2009-03-13 18:43:16 +01:00
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/*
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2011-01-29 10:13:54 +01:00
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* Copyright (C) 2003-2011 The Music Player Daemon Project
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2009-03-13 18:43:16 +01:00
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* http://www.musicpd.org
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2005-03-05 06:22:30 +01:00
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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2009-03-13 18:43:16 +01:00
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*
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* You should have received a copy of the GNU General Public License along
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* with this program; if not, write to the Free Software Foundation, Inc.,
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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2005-03-05 06:22:30 +01:00
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*/
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2009-11-12 09:12:38 +01:00
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#include "config.h"
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2011-09-17 08:54:28 +02:00
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#include "alsa_output_plugin.h"
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2009-11-12 09:12:38 +01:00
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#include "output_api.h"
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2009-03-14 11:36:59 +01:00
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#include "mixer_list.h"
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2012-03-21 19:54:35 +01:00
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#include "pcm_export.h"
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2005-03-05 06:22:30 +01:00
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2008-11-01 14:04:14 +01:00
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#include <glib.h>
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2008-10-26 21:58:37 +01:00
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#include <alsa/asoundlib.h>
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2005-03-05 06:22:30 +01:00
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2008-12-29 17:29:42 +01:00
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#undef G_LOG_DOMAIN
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#define G_LOG_DOMAIN "alsa"
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2005-03-05 06:22:30 +01:00
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#define ALSA_PCM_NEW_HW_PARAMS_API
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#define ALSA_PCM_NEW_SW_PARAMS_API
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2008-09-08 20:42:39 +02:00
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static const char default_device[] = "default";
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2008-12-01 22:37:05 +01:00
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enum {
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MPD_ALSA_BUFFER_TIME_US = 500000,
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};
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2006-07-24 03:38:51 +02:00
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#define MPD_ALSA_RETRY_NR 5
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2005-03-05 06:22:30 +01:00
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2006-07-20 18:02:40 +02:00
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typedef snd_pcm_sframes_t alsa_writei_t(snd_pcm_t * pcm, const void *buffer,
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snd_pcm_uframes_t size);
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2005-03-05 06:22:30 +01:00
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2009-01-25 13:05:16 +01:00
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struct alsa_data {
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2011-09-16 23:31:48 +02:00
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struct audio_output base;
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2012-03-21 19:54:35 +01:00
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struct pcm_export_state export;
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2012-03-21 19:09:22 +01:00
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2009-01-25 13:13:24 +01:00
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/** the configured name of the ALSA device; NULL for the
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default device */
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2008-11-01 14:04:14 +01:00
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char *device;
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2008-10-14 17:21:49 +02:00
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2009-01-25 13:13:24 +01:00
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/** use memory mapped I/O? */
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bool use_mmap;
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2012-03-21 10:36:19 +01:00
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/**
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* Enable DSD over USB according to the dCS suggested
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* standard?
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*
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* @see http://www.dcsltd.co.uk/page/assets/DSDoverUSB.pdf
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*/
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bool dsd_usb;
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2009-01-25 13:13:24 +01:00
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/** libasound's buffer_time setting (in microseconds) */
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unsigned int buffer_time;
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/** libasound's period_time setting (in microseconds) */
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unsigned int period_time;
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2008-10-14 17:21:49 +02:00
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/** the mode flags passed to snd_pcm_open */
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int mode;
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2009-01-25 13:13:24 +01:00
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/** the libasound PCM device handle */
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2009-01-25 13:05:16 +01:00
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snd_pcm_t *pcm;
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2009-01-25 13:13:24 +01:00
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/**
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* a pointer to the libasound writei() function, which is
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* snd_pcm_writei() or snd_pcm_mmap_writei(), depending on the
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* use_mmap configuration
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*/
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2006-07-20 18:02:40 +02:00
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alsa_writei_t *writei;
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2009-01-25 13:13:24 +01:00
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2012-03-22 01:01:11 +01:00
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/**
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* The size of one audio frame passed to method play().
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*/
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size_t in_frame_size;
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/**
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* The size of one audio frame passed to libasound.
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*/
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size_t out_frame_size;
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2009-11-09 22:22:31 +01:00
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/**
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* The size of one period, in number of frames.
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*/
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snd_pcm_uframes_t period_frames;
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/**
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* The number of frames written in the current period.
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*/
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snd_pcm_uframes_t period_position;
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2009-01-25 13:05:16 +01:00
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};
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2005-03-05 06:22:30 +01:00
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2009-02-26 22:04:59 +01:00
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/**
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* The quark used for GError.domain.
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*/
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static inline GQuark
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alsa_output_quark(void)
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{
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return g_quark_from_static_string("alsa_output");
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}
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2008-11-01 14:04:14 +01:00
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static const char *
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2009-01-25 13:05:16 +01:00
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alsa_device(const struct alsa_data *ad)
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2008-11-01 14:04:14 +01:00
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{
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return ad->device != NULL ? ad->device : default_device;
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}
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2009-01-25 13:05:16 +01:00
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static struct alsa_data *
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alsa_data_new(void)
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2006-07-20 18:02:40 +02:00
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{
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2009-01-25 13:05:16 +01:00
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struct alsa_data *ret = g_new(struct alsa_data, 1);
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2005-03-05 06:22:30 +01:00
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2008-10-14 17:21:49 +02:00
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ret->mode = 0;
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2005-03-05 06:22:30 +01:00
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ret->writei = snd_pcm_writei;
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return ret;
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}
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2008-10-12 11:47:33 +02:00
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static void
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2009-01-25 16:03:49 +01:00
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alsa_configure(struct alsa_data *ad, const struct config_param *param)
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2008-10-12 11:47:33 +02:00
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{
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2009-01-18 19:37:27 +01:00
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ad->device = config_dup_block_string(param, "device", NULL);
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2008-11-01 14:04:14 +01:00
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2009-01-25 13:05:16 +01:00
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ad->use_mmap = config_get_block_bool(param, "use_mmap", false);
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2009-01-17 20:23:56 +01:00
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2012-03-21 10:36:19 +01:00
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ad->dsd_usb = config_get_block_bool(param, "dsd_usb", false);
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2009-01-25 12:52:37 +01:00
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ad->buffer_time = config_get_block_unsigned(param, "buffer_time",
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MPD_ALSA_BUFFER_TIME_US);
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2009-03-08 03:55:01 +01:00
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ad->period_time = config_get_block_unsigned(param, "period_time", 0);
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2008-10-14 17:21:49 +02:00
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2008-10-14 22:37:27 +02:00
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#ifdef SND_PCM_NO_AUTO_RESAMPLE
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2009-01-17 20:23:56 +01:00
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if (!config_get_block_bool(param, "auto_resample", true))
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2008-10-14 17:21:49 +02:00
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ad->mode |= SND_PCM_NO_AUTO_RESAMPLE;
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2008-10-14 22:37:27 +02:00
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#endif
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2008-10-14 17:21:49 +02:00
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2008-10-14 22:37:27 +02:00
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#ifdef SND_PCM_NO_AUTO_CHANNELS
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2009-01-17 20:23:56 +01:00
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if (!config_get_block_bool(param, "auto_channels", true))
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2008-10-14 17:21:49 +02:00
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ad->mode |= SND_PCM_NO_AUTO_CHANNELS;
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2008-10-14 22:37:27 +02:00
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#endif
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2008-10-14 17:21:49 +02:00
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2008-10-14 22:37:27 +02:00
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#ifdef SND_PCM_NO_AUTO_FORMAT
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2009-01-17 20:23:56 +01:00
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if (!config_get_block_bool(param, "auto_format", true))
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2008-10-14 17:21:49 +02:00
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ad->mode |= SND_PCM_NO_AUTO_FORMAT;
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2008-10-14 22:37:27 +02:00
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#endif
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2008-10-12 11:47:33 +02:00
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}
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2011-09-16 23:31:48 +02:00
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static struct audio_output *
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alsa_init(const struct config_param *param, GError **error_r)
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2006-07-20 18:02:40 +02:00
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{
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2009-01-25 13:05:16 +01:00
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struct alsa_data *ad = alsa_data_new();
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2006-07-16 18:52:19 +02:00
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2011-09-16 23:31:48 +02:00
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if (!ao_base_init(&ad->base, &alsa_output_plugin, param, error_r)) {
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g_free(ad);
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return NULL;
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}
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2009-01-25 16:04:03 +01:00
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alsa_configure(ad, param);
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2005-03-05 06:22:30 +01:00
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2011-09-16 23:31:48 +02:00
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return &ad->base;
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2005-03-05 06:22:30 +01:00
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}
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2009-01-25 13:05:16 +01:00
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static void
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2011-09-16 23:31:48 +02:00
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alsa_finish(struct audio_output *ao)
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2006-07-20 18:02:40 +02:00
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{
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2011-09-16 23:31:48 +02:00
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struct alsa_data *ad = (struct alsa_data *)ao;
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2005-03-05 06:22:30 +01:00
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2011-09-16 23:31:48 +02:00
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ao_base_finish(&ad->base);
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2012-03-21 18:51:31 +01:00
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g_free(ad->device);
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g_free(ad);
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2009-04-21 22:47:12 +02:00
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/* free libasound's config cache */
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snd_config_update_free_global();
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2005-03-05 06:22:30 +01:00
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}
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2012-03-21 19:09:22 +01:00
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static bool
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alsa_output_enable(struct audio_output *ao, G_GNUC_UNUSED GError **error_r)
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{
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struct alsa_data *ad = (struct alsa_data *)ao;
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2012-03-21 19:54:35 +01:00
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pcm_export_init(&ad->export);
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2012-03-21 19:09:22 +01:00
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return true;
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}
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static void
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alsa_output_disable(struct audio_output *ao)
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{
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struct alsa_data *ad = (struct alsa_data *)ao;
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2012-03-21 19:54:35 +01:00
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pcm_export_deinit(&ad->export);
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2012-03-21 19:09:22 +01:00
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}
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2009-01-25 13:05:16 +01:00
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static bool
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alsa_test_default_device(void)
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2005-03-12 04:10:09 +01:00
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{
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2006-07-20 18:02:40 +02:00
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snd_pcm_t *handle;
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2005-03-12 04:10:09 +01:00
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2008-09-08 20:42:39 +02:00
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int ret = snd_pcm_open(&handle, default_device,
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SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
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2006-07-20 18:02:40 +02:00
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if (ret) {
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2008-12-29 17:29:42 +01:00
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g_message("Error opening default ALSA device: %s\n",
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snd_strerror(-ret));
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2008-10-29 20:40:27 +01:00
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return false;
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2006-07-20 18:02:40 +02:00
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} else
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snd_pcm_close(handle);
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2005-03-12 04:10:09 +01:00
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2008-10-29 20:40:27 +01:00
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return true;
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2005-03-12 04:10:09 +01:00
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}
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2009-01-25 13:05:16 +01:00
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static snd_pcm_format_t
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2010-01-16 17:45:56 +01:00
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get_bitformat(enum sample_format sample_format)
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2008-09-08 20:42:51 +02:00
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{
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2010-01-16 17:45:56 +01:00
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switch (sample_format) {
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2011-10-08 14:37:54 +02:00
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case SAMPLE_FORMAT_UNDEFINED:
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2012-03-01 00:25:08 +01:00
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case SAMPLE_FORMAT_DSD:
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2011-10-08 14:37:54 +02:00
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return SND_PCM_FORMAT_UNKNOWN;
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2009-11-10 17:11:34 +01:00
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case SAMPLE_FORMAT_S8:
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return SND_PCM_FORMAT_S8;
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case SAMPLE_FORMAT_S16:
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return SND_PCM_FORMAT_S16;
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case SAMPLE_FORMAT_S24_P32:
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return SND_PCM_FORMAT_S24;
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case SAMPLE_FORMAT_S32:
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return SND_PCM_FORMAT_S32;
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2011-10-08 10:25:06 +02:00
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case SAMPLE_FORMAT_FLOAT:
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return SND_PCM_FORMAT_FLOAT;
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2008-09-08 20:42:51 +02:00
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}
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2011-10-08 14:37:54 +02:00
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assert(false);
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return SND_PCM_FORMAT_UNKNOWN;
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2008-09-08 20:42:51 +02:00
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}
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2009-07-19 17:43:08 +02:00
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static snd_pcm_format_t
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byteswap_bitformat(snd_pcm_format_t fmt)
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{
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switch(fmt) {
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case SND_PCM_FORMAT_S16_LE: return SND_PCM_FORMAT_S16_BE;
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case SND_PCM_FORMAT_S24_LE: return SND_PCM_FORMAT_S24_BE;
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case SND_PCM_FORMAT_S32_LE: return SND_PCM_FORMAT_S32_BE;
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case SND_PCM_FORMAT_S16_BE: return SND_PCM_FORMAT_S16_LE;
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case SND_PCM_FORMAT_S24_BE: return SND_PCM_FORMAT_S24_LE;
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2010-01-16 18:35:10 +01:00
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case SND_PCM_FORMAT_S24_3BE:
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return SND_PCM_FORMAT_S24_3LE;
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case SND_PCM_FORMAT_S24_3LE:
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return SND_PCM_FORMAT_S24_3BE;
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2009-07-19 17:43:08 +02:00
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case SND_PCM_FORMAT_S32_BE: return SND_PCM_FORMAT_S32_LE;
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default: return SND_PCM_FORMAT_UNKNOWN;
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}
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}
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2010-01-16 17:14:30 +01:00
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2012-03-21 23:59:31 +01:00
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static snd_pcm_format_t
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alsa_to_packed_format(snd_pcm_format_t fmt)
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{
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switch (fmt) {
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case SND_PCM_FORMAT_S24_LE:
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return SND_PCM_FORMAT_S24_3LE;
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case SND_PCM_FORMAT_S24_BE:
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return SND_PCM_FORMAT_S24_3BE;
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default:
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return SND_PCM_FORMAT_UNKNOWN;
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}
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}
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static int
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alsa_try_format_or_packed(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
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snd_pcm_format_t fmt, bool *packed_r)
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{
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int err = snd_pcm_hw_params_set_format(pcm, hwparams, fmt);
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|
|
if (err == 0)
|
|
|
|
*packed_r = false;
|
|
|
|
|
|
|
|
if (err != -EINVAL)
|
|
|
|
return err;
|
|
|
|
|
|
|
|
fmt = alsa_to_packed_format(fmt);
|
|
|
|
if (fmt == SND_PCM_FORMAT_UNKNOWN)
|
|
|
|
return -EINVAL;
|
|
|
|
|
|
|
|
err = snd_pcm_hw_params_set_format(pcm, hwparams, fmt);
|
|
|
|
if (err == 0)
|
|
|
|
*packed_r = true;
|
|
|
|
|
|
|
|
return err;
|
|
|
|
}
|
|
|
|
|
2010-01-16 17:41:28 +01:00
|
|
|
/**
|
2012-03-22 00:23:07 +01:00
|
|
|
* Attempts to configure the specified sample format, and tries the
|
|
|
|
* reversed host byte order if was not supported.
|
2010-01-16 17:41:28 +01:00
|
|
|
*/
|
|
|
|
static int
|
|
|
|
alsa_output_try_format(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
|
2012-03-22 00:24:56 +01:00
|
|
|
enum sample_format sample_format,
|
2012-03-21 23:59:31 +01:00
|
|
|
bool *packed_r, bool *reverse_endian_r)
|
2010-01-16 17:41:28 +01:00
|
|
|
{
|
|
|
|
snd_pcm_format_t alsa_format = get_bitformat(sample_format);
|
|
|
|
if (alsa_format == SND_PCM_FORMAT_UNKNOWN)
|
|
|
|
return -EINVAL;
|
|
|
|
|
2012-03-21 23:59:31 +01:00
|
|
|
int err = alsa_try_format_or_packed(pcm, hwparams, alsa_format,
|
|
|
|
packed_r);
|
2012-03-22 00:24:56 +01:00
|
|
|
if (err == 0)
|
2012-03-22 00:23:07 +01:00
|
|
|
*reverse_endian_r = false;
|
2010-01-16 17:41:28 +01:00
|
|
|
|
2012-03-22 00:23:07 +01:00
|
|
|
if (err != -EINVAL)
|
|
|
|
return err;
|
2010-01-16 17:41:28 +01:00
|
|
|
|
2012-03-22 00:23:07 +01:00
|
|
|
alsa_format = byteswap_bitformat(alsa_format);
|
2010-01-16 17:41:28 +01:00
|
|
|
if (alsa_format == SND_PCM_FORMAT_UNKNOWN)
|
|
|
|
return -EINVAL;
|
|
|
|
|
2012-03-21 23:59:31 +01:00
|
|
|
err = alsa_try_format_or_packed(pcm, hwparams, alsa_format, packed_r);
|
2012-03-22 00:24:56 +01:00
|
|
|
if (err == 0)
|
2012-03-21 19:09:22 +01:00
|
|
|
*reverse_endian_r = true;
|
2010-01-16 18:08:13 +01:00
|
|
|
|
|
|
|
return err;
|
|
|
|
}
|
|
|
|
|
2009-02-25 22:01:30 +01:00
|
|
|
/**
|
2010-01-16 17:14:30 +01:00
|
|
|
* Configure a sample format, and probe other formats if that fails.
|
2009-02-25 22:01:30 +01:00
|
|
|
*/
|
2010-01-16 17:14:30 +01:00
|
|
|
static int
|
|
|
|
alsa_output_setup_format(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
|
2012-03-21 19:09:22 +01:00
|
|
|
struct audio_format *audio_format,
|
2012-03-21 23:59:31 +01:00
|
|
|
bool *packed_r, bool *reverse_endian_r)
|
2005-03-05 06:22:30 +01:00
|
|
|
{
|
2010-01-16 18:08:13 +01:00
|
|
|
/* try the input format first */
|
2005-03-05 06:22:30 +01:00
|
|
|
|
2012-03-22 00:24:56 +01:00
|
|
|
int err = alsa_output_try_format(pcm, hwparams, audio_format->format,
|
2012-03-21 23:59:31 +01:00
|
|
|
packed_r, reverse_endian_r);
|
2010-01-16 17:14:30 +01:00
|
|
|
|
2010-01-16 18:08:13 +01:00
|
|
|
/* if unsupported by the hardware, try other formats */
|
2010-01-16 17:14:30 +01:00
|
|
|
|
2010-01-16 18:08:13 +01:00
|
|
|
static const enum sample_format probe_formats[] = {
|
|
|
|
SAMPLE_FORMAT_S24_P32,
|
|
|
|
SAMPLE_FORMAT_S32,
|
|
|
|
SAMPLE_FORMAT_S16,
|
|
|
|
SAMPLE_FORMAT_S8,
|
|
|
|
SAMPLE_FORMAT_UNDEFINED,
|
|
|
|
};
|
2009-03-02 16:41:38 +01:00
|
|
|
|
2012-03-22 00:29:56 +01:00
|
|
|
for (unsigned i = 0;
|
|
|
|
err == -EINVAL && probe_formats[i] != SAMPLE_FORMAT_UNDEFINED;
|
|
|
|
++i) {
|
2012-03-22 00:24:56 +01:00
|
|
|
const enum sample_format mpd_format = probe_formats[i];
|
|
|
|
if (mpd_format == audio_format->format)
|
2010-01-16 18:08:13 +01:00
|
|
|
continue;
|
2010-01-16 17:41:28 +01:00
|
|
|
|
2012-03-22 00:24:56 +01:00
|
|
|
err = alsa_output_try_format(pcm, hwparams, mpd_format,
|
2012-03-21 23:59:31 +01:00
|
|
|
packed_r, reverse_endian_r);
|
2012-03-22 00:24:56 +01:00
|
|
|
if (err == 0)
|
|
|
|
audio_format->format = mpd_format;
|
2009-07-19 17:43:08 +02:00
|
|
|
}
|
2008-10-12 12:02:55 +02:00
|
|
|
|
2012-03-22 00:29:56 +01:00
|
|
|
return err;
|
2010-01-16 17:14:30 +01:00
|
|
|
}
|
|
|
|
|
|
|
|
/**
|
|
|
|
* Set up the snd_pcm_t object which was opened by the caller. Set up
|
|
|
|
* the configured settings and the audio format.
|
|
|
|
*/
|
|
|
|
static bool
|
|
|
|
alsa_setup(struct alsa_data *ad, struct audio_format *audio_format,
|
2012-03-27 00:32:10 +02:00
|
|
|
bool *packed_r, bool *reverse_endian_r, GError **error)
|
2010-01-16 17:14:30 +01:00
|
|
|
{
|
|
|
|
snd_pcm_hw_params_t *hwparams;
|
|
|
|
snd_pcm_sw_params_t *swparams;
|
|
|
|
unsigned int sample_rate = audio_format->sample_rate;
|
|
|
|
unsigned int channels = audio_format->channels;
|
|
|
|
snd_pcm_uframes_t alsa_buffer_size;
|
|
|
|
snd_pcm_uframes_t alsa_period_size;
|
|
|
|
int err;
|
|
|
|
const char *cmd = NULL;
|
|
|
|
int retry = MPD_ALSA_RETRY_NR;
|
|
|
|
unsigned int period_time, period_time_ro;
|
|
|
|
unsigned int buffer_time;
|
|
|
|
|
|
|
|
period_time_ro = period_time = ad->period_time;
|
|
|
|
configure_hw:
|
|
|
|
/* configure HW params */
|
|
|
|
snd_pcm_hw_params_alloca(&hwparams);
|
|
|
|
cmd = "snd_pcm_hw_params_any";
|
|
|
|
err = snd_pcm_hw_params_any(ad->pcm, hwparams);
|
|
|
|
if (err < 0)
|
|
|
|
goto error;
|
|
|
|
|
|
|
|
if (ad->use_mmap) {
|
|
|
|
err = snd_pcm_hw_params_set_access(ad->pcm, hwparams,
|
|
|
|
SND_PCM_ACCESS_MMAP_INTERLEAVED);
|
|
|
|
if (err < 0) {
|
|
|
|
g_warning("Cannot set mmap'ed mode on ALSA device \"%s\": %s\n",
|
|
|
|
alsa_device(ad), snd_strerror(-err));
|
|
|
|
g_warning("Falling back to direct write mode\n");
|
|
|
|
ad->use_mmap = false;
|
|
|
|
} else
|
|
|
|
ad->writei = snd_pcm_mmap_writei;
|
|
|
|
}
|
|
|
|
|
|
|
|
if (!ad->use_mmap) {
|
|
|
|
cmd = "snd_pcm_hw_params_set_access";
|
|
|
|
err = snd_pcm_hw_params_set_access(ad->pcm, hwparams,
|
|
|
|
SND_PCM_ACCESS_RW_INTERLEAVED);
|
|
|
|
if (err < 0)
|
|
|
|
goto error;
|
|
|
|
ad->writei = snd_pcm_writei;
|
|
|
|
}
|
|
|
|
|
2012-03-21 19:09:22 +01:00
|
|
|
err = alsa_output_setup_format(ad->pcm, hwparams, audio_format,
|
2012-03-27 00:32:10 +02:00
|
|
|
packed_r, reverse_endian_r);
|
2006-07-20 18:02:40 +02:00
|
|
|
if (err < 0) {
|
2009-02-26 22:04:59 +01:00
|
|
|
g_set_error(error, alsa_output_quark(), err,
|
2009-11-10 17:11:34 +01:00
|
|
|
"ALSA device \"%s\" does not support format %s: %s",
|
|
|
|
alsa_device(ad),
|
|
|
|
sample_format_to_string(audio_format->format),
|
2009-02-26 22:04:59 +01:00
|
|
|
snd_strerror(-err));
|
2009-02-25 22:01:30 +01:00
|
|
|
return false;
|
2005-03-05 06:22:30 +01:00
|
|
|
}
|
|
|
|
|
2012-03-26 23:49:16 +02:00
|
|
|
snd_pcm_format_t format;
|
|
|
|
if (snd_pcm_hw_params_get_format(hwparams, &format) == 0)
|
|
|
|
g_debug("format=%s (%s)", snd_pcm_format_name(format),
|
|
|
|
snd_pcm_format_description(format));
|
|
|
|
|
2009-01-25 13:05:16 +01:00
|
|
|
err = snd_pcm_hw_params_set_channels_near(ad->pcm, hwparams,
|
2006-07-20 18:02:40 +02:00
|
|
|
&channels);
|
|
|
|
if (err < 0) {
|
2009-02-26 22:04:59 +01:00
|
|
|
g_set_error(error, alsa_output_quark(), err,
|
|
|
|
"ALSA device \"%s\" does not support %i channels: %s",
|
|
|
|
alsa_device(ad), (int)audio_format->channels,
|
|
|
|
snd_strerror(-err));
|
2009-02-25 22:01:30 +01:00
|
|
|
return false;
|
2005-03-05 06:22:30 +01:00
|
|
|
}
|
2009-01-25 13:05:16 +01:00
|
|
|
audio_format->channels = (int8_t)channels;
|
2005-03-05 06:22:30 +01:00
|
|
|
|
2009-01-25 13:05:16 +01:00
|
|
|
err = snd_pcm_hw_params_set_rate_near(ad->pcm, hwparams,
|
2008-10-10 14:40:54 +02:00
|
|
|
&sample_rate, NULL);
|
|
|
|
if (err < 0 || sample_rate == 0) {
|
2009-02-26 22:04:59 +01:00
|
|
|
g_set_error(error, alsa_output_quark(), err,
|
|
|
|
"ALSA device \"%s\" does not support %u Hz audio",
|
|
|
|
alsa_device(ad), audio_format->sample_rate);
|
2009-02-25 22:01:30 +01:00
|
|
|
return false;
|
2005-03-05 06:22:30 +01:00
|
|
|
}
|
2009-01-25 13:05:16 +01:00
|
|
|
audio_format->sample_rate = sample_rate;
|
2005-03-05 06:22:30 +01:00
|
|
|
|
2010-11-05 08:25:21 +01:00
|
|
|
snd_pcm_uframes_t buffer_size_min, buffer_size_max;
|
|
|
|
snd_pcm_hw_params_get_buffer_size_min(hwparams, &buffer_size_min);
|
|
|
|
snd_pcm_hw_params_get_buffer_size_max(hwparams, &buffer_size_max);
|
|
|
|
unsigned buffer_time_min, buffer_time_max;
|
|
|
|
snd_pcm_hw_params_get_buffer_time_min(hwparams, &buffer_time_min, 0);
|
|
|
|
snd_pcm_hw_params_get_buffer_time_max(hwparams, &buffer_time_max, 0);
|
|
|
|
g_debug("buffer: size=%u..%u time=%u..%u",
|
|
|
|
(unsigned)buffer_size_min, (unsigned)buffer_size_max,
|
|
|
|
buffer_time_min, buffer_time_max);
|
|
|
|
|
|
|
|
snd_pcm_uframes_t period_size_min, period_size_max;
|
|
|
|
snd_pcm_hw_params_get_period_size_min(hwparams, &period_size_min, 0);
|
|
|
|
snd_pcm_hw_params_get_period_size_max(hwparams, &period_size_max, 0);
|
|
|
|
unsigned period_time_min, period_time_max;
|
|
|
|
snd_pcm_hw_params_get_period_time_min(hwparams, &period_time_min, 0);
|
|
|
|
snd_pcm_hw_params_get_period_time_max(hwparams, &period_time_max, 0);
|
|
|
|
g_debug("period: size=%u..%u time=%u..%u",
|
|
|
|
(unsigned)period_size_min, (unsigned)period_size_max,
|
|
|
|
period_time_min, period_time_max);
|
|
|
|
|
2008-10-11 12:52:48 +02:00
|
|
|
if (ad->buffer_time > 0) {
|
|
|
|
buffer_time = ad->buffer_time;
|
|
|
|
cmd = "snd_pcm_hw_params_set_buffer_time_near";
|
2009-01-25 13:05:16 +01:00
|
|
|
err = snd_pcm_hw_params_set_buffer_time_near(ad->pcm, hwparams,
|
2008-10-11 12:52:48 +02:00
|
|
|
&buffer_time, NULL);
|
|
|
|
if (err < 0)
|
|
|
|
goto error;
|
2009-03-08 04:11:30 +01:00
|
|
|
} else {
|
|
|
|
err = snd_pcm_hw_params_get_buffer_time(hwparams, &buffer_time,
|
|
|
|
NULL);
|
|
|
|
if (err < 0)
|
|
|
|
buffer_time = 0;
|
2008-10-11 12:52:48 +02:00
|
|
|
}
|
2006-07-16 18:52:06 +02:00
|
|
|
|
2009-03-08 03:55:01 +01:00
|
|
|
if (period_time_ro == 0 && buffer_time >= 10000) {
|
|
|
|
period_time_ro = period_time = buffer_time / 4;
|
|
|
|
|
|
|
|
g_debug("default period_time = buffer_time/4 = %u/4 = %u",
|
|
|
|
buffer_time, period_time);
|
|
|
|
}
|
|
|
|
|
2008-10-11 12:52:48 +02:00
|
|
|
if (period_time_ro > 0) {
|
|
|
|
period_time = period_time_ro;
|
|
|
|
cmd = "snd_pcm_hw_params_set_period_time_near";
|
2009-01-25 13:05:16 +01:00
|
|
|
err = snd_pcm_hw_params_set_period_time_near(ad->pcm, hwparams,
|
2008-10-11 12:52:48 +02:00
|
|
|
&period_time, NULL);
|
|
|
|
if (err < 0)
|
|
|
|
goto error;
|
|
|
|
}
|
2005-03-05 06:22:30 +01:00
|
|
|
|
2005-03-13 17:42:04 +01:00
|
|
|
cmd = "snd_pcm_hw_params";
|
2009-01-25 13:05:16 +01:00
|
|
|
err = snd_pcm_hw_params(ad->pcm, hwparams);
|
2008-10-11 12:52:48 +02:00
|
|
|
if (err == -EPIPE && --retry > 0 && period_time_ro > 0) {
|
2006-08-12 20:20:55 +02:00
|
|
|
period_time_ro = period_time_ro >> 1;
|
2006-07-24 03:38:51 +02:00
|
|
|
goto configure_hw;
|
|
|
|
} else if (err < 0)
|
2006-07-20 18:02:40 +02:00
|
|
|
goto error;
|
2006-07-24 03:38:51 +02:00
|
|
|
if (retry != MPD_ALSA_RETRY_NR)
|
2008-12-29 17:29:42 +01:00
|
|
|
g_debug("ALSA period_time set to %d\n", period_time);
|
2005-03-05 06:22:30 +01:00
|
|
|
|
2005-03-13 17:42:04 +01:00
|
|
|
cmd = "snd_pcm_hw_params_get_buffer_size";
|
2005-03-05 06:22:30 +01:00
|
|
|
err = snd_pcm_hw_params_get_buffer_size(hwparams, &alsa_buffer_size);
|
2006-07-20 18:02:40 +02:00
|
|
|
if (err < 0)
|
|
|
|
goto error;
|
2005-03-05 06:22:30 +01:00
|
|
|
|
2005-03-13 17:42:04 +01:00
|
|
|
cmd = "snd_pcm_hw_params_get_period_size";
|
2006-07-17 02:15:52 +02:00
|
|
|
err = snd_pcm_hw_params_get_period_size(hwparams, &alsa_period_size,
|
2006-07-20 18:02:40 +02:00
|
|
|
NULL);
|
|
|
|
if (err < 0)
|
|
|
|
goto error;
|
2005-03-05 06:22:30 +01:00
|
|
|
|
2005-03-05 06:45:39 +01:00
|
|
|
/* configure SW params */
|
2005-03-05 06:22:30 +01:00
|
|
|
snd_pcm_sw_params_alloca(&swparams);
|
|
|
|
|
2005-03-13 17:42:04 +01:00
|
|
|
cmd = "snd_pcm_sw_params_current";
|
2009-01-25 13:05:16 +01:00
|
|
|
err = snd_pcm_sw_params_current(ad->pcm, swparams);
|
2006-07-20 18:02:40 +02:00
|
|
|
if (err < 0)
|
|
|
|
goto error;
|
2005-03-13 17:42:04 +01:00
|
|
|
|
|
|
|
cmd = "snd_pcm_sw_params_set_start_threshold";
|
2009-01-25 13:05:16 +01:00
|
|
|
err = snd_pcm_sw_params_set_start_threshold(ad->pcm, swparams,
|
2006-07-20 18:02:40 +02:00
|
|
|
alsa_buffer_size -
|
|
|
|
alsa_period_size);
|
|
|
|
if (err < 0)
|
|
|
|
goto error;
|
2005-03-05 06:22:30 +01:00
|
|
|
|
2005-03-13 17:42:04 +01:00
|
|
|
cmd = "snd_pcm_sw_params_set_avail_min";
|
2009-01-25 13:05:16 +01:00
|
|
|
err = snd_pcm_sw_params_set_avail_min(ad->pcm, swparams,
|
2006-07-20 18:02:40 +02:00
|
|
|
alsa_period_size);
|
|
|
|
if (err < 0)
|
|
|
|
goto error;
|
2005-03-13 17:42:04 +01:00
|
|
|
|
|
|
|
cmd = "snd_pcm_sw_params";
|
2009-01-25 13:05:16 +01:00
|
|
|
err = snd_pcm_sw_params(ad->pcm, swparams);
|
2006-07-20 18:02:40 +02:00
|
|
|
if (err < 0)
|
|
|
|
goto error;
|
|
|
|
|
2009-03-03 22:19:37 +01:00
|
|
|
g_debug("buffer_size=%u period_size=%u",
|
|
|
|
(unsigned)alsa_buffer_size, (unsigned)alsa_period_size);
|
|
|
|
|
2011-07-20 06:54:51 +02:00
|
|
|
if (alsa_period_size == 0)
|
|
|
|
/* this works around a SIGFPE bug that occurred when
|
|
|
|
an ALSA driver indicated period_size==0; this
|
|
|
|
caused a division by zero in alsa_play(). By using
|
|
|
|
the fallback "1", we make sure that this won't
|
|
|
|
happen again. */
|
|
|
|
alsa_period_size = 1;
|
|
|
|
|
2009-11-09 22:22:31 +01:00
|
|
|
ad->period_frames = alsa_period_size;
|
|
|
|
ad->period_position = 0;
|
|
|
|
|
2008-10-29 20:40:27 +01:00
|
|
|
return true;
|
2005-03-05 06:22:30 +01:00
|
|
|
|
2006-08-20 12:13:54 +02:00
|
|
|
error:
|
2009-02-26 22:04:59 +01:00
|
|
|
g_set_error(error, alsa_output_quark(), err,
|
|
|
|
"Error opening ALSA device \"%s\" (%s): %s",
|
|
|
|
alsa_device(ad), cmd, snd_strerror(-err));
|
2009-02-25 22:01:30 +01:00
|
|
|
return false;
|
|
|
|
}
|
|
|
|
|
2012-03-21 10:36:19 +01:00
|
|
|
static bool
|
|
|
|
alsa_setup_dsd(struct alsa_data *ad, struct audio_format *audio_format,
|
2012-03-27 01:05:33 +02:00
|
|
|
bool *shift8_r, bool *packed_r, bool *reverse_endian_r,
|
|
|
|
GError **error_r)
|
2012-03-21 10:36:19 +01:00
|
|
|
{
|
|
|
|
assert(ad->dsd_usb);
|
|
|
|
assert(audio_format->format == SAMPLE_FORMAT_DSD);
|
|
|
|
|
|
|
|
/* pass 24 bit to alsa_setup() */
|
|
|
|
|
2012-03-27 00:22:47 +02:00
|
|
|
struct audio_format usb_format = *audio_format;
|
|
|
|
usb_format.format = SAMPLE_FORMAT_S24_P32;
|
|
|
|
usb_format.sample_rate /= 2;
|
2012-03-21 10:36:19 +01:00
|
|
|
|
2012-03-27 00:22:47 +02:00
|
|
|
const struct audio_format check = usb_format;
|
2012-03-21 10:36:19 +01:00
|
|
|
|
2012-03-27 00:22:47 +02:00
|
|
|
if (!alsa_setup(ad, &usb_format, packed_r, reverse_endian_r, error_r))
|
2012-03-21 10:36:19 +01:00
|
|
|
return false;
|
|
|
|
|
2012-03-27 01:05:33 +02:00
|
|
|
/* if the device allows only 32 bit, shift all DSD-over-USB
|
|
|
|
samples left by 8 bit and leave the lower 8 bit cleared;
|
|
|
|
the DSD-over-USB documentation does not specify whether
|
|
|
|
this is legal, but there is anecdotical evidence that this
|
|
|
|
is possible (and the only option for some devices) */
|
|
|
|
*shift8_r = usb_format.format == SAMPLE_FORMAT_S32;
|
|
|
|
if (usb_format.format == SAMPLE_FORMAT_S32)
|
|
|
|
usb_format.format = SAMPLE_FORMAT_S24_P32;
|
|
|
|
|
2012-03-27 00:22:47 +02:00
|
|
|
if (!audio_format_equals(&usb_format, &check)) {
|
2012-03-21 10:36:19 +01:00
|
|
|
/* no bit-perfect playback, which is required
|
|
|
|
for DSD over USB */
|
|
|
|
g_set_error(error_r, alsa_output_quark(), 0,
|
|
|
|
"Failed to configure DSD-over-USB on ALSA device \"%s\"",
|
|
|
|
alsa_device(ad));
|
|
|
|
return false;
|
|
|
|
}
|
|
|
|
|
|
|
|
return true;
|
|
|
|
}
|
|
|
|
|
|
|
|
static bool
|
|
|
|
alsa_setup_or_dsd(struct alsa_data *ad, struct audio_format *audio_format,
|
|
|
|
GError **error_r)
|
|
|
|
{
|
2012-03-27 01:05:33 +02:00
|
|
|
bool shift8 = false, packed, reverse_endian;
|
2012-03-21 10:36:19 +01:00
|
|
|
|
2012-03-27 00:22:47 +02:00
|
|
|
const bool dsd_usb = ad->dsd_usb &&
|
|
|
|
audio_format->format == SAMPLE_FORMAT_DSD;
|
|
|
|
const bool success = dsd_usb
|
2012-03-27 01:05:33 +02:00
|
|
|
? alsa_setup_dsd(ad, audio_format,
|
|
|
|
&shift8, &packed, &reverse_endian,
|
2012-03-27 00:32:10 +02:00
|
|
|
error_r)
|
|
|
|
: alsa_setup(ad, audio_format, &packed, &reverse_endian,
|
|
|
|
error_r);
|
|
|
|
if (!success)
|
|
|
|
return false;
|
|
|
|
|
2012-03-27 00:03:44 +02:00
|
|
|
pcm_export_open(&ad->export,
|
|
|
|
audio_format->format, audio_format->channels,
|
2012-03-27 01:05:33 +02:00
|
|
|
dsd_usb, shift8, packed, reverse_endian);
|
2012-03-27 00:32:10 +02:00
|
|
|
return true;
|
2012-03-21 10:36:19 +01:00
|
|
|
}
|
|
|
|
|
2009-02-25 22:01:30 +01:00
|
|
|
static bool
|
2011-09-16 23:31:48 +02:00
|
|
|
alsa_open(struct audio_output *ao, struct audio_format *audio_format, GError **error)
|
2009-02-25 22:01:30 +01:00
|
|
|
{
|
2011-09-16 23:31:48 +02:00
|
|
|
struct alsa_data *ad = (struct alsa_data *)ao;
|
2009-02-25 22:01:30 +01:00
|
|
|
int err;
|
|
|
|
bool success;
|
|
|
|
|
|
|
|
err = snd_pcm_open(&ad->pcm, alsa_device(ad),
|
|
|
|
SND_PCM_STREAM_PLAYBACK, ad->mode);
|
|
|
|
if (err < 0) {
|
2009-02-26 22:04:59 +01:00
|
|
|
g_set_error(error, alsa_output_quark(), err,
|
|
|
|
"Failed to open ALSA device \"%s\": %s",
|
|
|
|
alsa_device(ad), snd_strerror(err));
|
2009-02-25 22:01:30 +01:00
|
|
|
return false;
|
2005-03-13 17:42:04 +01:00
|
|
|
}
|
2009-02-25 22:01:30 +01:00
|
|
|
|
2012-03-26 23:49:16 +02:00
|
|
|
g_debug("opened %s type=%s", snd_pcm_name(ad->pcm),
|
|
|
|
snd_pcm_type_name(snd_pcm_type(ad->pcm)));
|
|
|
|
|
2012-03-21 10:36:19 +01:00
|
|
|
success = alsa_setup_or_dsd(ad, audio_format, error);
|
2009-02-25 22:01:30 +01:00
|
|
|
if (!success) {
|
2009-01-25 13:05:16 +01:00
|
|
|
snd_pcm_close(ad->pcm);
|
2009-02-25 22:01:30 +01:00
|
|
|
return false;
|
|
|
|
}
|
|
|
|
|
2012-03-22 01:01:11 +01:00
|
|
|
ad->in_frame_size = audio_format_frame_size(audio_format);
|
|
|
|
ad->out_frame_size = ad->export.pack24 ? 3 : ad->in_frame_size;
|
2009-02-25 22:01:30 +01:00
|
|
|
|
|
|
|
return true;
|
2005-03-05 06:22:30 +01:00
|
|
|
}
|
|
|
|
|
2009-01-25 13:05:16 +01:00
|
|
|
static int
|
|
|
|
alsa_recover(struct alsa_data *ad, int err)
|
2006-07-20 18:02:40 +02:00
|
|
|
{
|
|
|
|
if (err == -EPIPE) {
|
2008-12-29 17:29:42 +01:00
|
|
|
g_debug("Underrun on ALSA device \"%s\"\n", alsa_device(ad));
|
2006-07-20 18:02:40 +02:00
|
|
|
} else if (err == -ESTRPIPE) {
|
2008-12-29 17:29:42 +01:00
|
|
|
g_debug("ALSA device \"%s\" was suspended\n", alsa_device(ad));
|
2005-03-05 15:01:13 +01:00
|
|
|
}
|
|
|
|
|
2009-01-25 13:05:16 +01:00
|
|
|
switch (snd_pcm_state(ad->pcm)) {
|
2005-03-05 21:51:36 +01:00
|
|
|
case SND_PCM_STATE_PAUSED:
|
2009-01-25 13:05:16 +01:00
|
|
|
err = snd_pcm_pause(ad->pcm, /* disable */ 0);
|
2005-03-05 21:51:36 +01:00
|
|
|
break;
|
|
|
|
case SND_PCM_STATE_SUSPENDED:
|
2009-01-25 13:05:16 +01:00
|
|
|
err = snd_pcm_resume(ad->pcm);
|
2008-09-08 09:45:05 +02:00
|
|
|
if (err == -EAGAIN)
|
|
|
|
return 0;
|
|
|
|
/* fall-through to snd_pcm_prepare: */
|
2005-03-05 15:01:13 +01:00
|
|
|
case SND_PCM_STATE_SETUP:
|
|
|
|
case SND_PCM_STATE_XRUN:
|
2009-11-09 22:22:31 +01:00
|
|
|
ad->period_position = 0;
|
2009-01-25 13:05:16 +01:00
|
|
|
err = snd_pcm_prepare(ad->pcm);
|
2005-03-05 21:51:36 +01:00
|
|
|
break;
|
2006-07-16 18:52:29 +02:00
|
|
|
case SND_PCM_STATE_DISCONNECTED:
|
|
|
|
break;
|
2008-04-12 06:07:44 +02:00
|
|
|
/* this is no error, so just keep running */
|
|
|
|
case SND_PCM_STATE_RUNNING:
|
|
|
|
err = 0;
|
|
|
|
break;
|
2005-03-05 15:01:13 +01:00
|
|
|
default:
|
|
|
|
/* unknown state, do nothing */
|
|
|
|
break;
|
2005-03-05 06:22:30 +01:00
|
|
|
}
|
|
|
|
|
|
|
|
return err;
|
|
|
|
}
|
|
|
|
|
2009-10-29 15:59:40 +01:00
|
|
|
static void
|
2011-09-16 23:31:48 +02:00
|
|
|
alsa_drain(struct audio_output *ao)
|
2009-10-29 15:59:40 +01:00
|
|
|
{
|
2011-09-16 23:31:48 +02:00
|
|
|
struct alsa_data *ad = (struct alsa_data *)ao;
|
2009-10-29 15:59:40 +01:00
|
|
|
|
2009-11-09 22:22:31 +01:00
|
|
|
if (snd_pcm_state(ad->pcm) != SND_PCM_STATE_RUNNING)
|
|
|
|
return;
|
|
|
|
|
|
|
|
if (ad->period_position > 0) {
|
|
|
|
/* generate some silence to finish the partial
|
|
|
|
period */
|
|
|
|
snd_pcm_uframes_t nframes =
|
|
|
|
ad->period_frames - ad->period_position;
|
2012-03-22 01:01:11 +01:00
|
|
|
size_t nbytes = nframes * ad->out_frame_size;
|
2009-11-09 22:22:31 +01:00
|
|
|
void *buffer = g_malloc(nbytes);
|
|
|
|
snd_pcm_hw_params_t *params;
|
|
|
|
snd_pcm_format_t format;
|
|
|
|
unsigned channels;
|
|
|
|
|
|
|
|
snd_pcm_hw_params_alloca(¶ms);
|
|
|
|
snd_pcm_hw_params_current(ad->pcm, params);
|
|
|
|
snd_pcm_hw_params_get_format(params, &format);
|
|
|
|
snd_pcm_hw_params_get_channels(params, &channels);
|
|
|
|
|
|
|
|
snd_pcm_format_set_silence(format, buffer, nframes * channels);
|
|
|
|
ad->writei(ad->pcm, buffer, nframes);
|
|
|
|
g_free(buffer);
|
|
|
|
}
|
|
|
|
|
|
|
|
snd_pcm_drain(ad->pcm);
|
|
|
|
|
|
|
|
ad->period_position = 0;
|
2009-10-29 15:59:40 +01:00
|
|
|
}
|
|
|
|
|
2009-01-25 13:05:16 +01:00
|
|
|
static void
|
2011-09-16 23:31:48 +02:00
|
|
|
alsa_cancel(struct audio_output *ao)
|
2006-07-20 18:02:40 +02:00
|
|
|
{
|
2011-09-16 23:31:48 +02:00
|
|
|
struct alsa_data *ad = (struct alsa_data *)ao;
|
2005-12-12 04:22:27 +01:00
|
|
|
|
2009-11-09 22:22:31 +01:00
|
|
|
ad->period_position = 0;
|
|
|
|
|
2009-10-29 15:59:35 +01:00
|
|
|
snd_pcm_drop(ad->pcm);
|
2005-12-12 04:22:27 +01:00
|
|
|
}
|
|
|
|
|
2009-01-25 13:05:16 +01:00
|
|
|
static void
|
2011-09-16 23:31:48 +02:00
|
|
|
alsa_close(struct audio_output *ao)
|
2006-07-20 18:02:40 +02:00
|
|
|
{
|
2011-09-16 23:31:48 +02:00
|
|
|
struct alsa_data *ad = (struct alsa_data *)ao;
|
2005-03-05 15:01:13 +01:00
|
|
|
|
2009-03-10 21:25:45 +01:00
|
|
|
snd_pcm_close(ad->pcm);
|
2005-03-05 15:01:13 +01:00
|
|
|
}
|
|
|
|
|
2009-02-23 09:29:56 +01:00
|
|
|
static size_t
|
2011-09-16 23:31:48 +02:00
|
|
|
alsa_play(struct audio_output *ao, const void *chunk, size_t size,
|
|
|
|
GError **error)
|
2005-03-05 06:22:30 +01:00
|
|
|
{
|
2011-09-16 23:31:48 +02:00
|
|
|
struct alsa_data *ad = (struct alsa_data *)ao;
|
2005-03-05 06:22:30 +01:00
|
|
|
|
2012-03-22 01:01:11 +01:00
|
|
|
assert(size % ad->in_frame_size == 0);
|
|
|
|
|
2012-03-21 19:54:35 +01:00
|
|
|
chunk = pcm_export(&ad->export, chunk, size, &size);
|
2012-03-21 19:09:22 +01:00
|
|
|
|
2012-03-22 01:01:11 +01:00
|
|
|
assert(size % ad->out_frame_size == 0);
|
|
|
|
|
|
|
|
size /= ad->out_frame_size;
|
2005-03-05 06:45:39 +01:00
|
|
|
|
2009-02-23 09:29:56 +01:00
|
|
|
while (true) {
|
2009-03-10 21:31:13 +01:00
|
|
|
snd_pcm_sframes_t ret = ad->writei(ad->pcm, chunk, size);
|
2009-11-09 22:22:31 +01:00
|
|
|
if (ret > 0) {
|
|
|
|
ad->period_position = (ad->period_position + ret)
|
|
|
|
% ad->period_frames;
|
2012-03-27 00:03:44 +02:00
|
|
|
return pcm_export_source_size(&ad->export,
|
|
|
|
ret * ad->in_frame_size);
|
2009-11-09 22:22:31 +01:00
|
|
|
}
|
2009-02-23 09:29:56 +01:00
|
|
|
|
|
|
|
if (ret < 0 && ret != -EAGAIN && ret != -EINTR &&
|
|
|
|
alsa_recover(ad, ret) < 0) {
|
2009-02-26 22:04:59 +01:00
|
|
|
g_set_error(error, alsa_output_quark(), errno,
|
|
|
|
"%s", snd_strerror(-errno));
|
2009-02-23 09:29:56 +01:00
|
|
|
return 0;
|
2005-03-05 06:22:30 +01:00
|
|
|
}
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
2011-09-17 20:04:18 +02:00
|
|
|
const struct audio_output_plugin alsa_output_plugin = {
|
2008-09-29 15:55:17 +02:00
|
|
|
.name = "alsa",
|
2009-01-25 13:05:16 +01:00
|
|
|
.test_default_device = alsa_test_default_device,
|
|
|
|
.init = alsa_init,
|
|
|
|
.finish = alsa_finish,
|
2012-03-21 19:09:22 +01:00
|
|
|
.enable = alsa_output_enable,
|
|
|
|
.disable = alsa_output_disable,
|
2009-01-25 13:05:16 +01:00
|
|
|
.open = alsa_open,
|
|
|
|
.play = alsa_play,
|
2009-10-29 15:59:40 +01:00
|
|
|
.drain = alsa_drain,
|
2009-01-25 13:05:16 +01:00
|
|
|
.cancel = alsa_cancel,
|
|
|
|
.close = alsa_close,
|
2009-10-20 21:23:05 +02:00
|
|
|
|
|
|
|
.mixer_plugin = &alsa_mixer_plugin,
|
2005-03-05 06:22:30 +01:00
|
|
|
};
|