X-Git-Url: https://www.chiark.greenend.org.uk/ucgi/~mdw/git/disorder/blobdiff_plain/8e3fe3d8ea9f5e75b6f1be4e4ae8cf2c16db88e3..1f3d2413174c3187de3fd8701d3ba26a0d1e4489:/clients/playrtp.c
diff --git a/clients/playrtp.c b/clients/playrtp.c
index 86a248c..aca1532 100644
--- a/clients/playrtp.c
+++ b/clients/playrtp.c
@@ -1,21 +1,19 @@
/*
* This file is part of DisOrder.
- * Copyright (C) 2007 Richard Kettlewell
+ * Copyright (C) 2007, 2008 Richard Kettlewell
*
- * This program is free software; you can redistribute it and/or modify
+ * 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
+ * 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.
- *
+ * 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
+ * along with this program. If not, see .
*/
/** @file clients/playrtp.c
* @brief RTP player
@@ -34,11 +32,12 @@
*
* 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
@@ -48,12 +47,9 @@
* - it is safe to read uint32_t values without a lock protecting them
*/
-#include
-#include "types.h"
+#include "common.h"
#include
-#include
-#include
#include
#include
#include
@@ -61,8 +57,13 @@
#include
#include
#include
-#include
-#include
+#include
+#include
+#include
+#include
+#include
+#include
+#include
#include "log.h"
#include "mem.h"
@@ -74,17 +75,18 @@
#include "vector.h"
#include "heap.h"
#include "timeval.h"
+#include "client.h"
#include "playrtp.h"
-
-#if HAVE_COREAUDIO_AUDIOHARDWARE_H
-# include
-#endif
-#if API_ALSA
-#include
-#endif
+#include "inputline.h"
+#include "version.h"
#define readahead linux_headers_are_borked
+/** @brief Obsolete synonym */
+#ifndef IPV6_JOIN_GROUP
+# define IPV6_JOIN_GROUP IPV6_ADD_MEMBERSHIP
+#endif
+
/** @brief RTP socket */
static int rtpfd;
@@ -97,7 +99,7 @@ const char *device;
/** @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
*
@@ -166,8 +168,43 @@ pthread_mutex_t lock = PTHREAD_MUTEX_INITIALIZER;
/** @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;
+
HEAP_DEFINE(pheap, struct packet *, lt_packet);
+/** @brief Control socket or NULL */
+const char *control_socket;
+
+/** @brief Buffer for debugging dump
+ *
+ * The debug dump is enabled by the @c --dump option. It records the last 20s
+ * of audio to the specified file (which will be about 3.5Mbytes). The file is
+ * written as as ring buffer, so the start point will progress through it.
+ *
+ * Use clients/dump2wav to convert this to a WAV file, which can then be loaded
+ * into (e.g.) Audacity for further inspection.
+ *
+ * All three backends (ALSA, OSS, Core Audio) now support this option.
+ *
+ * The idea is to allow the user a few seconds to react to an audible artefact.
+ */
+int16_t *dump_buffer;
+
+/** @brief Current index within debugging dump */
+size_t dump_index;
+
+/** @brief Size of debugging dump in samples */
+size_t dump_size = 44100/*Hz*/ * 2/*channels*/ * 20/*seconds*/;
+
static const struct option options[] = {
{ "help", no_argument, 0, 'h' },
{ "version", no_argument, 0, 'V' },
@@ -177,10 +214,87 @@ static const struct option options[] = {
{ "max", required_argument, 0, 'x' },
{ "buffer", required_argument, 0, 'b' },
{ "rcvbuf", required_argument, 0, 'R' },
- { "multicast", required_argument, 0, 'M' },
+#if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
+ { "oss", no_argument, 0, 'o' },
+#endif
+#if HAVE_ALSA_ASOUNDLIB_H
+ { "alsa", no_argument, 0, 'a' },
+#endif
+#if HAVE_COREAUDIO_AUDIOHARDWARE_H
+ { "core-audio", no_argument, 0, 'c' },
+#endif
+ { "dump", required_argument, 0, 'r' },
+ { "socket", required_argument, 0, 's' },
+ { "config", required_argument, 0, 'C' },
{ 0, 0, 0, 0 }
};
+/** @brief Control thread
+ *
+ * This thread is responsible for accepting control commands from Disobedience
+ * (or other controllers) over an AF_UNIX stream socket with a path specified
+ * by the @c --socket option. The protocol uses simple string commands and
+ * replies:
+ *
+ * - @c stop will shut the player down
+ * - @c query will send back the reply @c running
+ * - anything else is ignored
+ *
+ * Commands and response strings terminated by shutting down the connection or
+ * by a newline. No attempt is made to multiplex multiple clients so it is
+ * important that the command be sent as soon as the connection is made - it is
+ * assumed that both parties to the protocol are entirely cooperating with one
+ * another.
+ */
+static void *control_thread(void attribute((unused)) *arg) {
+ struct sockaddr_un sa;
+ int sfd, cfd;
+ char *line;
+ socklen_t salen;
+ FILE *fp;
+
+ assert(control_socket);
+ unlink(control_socket);
+ memset(&sa, 0, sizeof sa);
+ sa.sun_family = AF_UNIX;
+ strcpy(sa.sun_path, control_socket);
+ sfd = xsocket(PF_UNIX, SOCK_STREAM, 0);
+ if(bind(sfd, (const struct sockaddr *)&sa, sizeof sa) < 0)
+ fatal(errno, "error binding to %s", control_socket);
+ if(listen(sfd, 128) < 0)
+ fatal(errno, "error calling listen on %s", control_socket);
+ info("listening on %s", control_socket);
+ for(;;) {
+ salen = sizeof sa;
+ cfd = accept(sfd, (struct sockaddr *)&sa, &salen);
+ if(cfd < 0) {
+ switch(errno) {
+ case EINTR:
+ case EAGAIN:
+ break;
+ default:
+ fatal(errno, "error calling accept on %s", control_socket);
+ }
+ }
+ if(!(fp = fdopen(cfd, "r+"))) {
+ error(errno, "error calling fdopen for %s connection", control_socket);
+ close(cfd);
+ continue;
+ }
+ if(!inputline(control_socket, fp, &line, '\n')) {
+ if(!strcmp(line, "stop")) {
+ info("stopped via %s", control_socket);
+ exit(0); /* terminate immediately */
+ }
+ if(!strcmp(line, "query"))
+ fprintf(fp, "running");
+ xfree(line);
+ }
+ if(fclose(fp) < 0)
+ error(errno, "error closing %s connection", control_socket);
+ }
+}
+
/** @brief Drop the first packet
*
* Assumes that @ref lock is held.
@@ -189,7 +303,7 @@ static void drop_first_packet(void) {
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);
}
}
@@ -207,8 +321,9 @@ static void *queue_thread(void attribute((unused)) *arg) {
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)
@@ -243,7 +358,7 @@ static void *queue_thread(void attribute((unused)) *arg) {
* 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;
@@ -255,7 +370,7 @@ static void *listen_thread(void attribute((unused)) *arg) {
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;
@@ -282,6 +397,9 @@ static void *listen_thread(void attribute((unused)) *arg) {
timestamp, next_timestamp);
continue;
}
+ /* Ignore packets with the extension bit set. */
+ if(header.vpxcc & 0x10)
+ continue;
p->next = 0;
p->flags = 0;
p->timestamp = timestamp;
@@ -306,8 +424,9 @@ static void *listen_thread(void attribute((unused)) *arg) {
* 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 */
@@ -322,28 +441,17 @@ static void *listen_thread(void attribute((unused)) *arg) {
}
}
-/** @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...");
- while(nsamples < readahead)
+ while(nsamples < readahead) {
pthread_cond_wait(&cond, &lock);
+ }
next_timestamp = pheap_first(&packets)->timestamp;
active = 1;
}
@@ -357,7 +465,7 @@ static void fill_buffer(void) {
*
* 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)) {
@@ -371,214 +479,6 @@ static struct packet *next_packet(void) {
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:
@@ -590,120 +490,16 @@ static void alsa_reset(int hard_reset) {
*/
static void play_rtp(void) {
pthread_t ltid;
+ int err;
/* We receive and convert audio data in a background thread */
- pthread_create(<id, 0, listen_thread, 0);
+ 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 */
- 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
+ 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 */
@@ -716,7 +512,16 @@ static void help(void) {
" --buffer, -b FRAMES Buffer high water mark\n"
" --max, -x FRAMES Buffer maximum size\n"
" --rcvbuf, -R BYTES Socket receive buffer size\n"
- " --multicast, -M GROUP Join multicast group\n"
+ " --config, -C PATH Set configuration file\n"
+#if HAVE_ALSA_ASOUNDLIB_H
+ " --alsa, -a Use ALSA to play audio\n"
+#endif
+#if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
+ " --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"
);
@@ -724,41 +529,39 @@ static void help(void) {
exit(0);
}
-/* display version number and terminate */
-static void version(void) {
- xprintf("disorder-playrtp version %s\n", disorder_version_string);
- xfclose(stdout);
- exit(0);
-}
-
int main(int argc, char **argv) {
- int n;
+ int n, err;
struct addrinfo *res;
struct stringlist sl;
char *sockname;
int rcvbuf, target_rcvbuf = 131072;
socklen_t len;
- char *multicast_group = 0;
struct ip_mreq mreq;
struct ipv6_mreq mreq6;
+ disorder_client *c;
+ char *address, *port;
+ int is_multicast;
+ union any_sockaddr {
+ struct sockaddr sa;
+ struct sockaddr_in in;
+ struct sockaddr_in6 in6;
+ };
+ union any_sockaddr mgroup;
+ const char *dumpfile = 0;
static const struct addrinfo prefs = {
- AI_PASSIVE,
- PF_INET,
- SOCK_DGRAM,
- IPPROTO_UDP,
- 0,
- 0,
- 0,
- 0
+ .ai_flags = AI_PASSIVE,
+ .ai_family = PF_INET,
+ .ai_socktype = SOCK_DGRAM,
+ .ai_protocol = IPPROTO_UDP
};
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:aocC:r", options, 0)) >= 0) {
switch(n) {
case 'h': help();
- case 'V': version();
+ case 'V': version("disorder-playrtp");
case 'd': debugging = 1; break;
case 'D': device = optarg; break;
case 'm': minbuffer = 2 * atol(optarg); break;
@@ -766,40 +569,92 @@ int main(int argc, char **argv) {
case 'x': maxbuffer = 2 * atol(optarg); break;
case 'L': logfp = fopen(optarg, "w"); break;
case 'R': target_rcvbuf = atoi(optarg); break;
- case 'M': multicast_group = optarg; break;
+#if HAVE_ALSA_ASOUNDLIB_H
+ case 'a': backend = playrtp_alsa; break;
+#endif
+#if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
+ case 'o': backend = playrtp_oss; break;
+#endif
+#if HAVE_COREAUDIO_AUDIOHARDWARE_H
+ case 'c': backend = playrtp_coreaudio; break;
+#endif
+ case 'C': configfile = optarg; break;
+ case 's': control_socket = optarg; break;
+ case 'r': dumpfile = optarg; break;
default: fatal(0, "invalid option");
}
}
+ if(config_read(0)) fatal(0, "cannot read configuration");
if(!maxbuffer)
maxbuffer = 4 * readahead;
argc -= optind;
argv += optind;
- if(argc < 1 || argc > 2)
- fatal(0, "usage: disorder-playrtp [OPTIONS] ADDRESS [PORT]");
- sl.n = argc;
- sl.s = argv;
- /* Listen for inbound audio data */
+ switch(argc) {
+ case 0:
+ /* Get configuration from server */
+ if(!(c = disorder_new(1))) exit(EXIT_FAILURE);
+ if(disorder_connect(c)) exit(EXIT_FAILURE);
+ if(disorder_rtp_address(c, &address, &port)) exit(EXIT_FAILURE);
+ sl.n = 2;
+ sl.s = xcalloc(2, sizeof *sl.s);
+ sl.s[0] = address;
+ sl.s[1] = port;
+ break;
+ case 1:
+ case 2:
+ /* Use command-line ADDRESS+PORT or just PORT */
+ sl.n = argc;
+ sl.s = argv;
+ break;
+ default:
+ fatal(0, "usage: disorder-playrtp [OPTIONS] [[ADDRESS] PORT]");
+ }
+ /* Look up address and port */
if(!(res = get_address(&sl, &prefs, &sockname)))
exit(1);
+ /* Create the socket */
if((rtpfd = socket(res->ai_family,
res->ai_socktype,
res->ai_protocol)) < 0)
fatal(errno, "error creating socket");
+ /* Stash the multicast group address */
+ if((is_multicast = multicast(res->ai_addr))) {
+ memcpy(&mgroup, res->ai_addr, res->ai_addrlen);
+ switch(res->ai_addr->sa_family) {
+ case AF_INET:
+ mgroup.in.sin_port = 0;
+ break;
+ case AF_INET6:
+ mgroup.in6.sin6_port = 0;
+ break;
+ }
+ }
+ /* Bind to 0/port */
+ switch(res->ai_addr->sa_family) {
+ case AF_INET:
+ memset(&((struct sockaddr_in *)res->ai_addr)->sin_addr, 0,
+ sizeof (struct in_addr));
+ break;
+ case AF_INET6:
+ memset(&((struct sockaddr_in6 *)res->ai_addr)->sin6_addr, 0,
+ sizeof (struct in6_addr));
+ break;
+ default:
+ fatal(0, "unsupported family %d", (int)res->ai_addr->sa_family);
+ }
if(bind(rtpfd, res->ai_addr, res->ai_addrlen) < 0)
fatal(errno, "error binding socket to %s", sockname);
- if(multicast_group) {
- if((n = getaddrinfo(multicast_group, 0, &prefs, &res)))
- fatal(0, "getaddrinfo %s: %s", multicast_group, gai_strerror(n));
- switch(res->ai_family) {
+ if(is_multicast) {
+ switch(mgroup.sa.sa_family) {
case PF_INET:
- mreq.imr_multiaddr = ((struct sockaddr_in *)res->ai_addr)->sin_addr;
+ mreq.imr_multiaddr = mgroup.in.sin_addr;
mreq.imr_interface.s_addr = 0; /* use primary interface */
if(setsockopt(rtpfd, IPPROTO_IP, IP_ADD_MEMBERSHIP,
&mreq, sizeof mreq) < 0)
fatal(errno, "error calling setsockopt IP_ADD_MEMBERSHIP");
break;
case PF_INET6:
- mreq6.ipv6mr_multiaddr = ((struct sockaddr_in6 *)res->ai_addr)->sin6_addr;
+ mreq6.ipv6mr_multiaddr = mgroup.in6.sin6_addr;
memset(&mreq6.ipv6mr_interface, 0, sizeof mreq6.ipv6mr_interface);
if(setsockopt(rtpfd, IPPROTO_IPV6, IPV6_JOIN_GROUP,
&mreq6, sizeof mreq6) < 0)
@@ -808,7 +663,10 @@ int main(int argc, char **argv) {
default:
fatal(0, "unsupported address family %d", res->ai_family);
}
- }
+ info("listening on %s multicast group %s",
+ format_sockaddr(res->ai_addr), format_sockaddr(&mgroup.sa));
+ } else
+ info("listening on %s", format_sockaddr(res->ai_addr));
len = sizeof rcvbuf;
if(getsockopt(rtpfd, SOL_SOCKET, SO_RCVBUF, &rcvbuf, &len) < 0)
fatal(errno, "error calling getsockopt SO_RCVBUF");
@@ -825,6 +683,33 @@ int main(int argc, char **argv) {
info("default socket receive buffer %d", rcvbuf);
if(logfp)
info("WARNING: -L option can impact performance");
+ if(control_socket) {
+ pthread_t tid;
+
+ if((err = pthread_create(&tid, 0, control_thread, 0)))
+ fatal(err, "pthread_create control_thread");
+ }
+ if(dumpfile) {
+ int fd;
+ unsigned char buffer[65536];
+ size_t written;
+
+ if((fd = open(dumpfile, O_RDWR|O_TRUNC|O_CREAT, 0666)) < 0)
+ fatal(errno, "opening %s", dumpfile);
+ /* Fill with 0s to a suitable size */
+ memset(buffer, 0, sizeof buffer);
+ for(written = 0; written < dump_size * sizeof(int16_t);
+ written += sizeof buffer) {
+ if(write(fd, buffer, sizeof buffer) < 0)
+ fatal(errno, "clearing %s", dumpfile);
+ }
+ /* Map the buffer into memory for convenience */
+ dump_buffer = mmap(0, dump_size * sizeof(int16_t), PROT_READ|PROT_WRITE,
+ MAP_SHARED, fd, 0);
+ if(dump_buffer == (void *)-1)
+ fatal(errno, "mapping %s", dumpfile);
+ info("dumping to %s", dumpfile);
+ }
play_rtp();
return 0;
}