572 lines
20 KiB
C
572 lines
20 KiB
C
/*
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* libalsa output driver. This file is part of Shairport.
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* Copyright (c) Muffinman, Skaman 2013
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* All rights reserved.
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*
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* Permission is hereby granted, free of charge, to any person
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* obtaining a copy of this software and associated documentation
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* files (the "Software"), to deal in the Software without
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* restriction, including without limitation the rights to use,
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* copy, modify, merge, publish, distribute, sublicense, and/or
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* sell copies of the Software, and to permit persons to whom the
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* Software is furnished to do so, subject to the following conditions:
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*
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* The above copyright notice and this permission notice shall be
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* included in all copies or substantial portions of the Software.
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*
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
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* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES
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* OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND
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* NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT
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* HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY,
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* WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
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* FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR
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* OTHER DEALINGS IN THE SOFTWARE.
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*/
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#define ALSA_PCM_NEW_HW_PARAMS_API
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#include <stdio.h>
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#include <unistd.h>
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#include <memory.h>
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#include <math.h>
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#include <pthread.h>
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#include <alsa/asoundlib.h>
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#include "common.h"
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#include "audio.h"
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static void help(void);
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static int init(int argc, char **argv);
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static void deinit(void);
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static void start(int sample_rate);
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static void play(short buf[], int samples);
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static void stop(void);
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static void flush(void);
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static uint32_t delay(void);
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static void volume(double vol);
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static void linear_volume(double vol);
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static void parameters(audio_parameters *info);
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static void mute(int do_mute);
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static double set_volume;
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audio_output audio_alsa = {
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.name = "alsa",
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.help = &help,
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.init = &init,
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.deinit = &deinit,
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.start = &start,
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.stop = &stop,
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.flush = &flush,
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.delay = &delay,
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.play = &play,
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.mute = NULL, // to be set later on...
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.volume = NULL, // to be set later on...
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.parameters = NULL // to be set later on...
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};
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static pthread_mutex_t alsa_mutex = PTHREAD_MUTEX_INITIALIZER;
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static unsigned int desired_sample_rate;
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static snd_pcm_t *alsa_handle = NULL;
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static snd_pcm_hw_params_t *alsa_params = NULL;
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static snd_mixer_t *alsa_mix_handle = NULL;
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static snd_mixer_elem_t *alsa_mix_elem = NULL;
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static snd_mixer_selem_id_t *alsa_mix_sid = NULL;
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static long alsa_mix_minv, alsa_mix_maxv;
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static long alsa_mix_mindb, alsa_mix_maxdb;
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static char *alsa_out_dev = "default";
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static char *alsa_mix_dev = NULL;
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static char *alsa_mix_ctrl = "Master";
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static int alsa_mix_index = 0;
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static int hardware_mixer = 0;
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static int play_number;
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static int64_t accumulated_delay, accumulated_da_delay;
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static void help(void) {
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printf(" -d output-device set the output device [default*|...]\n"
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" -m mixer-device set the mixer device ['output-device'*|...]\n"
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" -c mixer-control set the mixer control [Master*|...]\n"
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" -i mixer-index set the mixer index [0*|...]\n"
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" *) default option\n");
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}
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int open_mixer() {
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if (hardware_mixer) {
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debug(2, "Open Mixer");
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int ret = 0;
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snd_mixer_selem_id_alloca(&alsa_mix_sid);
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snd_mixer_selem_id_set_index(alsa_mix_sid, alsa_mix_index);
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snd_mixer_selem_id_set_name(alsa_mix_sid, alsa_mix_ctrl);
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if ((snd_mixer_open(&alsa_mix_handle, 0)) < 0)
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die("Failed to open mixer");
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debug(3, "Mixer device name is \"%s\".", alsa_mix_dev);
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if ((snd_mixer_attach(alsa_mix_handle, alsa_mix_dev)) < 0)
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die("Failed to attach mixer");
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if ((snd_mixer_selem_register(alsa_mix_handle, NULL, NULL)) < 0)
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die("Failed to register mixer element");
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ret = snd_mixer_load(alsa_mix_handle);
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if (ret < 0)
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die("Failed to load mixer element");
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debug(3, "Mixer Control name is \"%s\".", alsa_mix_ctrl);
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alsa_mix_elem = snd_mixer_find_selem(alsa_mix_handle, alsa_mix_sid);
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if (!alsa_mix_elem)
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die("Failed to find mixer element");
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}
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}
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static int init(int argc, char **argv) {
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pthread_mutex_lock(&alsa_mutex);
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// debug(2,"audio_alsa init called.");
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const char *str;
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int value;
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config.audio_backend_latency_offset = 0; // this is the default for ALSA
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config.audio_backend_buffer_desired_length =
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6615; // default for alsa with a software mixer
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// get settings from settings file first, allow them to be overridden by
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// command line options
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if (config.cfg != NULL) {
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/* Get the desired buffer size setting. */
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if (config_lookup_int(config.cfg,
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"alsa.audio_backend_buffer_desired_length", &value)) {
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if ((value < 0) || (value > 66150))
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die("Invalid alsa audio backend buffer desired length \"%d\". It "
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"should be between 0 and "
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"66150, default is 6615",
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value);
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else {
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config.audio_backend_buffer_desired_length = value;
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}
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}
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/* Get the latency offset. */
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if (config_lookup_int(config.cfg, "alsa.audio_backend_latency_offset",
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&value)) {
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if ((value < -66150) || (value > 66150))
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die("Invalid alsa audio backend buffer latency offset \"%d\". It "
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"should be between -66150 and +66150, default is 0",
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value);
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else
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config.audio_backend_latency_offset = value;
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}
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/* Get the Output Device Name. */
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if (config_lookup_string(config.cfg, "alsa.output_device", &str)) {
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alsa_out_dev = (char *)str;
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}
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/* Get the Mixer Type setting. */
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if (config_lookup_string(config.cfg, "alsa.mixer_type", &str)) {
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inform("The alsa mixer_type setting is deprecated and has been ignored. "
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"FYI, using the \"mixer_control_name\" setting automatically "
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"chooses a hardware mixer.");
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}
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/* Get the Mixer Device Name. */
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if (config_lookup_string(config.cfg, "alsa.mixer_device", &str)) {
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alsa_mix_dev = (char *)str;
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}
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/* Get the Mixer Control Name. */
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if (config_lookup_string(config.cfg, "alsa.mixer_control_name", &str)) {
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alsa_mix_ctrl = (char *)str;
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hardware_mixer = 1;
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}
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}
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optind = 1; // optind=0 is equivalent to optind=1 plus special behaviour
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argv--; // so we shift the arguments to satisfy getopt()
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argc++;
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// some platforms apparently require optreset = 1; - which?
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int opt;
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while ((opt = getopt(argc, argv, "d:t:m:c:i:")) > 0) {
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switch (opt) {
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case 'd':
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alsa_out_dev = optarg;
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break;
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case 't':
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inform("The alsa backend -t option is deprecated and has been ignored. "
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"FYI, using the -c option automatically chooses a hardware "
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"mixer.");
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break;
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case 'm':
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alsa_mix_dev = optarg;
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break;
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case 'c':
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alsa_mix_ctrl = optarg;
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hardware_mixer = 1;
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break;
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case 'i':
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alsa_mix_index = strtol(optarg, NULL, 10);
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break;
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default:
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help();
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die("Invalid audio option -%c specified", opt);
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}
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}
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if (optind < argc)
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die("Invalid audio argument: %s", argv[optind]);
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debug(1, "Output device name is \"%s\".", alsa_out_dev);
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if (hardware_mixer) {
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if (alsa_mix_dev == NULL)
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alsa_mix_dev = alsa_out_dev;
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// Open mixer
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open_mixer();
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if (snd_mixer_selem_get_playback_volume_range(alsa_mix_elem, &alsa_mix_minv,
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&alsa_mix_maxv) < 0)
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debug(1, "Can't read mixer's [linear] min and max volumes.");
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else {
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if (snd_mixer_selem_get_playback_dB_range(alsa_mix_elem, &alsa_mix_mindb,
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&alsa_mix_maxdb) == 0) {
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audio_alsa.volume =
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&volume; // insert the volume function now we know it can do dB stuff
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audio_alsa.parameters = ¶meters; // likewise the parameters stuff
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if (alsa_mix_mindb == SND_CTL_TLV_DB_GAIN_MUTE) {
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// Raspberry Pi does this
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debug(1, "Lowest dB value is a mute.");
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if (snd_mixer_selem_ask_playback_vol_dB(
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alsa_mix_elem, alsa_mix_minv + 1, &alsa_mix_mindb) == 0)
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debug(1, "Can't get dB value corresponding to a \"volume\" of 1.");
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}
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debug(1, "Hardware mixer has dB volume from %f to %f.",
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(1.0 * alsa_mix_mindb) / 100.0, (1.0 * alsa_mix_maxdb) / 100.0);
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} else {
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// use the linear scale and do the db conversion ourselves
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debug(1, "note: the hardware mixer specified -- \"%s\" -- does not have "
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"a dB volume scale, so it can't be used.",
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alsa_mix_ctrl);
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/*
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debug(1, "Min and max volumes are %d and
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%d.",alsa_mix_minv,alsa_mix_maxv);
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alsa_mix_maxdb = 0;
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if ((alsa_mix_maxv!=0) && (alsa_mix_minv!=0))
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alsa_mix_mindb =
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-20*100*(log10(alsa_mix_maxv*1.0)-log10(alsa_mix_minv*1.0));
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else if (alsa_mix_maxv!=0)
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alsa_mix_mindb = -20*100*log10(alsa_mix_maxv*1.0);
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audio_alsa.volume = &linear_volume; // insert the linear volume function
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audio_alsa.parameters = ¶meters; // likewise the parameters stuff
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debug(1,"Max and min dB calculated are %d and
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%d.",alsa_mix_maxdb,alsa_mix_mindb);
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*/
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}
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}
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if (snd_mixer_selem_has_playback_switch(alsa_mix_elem)) {
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audio_alsa.mute =
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&mute; // insert the mute function now we know it can do muting stuff
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debug(1, "Has mute ability.");
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}
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snd_mixer_close(alsa_mix_handle);
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}
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alsa_mix_handle = NULL;
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pthread_mutex_unlock(&alsa_mutex);
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return 0;
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}
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static void deinit(void) {
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// debug(2,"audio_alsa deinit called.");
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stop();
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if (hardware_mixer && alsa_mix_handle) {
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snd_mixer_close(alsa_mix_handle);
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}
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}
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int open_alsa_device(void) {
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const snd_pcm_uframes_t minimal_buffer_headroom =
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352 * 2; // we accept this much headroom in the hardware buffer, but we'll
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// accept less
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const snd_pcm_uframes_t requested_buffer_headroom =
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minimal_buffer_headroom + 2048; // we ask for this much headroom in the
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// hardware buffer, but we'll accept less
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int ret, dir = 0;
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unsigned int my_sample_rate = desired_sample_rate;
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// snd_pcm_uframes_t frames = 441 * 10;
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snd_pcm_uframes_t buffer_size, actual_buffer_length;
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ret = snd_pcm_open(&alsa_handle, alsa_out_dev, SND_PCM_STREAM_PLAYBACK, 0);
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if (ret < 0)
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return (ret);
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// die("Alsa initialization failed: unable to open pcm device: %s.",
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// snd_strerror(ret));
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snd_pcm_hw_params_alloca(&alsa_params);
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ret = snd_pcm_hw_params_any(alsa_handle, alsa_params);
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if (ret < 0) {
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die("audio_alsa: Broken configuration for device \"%s\": no configurations "
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"available",
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alsa_out_dev);
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}
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ret = snd_pcm_hw_params_set_access(alsa_handle, alsa_params,
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SND_PCM_ACCESS_RW_INTERLEAVED);
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if (ret < 0) {
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die("audio_alsa: Access type not available for device \"%s\": %s",
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alsa_out_dev, snd_strerror(ret));
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}
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ret = snd_pcm_hw_params_set_format(alsa_handle, alsa_params,
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SND_PCM_FORMAT_S16);
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if (ret < 0) {
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die("audio_alsa: Sample format not available for device \"%s\": %s",
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alsa_out_dev, snd_strerror(ret));
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}
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ret = snd_pcm_hw_params_set_channels(alsa_handle, alsa_params, 2);
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if (ret < 0) {
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die("audio_alsa: Channels count (2) not available for device \"%s\": %s",
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alsa_out_dev, snd_strerror(ret));
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}
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ret = snd_pcm_hw_params_set_rate_near(alsa_handle, alsa_params,
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&my_sample_rate, &dir);
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if (ret < 0) {
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die("audio_alsa: Rate %iHz not available for playback: %s",
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desired_sample_rate, snd_strerror(ret));
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}
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// snd_pcm_hw_params_set_period_size_near(alsa_handle, alsa_params, &frames,
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// &dir);
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// snd_pcm_hw_params_set_buffer_size_near(alsa_handle, alsa_params,
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// &buffer_size);
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ret = snd_pcm_hw_params(alsa_handle, alsa_params);
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if (ret < 0) {
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die("audio_alsa: Unable to set hw parameters for device \"%s\": %s.",
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alsa_out_dev, snd_strerror(ret));
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}
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if (my_sample_rate != desired_sample_rate) {
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die("Can't set the D/A converter to %d.", desired_sample_rate);
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}
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ret = snd_pcm_hw_params_get_buffer_size(alsa_params, &actual_buffer_length);
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if (ret < 0) {
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die("audio_alsa: Unable to get hw buffer length for device \"%s\": %s.",
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alsa_out_dev, snd_strerror(ret));
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}
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if (actual_buffer_length <
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config.audio_backend_buffer_desired_length + minimal_buffer_headroom) {
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// the dac buffer is too small, so let's try to set it
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buffer_size =
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config.audio_backend_buffer_desired_length + requested_buffer_headroom;
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ret = snd_pcm_hw_params_set_buffer_size_near(alsa_handle, alsa_params,
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&buffer_size);
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if (ret < 0)
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die("audio_alsa: Unable to set hw buffer size to %lu for device \"%s\": "
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"%s.",
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config.audio_backend_buffer_desired_length +
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requested_buffer_headroom,
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alsa_out_dev, snd_strerror(ret));
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if (config.audio_backend_buffer_desired_length + minimal_buffer_headroom >
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buffer_size) {
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die("audio_alsa: Can't set hw buffer size to %lu or more for device "
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"\"%s\". Requested size: %lu, granted size: %lu.",
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config.audio_backend_buffer_desired_length + minimal_buffer_headroom,
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alsa_out_dev, config.audio_backend_buffer_desired_length +
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requested_buffer_headroom,
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buffer_size);
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}
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}
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return (0);
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}
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static void start(int sample_rate) {
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// debug(2,"audio_alsa start called.");
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if (sample_rate != 44100)
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die("Unexpected sample rate %d -- only 44,100 supported!", sample_rate);
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desired_sample_rate = sample_rate; // must be a variable
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}
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static uint32_t delay() {
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// debug(3,"audio_alsa delay called.");
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if (alsa_handle == NULL) {
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return 0;
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} else {
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pthread_mutex_lock(&alsa_mutex);
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snd_pcm_sframes_t current_avail, current_delay = 0;
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int derr, ignore;
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if (snd_pcm_state(alsa_handle) == SND_PCM_STATE_RUNNING) {
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derr = snd_pcm_avail_delay(alsa_handle, ¤t_avail, ¤t_delay);
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// current_avail not used
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if (derr != 0) {
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ignore = snd_pcm_recover(alsa_handle, derr, 0);
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debug(1, "Error %d in delay(): %s. Delay reported is %d frames.", derr,
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snd_strerror(derr), current_delay);
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current_delay = -1;
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}
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} else if (snd_pcm_state(alsa_handle) == SND_PCM_STATE_PREPARED) {
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current_delay = 0;
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} else {
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if (snd_pcm_state(alsa_handle) == SND_PCM_STATE_XRUN)
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current_delay = 0;
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else {
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current_delay = -1;
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debug(1, "Error -- ALSA delay(): bad state: %d.",
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snd_pcm_state(alsa_handle));
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}
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if ((derr = snd_pcm_prepare(alsa_handle))) {
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ignore = snd_pcm_recover(alsa_handle, derr, 0);
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debug(1, "Error preparing after delay error: %s.", snd_strerror(derr));
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current_delay = -1;
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}
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}
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pthread_mutex_unlock(&alsa_mutex);
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return current_delay;
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}
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}
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static void play(short buf[], int samples) {
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// debug(3,"audio_alsa play called.");
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int ret = 0;
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if (alsa_handle == NULL) {
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pthread_mutex_lock(&alsa_mutex);
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ret = open_alsa_device();
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open_mixer();
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pthread_mutex_unlock(&alsa_mutex);
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if ((ret == 0) && (audio_alsa.volume))
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audio_alsa.volume(set_volume);
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}
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if (ret == 0) {
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pthread_mutex_lock(&alsa_mutex);
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snd_pcm_sframes_t current_delay = 0;
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int err, ignore;
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if ((snd_pcm_state(alsa_handle) == SND_PCM_STATE_PREPARED) ||
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(snd_pcm_state(alsa_handle) == SND_PCM_STATE_RUNNING)) {
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err = snd_pcm_writei(alsa_handle, (char *)buf, samples);
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if (err < 0) {
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ignore = snd_pcm_recover(alsa_handle, err, 0);
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debug(1, "Error %d writing %d samples in play() %s.", err, samples,
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snd_strerror(err));
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}
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} else {
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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) {
|
|
// debug(2,"audio_alsa flush called.");
|
|
pthread_mutex_lock(&alsa_mutex);
|
|
int derr;
|
|
if (hardware_mixer && alsa_mix_handle) {
|
|
snd_mixer_close(alsa_mix_handle);
|
|
alsa_mix_handle = NULL;
|
|
}
|
|
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;
|
|
}
|
|
pthread_mutex_unlock(&alsa_mutex);
|
|
}
|
|
|
|
static void stop(void) {
|
|
// debug(2,"audio_alsa stop called.");
|
|
// 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->minimum_volume_dB = alsa_mix_mindb;
|
|
info->maximum_volume_dB = alsa_mix_maxdb;
|
|
}
|
|
|
|
static void volume(double vol) {
|
|
pthread_mutex_lock(&alsa_mutex);
|
|
debug(2, "Setting volume db to %f.", vol);
|
|
set_volume = vol;
|
|
if (hardware_mixer && alsa_mix_handle) {
|
|
if (snd_mixer_selem_set_playback_dB_all(alsa_mix_elem, vol, 0) != 0) {
|
|
debug(1, "Can't set playback volume accurately to %f dB.", vol);
|
|
if (snd_mixer_selem_set_playback_dB_all(alsa_mix_elem, vol, -1) != 0)
|
|
if (snd_mixer_selem_set_playback_dB_all(alsa_mix_elem, vol, 1) != 0)
|
|
die("Failed to set playback dB volume");
|
|
}
|
|
}
|
|
pthread_mutex_unlock(&alsa_mutex);
|
|
}
|
|
|
|
static void linear_volume(double vol) {
|
|
debug(2, "Setting linear volume to %f.", vol);
|
|
set_volume = vol;
|
|
if (hardware_mixer && alsa_mix_handle) {
|
|
double linear_volume = pow(10, vol);
|
|
// debug(1,"Linear volume is %f.",linear_volume);
|
|
long int_vol =
|
|
alsa_mix_minv + (alsa_mix_maxv - alsa_mix_minv) * linear_volume;
|
|
// debug(1,"Setting volume to %ld, for volume input of %f.",int_vol,vol);
|
|
if (alsa_mix_handle) {
|
|
if (snd_mixer_selem_set_playback_volume_all(alsa_mix_elem, int_vol) != 0)
|
|
die("Failed to set playback volume");
|
|
}
|
|
}
|
|
}
|
|
|
|
static void mute(int do_mute) {
|
|
pthread_mutex_lock(&alsa_mutex);
|
|
// debug(2,"audio_alsa mute called.");
|
|
if (hardware_mixer && alsa_mix_handle) {
|
|
if (do_mute) {
|
|
// debug(1,"Mute");
|
|
snd_mixer_selem_set_playback_switch_all(alsa_mix_elem, 0);
|
|
} else {
|
|
// debug(1,"Unmute");
|
|
snd_mixer_selem_set_playback_switch_all(alsa_mix_elem, 1);
|
|
}
|
|
}
|
|
pthread_mutex_unlock(&alsa_mutex);
|
|
}
|