audio_alsa.c

/*
 * libalsa output driver. This file is part of Shairport.
 * Copyright (c) Muffinman, Skaman 2013
 * All rights reserved.
 *
 * Permission is hereby granted, free of charge, to any person
 * obtaining a copy of this software and associated documentation
 * files (the "Software"), to deal in the Software without
 * restriction, including without limitation the rights to use,
 * copy, modify, merge, publish, distribute, sublicense, and/or
 * sell copies of the Software, and to permit persons to whom the
 * Software is furnished to do so, subject to the following conditions:
 *
 * The above copyright notice and this permission notice shall be
 * included in all copies or substantial portions of the Software.
 *
 * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
 * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES
 * OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND
 * NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT
 * HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY,
 * WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
 * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR
 * OTHER DEALINGS IN THE SOFTWARE.
 */

#define ALSA_PCM_NEW_HW_PARAMS_API

#include <stdio.h>
#include <unistd.h>
#include <memory.h>
#include <pthread.h>
#include <alsa/asoundlib.h>
#include "common.h"
#include "audio.h"

static void help(void);
static int init(int argc, char **argv);
static void deinit(void);
static void start(int sample_rate);
static void play(short buf[], int samples);
static void stop(void);
static void flush(void);
static uint32_t delay(void);
static void volume(double vol);
static void parameters(audio_parameters *info);
static int has_mute = 0;
static int has_db_vol = 0;
static double set_volume;

audio_output audio_alsa = {
    .name = "alsa",
    .help = &help,
    .init = &init,
    .deinit = &deinit,
    .start = &start,
    .stop = &stop,
    .flush = &flush,
    .delay = &delay,
    .play = &play,
    .volume = NULL,    // to be set later on...
    .parameters = NULL // to be set later on...
};

static pthread_mutex_t alsa_mutex = PTHREAD_MUTEX_INITIALIZER;

static unsigned int desired_sample_rate;

static snd_pcm_t *alsa_handle = NULL;
static snd_pcm_hw_params_t *alsa_params = NULL;

static snd_mixer_t *alsa_mix_handle = NULL;
static snd_mixer_elem_t *alsa_mix_elem = NULL;
static snd_mixer_selem_id_t *alsa_mix_sid = NULL;
static long alsa_mix_minv, alsa_mix_maxv;
static long alsa_mix_mindb, alsa_mix_maxdb;

static char *alsa_out_dev = "default";
static char *alsa_mix_dev = NULL;
static char *alsa_mix_ctrl = "Master";
static int alsa_mix_index = 0;

static int play_number;
static int64_t accumulated_delay, accumulated_da_delay;

static void help(void) {
  printf("    -d output-device    set the output device [default*|...]\n"
         "    -t mixer-type       set the mixer type [software*|hardware]\n"
         "    -m mixer-device     set the mixer device ['output-device'*|...]\n"
         "    -c mixer-control    set the mixer control [Master*|...]\n"
         "    -i mixer-index      set the mixer index [0*|...]\n"
         "    *) default option\n");
}

static int init(int argc, char **argv) {
  const char *str;
  int value;

  int hardware_mixer = 0;

  config.audio_backend_latency_offset = 0;           // this is the default for ALSA
  config.audio_backend_buffer_desired_length = 6615; // default for alsa with a software mixer
  
  

  // get settings from settings file first, allow them to be over-ridden by command line options

  if (config.cfg != NULL) {
    /* Get the desired buffer size setting. */
    if (config_lookup_int(config.cfg, "alsa.audio_backend_buffer_desired_length_software", &value)) {
      if ((value < 0) || (value > 66150))
        die("Invalid alsa audio backend buffer desired length (software) \"%d\". It should be between 0 and "
            "66150, default is 6615",
            value);
      else {
        config.audio_backend_buffer_desired_length = value;
      }
    }
    
    /* Get the latency offset. */
    if (config_lookup_int(config.cfg, "alsa.audio_backend_latency_offset", &value)) {
      if ((value < -66150) || (value > 66150))
        die("Invalid alsa audio backend buffer latency offset \"%d\". It should be between -66150 and +66150, default is 0",
            value);
      else
        config.audio_backend_latency_offset = value;
    }

    /* Get the Output Device Name. */
    if (config_lookup_string(config.cfg, "alsa.output_device", &str)) {
      alsa_out_dev = (char *)str;
    }

    /* Get the Mixer Type setting. */
    if (config_lookup_string(config.cfg, "alsa.mixer_type", &str)) {
      if (strcasecmp(str, "software") == 0)
        hardware_mixer = 0;
      else if (strcasecmp(str, "hardware") == 0)
        hardware_mixer = 1;
      else
        die("Invalid alsa mixer option choice \"%s\". It should be \"software\" or \"hardware\"");
    }

    /* Get the Mixer Device Name. */
    if (config_lookup_string(config.cfg, "alsa.mixer_device", &str)) {
      alsa_mix_dev = (char *)str;
    }

    /* Get the Mixer Control Name. */
    if (config_lookup_string(config.cfg, "alsa.mixer_control_name", &str)) {
      alsa_mix_ctrl = (char *)str;
    }
  }

  optind = 1; // optind=0 is equivalent to optind=1 plus special behaviour
  argv--;     // so we shift the arguments to satisfy getopt()
  argc++;
  // some platforms apparently require optreset = 1; - which?
  int opt;
  while ((opt = getopt(argc, argv, "d:t:m:c:i:")) > 0) {
    switch (opt) {
    case 'd':
      alsa_out_dev = optarg;
      break;
    case 't':
      if (strcmp(optarg, "hardware") == 0)
        hardware_mixer = 1;
      break;
    case 'm':
      alsa_mix_dev = optarg;
      break;
    case 'c':
      alsa_mix_ctrl = optarg;
      break;
    case 'i':
      alsa_mix_index = strtol(optarg, NULL, 10);
      break;
    default:
      help();
      die("Invalid audio option -%c specified", opt);
    }
  }

  if (optind < argc)
    die("Invalid audio argument: %s", argv[optind]);

  if (!hardware_mixer)
    return 0;

  if (alsa_mix_dev == NULL)
    alsa_mix_dev = alsa_out_dev;

  int ret = 0;

  snd_mixer_selem_id_alloca(&alsa_mix_sid);
  snd_mixer_selem_id_set_index(alsa_mix_sid, alsa_mix_index);
  snd_mixer_selem_id_set_name(alsa_mix_sid, alsa_mix_ctrl);

  if ((snd_mixer_open(&alsa_mix_handle, 0)) < 0)
    die("Failed to open mixer");
  if ((snd_mixer_attach(alsa_mix_handle, alsa_mix_dev)) < 0)
    die("Failed to attach mixer");
  if ((snd_mixer_selem_register(alsa_mix_handle, NULL, NULL)) < 0)
    die("Failed to register mixer element");

  ret = snd_mixer_load(alsa_mix_handle);
  if (ret < 0)
    die("Failed to load mixer element");
  alsa_mix_elem = snd_mixer_find_selem(alsa_mix_handle, alsa_mix_sid);
  if (!alsa_mix_elem)
    die("Failed to find mixer element");
  if (snd_mixer_selem_get_playback_volume_range(alsa_mix_elem, &alsa_mix_minv, &alsa_mix_maxv) < 0)
    debug(1, "Can't read mixer's [linear] min and max volumes.");
  else {
    if ((snd_mixer_selem_ask_playback_vol_dB(alsa_mix_elem, alsa_mix_minv, &alsa_mix_mindb) == 0) &&
        (snd_mixer_selem_ask_playback_vol_dB(alsa_mix_elem, alsa_mix_maxv, &alsa_mix_maxdb) == 0)) {
      has_db_vol = 1;
      audio_alsa.volume = &volume; // insert the volume function now we know it can do dB stuff
      audio_alsa.parameters = &parameters; // likewise the parameters stuff
      if (alsa_mix_mindb == -9999999) {
        // trying to say that the lowest vol is mute, maybe? Raspberry Pi does this
        if (snd_mixer_selem_ask_playback_vol_dB(alsa_mix_elem, alsa_mix_minv + 1,
                                                &alsa_mix_mindb) == 0)
          debug(1, "Can't get dB value corresponding to a \"volume\" of 1.");
      }
      debug(1, "Hardware mixer has dB volume from %f to %f.", (1.0 * alsa_mix_mindb) / 100.0,
            (1.0 * alsa_mix_maxdb) / 100.0);
    } else {
      debug(1, "Hardware mixer does not have dB volume -- not used.");
    }
  }
  if (snd_mixer_selem_has_playback_switch(alsa_mix_elem)) {
    has_mute = 1;
    debug(1, "Has mute ability.");
  }
  return 0;
}

static void deinit(void) {
  stop();
  if (alsa_mix_handle) {
    snd_mixer_close(alsa_mix_handle);
  }
}

int open_alsa_device(void) {

  int ret, dir = 0;
  unsigned int my_sample_rate = desired_sample_rate;
  snd_pcm_uframes_t frames = 441 * 10;
  snd_pcm_uframes_t buffer_size = frames * 4;
  ret = snd_pcm_open(&alsa_handle, alsa_out_dev, SND_PCM_STREAM_PLAYBACK, 0);
  if (ret < 0)
    return (ret);
  // die("Alsa initialization failed: unable to open pcm device: %s.", snd_strerror(ret));

  snd_pcm_hw_params_alloca(&alsa_params);
  snd_pcm_hw_params_any(alsa_handle, alsa_params);
  snd_pcm_hw_params_set_access(alsa_handle, alsa_params, SND_PCM_ACCESS_RW_INTERLEAVED);
  snd_pcm_hw_params_set_format(alsa_handle, alsa_params, SND_PCM_FORMAT_S16);
  snd_pcm_hw_params_set_channels(alsa_handle, alsa_params, 2);
  snd_pcm_hw_params_set_rate_near(alsa_handle, alsa_params, &my_sample_rate, &dir);
  // snd_pcm_hw_params_set_period_size_near(alsa_handle, alsa_params, &frames, &dir);
  // snd_pcm_hw_params_set_buffer_size_near(alsa_handle, alsa_params, &buffer_size);
  ret = snd_pcm_hw_params(alsa_handle, alsa_params);
  if (ret < 0) {
    die("unable to set hw parameters: %s.", snd_strerror(ret));
  }
  if (my_sample_rate != desired_sample_rate) {
    die("Can't set the D/A converter to %d -- set to %d instead./n", desired_sample_rate,
        my_sample_rate);
  }
  return (0);
}

static void start(int sample_rate) {
  if (sample_rate != 44100)
    die("Unexpected sample rate %d -- only 44,100 supported!", sample_rate);
  desired_sample_rate = sample_rate; // must be a variable
}

static uint32_t delay() {
  if (alsa_handle == NULL) {
    return 0;
  } else {
    pthread_mutex_lock(&alsa_mutex);
    snd_pcm_sframes_t current_avail, current_delay = 0;
    int derr, ignore;
    if (snd_pcm_state(alsa_handle) == SND_PCM_STATE_RUNNING) {
      derr = snd_pcm_avail_delay(alsa_handle, &current_avail, &current_delay);
      // current_avail not used
      if (derr != 0) {
        ignore = snd_pcm_recover(alsa_handle, derr, 0);
        debug(1, "Error %d in delay(): %s. Delay reported is %d frames.", derr, snd_strerror(derr),
              current_delay);
        current_delay = -1;
      }
    } else if (snd_pcm_state(alsa_handle) == SND_PCM_STATE_PREPARED) {
      current_delay = 0;
    } else {
      if (snd_pcm_state(alsa_handle) == SND_PCM_STATE_XRUN)
        current_delay = 0;
      else {
        current_delay = -1;
        debug(1, "Error -- ALSA delay(): bad state: %d.", snd_pcm_state(alsa_handle));
      }
      if (derr = snd_pcm_prepare(alsa_handle)) {
        ignore = snd_pcm_recover(alsa_handle, derr, 0);
        debug(1, "Error preparing after delay error: %s.", snd_strerror(derr));
        current_delay = -1;
      }
    }
    pthread_mutex_unlock(&alsa_mutex);
    return current_delay;
  }
}

static void play(short buf[], int samples) {
  int ret = 0;
  if (alsa_handle == NULL) {
    ret = open_alsa_device();
    if ((ret == 0) && (audio_alsa.volume))
      volume(set_volume);
  }
  if (ret == 0) {
    pthread_mutex_lock(&alsa_mutex);
    snd_pcm_sframes_t current_delay = 0;
    int err, ignore;
    if ((snd_pcm_state(alsa_handle) == SND_PCM_STATE_PREPARED) ||
        (snd_pcm_state(alsa_handle) == SND_PCM_STATE_RUNNING)) {
      err = snd_pcm_writei(alsa_handle, (char *)buf, samples);
      if (err < 0) {
        ignore = snd_pcm_recover(alsa_handle, err, 0);
        debug(1, "Error %d writing %d samples in play() %s.", err, samples, snd_strerror(err));
      }
    } else {
      debug(1, "Error -- ALSA device in incorrect state (%d) for play.",
            snd_pcm_state(alsa_handle));
      if (err = snd_pcm_prepare(alsa_handle)) {
        ignore = snd_pcm_recover(alsa_handle, err, 0);
        debug(1, "Error preparing after play error: %s.", snd_strerror(err));
      }
    }
    pthread_mutex_unlock(&alsa_mutex);
  }
}

static void flush(void) {
  int derr;
  if (alsa_handle) {
    // debug(1,"Dropping frames for flush...");
    if (derr = snd_pcm_drop(alsa_handle))
      debug(1, "Error dropping frames: %s.", snd_strerror(derr));
    // debug(1,"Dropped frames ok. State is %d.",snd_pcm_state(alsa_handle));
    if (derr = snd_pcm_prepare(alsa_handle))
      debug(1, "Error preparing after flush: %s.", snd_strerror(derr));
    // debug(1,"Frames successfully dropped.");
    /*
    if (snd_pcm_state(alsa_handle)==SND_PCM_STATE_PREPARED)
      debug(1,"Flush returns to SND_PCM_STATE_PREPARED state.");
    if (snd_pcm_state(alsa_handle)==SND_PCM_STATE_RUNNING)
      debug(1,"Flush returns to SND_PCM_STATE_RUNNING state.");
    */
    if (!((snd_pcm_state(alsa_handle) == SND_PCM_STATE_PREPARED) ||
          (snd_pcm_state(alsa_handle) == SND_PCM_STATE_RUNNING)))
      debug(1, "Flush returning unexpected state -- %d.", snd_pcm_state(alsa_handle));

    // flush also closes the device
    snd_pcm_close(alsa_handle);
    alsa_handle = NULL;
  }
}

static void stop(void) {
  if (alsa_handle != 0)
    // when we want to stop, we want the alsa device
    // to be closed immediately -- we may even be killing the thread, so we don't wish to wait
    // so we should flush first
    flush(); // flush will also close the device
             // close_alsa_device();
}

static void parameters(audio_parameters *info) {
  info->has_true_mute = has_mute;
  info->is_muted = ((has_mute) && (set_volume == -144.0));
  info->minimum_volume_dB = alsa_mix_mindb;
  info->maximum_volume_dB = alsa_mix_maxdb;
  info->airplay_volume = set_volume;
  info->current_volume_dB = vol2attn(set_volume, alsa_mix_maxdb, alsa_mix_mindb);
}

static void volume(double vol) {
  set_volume = vol;
  double vol_setting = vol2attn(vol, alsa_mix_maxdb, alsa_mix_mindb);
  // debug(1,"Setting volume db to %f, for volume input of %f.",vol_setting/100,vol);
  if (snd_mixer_selem_set_playback_dB_all(alsa_mix_elem, vol_setting, -1) != 0)
    die("Failed to set playback dB volume");
  if (has_mute)
    snd_mixer_selem_set_playback_switch_all(alsa_mix_elem, (vol != -144.0));
}