From 1c3f1e73710d27fb5ae5b4c15798d03e89e74363 Mon Sep 17 00:00:00 2001 Message-Id: <1c3f1e73710d27fb5ae5b4c15798d03e89e74363.1713962442.git.mdw@distorted.org.uk> From: Mark Wooding Date: Mon, 24 Sep 2007 16:02:49 +0100 Subject: [PATCH] split backends out into their own speaker-*.c Organization: Straylight/Edgeware From: rjk@greenend.org.uk <> --- server/Makefile.am | 5 +- server/speaker-alsa.c | 287 ++++++++++++++++++ server/speaker-command.c | 130 ++++++++ server/speaker-network.c | 313 +++++++++++++++++++ server/speaker.c | 640 ++------------------------------------- server/speaker.h | 14 + 6 files changed, 775 insertions(+), 614 deletions(-) create mode 100644 server/speaker-alsa.c create mode 100644 server/speaker-command.c create mode 100644 server/speaker-network.c diff --git a/server/Makefile.am b/server/Makefile.am index 290393b..92f3de2 100644 --- a/server/Makefile.am +++ b/server/Makefile.am @@ -45,7 +45,10 @@ disorder_deadlock_LDADD=$(LIBOBJS) ../lib/libdisorder.a \ $(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 diff --git a/server/speaker-alsa.c b/server/speaker-alsa.c new file mode 100644 index 0000000..fa9e6c3 --- /dev/null +++ b/server/speaker-alsa.c @@ -0,0 +1,287 @@ +/* + * 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 + +#if API_ALSA + +#include "types.h" + +#include +#include +#include + +#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: +*/ diff --git a/server/speaker-command.c b/server/speaker-command.c new file mode 100644 index 0000000..8a8f63c --- /dev/null +++ b/server/speaker-command.c @@ -0,0 +1,130 @@ +/* + * 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 +#include "types.h" + +#include +#include + +#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: +*/ diff --git a/server/speaker-network.c b/server/speaker-network.c new file mode 100644 index 0000000..3d3bbc0 --- /dev/null +++ b/server/speaker-network.c @@ -0,0 +1,313 @@ +/* + * 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 +#include "types.h" + +#include +#include +#include +#include +#include +#include +#include + +#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: +*/ diff --git a/server/speaker.c b/server/speaker.c index aebd930..1dc90e4 100644 --- a/server/speaker.c +++ b/server/speaker.c @@ -72,10 +72,6 @@ #include #include #include -#include -#include -#include -#include #include "configuration.h" #include "syscalls.h" @@ -84,31 +80,25 @@ #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 -#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; @@ -121,52 +111,11 @@ 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; @@ -258,8 +207,8 @@ static void acquire(struct track *t, int fd) { } /** @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 @@ -419,7 +368,7 @@ static void idle(void) { } /** @brief Abandon the current track */ -static void abandon(void) { +void abandon(void) { struct speaker_message sm; D(("abandon")); @@ -453,7 +402,7 @@ static void activate(void) { 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 @@ -498,7 +447,7 @@ static void play(size_t frames) { 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, @@ -510,7 +459,7 @@ static void play(size_t frames) { 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; @@ -518,7 +467,7 @@ static void play(size_t 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; @@ -556,7 +505,7 @@ static void reap(int __attribute__((unused)) sig) { 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; @@ -565,549 +514,14 @@ static int addfd(int fd, int 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 @@ -1299,12 +713,12 @@ int main(int argc, char **argv) { /* 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(); diff --git a/server/speaker.h b/server/speaker.h index 8bc73a3..53230cb 100644 --- a/server/speaker.h +++ b/server/speaker.h @@ -207,6 +207,20 @@ extern ao_sample_format device_format; 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]