chiark / gitweb /
split backends out into their own speaker-*.c
[disorder] / server / speaker-alsa.c
diff --git a/server/speaker-alsa.c b/server/speaker-alsa.c
new file mode 100644 (file)
index 0000000..fa9e6c3
--- /dev/null
@@ -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 <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:
+*/