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-rw-r--r--src/output/alsa_output_plugin.c819
1 files changed, 0 insertions, 819 deletions
diff --git a/src/output/alsa_output_plugin.c b/src/output/alsa_output_plugin.c
deleted file mode 100644
index d8b184273..000000000
--- a/src/output/alsa_output_plugin.c
+++ /dev/null
@@ -1,819 +0,0 @@
-/*
- * Copyright (C) 2003-2011 The Music Player Daemon Project
- * http://www.musicpd.org
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License along
- * with this program; if not, write to the Free Software Foundation, Inc.,
- * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- */
-
-#include "config.h"
-#include "alsa_output_plugin.h"
-#include "output_api.h"
-#include "mixer_list.h"
-#include "pcm_export.h"
-
-#include <glib.h>
-#include <alsa/asoundlib.h>
-
-#undef G_LOG_DOMAIN
-#define G_LOG_DOMAIN "alsa"
-
-#define ALSA_PCM_NEW_HW_PARAMS_API
-#define ALSA_PCM_NEW_SW_PARAMS_API
-
-static const char default_device[] = "default";
-
-enum {
- MPD_ALSA_BUFFER_TIME_US = 500000,
-};
-
-#define MPD_ALSA_RETRY_NR 5
-
-typedef snd_pcm_sframes_t alsa_writei_t(snd_pcm_t * pcm, const void *buffer,
- snd_pcm_uframes_t size);
-
-struct alsa_data {
- struct audio_output base;
-
- struct pcm_export_state export;
-
- /** the configured name of the ALSA device; NULL for the
- default device */
- char *device;
-
- /** use memory mapped I/O? */
- bool use_mmap;
-
- /**
- * Enable DSD over USB according to the dCS suggested
- * standard?
- *
- * @see http://www.dcsltd.co.uk/page/assets/DSDoverUSB.pdf
- */
- bool dsd_usb;
-
- /** libasound's buffer_time setting (in microseconds) */
- unsigned int buffer_time;
-
- /** libasound's period_time setting (in microseconds) */
- unsigned int period_time;
-
- /** the mode flags passed to snd_pcm_open */
- int mode;
-
- /** the libasound PCM device handle */
- snd_pcm_t *pcm;
-
- /**
- * a pointer to the libasound writei() function, which is
- * snd_pcm_writei() or snd_pcm_mmap_writei(), depending on the
- * use_mmap configuration
- */
- alsa_writei_t *writei;
-
- /**
- * The size of one audio frame passed to method play().
- */
- size_t in_frame_size;
-
- /**
- * The size of one audio frame passed to libasound.
- */
- size_t out_frame_size;
-
- /**
- * The size of one period, in number of frames.
- */
- snd_pcm_uframes_t period_frames;
-
- /**
- * The number of frames written in the current period.
- */
- snd_pcm_uframes_t period_position;
-};
-
-/**
- * The quark used for GError.domain.
- */
-static inline GQuark
-alsa_output_quark(void)
-{
- return g_quark_from_static_string("alsa_output");
-}
-
-static const char *
-alsa_device(const struct alsa_data *ad)
-{
- return ad->device != NULL ? ad->device : default_device;
-}
-
-static struct alsa_data *
-alsa_data_new(void)
-{
- struct alsa_data *ret = g_new(struct alsa_data, 1);
-
- ret->mode = 0;
- ret->writei = snd_pcm_writei;
-
- return ret;
-}
-
-static void
-alsa_configure(struct alsa_data *ad, const struct config_param *param)
-{
- ad->device = config_dup_block_string(param, "device", NULL);
-
- ad->use_mmap = config_get_block_bool(param, "use_mmap", false);
-
- ad->dsd_usb = config_get_block_bool(param, "dsd_usb", false);
-
- ad->buffer_time = config_get_block_unsigned(param, "buffer_time",
- MPD_ALSA_BUFFER_TIME_US);
- ad->period_time = config_get_block_unsigned(param, "period_time", 0);
-
-#ifdef SND_PCM_NO_AUTO_RESAMPLE
- if (!config_get_block_bool(param, "auto_resample", true))
- ad->mode |= SND_PCM_NO_AUTO_RESAMPLE;
-#endif
-
-#ifdef SND_PCM_NO_AUTO_CHANNELS
- if (!config_get_block_bool(param, "auto_channels", true))
- ad->mode |= SND_PCM_NO_AUTO_CHANNELS;
-#endif
-
-#ifdef SND_PCM_NO_AUTO_FORMAT
- if (!config_get_block_bool(param, "auto_format", true))
- ad->mode |= SND_PCM_NO_AUTO_FORMAT;
-#endif
-}
-
-static struct audio_output *
-alsa_init(const struct config_param *param, GError **error_r)
-{
- struct alsa_data *ad = alsa_data_new();
-
- if (!ao_base_init(&ad->base, &alsa_output_plugin, param, error_r)) {
- g_free(ad);
- return NULL;
- }
-
- alsa_configure(ad, param);
-
- return &ad->base;
-}
-
-static void
-alsa_finish(struct audio_output *ao)
-{
- struct alsa_data *ad = (struct alsa_data *)ao;
-
- ao_base_finish(&ad->base);
-
- g_free(ad->device);
- g_free(ad);
-
- /* free libasound's config cache */
- snd_config_update_free_global();
-}
-
-static bool
-alsa_output_enable(struct audio_output *ao, G_GNUC_UNUSED GError **error_r)
-{
- struct alsa_data *ad = (struct alsa_data *)ao;
-
- pcm_export_init(&ad->export);
- return true;
-}
-
-static void
-alsa_output_disable(struct audio_output *ao)
-{
- struct alsa_data *ad = (struct alsa_data *)ao;
-
- pcm_export_deinit(&ad->export);
-}
-
-static bool
-alsa_test_default_device(void)
-{
- snd_pcm_t *handle;
-
- int ret = snd_pcm_open(&handle, default_device,
- SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
- if (ret) {
- g_message("Error opening default ALSA device: %s\n",
- snd_strerror(-ret));
- return false;
- } else
- snd_pcm_close(handle);
-
- return true;
-}
-
-static snd_pcm_format_t
-get_bitformat(enum sample_format sample_format)
-{
- switch (sample_format) {
- case SAMPLE_FORMAT_UNDEFINED:
- case SAMPLE_FORMAT_DSD:
- return SND_PCM_FORMAT_UNKNOWN;
-
- case SAMPLE_FORMAT_S8:
- return SND_PCM_FORMAT_S8;
-
- case SAMPLE_FORMAT_S16:
- return SND_PCM_FORMAT_S16;
-
- case SAMPLE_FORMAT_S24_P32:
- return SND_PCM_FORMAT_S24;
-
- case SAMPLE_FORMAT_S32:
- return SND_PCM_FORMAT_S32;
-
- case SAMPLE_FORMAT_FLOAT:
- return SND_PCM_FORMAT_FLOAT;
- }
-
- assert(false);
- return SND_PCM_FORMAT_UNKNOWN;
-}
-
-static snd_pcm_format_t
-byteswap_bitformat(snd_pcm_format_t fmt)
-{
- switch(fmt) {
- case SND_PCM_FORMAT_S16_LE: return SND_PCM_FORMAT_S16_BE;
- case SND_PCM_FORMAT_S24_LE: return SND_PCM_FORMAT_S24_BE;
- case SND_PCM_FORMAT_S32_LE: return SND_PCM_FORMAT_S32_BE;
- case SND_PCM_FORMAT_S16_BE: return SND_PCM_FORMAT_S16_LE;
- case SND_PCM_FORMAT_S24_BE: return SND_PCM_FORMAT_S24_LE;
-
- case SND_PCM_FORMAT_S24_3BE:
- return SND_PCM_FORMAT_S24_3LE;
-
- case SND_PCM_FORMAT_S24_3LE:
- return SND_PCM_FORMAT_S24_3BE;
-
- case SND_PCM_FORMAT_S32_BE: return SND_PCM_FORMAT_S32_LE;
- default: return SND_PCM_FORMAT_UNKNOWN;
- }
-}
-
-static snd_pcm_format_t
-alsa_to_packed_format(snd_pcm_format_t fmt)
-{
- switch (fmt) {
- case SND_PCM_FORMAT_S24_LE:
- return SND_PCM_FORMAT_S24_3LE;
-
- case SND_PCM_FORMAT_S24_BE:
- return SND_PCM_FORMAT_S24_3BE;
-
- default:
- return SND_PCM_FORMAT_UNKNOWN;
- }
-}
-
-static int
-alsa_try_format_or_packed(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
- snd_pcm_format_t fmt, bool *packed_r)
-{
- int err = snd_pcm_hw_params_set_format(pcm, hwparams, fmt);
- 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;
-}
-
-/**
- * Attempts to configure the specified sample format, and tries the
- * reversed host byte order if was not supported.
- */
-static int
-alsa_output_try_format(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
- enum sample_format sample_format,
- bool *packed_r, bool *reverse_endian_r)
-{
- snd_pcm_format_t alsa_format = get_bitformat(sample_format);
- if (alsa_format == SND_PCM_FORMAT_UNKNOWN)
- return -EINVAL;
-
- int err = alsa_try_format_or_packed(pcm, hwparams, alsa_format,
- packed_r);
- if (err == 0)
- *reverse_endian_r = false;
-
- if (err != -EINVAL)
- return err;
-
- alsa_format = byteswap_bitformat(alsa_format);
- if (alsa_format == SND_PCM_FORMAT_UNKNOWN)
- return -EINVAL;
-
- err = alsa_try_format_or_packed(pcm, hwparams, alsa_format, packed_r);
- if (err == 0)
- *reverse_endian_r = true;
-
- return err;
-}
-
-/**
- * Configure a sample format, and probe other formats if that fails.
- */
-static int
-alsa_output_setup_format(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
- struct audio_format *audio_format,
- bool *packed_r, bool *reverse_endian_r)
-{
- /* try the input format first */
-
- int err = alsa_output_try_format(pcm, hwparams, audio_format->format,
- packed_r, reverse_endian_r);
-
- /* if unsupported by the hardware, try other formats */
-
- static const enum sample_format probe_formats[] = {
- SAMPLE_FORMAT_S24_P32,
- SAMPLE_FORMAT_S32,
- SAMPLE_FORMAT_S16,
- SAMPLE_FORMAT_S8,
- SAMPLE_FORMAT_UNDEFINED,
- };
-
- for (unsigned i = 0;
- err == -EINVAL && probe_formats[i] != SAMPLE_FORMAT_UNDEFINED;
- ++i) {
- const enum sample_format mpd_format = probe_formats[i];
- if (mpd_format == audio_format->format)
- continue;
-
- err = alsa_output_try_format(pcm, hwparams, mpd_format,
- packed_r, reverse_endian_r);
- if (err == 0)
- audio_format->format = mpd_format;
- }
-
- return err;
-}
-
-/**
- * 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,
- bool *packed_r, bool *reverse_endian_r, GError **error)
-{
- 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;
- }
-
- err = alsa_output_setup_format(ad->pcm, hwparams, audio_format,
- packed_r, reverse_endian_r);
- if (err < 0) {
- g_set_error(error, alsa_output_quark(), err,
- "ALSA device \"%s\" does not support format %s: %s",
- alsa_device(ad),
- sample_format_to_string(audio_format->format),
- snd_strerror(-err));
- return false;
- }
-
- 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));
-
- err = snd_pcm_hw_params_set_channels_near(ad->pcm, hwparams,
- &channels);
- if (err < 0) {
- 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));
- return false;
- }
- audio_format->channels = (int8_t)channels;
-
- err = snd_pcm_hw_params_set_rate_near(ad->pcm, hwparams,
- &sample_rate, NULL);
- if (err < 0 || sample_rate == 0) {
- g_set_error(error, alsa_output_quark(), err,
- "ALSA device \"%s\" does not support %u Hz audio",
- alsa_device(ad), audio_format->sample_rate);
- return false;
- }
- audio_format->sample_rate = sample_rate;
-
- 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);
-
- if (ad->buffer_time > 0) {
- buffer_time = ad->buffer_time;
- cmd = "snd_pcm_hw_params_set_buffer_time_near";
- err = snd_pcm_hw_params_set_buffer_time_near(ad->pcm, hwparams,
- &buffer_time, NULL);
- if (err < 0)
- goto error;
- } else {
- err = snd_pcm_hw_params_get_buffer_time(hwparams, &buffer_time,
- NULL);
- if (err < 0)
- buffer_time = 0;
- }
-
- 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);
- }
-
- if (period_time_ro > 0) {
- period_time = period_time_ro;
- cmd = "snd_pcm_hw_params_set_period_time_near";
- err = snd_pcm_hw_params_set_period_time_near(ad->pcm, hwparams,
- &period_time, NULL);
- if (err < 0)
- goto error;
- }
-
- cmd = "snd_pcm_hw_params";
- err = snd_pcm_hw_params(ad->pcm, hwparams);
- if (err == -EPIPE && --retry > 0 && period_time_ro > 0) {
- period_time_ro = period_time_ro >> 1;
- goto configure_hw;
- } else if (err < 0)
- goto error;
- if (retry != MPD_ALSA_RETRY_NR)
- g_debug("ALSA period_time set to %d\n", period_time);
-
- cmd = "snd_pcm_hw_params_get_buffer_size";
- err = snd_pcm_hw_params_get_buffer_size(hwparams, &alsa_buffer_size);
- if (err < 0)
- goto error;
-
- cmd = "snd_pcm_hw_params_get_period_size";
- err = snd_pcm_hw_params_get_period_size(hwparams, &alsa_period_size,
- NULL);
- if (err < 0)
- goto error;
-
- /* configure SW params */
- snd_pcm_sw_params_alloca(&swparams);
-
- cmd = "snd_pcm_sw_params_current";
- err = snd_pcm_sw_params_current(ad->pcm, swparams);
- if (err < 0)
- goto error;
-
- cmd = "snd_pcm_sw_params_set_start_threshold";
- err = snd_pcm_sw_params_set_start_threshold(ad->pcm, swparams,
- alsa_buffer_size -
- alsa_period_size);
- if (err < 0)
- goto error;
-
- cmd = "snd_pcm_sw_params_set_avail_min";
- err = snd_pcm_sw_params_set_avail_min(ad->pcm, swparams,
- alsa_period_size);
- if (err < 0)
- goto error;
-
- cmd = "snd_pcm_sw_params";
- err = snd_pcm_sw_params(ad->pcm, swparams);
- if (err < 0)
- goto error;
-
- g_debug("buffer_size=%u period_size=%u",
- (unsigned)alsa_buffer_size, (unsigned)alsa_period_size);
-
- 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;
-
- ad->period_frames = alsa_period_size;
- ad->period_position = 0;
-
- return true;
-
-error:
- g_set_error(error, alsa_output_quark(), err,
- "Error opening ALSA device \"%s\" (%s): %s",
- alsa_device(ad), cmd, snd_strerror(-err));
- return false;
-}
-
-static bool
-alsa_setup_dsd(struct alsa_data *ad, struct audio_format *audio_format,
- bool *shift8_r, bool *packed_r, bool *reverse_endian_r,
- GError **error_r)
-{
- assert(ad->dsd_usb);
- assert(audio_format->format == SAMPLE_FORMAT_DSD);
-
- /* pass 24 bit to alsa_setup() */
-
- struct audio_format usb_format = *audio_format;
- usb_format.format = SAMPLE_FORMAT_S24_P32;
- usb_format.sample_rate /= 2;
-
- const struct audio_format check = usb_format;
-
- if (!alsa_setup(ad, &usb_format, packed_r, reverse_endian_r, error_r))
- return false;
-
- /* 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;
-
- if (!audio_format_equals(&usb_format, &check)) {
- /* 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)
-{
- bool shift8 = false, packed, reverse_endian;
-
- const bool dsd_usb = ad->dsd_usb &&
- audio_format->format == SAMPLE_FORMAT_DSD;
- const bool success = dsd_usb
- ? alsa_setup_dsd(ad, audio_format,
- &shift8, &packed, &reverse_endian,
- error_r)
- : alsa_setup(ad, audio_format, &packed, &reverse_endian,
- error_r);
- if (!success)
- return false;
-
- pcm_export_open(&ad->export,
- audio_format->format, audio_format->channels,
- dsd_usb, shift8, packed, reverse_endian);
- return true;
-}
-
-static bool
-alsa_open(struct audio_output *ao, struct audio_format *audio_format, GError **error)
-{
- struct alsa_data *ad = (struct alsa_data *)ao;
- int err;
- bool success;
-
- err = snd_pcm_open(&ad->pcm, alsa_device(ad),
- SND_PCM_STREAM_PLAYBACK, ad->mode);
- if (err < 0) {
- g_set_error(error, alsa_output_quark(), err,
- "Failed to open ALSA device \"%s\": %s",
- alsa_device(ad), snd_strerror(err));
- return false;
- }
-
- g_debug("opened %s type=%s", snd_pcm_name(ad->pcm),
- snd_pcm_type_name(snd_pcm_type(ad->pcm)));
-
- success = alsa_setup_or_dsd(ad, audio_format, error);
- if (!success) {
- snd_pcm_close(ad->pcm);
- return false;
- }
-
- ad->in_frame_size = audio_format_frame_size(audio_format);
- ad->out_frame_size = pcm_export_frame_size(&ad->export, audio_format);
-
- return true;
-}
-
-static int
-alsa_recover(struct alsa_data *ad, int err)
-{
- if (err == -EPIPE) {
- g_debug("Underrun on ALSA device \"%s\"\n", alsa_device(ad));
- } else if (err == -ESTRPIPE) {
- g_debug("ALSA device \"%s\" was suspended\n", alsa_device(ad));
- }
-
- switch (snd_pcm_state(ad->pcm)) {
- case SND_PCM_STATE_PAUSED:
- err = snd_pcm_pause(ad->pcm, /* disable */ 0);
- break;
- case SND_PCM_STATE_SUSPENDED:
- err = snd_pcm_resume(ad->pcm);
- if (err == -EAGAIN)
- return 0;
- /* fall-through to snd_pcm_prepare: */
- case SND_PCM_STATE_SETUP:
- case SND_PCM_STATE_XRUN:
- ad->period_position = 0;
- err = snd_pcm_prepare(ad->pcm);
- break;
- case SND_PCM_STATE_DISCONNECTED:
- break;
- /* this is no error, so just keep running */
- case SND_PCM_STATE_RUNNING:
- err = 0;
- break;
- default:
- /* unknown state, do nothing */
- break;
- }
-
- return err;
-}
-
-static void
-alsa_drain(struct audio_output *ao)
-{
- struct alsa_data *ad = (struct alsa_data *)ao;
-
- 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;
- size_t nbytes = nframes * ad->out_frame_size;
- void *buffer = g_malloc(nbytes);
- snd_pcm_hw_params_t *params;
- snd_pcm_format_t format;
- unsigned channels;
-
- snd_pcm_hw_params_alloca(&params);
- 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;
-}
-
-static void
-alsa_cancel(struct audio_output *ao)
-{
- struct alsa_data *ad = (struct alsa_data *)ao;
-
- ad->period_position = 0;
-
- snd_pcm_drop(ad->pcm);
-}
-
-static void
-alsa_close(struct audio_output *ao)
-{
- struct alsa_data *ad = (struct alsa_data *)ao;
-
- snd_pcm_close(ad->pcm);
-}
-
-static size_t
-alsa_play(struct audio_output *ao, const void *chunk, size_t size,
- GError **error)
-{
- struct alsa_data *ad = (struct alsa_data *)ao;
-
- assert(size % ad->in_frame_size == 0);
-
- chunk = pcm_export(&ad->export, chunk, size, &size);
-
- assert(size % ad->out_frame_size == 0);
-
- size /= ad->out_frame_size;
-
- while (true) {
- snd_pcm_sframes_t ret = ad->writei(ad->pcm, chunk, size);
- if (ret > 0) {
- ad->period_position = (ad->period_position + ret)
- % ad->period_frames;
-
- size_t bytes_written = ret * ad->out_frame_size;
- return pcm_export_source_size(&ad->export,
- bytes_written);
- }
-
- if (ret < 0 && ret != -EAGAIN && ret != -EINTR &&
- alsa_recover(ad, ret) < 0) {
- g_set_error(error, alsa_output_quark(), errno,
- "%s", snd_strerror(-errno));
- return 0;
- }
- }
-}
-
-const struct audio_output_plugin alsa_output_plugin = {
- .name = "alsa",
- .test_default_device = alsa_test_default_device,
- .init = alsa_init,
- .finish = alsa_finish,
- .enable = alsa_output_enable,
- .disable = alsa_output_disable,
- .open = alsa_open,
- .play = alsa_play,
- .drain = alsa_drain,
- .cancel = alsa_cancel,
- .close = alsa_close,
-
- .mixer_plugin = &alsa_mixer_plugin,
-};
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-05 12:38:40 +0000