pthread_create(<id, 0, listen_thread, 0)))
+ fatal(err, "pthread_create listen_thread");
+ /* We have a second thread to add received packets to the queue */
+ if((err = pthread_create(<id, 0, queue_thread, 0)))
+ fatal(err, "pthread_create queue_thread");
+ 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;
+ backend->activate();
+ /* 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 */
+ backend->deactivate();
+ active = 0;
+ /* Go back round */
+ }
return 0;
}
void playrtp_fill_buffer(void);
struct packet *playrtp_next_packet(void);
-extern const char *device;
extern struct packet *received_packets;
extern struct packet **received_tail;
extern pthread_mutex_t receive_lock;
#
# This file is part of DisOrder.
-# Copyright (C) 2004-2008 Richard Kettlewell
+# Copyright (C) 2004-2009 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
$(TRACKDB) trackdb.h trackdb-int.h \
trackname.c trackorder.c trackname.h \
tracksort.c \
+ uaudio.c uaudio.h \
+ uaudio-oss.c uaudio-coreaudio.c uaudio-rtp.c \
url.h url.c \
user.h user.c \
unicode.h unicode.c \
--- /dev/null
+/*
+ * This file is part of DisOrder.
+ * Copyright (C) 2009 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 3 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, see <http://www.gnu.org/licenses/>.
+ */
+/** @file lib/uaudio-coreaudio.c
+ * @brief Support for Core Audio backend */
+#include "common.h"
+
+#if HAVE_COREAUDIO_AUDIOHARDWARE_H
+
+#include "coreaudio.h"
+#include "uaudio.h"
+#include "mem.h"
+#include "log.h"
+#include "syscalls.h"
+
+/** @brief Callback to request sample data */
+static uaudio_callback *coreaudio_callback;
+
+/** @brief Userdata for @ref coreaudio_callback */
+static void *coreaudio_userdata;
+
+/** @brief Core Audio device ID */
+static AudioDeviceID coreaudio_adid;
+
+/** @brief Core Audio option names */
+static const char *const coreaudio_options[] = {
+ "device",
+ NULL
+};
+
+/** @brief Callback from Core Audio
+ *
+ * Core Audio demands floating point samples but we provide integers.
+ * So there is a conversion step in here.
+ */
+static OSStatus coreaudio_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) {
+ /* Number of buffers we must fill */
+ unsigned nbuffers = outOutputData->mNumberBuffers;
+ /* Pointer to buffer to fill */
+ AudioBuffer *ab = outOutputData->mBuffers;
+
+ while(nbuffers > 0) {
+ /* Where to store converted sample data */
+ float *samples = ab->mData;
+ /* Number of samples left to fill */
+ size_t nsamples = ab->mDataByteSize / sizeof (float);
+
+ while(nsamples > 0) {
+ /* Integer-format input buffer */
+ int16_t input[1024], *ptr = input;
+ /* How many samples we'll ask for */
+ const int ask = nsamples > 1024 ? 1024 : (int)nsamples;
+ /* How many we get */
+ int got;
+
+ got = coreaudio_callback(input, ask, coreaudio_userdata);
+ /* Convert the samples and store in the output buffer */
+ nsamples -= got;
+ while(got > 0) {
+ --got;
+ *samples++ = *ptr++ * (0.5 / 32767);
+ }
+ }
+ /* Move on to the next buffer */
+ ++ab;
+ --nbuffers;
+ }
+ return 0;
+}
+
+static void coreaudio_start(uaudio_callback *callback,
+ void *userdata) {
+ OSStatus status;
+ UInt32 propertySize;
+ AudioStreamBasicDescription asbd;
+ const char *device;
+
+ coreaudio_callback = callback;
+ coreaudio_userdata = userdata;
+ device = uaudio_get("device");
+ coreaudio_adid = coreaudio_getdevice(device);
+ propertySize = sizeof asbd;
+ status = AudioDeviceGetProperty(coreaudio_adid, 0, false,
+ kAudioDevicePropertyStreamFormat,
+ &propertySize, &asbd);
+ if(status)
+ coreaudio_fatal(status, "AudioHardwareGetProperty");
+ 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)
+ disorder_fatal(0, "audio device does not support kAudioFormatLinearPCM");
+ status = AudioDeviceAddIOProc(coreaudio_adid, coreaudio_adioproc, 0);
+ if(status)
+ coreaudio_fatal(status, "AudioDeviceAddIOProc");
+}
+
+static void coreaudio_stop(void) {
+}
+
+static void coreaudio_activate(void) {
+ OSStatus status;
+
+ status = AudioDeviceStart(coreaudio_adid, coreaudio_adioproc);
+ if(status)
+ coreaudio_fatal(status, "AudioDeviceStart");
+}
+
+static void coreaudio_deactivate(void) {
+ OSStatus status;
+
+ status = AudioDeviceStop(coreaudio_adid, coreaudio_adioproc);
+ if(status)
+ coreaudio_fatal(status, "AudioDeviceStop");
+}
+
+const struct uaudio uaudio_coreaudio = {
+ .name = "coreaudio",
+ .options = coreaudio_options,
+ .start = coreaudio_start,
+ .stop = coreaudio_stop,
+ .activate = coreaudio_activate,
+ .deactivate = coreaudio_deactivate
+};
+
+#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) 2009 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 3 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, see <http://www.gnu.org/licenses/>.
+ */
+/** @file lib/uaudio-oss.c
+ * @brief Support for OSS backend */
+#include "common.h"
+
+#if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
+
+#if HAVE_SYS_SOUNDCARD_H
+# include <sys/soundcard.h>
+#endif
+#include <sys/ioctl.h>
+#include <pthread.h>
+#include <fcntl.h>
+#include <unistd.h>
+#include <errno.h>
+#include <arpa/inet.h>
+
+#include "mem.h"
+#include "log.h"
+#include "syscalls.h"
+
+static int oss_fd = -1;
+static pthread_t oss_thread;
+static uaudio_callback *oss_callback;
+static int oss_started;
+static int oss_activated;
+static int oss_going;
+static pthread_mutex_t oss_lock = PTHREAD_MUTEX_INITIALIZER;
+static pthread_cond_t oss_cond = PTHREAD_COND_INITIALIZER;
+static int oss_bufsize;
+
+static const char *const oss_options[] = {
+ "device",
+ NULL
+};
+
+static void *oss_thread(void *userdata) {
+ int16_t *buffer;
+ pthread_mutex_lock(&oss_lock);
+ buffer = xmalloc(XXX);
+ while(oss_started) {
+ while(oss_started && !oss_activated)
+ pthread_cond_wait(&oss_cond, &oss_lock);
+ if(!oss_started)
+ break;
+ /* We are definitely active now */
+ oss_going = 1;
+ pthread_cond_signal(&oss_cond);
+ while(oss_activated) {
+ const int nsamples = oss_callback(buffer, oss_bufsizeXXX, userdata);;
+ if(write(oss_fd, buffer, XXX) < 0)
+ fatal(errno, "error playing audio");
+ // TODO short writes!
+ }
+ oss_going = 0;
+ pthread_cond_signal(&oss_cond);
+ }
+ pthread_mutex_unlock(&oss_lock);
+ return NULL;
+}
+
+static void oss_start(uaudio_callback *callback,
+ void *userdata) {
+ int e;
+ oss_callback = callback;
+ if((e = pthread_create(&oss_thread,
+ NULL,
+ oss_thread_fn,
+ userdata)))
+ fatal(e, "pthread_create");
+}
+
+static void oss_stop(void) {
+ void *result;
+
+ oss_deactivate();
+ pthread_mutex_lock(&oss_lock);
+ oss_started = 0;
+ pthread_cond_signal(&oss_cond);
+ pthread_mutex_unlock(&oss_lock);
+ pthread_join(oss_thread, &result);
+}
+
+static void oss_activate(void) {
+ pthread_mutex_lock(&oss_lock);
+ if(!oss_activated) {
+ const char *device = uaudio_get(&uadiuo_oss, "device");
+
+#if EMPEG_HOST
+ if(!device || !*device || !strcmp(device, "default")) {
+ device "/dev/audio";
+#else
+ if(!device || !*device || !strcmp(device, "default")) {
+ if(access("/dev/dsp", W_OK) == 0)
+ device = "/dev/dsp";
+ else
+ device = "/dev/audio";
+ }
+#endif
+ if((oss_fd = open(device, O_WRONLY, 0)) < 0)
+ fatal(errno, "error opening %s", device);
+#if EMPEG_HOST
+ /* empeg audio driver only knows /dev/audio, only supports the equivalent
+ * of AFMT_S16_NE, has a fixed buffer size, and does not support the
+ * SNDCTL_ ioctls. */
+ oss_bufsize = 4608;
+#else
+ int stereo = (config->sample_format.channels == 2), format, rate;
+ if(ioctl(ossfd, SNDCTL_DSP_STEREO, &stereo) < 0) {
+ error(errno, "error calling ioctl SNDCTL_DSP_STEREO %d", stereo);
+ goto failed;
+ }
+ /* TODO we need to think about where we decide this */
+ if(config->sample_format.bits == 8)
+ format = AFMT_U8;
+ else if(config->sample_format.bits == 16)
+ format = (config->sample_format.endian == ENDIAN_LITTLE
+ ? AFMT_S16_LE : AFMT_S16_BE);
+ else
+ fatal(0, "unsupported sample_format for oss backend");
+ if(ioctl(oss_fd, SNDCTL_DSP_SETFMT, &format) < 0)
+ fatal(errno, "error calling ioctl SNDCTL_DSP_SETFMT %#x", format);
+ rate = config->sample_format.rate;
+ if(ioctl(oss_fd, SNDCTL_DSP_SPEED, &rate) < 0)
+ fatal(errno, "error calling ioctl SNDCTL_DSP_SPEED %d", rate);
+ if((unsigned)rate != config->sample_format.rate)
+ error(0, "asked for %luHz, got %dHz",
+ (unsigned long)config->sample_format.rate, rate);
+ if(ioctl(oss_fd, SNDCTL_DSP_GETBLKSIZE, &oss_bufsize) < 0) {
+ error(errno, "ioctl SNDCTL_DSP_GETBLKSIZE");
+ oss_bufsize = 2048; /* guess */
+ }
+#endif
+ oss_activated = 1;
+ pthread_cond_signal(&oss_cond);
+ while(!oss_going)
+ pthread_cond_wait(&oss_cond, &oss_lock);
+ }
+ pthread_mutex_unlock(&oss_lock);
+}
+
+static void oss_deactivate(void) {
+ pthread_mutex_lock(&oss_lock);
+ if(oss_activated) {
+ oss_activated = 0;
+ pthread_cond_signal(&oss_cond);
+ while(oss_going)
+ pthread_cond_wait(&oss_cond, &oss_lock);
+ close(oss_fd);
+ oss_fd = -1;
+ }
+ pthread_mutex_unlock(&oss_lock);
+}
+
+const struct uaudio uaudio_oss = {
+ .name = "oss",
+ .options = oss_options,
+ .start = oss_start,
+ .stop = oss_stop,
+ .activate = oss_activate,
+ .deactivate = oss_deactivate
+};
+
+#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) 2009 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 3 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, see <http://www.gnu.org/licenses/>.
+ */
+/** @file lib/uaudio-oss.c
+ * @brief Support for RTP network play backend */
+#include "common.h"
+
+#include <pthread.h>
+
+#include "uaudio.h"
+#include "mem.h"
+#include "log.h"
+#include "syscalls.h"
+
+static const char *const rtp_options[] = {
+ NULL
+};
+
+static void rtp_start(uaudio_callback *callback,
+ void *userdata) {
+ (void)callback;
+ (void)userdata;
+ /* TODO */
+}
+
+static void rtp_stop(void) {
+ /* TODO */
+}
+
+static void rtp_activate(void) {
+ /* TODO */
+}
+
+static void rtp_deactivate(void) {
+ /* TODO */
+}
+
+const struct uaudio uaudio_rtp = {
+ .name = "rtp",
+ .options = rtp_options,
+ .start = rtp_start,
+ .stop = rtp_stop,
+ .activate = rtp_activate,
+ .deactivate = rtp_deactivate
+};
+
+/*
+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) 2009 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 3 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, see <http://www.gnu.org/licenses/>.
+ */
+
+/** @file lib/uaudio.c
+ * @brief Uniform audio interface
+ */
+
+#include "common.h"
+#include "uaudio.h"
+#include "hash.h"
+#include "mem.h"
+
+/** @brief Options for chosen uaudio API */
+static hash *uaudio_options;
+
+/** @brief Set a uaudio option */
+void uaudio_set(const char *name, const char *value) {
+ if(!uaudio_options)
+ uaudio_options = hash_new(sizeof(char *));
+ value = xstrdup(value);
+ hash_add(uaudio_options, name, &value, HASH_INSERT_OR_REPLACE);
+}
+
+/** @brief Set a uaudio option */
+const char *uaudio_get(const char *name) {
+ const char *value = (uaudio_options ?
+ *(char **)hash_find(uaudio_options, name)
+ : NULL);
+ return value ? xstrdup(value) : NULL;
+}
+
+/** @brief List of known APIs
+ *
+ * Terminated by a null pointer.
+ *
+ * The first one will be used as a default, so putting ALSA before OSS
+ * constitutes a policy decision.
+ */
+const struct uaudio *uaudio_apis[] = {
+#if HAVE_COREAUDIO_AUDIOHARDWARE_H
+ &uaudio_coreaudio,
+#endif
+#if HAVE_ALSA_ASOUNDLIB_H
+ &uaudio_alsa,
+#endif
+#if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
+ &uaudio_oss,
+#endif
+ &uaudio_rtp,
+ NULL,
+};
+
+/*
+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) 2009 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 3 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, see <http://www.gnu.org/licenses/>.
+ */
+
+/** @file lib/uaudio.h
+ * @brief Uniform audio interface
+ */
+
+#ifndef UAUDIO_H
+#define UAUDIO_H
+
+/** @brief Callback to get audio data
+ * @param buffer Where to put audio data
+ * @param max_samples How many samples to supply
+ * @param userdata As passed to uaudio_open()
+ * @return Number of samples filled
+ *
+ * One sample is a single 16-bit signed value.
+ */
+typedef size_t uaudio_callback(int16_t *buffer,
+ size_t max_samples,
+ void *userdata);
+
+/** @brief Audio API definition */
+struct uaudio {
+ /** @brief Name of this API */
+ const char *name;
+
+ /** @brief List of options, terminated by NULL */
+ const char *const *options;
+
+ /** @brief Do slow setup
+ * @param ua Handle returned by uaudio_open()
+ * @param callback Called for audio data
+ * @param userdata Passed to @p callback
+ *
+ * This does resource-intensive setup for the output device.
+ *
+ * For instance it might open mixable audio devices or network sockets. It
+ * will create any background thread required. However, it must not exclude
+ * other processes from outputting sound.
+ */
+ void (*start)(uaudio_callback *callback,
+ void *userdata);
+
+ /** @brief Tear down
+ * @param ua Handle returned by uaudio_open()
+ *
+ * This undoes the effect of @c start.
+ */
+ void (*stop)(void);
+
+ /** @brief Enable output
+ *
+ * A background thread will start calling @c callback as set by @c
+ * start and playing the audio data received from it.
+ */
+ void (*activate)(void);
+
+ /** @brief Disable output
+ *
+ * The background thread will stop calling @c callback.
+ */
+ void (*deactivate)(void);
+
+};
+
+void uaudio_set(const char *name, const char *value);
+const char *uaudio_get(const char *name);
+
+#if HAVE_COREAUDIO_AUDIOHARDWARE_H
+extern const struct uaudio uaudio_coreaudio;
+#endif
+
+#if HAVE_ALSA_ASOUNDLIB_H
+extern const struct uaudio uaudio_alsa;
+#endif
+
+#if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
+extern const struct uaudio uaudio_oss;
+#endif
+
+extern const struct uaudio uaudio_rtp;
+
+extern const struct uaudio *uaudio_apis[];
+
+#endif /* UAUDIO_H */
+
+/*
+Local Variables:
+c-basic-offset:2
+comment-column:40
+fill-column:79
+indent-tabs-mode:nil
+End:
+*/