$(LIBDB) $(LIBPCRE) $(LIBICONV)
disorder_deadlock_DEPENDENCIES=../lib/libdisorder.a
-disorder_speaker_SOURCES=speaker.c speaker.h
+disorder_speaker_SOURCES=speaker.c speaker.h \
+ speaker-command.c \
+ speaker-network.c \
+ speaker-alsa.c
disorder_speaker_LDADD=$(LIBOBJS) ../lib/libdisorder.a \
$(LIBASOUND) $(LIBPCRE) $(LIBICONV) $(LIBGCRYPT)
disorder_speaker_DEPENDENCIES=../lib/libdisorder.a
--- /dev/null
+/*
+ * This file is part of DisOrder
+ * Copyright (C) 2005, 2006, 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 server/speaker-alsa.c
+ * @brief Support for @ref BACKEND_ALSA */
+
+#include <config.h>
+
+#if API_ALSA
+
+#include "types.h"
+
+#include <unistd.h>
+#include <poll.h>
+#include <alsa/asoundlib.h>
+
+#include "configuration.h"
+#include "syscalls.h"
+#include "log.h"
+#include "speaker-protocol.h"
+#include "speaker.h"
+
+/** @brief The current PCM handle */
+static snd_pcm_t *pcm;
+
+/** @brief Last seen buffer size */
+static snd_pcm_uframes_t last_pcm_bufsize;
+
+/** @brief ALSA backend initialization */
+static void alsa_init(void) {
+ info("selected ALSA backend");
+}
+
+/** @brief Log ALSA parameters */
+static void log_params(snd_pcm_hw_params_t *hwparams,
+ snd_pcm_sw_params_t *swparams) {
+ snd_pcm_uframes_t f;
+ unsigned u;
+
+ return; /* too verbose */
+ if(hwparams) {
+ /* TODO */
+ }
+ if(swparams) {
+ snd_pcm_sw_params_get_silence_size(swparams, &f);
+ info("sw silence_size=%lu", (unsigned long)f);
+ snd_pcm_sw_params_get_silence_threshold(swparams, &f);
+ info("sw silence_threshold=%lu", (unsigned long)f);
+ snd_pcm_sw_params_get_sleep_min(swparams, &u);
+ info("sw sleep_min=%lu", (unsigned long)u);
+ snd_pcm_sw_params_get_start_threshold(swparams, &f);
+ info("sw start_threshold=%lu", (unsigned long)f);
+ snd_pcm_sw_params_get_stop_threshold(swparams, &f);
+ info("sw stop_threshold=%lu", (unsigned long)f);
+ snd_pcm_sw_params_get_xfer_align(swparams, &f);
+ info("sw xfer_align=%lu", (unsigned long)f);
+ }
+}
+
+/** @brief ALSA deactivation */
+static void alsa_deactivate(void) {
+ if(pcm) {
+ int err;
+
+ if((err = snd_pcm_nonblock(pcm, 0)) < 0)
+ fatal(0, "error calling snd_pcm_nonblock: %d", err);
+ D(("draining pcm"));
+ snd_pcm_drain(pcm);
+ D(("closing pcm"));
+ snd_pcm_close(pcm);
+ pcm = 0;
+ device_state = device_closed;
+ D(("released audio device"));
+ }
+}
+
+/** @brief ALSA backend activation */
+static void alsa_activate(void) {
+ /* If we need to change format then close the current device. */
+ if(pcm && !formats_equal(&playing->format, &device_format))
+ alsa_deactivate();
+ /* Now if the sound device is open it must have the right format */
+ if(!pcm) {
+ snd_pcm_hw_params_t *hwparams;
+ snd_pcm_sw_params_t *swparams;
+ snd_pcm_uframes_t pcm_bufsize;
+ int err;
+ int sample_format = 0;
+ unsigned rate;
+
+ D(("snd_pcm_open"));
+ if((err = snd_pcm_open(&pcm,
+ config->device,
+ SND_PCM_STREAM_PLAYBACK,
+ SND_PCM_NONBLOCK))) {
+ error(0, "error from snd_pcm_open: %d", err);
+ goto error;
+ }
+ snd_pcm_hw_params_alloca(&hwparams);
+ D(("set up hw params"));
+ 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);
+ switch(playing->format.bits) {
+ case 8:
+ sample_format = SND_PCM_FORMAT_S8;
+ break;
+ case 16:
+ switch(playing->format.byte_format) {
+ case AO_FMT_NATIVE: sample_format = SND_PCM_FORMAT_S16; break;
+ case AO_FMT_LITTLE: sample_format = SND_PCM_FORMAT_S16_LE; break;
+ case AO_FMT_BIG: sample_format = SND_PCM_FORMAT_S16_BE; break;
+ error(0, "unrecognized byte format %d", playing->format.byte_format);
+ goto fatal;
+ }
+ break;
+ default:
+ error(0, "unsupported sample size %d", playing->format.bits);
+ goto fatal;
+ }
+ if((err = snd_pcm_hw_params_set_format(pcm, hwparams,
+ sample_format)) < 0) {
+ error(0, "error from snd_pcm_hw_params_set_format (%d): %d",
+ sample_format, err);
+ goto fatal;
+ }
+ rate = playing->format.rate;
+ if((err = snd_pcm_hw_params_set_rate_near(pcm, hwparams, &rate, 0)) < 0) {
+ error(0, "error from snd_pcm_hw_params_set_rate (%d): %d",
+ playing->format.rate, err);
+ goto fatal;
+ }
+ if(rate != (unsigned)playing->format.rate)
+ info("want rate %d, got %u", playing->format.rate, rate);
+ if((err = snd_pcm_hw_params_set_channels(pcm, hwparams,
+ playing->format.channels)) < 0) {
+ error(0, "error from snd_pcm_hw_params_set_channels (%d): %d",
+ playing->format.channels, err);
+ goto fatal;
+ }
+ pcm_bufsize = 3 * FRAMES;
+ 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",
+ 3 * FRAMES, err);
+ if(pcm_bufsize != 3 * FRAMES && pcm_bufsize != last_pcm_bufsize)
+ info("asked for PCM buffer of %d frames, got %d",
+ 3 * FRAMES, (int)pcm_bufsize);
+ last_pcm_bufsize = pcm_bufsize;
+ if((err = snd_pcm_hw_params(pcm, hwparams)) < 0)
+ fatal(0, "error calling snd_pcm_hw_params: %d", err);
+ D(("set up sw params"));
+ 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, FRAMES)) < 0)
+ fatal(0, "error calling snd_pcm_sw_params_set_avail_min %d: %d",
+ FRAMES, err);
+ if((err = snd_pcm_sw_params(pcm, swparams)) < 0)
+ fatal(0, "error calling snd_pcm_sw_params: %d", err);
+ device_format = playing->format;
+ D(("acquired audio device"));
+ log_params(hwparams, swparams);
+ device_state = device_open;
+ }
+ return;
+fatal:
+ abandon();
+error:
+ /* We assume the error is temporary and that we'll retry in a bit. */
+ if(pcm) {
+ snd_pcm_close(pcm);
+ pcm = 0;
+ device_state = device_error;
+ }
+ return;
+}
+
+/** @brief Play via ALSA */
+static size_t alsa_play(size_t frames) {
+ snd_pcm_sframes_t pcm_written_frames;
+ int err;
+
+ pcm_written_frames = snd_pcm_writei(pcm,
+ playing->buffer + playing->start,
+ frames);
+ D(("actually play %zu frames, wrote %d",
+ frames, (int)pcm_written_frames));
+ if(pcm_written_frames < 0) {
+ switch(pcm_written_frames) {
+ case -EPIPE: /* underrun */
+ error(0, "snd_pcm_writei reports underrun");
+ if((err = snd_pcm_prepare(pcm)) < 0)
+ fatal(0, "error calling snd_pcm_prepare: %d", err);
+ return 0;
+ case -EAGAIN:
+ return 0;
+ default:
+ fatal(0, "error calling snd_pcm_writei: %d",
+ (int)pcm_written_frames);
+ }
+ } else
+ return pcm_written_frames;
+}
+
+static int alsa_slots, alsa_nslots = -1;
+
+/** @brief Fill in poll fd array for ALSA */
+static void alsa_beforepoll(void) {
+ /* We send sample data to ALSA as fast as it can accept it, relying on
+ * the fact that it has a relatively small buffer to minimize pause
+ * latency. */
+ int retry = 3, err;
+
+ alsa_slots = fdno;
+ do {
+ retry = 0;
+ alsa_nslots = snd_pcm_poll_descriptors(pcm, &fds[fdno], NFDS - fdno);
+ if((alsa_nslots <= 0
+ || !(fds[alsa_slots].events & POLLOUT))
+ && snd_pcm_state(pcm) == SND_PCM_STATE_XRUN) {
+ error(0, "underrun detected after call to snd_pcm_poll_descriptors()");
+ if((err = snd_pcm_prepare(pcm)))
+ fatal(0, "error calling snd_pcm_prepare: %d", err);
+ } else
+ break;
+ } while(retry-- > 0);
+ if(alsa_nslots >= 0)
+ fdno += alsa_nslots;
+}
+
+/** @brief Process poll() results for ALSA */
+static int alsa_ready(void) {
+ int err;
+
+ unsigned short alsa_revents;
+
+ if((err = snd_pcm_poll_descriptors_revents(pcm,
+ &fds[alsa_slots],
+ alsa_nslots,
+ &alsa_revents)) < 0)
+ fatal(0, "error calling snd_pcm_poll_descriptors_revents: %d", err);
+ if(alsa_revents & (POLLOUT | POLLERR))
+ return 1;
+ else
+ return 0;
+}
+
+const struct speaker_backend alsa_backend = {
+ BACKEND_ALSA,
+ 0,
+ alsa_init,
+ alsa_activate,
+ alsa_play,
+ alsa_deactivate,
+ alsa_beforepoll,
+ alsa_ready
+};
+
+#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) 2005, 2006, 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 server/speaker-command.c
+ * @brief Support for @ref BACKEND_COMMAND */
+
+#include <config.h>
+#include "types.h"
+
+#include <unistd.h>
+#include <poll.h>
+
+#include "configuration.h"
+#include "syscalls.h"
+#include "log.h"
+#include "speaker-protocol.h"
+#include "speaker.h"
+
+/** @brief Pipe to subprocess
+ *
+ * This is the file descriptor to write to for @ref BACKEND_COMMAND.
+ */
+static int cmdfd = -1;
+
+/** @brief poll array slot for @ref cmdfd
+ *
+ * Set by command_beforepoll().
+ */
+static int cmdfd_slot;
+
+/** @brief Start the subprocess for @ref BACKEND_COMMAND */
+static void fork_cmd(void) {
+ pid_t cmdpid;
+ int pfd[2];
+ if(cmdfd != -1) close(cmdfd);
+ xpipe(pfd);
+ cmdpid = xfork();
+ if(!cmdpid) {
+ signal(SIGPIPE, SIG_DFL);
+ xdup2(pfd[0], 0);
+ close(pfd[0]);
+ close(pfd[1]);
+ execl("/bin/sh", "sh", "-c", config->speaker_command, (char *)0);
+ fatal(errno, "error execing /bin/sh");
+ }
+ close(pfd[0]);
+ cmdfd = pfd[1];
+ D(("forked cmd %d, fd = %d", cmdpid, cmdfd));
+}
+
+/** @brief Command backend initialization */
+static void command_init(void) {
+ info("selected command backend");
+ fork_cmd();
+}
+
+/** @brief Play to a subprocess */
+static size_t command_play(size_t frames) {
+ size_t bytes = frames * device_bpf;
+ int written_bytes;
+
+ written_bytes = write(cmdfd, playing->buffer + playing->start, bytes);
+ D(("actually play %zu bytes, wrote %d",
+ bytes, written_bytes));
+ if(written_bytes < 0) {
+ switch(errno) {
+ case EPIPE:
+ error(0, "hmm, command died; trying another");
+ fork_cmd();
+ return 0;
+ case EAGAIN:
+ return 0;
+ default:
+ fatal(errno, "error writing to subprocess");
+ }
+ } else
+ return written_bytes / device_bpf;
+}
+
+/** @brief Update poll array for writing to subprocess */
+static void command_beforepoll(void) {
+ /* We send sample data to the subprocess as fast as it can accept it.
+ * This isn't ideal as pause latency can be very high as a result. */
+ if(cmdfd >= 0)
+ cmdfd_slot = addfd(cmdfd, POLLOUT);
+}
+
+/** @brief Process poll() results for subprocess play */
+static int command_ready(void) {
+ if(fds[cmdfd_slot].revents & (POLLOUT | POLLERR))
+ return 1;
+ else
+ return 0;
+}
+
+const struct speaker_backend command_backend = {
+ BACKEND_COMMAND,
+ FIXED_FORMAT,
+ command_init,
+ 0, /* activate */
+ command_play,
+ 0, /* deactivate */
+ command_beforepoll,
+ command_ready
+};
+
+/*
+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) 2005, 2006, 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 server/speaker-network.c
+ * @brief Support for @ref BACKEND_NETWORK */
+
+#include <config.h>
+#include "types.h"
+
+#include <unistd.h>
+#include <poll.h>
+#include <netdb.h>
+#include <gcrypt.h>
+#include <sys/socket.h>
+#include <sys/uio.h>
+#include <assert.h>
+
+#include "configuration.h"
+#include "syscalls.h"
+#include "log.h"
+#include "addr.h"
+#include "timeval.h"
+#include "rtp.h"
+#include "speaker-protocol.h"
+#include "speaker.h"
+
+/** @brief Network socket
+ *
+ * This is the file descriptor to write to for @ref BACKEND_NETWORK.
+ */
+static int bfd = -1;
+
+/** @brief RTP timestamp
+ *
+ * This counts the number of samples played (NB not the number of frames
+ * played).
+ *
+ * The timestamp in the packet header is only 32 bits wide. With 44100Hz
+ * stereo, that only gives about half a day before wrapping, which is not
+ * particularly convenient for certain debugging purposes. Therefore the
+ * timestamp is maintained as a 64-bit integer, giving around six million years
+ * before wrapping, and truncated to 32 bits when transmitting.
+ */
+static uint64_t rtp_time;
+
+/** @brief RTP base timestamp
+ *
+ * This is the real time correspoding to an @ref rtp_time of 0. It is used
+ * to recalculate the timestamp after idle periods.
+ */
+static struct timeval rtp_time_0;
+
+/** @brief RTP packet sequence number */
+static uint16_t rtp_seq;
+
+/** @brief RTP SSRC */
+static uint32_t rtp_id;
+
+/** @brief Error counter */
+static int audio_errors;
+
+/** @brief Network backend initialization */
+static void network_init(void) {
+ struct addrinfo *res, *sres;
+ static const struct addrinfo pref = {
+ 0,
+ PF_INET,
+ SOCK_DGRAM,
+ IPPROTO_UDP,
+ 0,
+ 0,
+ 0,
+ 0
+ };
+ static const struct addrinfo prefbind = {
+ AI_PASSIVE,
+ PF_INET,
+ SOCK_DGRAM,
+ IPPROTO_UDP,
+ 0,
+ 0,
+ 0,
+ 0
+ };
+ static const int one = 1;
+ int sndbuf, target_sndbuf = 131072;
+ socklen_t len;
+ char *sockname, *ssockname;
+
+ res = get_address(&config->broadcast, &pref, &sockname);
+ if(!res) exit(-1);
+ if(config->broadcast_from.n) {
+ sres = get_address(&config->broadcast_from, &prefbind, &ssockname);
+ if(!sres) exit(-1);
+ } else
+ sres = 0;
+ if((bfd = socket(res->ai_family,
+ res->ai_socktype,
+ res->ai_protocol)) < 0)
+ fatal(errno, "error creating broadcast socket");
+ if(setsockopt(bfd, SOL_SOCKET, SO_BROADCAST, &one, sizeof one) < 0)
+ fatal(errno, "error setting SO_BROADCAST on broadcast socket");
+ len = sizeof sndbuf;
+ if(getsockopt(bfd, SOL_SOCKET, SO_SNDBUF,
+ &sndbuf, &len) < 0)
+ fatal(errno, "error getting SO_SNDBUF");
+ if(target_sndbuf > sndbuf) {
+ if(setsockopt(bfd, SOL_SOCKET, SO_SNDBUF,
+ &target_sndbuf, sizeof target_sndbuf) < 0)
+ error(errno, "error setting SO_SNDBUF to %d", target_sndbuf);
+ else
+ info("changed socket send buffer size from %d to %d",
+ sndbuf, target_sndbuf);
+ } else
+ info("default socket send buffer is %d",
+ sndbuf);
+ /* We might well want to set additional broadcast- or multicast-related
+ * options here */
+ if(sres && bind(bfd, sres->ai_addr, sres->ai_addrlen) < 0)
+ fatal(errno, "error binding broadcast socket to %s", ssockname);
+ if(connect(bfd, res->ai_addr, res->ai_addrlen) < 0)
+ fatal(errno, "error connecting broadcast socket to %s", sockname);
+ /* Select an SSRC */
+ gcry_randomize(&rtp_id, sizeof rtp_id, GCRY_STRONG_RANDOM);
+ info("selected network backend, sending to %s", sockname);
+ if(config->sample_format.byte_format != AO_FMT_BIG) {
+ info("forcing big-endian sample format");
+ config->sample_format.byte_format = AO_FMT_BIG;
+ }
+}
+
+/** @brief Play over the network */
+static size_t network_play(size_t frames) {
+ struct rtp_header header;
+ struct iovec vec[2];
+ size_t bytes = frames * device_bpf, written_frames;
+ int written_bytes;
+ /* We transmit using RTP (RFC3550) and attempt to conform to the internet
+ * AVT profile (RFC3551). */
+
+ if(idled) {
+ /* There may have been a gap. Fix up the RTP time accordingly. */
+ struct timeval now;
+ uint64_t delta;
+ uint64_t target_rtp_time;
+
+ /* Find the current time */
+ xgettimeofday(&now, 0);
+ /* Find the number of microseconds elapsed since rtp_time=0 */
+ delta = tvsub_us(now, rtp_time_0);
+ assert(delta <= UINT64_MAX / 88200);
+ target_rtp_time = (delta * playing->format.rate
+ * playing->format.channels) / 1000000;
+ /* Overflows at ~6 years uptime with 44100Hz stereo */
+
+ /* rtp_time is the number of samples we've played. NB that we play
+ * RTP_AHEAD_MS ahead of ourselves, so it may legitimately be ahead of
+ * the value we deduce from time comparison.
+ *
+ * Suppose we have 1s track started at t=0, and another track begins to
+ * play at t=2s. Suppose RTP_AHEAD_MS=1000 and 44100Hz stereo. In that
+ * case we'll send 1s of audio as fast as we can, giving rtp_time=88200.
+ * rtp_time stops at this point.
+ *
+ * At t=2s we'll have calculated target_rtp_time=176400. In this case we
+ * set rtp_time=176400 and the player can correctly conclude that it
+ * should leave 1s between the tracks.
+ *
+ * Suppose instead that the second track arrives at t=0.5s, and that
+ * we've managed to transmit the whole of the first track already. We'll
+ * have target_rtp_time=44100.
+ *
+ * The desired behaviour is to play the second track back to back with
+ * first. In this case therefore we do not modify rtp_time.
+ *
+ * Is it ever right to reduce rtp_time? No; for that would imply
+ * transmitting packets with overlapping timestamp ranges, which does not
+ * make sense.
+ */
+ target_rtp_time &= ~(uint64_t)1; /* stereo! */
+ if(target_rtp_time > rtp_time) {
+ /* More time has elapsed than we've transmitted samples. That implies
+ * we've been 'sending' silence. */
+ info("advancing rtp_time by %"PRIu64" samples",
+ target_rtp_time - rtp_time);
+ rtp_time = target_rtp_time;
+ } else if(target_rtp_time < rtp_time) {
+ const int64_t samples_ahead = ((uint64_t)RTP_AHEAD_MS
+ * config->sample_format.rate
+ * config->sample_format.channels
+ / 1000);
+
+ if(target_rtp_time + samples_ahead < rtp_time) {
+ info("reversing rtp_time by %"PRIu64" samples",
+ rtp_time - target_rtp_time);
+ }
+ }
+ }
+ header.vpxcc = 2 << 6; /* V=2, P=0, X=0, CC=0 */
+ header.seq = htons(rtp_seq++);
+ header.timestamp = htonl((uint32_t)rtp_time);
+ header.ssrc = rtp_id;
+ header.mpt = (idled ? 0x80 : 0x00) | 10;
+ /* 10 = L16 = 16-bit x 2 x 44100KHz. We ought to deduce this value from
+ * the sample rate (in a library somewhere so that configuration.c can rule
+ * out invalid rates).
+ */
+ idled = 0;
+ if(bytes > NETWORK_BYTES - sizeof header) {
+ bytes = NETWORK_BYTES - sizeof header;
+ /* Always send a whole number of frames */
+ bytes -= bytes % device_bpf;
+ }
+ /* "The RTP clock rate used for generating the RTP timestamp is independent
+ * of the number of channels and the encoding; it equals the number of
+ * sampling periods per second. For N-channel encodings, each sampling
+ * period (say, 1/8000 of a second) generates N samples. (This terminology
+ * is standard, but somewhat confusing, as the total number of samples
+ * generated per second is then the sampling rate times the channel
+ * count.)"
+ */
+ vec[0].iov_base = (void *)&header;
+ vec[0].iov_len = sizeof header;
+ vec[1].iov_base = playing->buffer + playing->start;
+ vec[1].iov_len = bytes;
+ do {
+ written_bytes = writev(bfd, vec, 2);
+ } while(written_bytes < 0 && errno == EINTR);
+ if(written_bytes < 0) {
+ error(errno, "error transmitting audio data");
+ ++audio_errors;
+ if(audio_errors == 10)
+ fatal(0, "too many audio errors");
+ return 0;
+ } else
+ audio_errors /= 2;
+ written_bytes -= sizeof (struct rtp_header);
+ written_frames = written_bytes / device_bpf;
+ /* Advance RTP's notion of the time */
+ rtp_time += written_frames * playing->format.channels;
+ return written_frames;
+}
+
+static int bfd_slot;
+
+/** @brief Set up poll array for network play */
+static void network_beforepoll(void) {
+ struct timeval now;
+ uint64_t target_us;
+ uint64_t target_rtp_time;
+ const int64_t samples_ahead = ((uint64_t)RTP_AHEAD_MS
+ * config->sample_format.rate
+ * config->sample_format.channels
+ / 1000);
+
+ /* If we're starting then initialize the base time */
+ if(!rtp_time)
+ xgettimeofday(&rtp_time_0, 0);
+ /* We send audio data whenever we get RTP_AHEAD seconds or more
+ * behind */
+ xgettimeofday(&now, 0);
+ target_us = tvsub_us(now, rtp_time_0);
+ assert(target_us <= UINT64_MAX / 88200);
+ target_rtp_time = (target_us * config->sample_format.rate
+ * config->sample_format.channels)
+ / 1000000;
+ if((int64_t)(rtp_time - target_rtp_time) < samples_ahead)
+ bfd_slot = addfd(bfd, POLLOUT);
+}
+
+/** @brief Process poll() results for network play */
+static int network_ready(void) {
+ if(fds[bfd_slot].revents & (POLLOUT | POLLERR))
+ return 1;
+ else
+ return 0;
+}
+
+const struct speaker_backend network_backend = {
+ BACKEND_NETWORK,
+ FIXED_FORMAT,
+ network_init,
+ 0, /* activate */
+ network_play,
+ 0, /* deactivate */
+ network_beforepoll,
+ network_ready
+};
+
+/*
+Local Variables:
+c-basic-offset:2
+comment-column:40
+fill-column:79
+indent-tabs-mode:nil
+End:
+*/
#include <time.h>
#include <fcntl.h>
#include <poll.h>
-#include <sys/socket.h>
-#include <netdb.h>
-#include <gcrypt.h>
-#include <sys/uio.h>
#include "configuration.h"
#include "syscalls.h"
#include "mem.h"
#include "speaker-protocol.h"
#include "user.h"
-#include "addr.h"
-#include "timeval.h"
-#include "rtp.h"
#include "speaker.h"
-#if API_ALSA
-#include <alsa/asoundlib.h>
-#endif
-
/** @brief Linked list of all prepared tracks */
struct track *tracks;
/** @brief Playing track, or NULL */
struct track *playing;
+/** @brief Number of bytes pre frame */
+size_t device_bpf;
+
+/** @brief Array of file descriptors for poll() */
+struct pollfd fds[NFDS];
+
+/** @brief Next free slot in @ref fds */
+int fdno;
+
static time_t last_report; /* when we last reported */
static int paused; /* pause status */
-static size_t bpf; /* bytes per frame */
-static struct pollfd fds[NFDS]; /* if we need more than that */
-static int fdno; /* fd number */
-#if API_ALSA
-/** @brief The current PCM handle */
-static snd_pcm_t *pcm;
-static snd_pcm_uframes_t last_pcm_bufsize; /* last seen buffer size */
-#endif
/** @brief The current device state */
enum device_states device_state;
*/
ao_sample_format device_format;
-/** @brief Pipe to subprocess
- *
- * This is the file descriptor to write to for @ref BACKEND_COMMAND.
- */
-static int cmdfd = -1;
-
-/** @brief Network socket
- *
- * This is the file descriptor to write to for @ref BACKEND_NETWORK.
- */
-static int bfd = -1;
-
-/** @brief RTP timestamp
- *
- * This counts the number of samples played (NB not the number of frames
- * played).
- *
- * The timestamp in the packet header is only 32 bits wide. With 44100Hz
- * stereo, that only gives about half a day before wrapping, which is not
- * particularly convenient for certain debugging purposes. Therefore the
- * timestamp is maintained as a 64-bit integer, giving around six million years
- * before wrapping, and truncated to 32 bits when transmitting.
- */
-static uint64_t rtp_time;
-
-/** @brief RTP base timestamp
- *
- * This is the real time correspoding to an @ref rtp_time of 0. It is used
- * to recalculate the timestamp after idle periods.
- */
-static struct timeval rtp_time_0;
-
-/** @brief RTP packet sequence number */
-static uint16_t rtp_seq;
-
-/** @brief RTP SSRC */
-static uint32_t rtp_id;
-
/** @brief Set when idled
*
* This is set when the sound device is deliberately closed by idle().
*/
-static int idled; /* set when idled */
-
-/** @brief Error counter */
-static int audio_errors;
+int idled;
/** @brief Selected backend */
static const struct speaker_backend *backend;
}
/** @brief Return true if A and B denote identical libao formats, else false */
-static int formats_equal(const ao_sample_format *a,
- const ao_sample_format *b) {
+int formats_equal(const ao_sample_format *a,
+ const ao_sample_format *b) {
return (a->bits == b->bits
&& a->rate == b->rate
&& a->channels == b->channels
}
/** @brief Abandon the current track */
-static void abandon(void) {
+void abandon(void) {
struct speaker_message sm;
D(("abandon"));
device_state = device_open;
}
if(device_state == device_open)
- bpf = bytes_per_frame(&device_format);
+ device_bpf = bytes_per_frame(&device_format);
}
/** @brief Check whether the current track has finished
if(device_state != device_open)
return;
}
- D(("play: play %zu/%zu%s %dHz %db %dc", frames, playing->used / bpf,
+ D(("play: play %zu/%zu%s %dHz %db %dc", frames, playing->used / device_bpf,
playing->eof ? " EOF" : "",
playing->format.rate,
playing->format.bits,
else
/* The ring buffer is not wrapped, can play the lot */
avail_bytes = playing->used;
- avail_frames = avail_bytes / bpf;
+ avail_frames = avail_bytes / device_bpf;
/* Only play up to the requested amount */
if(avail_frames > frames)
avail_frames = frames;
return;
/* Play it, Sam */
written_frames = backend->play(avail_frames);
- written_bytes = written_frames * bpf;
+ written_bytes = written_frames * device_bpf;
/* written_bytes and written_frames had better both be set and correct by
* this point */
playing->start += written_bytes;
signal(SIGCHLD, reap);
}
-static int addfd(int fd, int events) {
+int addfd(int fd, int events) {
if(fdno < NFDS) {
fds[fdno].fd = fd;
fds[fdno].events = events;
return -1;
}
-#if API_ALSA
-/** @brief ALSA backend initialization */
-static void alsa_init(void) {
- info("selected ALSA backend");
-}
-
-/** @brief Log ALSA parameters */
-static void log_params(snd_pcm_hw_params_t *hwparams,
- snd_pcm_sw_params_t *swparams) {
- snd_pcm_uframes_t f;
- unsigned u;
-
- return; /* too verbose */
- if(hwparams) {
- /* TODO */
- }
- if(swparams) {
- snd_pcm_sw_params_get_silence_size(swparams, &f);
- info("sw silence_size=%lu", (unsigned long)f);
- snd_pcm_sw_params_get_silence_threshold(swparams, &f);
- info("sw silence_threshold=%lu", (unsigned long)f);
- snd_pcm_sw_params_get_sleep_min(swparams, &u);
- info("sw sleep_min=%lu", (unsigned long)u);
- snd_pcm_sw_params_get_start_threshold(swparams, &f);
- info("sw start_threshold=%lu", (unsigned long)f);
- snd_pcm_sw_params_get_stop_threshold(swparams, &f);
- info("sw stop_threshold=%lu", (unsigned long)f);
- snd_pcm_sw_params_get_xfer_align(swparams, &f);
- info("sw xfer_align=%lu", (unsigned long)f);
- }
-}
-
-/** @brief ALSA deactivation */
-static void alsa_deactivate(void) {
- if(pcm) {
- int err;
-
- if((err = snd_pcm_nonblock(pcm, 0)) < 0)
- fatal(0, "error calling snd_pcm_nonblock: %d", err);
- D(("draining pcm"));
- snd_pcm_drain(pcm);
- D(("closing pcm"));
- snd_pcm_close(pcm);
- pcm = 0;
- device_state = device_closed;
- D(("released audio device"));
- }
-}
-
-/** @brief ALSA backend activation */
-static void alsa_activate(void) {
- /* If we need to change format then close the current device. */
- if(pcm && !formats_equal(&playing->format, &device_format))
- alsa_deactivate();
- /* Now if the sound device is open it must have the right format */
- if(!pcm) {
- snd_pcm_hw_params_t *hwparams;
- snd_pcm_sw_params_t *swparams;
- snd_pcm_uframes_t pcm_bufsize;
- int err;
- int sample_format = 0;
- unsigned rate;
-
- D(("snd_pcm_open"));
- if((err = snd_pcm_open(&pcm,
- config->device,
- SND_PCM_STREAM_PLAYBACK,
- SND_PCM_NONBLOCK))) {
- error(0, "error from snd_pcm_open: %d", err);
- goto error;
- }
- snd_pcm_hw_params_alloca(&hwparams);
- D(("set up hw params"));
- 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);
- switch(playing->format.bits) {
- case 8:
- sample_format = SND_PCM_FORMAT_S8;
- break;
- case 16:
- switch(playing->format.byte_format) {
- case AO_FMT_NATIVE: sample_format = SND_PCM_FORMAT_S16; break;
- case AO_FMT_LITTLE: sample_format = SND_PCM_FORMAT_S16_LE; break;
- case AO_FMT_BIG: sample_format = SND_PCM_FORMAT_S16_BE; break;
- error(0, "unrecognized byte format %d", playing->format.byte_format);
- goto fatal;
- }
- break;
- default:
- error(0, "unsupported sample size %d", playing->format.bits);
- goto fatal;
- }
- if((err = snd_pcm_hw_params_set_format(pcm, hwparams,
- sample_format)) < 0) {
- error(0, "error from snd_pcm_hw_params_set_format (%d): %d",
- sample_format, err);
- goto fatal;
- }
- rate = playing->format.rate;
- if((err = snd_pcm_hw_params_set_rate_near(pcm, hwparams, &rate, 0)) < 0) {
- error(0, "error from snd_pcm_hw_params_set_rate (%d): %d",
- playing->format.rate, err);
- goto fatal;
- }
- if(rate != (unsigned)playing->format.rate)
- info("want rate %d, got %u", playing->format.rate, rate);
- if((err = snd_pcm_hw_params_set_channels(pcm, hwparams,
- playing->format.channels)) < 0) {
- error(0, "error from snd_pcm_hw_params_set_channels (%d): %d",
- playing->format.channels, err);
- goto fatal;
- }
- pcm_bufsize = 3 * FRAMES;
- 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",
- 3 * FRAMES, err);
- if(pcm_bufsize != 3 * FRAMES && pcm_bufsize != last_pcm_bufsize)
- info("asked for PCM buffer of %d frames, got %d",
- 3 * FRAMES, (int)pcm_bufsize);
- last_pcm_bufsize = pcm_bufsize;
- if((err = snd_pcm_hw_params(pcm, hwparams)) < 0)
- fatal(0, "error calling snd_pcm_hw_params: %d", err);
- D(("set up sw params"));
- 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, FRAMES)) < 0)
- fatal(0, "error calling snd_pcm_sw_params_set_avail_min %d: %d",
- FRAMES, err);
- if((err = snd_pcm_sw_params(pcm, swparams)) < 0)
- fatal(0, "error calling snd_pcm_sw_params: %d", err);
- device_format = playing->format;
- D(("acquired audio device"));
- log_params(hwparams, swparams);
- device_state = device_open;
- }
- return;
-fatal:
- abandon();
-error:
- /* We assume the error is temporary and that we'll retry in a bit. */
- if(pcm) {
- snd_pcm_close(pcm);
- pcm = 0;
- device_state = device_error;
- }
- return;
-}
-
-/** @brief Play via ALSA */
-static size_t alsa_play(size_t frames) {
- snd_pcm_sframes_t pcm_written_frames;
- int err;
-
- pcm_written_frames = snd_pcm_writei(pcm,
- playing->buffer + playing->start,
- frames);
- D(("actually play %zu frames, wrote %d",
- frames, (int)pcm_written_frames));
- if(pcm_written_frames < 0) {
- switch(pcm_written_frames) {
- case -EPIPE: /* underrun */
- error(0, "snd_pcm_writei reports underrun");
- if((err = snd_pcm_prepare(pcm)) < 0)
- fatal(0, "error calling snd_pcm_prepare: %d", err);
- return 0;
- case -EAGAIN:
- return 0;
- default:
- fatal(0, "error calling snd_pcm_writei: %d",
- (int)pcm_written_frames);
- }
- } else
- return pcm_written_frames;
-}
-
-static int alsa_slots, alsa_nslots = -1;
-
-/** @brief Fill in poll fd array for ALSA */
-static void alsa_beforepoll(void) {
- /* We send sample data to ALSA as fast as it can accept it, relying on
- * the fact that it has a relatively small buffer to minimize pause
- * latency. */
- int retry = 3, err;
-
- alsa_slots = fdno;
- do {
- retry = 0;
- alsa_nslots = snd_pcm_poll_descriptors(pcm, &fds[fdno], NFDS - fdno);
- if((alsa_nslots <= 0
- || !(fds[alsa_slots].events & POLLOUT))
- && snd_pcm_state(pcm) == SND_PCM_STATE_XRUN) {
- error(0, "underrun detected after call to snd_pcm_poll_descriptors()");
- if((err = snd_pcm_prepare(pcm)))
- fatal(0, "error calling snd_pcm_prepare: %d", err);
- } else
- break;
- } while(retry-- > 0);
- if(alsa_nslots >= 0)
- fdno += alsa_nslots;
-}
-
-/** @brief Process poll() results for ALSA */
-static int alsa_ready(void) {
- int err;
-
- unsigned short alsa_revents;
-
- if((err = snd_pcm_poll_descriptors_revents(pcm,
- &fds[alsa_slots],
- alsa_nslots,
- &alsa_revents)) < 0)
- fatal(0, "error calling snd_pcm_poll_descriptors_revents: %d", err);
- if(alsa_revents & (POLLOUT | POLLERR))
- return 1;
- else
- return 0;
-}
-#endif
-
-/** @brief Start the subprocess for @ref BACKEND_COMMAND */
-static void fork_cmd(void) {
- pid_t cmdpid;
- int pfd[2];
- if(cmdfd != -1) close(cmdfd);
- xpipe(pfd);
- cmdpid = xfork();
- if(!cmdpid) {
- signal(SIGPIPE, SIG_DFL);
- xdup2(pfd[0], 0);
- close(pfd[0]);
- close(pfd[1]);
- execl("/bin/sh", "sh", "-c", config->speaker_command, (char *)0);
- fatal(errno, "error execing /bin/sh");
- }
- close(pfd[0]);
- cmdfd = pfd[1];
- D(("forked cmd %d, fd = %d", cmdpid, cmdfd));
-}
-
-/** @brief Command backend initialization */
-static void command_init(void) {
- info("selected command backend");
- fork_cmd();
-}
-
-/** @brief Play to a subprocess */
-static size_t command_play(size_t frames) {
- size_t bytes = frames * bpf;
- int written_bytes;
-
- written_bytes = write(cmdfd, playing->buffer + playing->start, bytes);
- D(("actually play %zu bytes, wrote %d",
- bytes, written_bytes));
- if(written_bytes < 0) {
- switch(errno) {
- case EPIPE:
- error(0, "hmm, command died; trying another");
- fork_cmd();
- return 0;
- case EAGAIN:
- return 0;
- default:
- fatal(errno, "error writing to subprocess");
- }
- } else
- return written_bytes / bpf;
-}
-
-static int cmdfd_slot;
-
-/** @brief Update poll array for writing to subprocess */
-static void command_beforepoll(void) {
- /* We send sample data to the subprocess as fast as it can accept it.
- * This isn't ideal as pause latency can be very high as a result. */
- if(cmdfd >= 0)
- cmdfd_slot = addfd(cmdfd, POLLOUT);
-}
-
-/** @brief Process poll() results for subprocess play */
-static int command_ready(void) {
- if(fds[cmdfd_slot].revents & (POLLOUT | POLLERR))
- return 1;
- else
- return 0;
-}
-
-/** @brief Network backend initialization */
-static void network_init(void) {
- struct addrinfo *res, *sres;
- static const struct addrinfo pref = {
- 0,
- PF_INET,
- SOCK_DGRAM,
- IPPROTO_UDP,
- 0,
- 0,
- 0,
- 0
- };
- static const struct addrinfo prefbind = {
- AI_PASSIVE,
- PF_INET,
- SOCK_DGRAM,
- IPPROTO_UDP,
- 0,
- 0,
- 0,
- 0
- };
- static const int one = 1;
- int sndbuf, target_sndbuf = 131072;
- socklen_t len;
- char *sockname, *ssockname;
-
- res = get_address(&config->broadcast, &pref, &sockname);
- if(!res) exit(-1);
- if(config->broadcast_from.n) {
- sres = get_address(&config->broadcast_from, &prefbind, &ssockname);
- if(!sres) exit(-1);
- } else
- sres = 0;
- if((bfd = socket(res->ai_family,
- res->ai_socktype,
- res->ai_protocol)) < 0)
- fatal(errno, "error creating broadcast socket");
- if(setsockopt(bfd, SOL_SOCKET, SO_BROADCAST, &one, sizeof one) < 0)
- fatal(errno, "error setting SO_BROADCAST on broadcast socket");
- len = sizeof sndbuf;
- if(getsockopt(bfd, SOL_SOCKET, SO_SNDBUF,
- &sndbuf, &len) < 0)
- fatal(errno, "error getting SO_SNDBUF");
- if(target_sndbuf > sndbuf) {
- if(setsockopt(bfd, SOL_SOCKET, SO_SNDBUF,
- &target_sndbuf, sizeof target_sndbuf) < 0)
- error(errno, "error setting SO_SNDBUF to %d", target_sndbuf);
- else
- info("changed socket send buffer size from %d to %d",
- sndbuf, target_sndbuf);
- } else
- info("default socket send buffer is %d",
- sndbuf);
- /* We might well want to set additional broadcast- or multicast-related
- * options here */
- if(sres && bind(bfd, sres->ai_addr, sres->ai_addrlen) < 0)
- fatal(errno, "error binding broadcast socket to %s", ssockname);
- if(connect(bfd, res->ai_addr, res->ai_addrlen) < 0)
- fatal(errno, "error connecting broadcast socket to %s", sockname);
- /* Select an SSRC */
- gcry_randomize(&rtp_id, sizeof rtp_id, GCRY_STRONG_RANDOM);
- info("selected network backend, sending to %s", sockname);
- if(config->sample_format.byte_format != AO_FMT_BIG) {
- info("forcing big-endian sample format");
- config->sample_format.byte_format = AO_FMT_BIG;
- }
-}
-
-/** @brief Play over the network */
-static size_t network_play(size_t frames) {
- struct rtp_header header;
- struct iovec vec[2];
- size_t bytes = frames * bpf, written_frames;
- int written_bytes;
- /* We transmit using RTP (RFC3550) and attempt to conform to the internet
- * AVT profile (RFC3551). */
-
- if(idled) {
- /* There may have been a gap. Fix up the RTP time accordingly. */
- struct timeval now;
- uint64_t delta;
- uint64_t target_rtp_time;
-
- /* Find the current time */
- xgettimeofday(&now, 0);
- /* Find the number of microseconds elapsed since rtp_time=0 */
- delta = tvsub_us(now, rtp_time_0);
- assert(delta <= UINT64_MAX / 88200);
- target_rtp_time = (delta * playing->format.rate
- * playing->format.channels) / 1000000;
- /* Overflows at ~6 years uptime with 44100Hz stereo */
-
- /* rtp_time is the number of samples we've played. NB that we play
- * RTP_AHEAD_MS ahead of ourselves, so it may legitimately be ahead of
- * the value we deduce from time comparison.
- *
- * Suppose we have 1s track started at t=0, and another track begins to
- * play at t=2s. Suppose RTP_AHEAD_MS=1000 and 44100Hz stereo. In that
- * case we'll send 1s of audio as fast as we can, giving rtp_time=88200.
- * rtp_time stops at this point.
- *
- * At t=2s we'll have calculated target_rtp_time=176400. In this case we
- * set rtp_time=176400 and the player can correctly conclude that it
- * should leave 1s between the tracks.
- *
- * Suppose instead that the second track arrives at t=0.5s, and that
- * we've managed to transmit the whole of the first track already. We'll
- * have target_rtp_time=44100.
- *
- * The desired behaviour is to play the second track back to back with
- * first. In this case therefore we do not modify rtp_time.
- *
- * Is it ever right to reduce rtp_time? No; for that would imply
- * transmitting packets with overlapping timestamp ranges, which does not
- * make sense.
- */
- target_rtp_time &= ~(uint64_t)1; /* stereo! */
- if(target_rtp_time > rtp_time) {
- /* More time has elapsed than we've transmitted samples. That implies
- * we've been 'sending' silence. */
- info("advancing rtp_time by %"PRIu64" samples",
- target_rtp_time - rtp_time);
- rtp_time = target_rtp_time;
- } else if(target_rtp_time < rtp_time) {
- const int64_t samples_ahead = ((uint64_t)RTP_AHEAD_MS
- * config->sample_format.rate
- * config->sample_format.channels
- / 1000);
-
- if(target_rtp_time + samples_ahead < rtp_time) {
- info("reversing rtp_time by %"PRIu64" samples",
- rtp_time - target_rtp_time);
- }
- }
- }
- header.vpxcc = 2 << 6; /* V=2, P=0, X=0, CC=0 */
- header.seq = htons(rtp_seq++);
- header.timestamp = htonl((uint32_t)rtp_time);
- header.ssrc = rtp_id;
- header.mpt = (idled ? 0x80 : 0x00) | 10;
- /* 10 = L16 = 16-bit x 2 x 44100KHz. We ought to deduce this value from
- * the sample rate (in a library somewhere so that configuration.c can rule
- * out invalid rates).
- */
- idled = 0;
- if(bytes > NETWORK_BYTES - sizeof header) {
- bytes = NETWORK_BYTES - sizeof header;
- /* Always send a whole number of frames */
- bytes -= bytes % bpf;
- }
- /* "The RTP clock rate used for generating the RTP timestamp is independent
- * of the number of channels and the encoding; it equals the number of
- * sampling periods per second. For N-channel encodings, each sampling
- * period (say, 1/8000 of a second) generates N samples. (This terminology
- * is standard, but somewhat confusing, as the total number of samples
- * generated per second is then the sampling rate times the channel
- * count.)"
- */
- vec[0].iov_base = (void *)&header;
- vec[0].iov_len = sizeof header;
- vec[1].iov_base = playing->buffer + playing->start;
- vec[1].iov_len = bytes;
- do {
- written_bytes = writev(bfd, vec, 2);
- } while(written_bytes < 0 && errno == EINTR);
- if(written_bytes < 0) {
- error(errno, "error transmitting audio data");
- ++audio_errors;
- if(audio_errors == 10)
- fatal(0, "too many audio errors");
- return 0;
- } else
- audio_errors /= 2;
- written_bytes -= sizeof (struct rtp_header);
- written_frames = written_bytes / bpf;
- /* Advance RTP's notion of the time */
- rtp_time += written_frames * playing->format.channels;
- return written_frames;
-}
-
-static int bfd_slot;
-
-/** @brief Set up poll array for network play */
-static void network_beforepoll(void) {
- struct timeval now;
- uint64_t target_us;
- uint64_t target_rtp_time;
- const int64_t samples_ahead = ((uint64_t)RTP_AHEAD_MS
- * config->sample_format.rate
- * config->sample_format.channels
- / 1000);
-
- /* If we're starting then initialize the base time */
- if(!rtp_time)
- xgettimeofday(&rtp_time_0, 0);
- /* We send audio data whenever we get RTP_AHEAD seconds or more
- * behind */
- xgettimeofday(&now, 0);
- target_us = tvsub_us(now, rtp_time_0);
- assert(target_us <= UINT64_MAX / 88200);
- target_rtp_time = (target_us * config->sample_format.rate
- * config->sample_format.channels)
- / 1000000;
- if((int64_t)(rtp_time - target_rtp_time) < samples_ahead)
- bfd_slot = addfd(bfd, POLLOUT);
-}
-
-/** @brief Process poll() results for network play */
-static int network_ready(void) {
- if(fds[bfd_slot].revents & (POLLOUT | POLLERR))
- return 1;
- else
- return 0;
-}
-
/** @brief Table of speaker backends */
-static const struct speaker_backend backends[] = {
+static const struct speaker_backend *backends[] = {
#if API_ALSA
- {
- BACKEND_ALSA,
- 0,
- alsa_init,
- alsa_activate,
- alsa_play,
- alsa_deactivate,
- alsa_beforepoll,
- alsa_ready
- },
+ &alsa_backend,
#endif
- {
- BACKEND_COMMAND,
- FIXED_FORMAT,
- command_init,
- 0, /* activate */
- command_play,
- 0, /* deactivate */
- command_beforepoll,
- command_ready
- },
- {
- BACKEND_NETWORK,
- FIXED_FORMAT,
- network_init,
- 0, /* activate */
- network_play,
- 0, /* deactivate */
- network_beforepoll,
- network_ready
- },
- { -1, 0, 0, 0, 0, 0, 0, 0 } /* end of list */
+ &command_backend,
+ &network_backend,
+ 0
};
/** @brief Return nonzero if we want to play some audio
/* make sure we're not root, whatever the config says */
if(getuid() == 0 || geteuid() == 0) fatal(0, "do not run as root");
/* identify the backend used to play */
- for(n = 0; backends[n].backend != -1; ++n)
- if(backends[n].backend == config->speaker_backend)
+ for(n = 0; backends[n]; ++n)
+ if(backends[n]->backend == config->speaker_backend)
break;
- if(backends[n].backend == -1)
+ if(!backends[n])
fatal(0, "unsupported backend %d", config->speaker_backend);
- backend = &backends[n];
+ backend = backends[n];
/* backend-specific initialization */
backend->init();
mainloop();
extern struct track *tracks;
extern struct track *playing;
+extern const struct speaker_backend network_backend;
+extern const struct speaker_backend alsa_backend;
+extern const struct speaker_backend command_backend;
+
+extern struct pollfd fds[NFDS];
+extern int fdno;
+extern size_t device_bpf;
+extern int idled;
+
+int addfd(int fd, int events);
+int formats_equal(const ao_sample_format *a,
+ const ao_sample_format *b);
+void abandon(void);
+
#endif /* SPEAKER_H */
/*
~mdw / disorder / commitdiff