output/{alsa,oss}: convert to C++
This commit is contained in:
Max Kellermann
parent
76417d4446
commit
26a9ce7b29
@ -809,7 +809,8 @@ libmixer_plugins_a_CPPFLAGS = $(AM_CPPFLAGS) \
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if HAVE_ALSA
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liboutput_plugins_a_SOURCES += \
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src/output/alsa_output_plugin.c src/output/alsa_output_plugin.h
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src/output/AlsaOutputPlugin.cxx \
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src/output/AlsaOutputPlugin.hxx
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libmixer_plugins_a_SOURCES += src/mixer/AlsaMixerPlugin.cxx
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endif
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@ -851,8 +852,9 @@ endif
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if HAVE_OSS
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liboutput_plugins_a_SOURCES += \
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src/output/oss_output_plugin.c src/output/oss_output_plugin.h
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libmixer_plugins_a_SOURCES += src/mixer/oss_mixer_plugin.c
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src/output/OssOutputPlugin.cxx \
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src/output/OssOutputPlugin.hxx
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libmixer_plugins_a_SOURCES += src/mixer/OssMixerPlugin.cxx
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endif
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if HAVE_OPENAL
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@ -20,7 +20,7 @@
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#include "config.h"
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#include "OutputList.hxx"
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#include "output_api.h"
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#include "output/alsa_output_plugin.h"
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#include "output/AlsaOutputPlugin.hxx"
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#include "output/ao_output_plugin.h"
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#include "output/ffado_output_plugin.h"
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#include "output/fifo_output_plugin.h"
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@ -29,7 +29,7 @@
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#include "output/mvp_output_plugin.h"
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#include "output/null_output_plugin.h"
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#include "output/openal_output_plugin.h"
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#include "output/oss_output_plugin.h"
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#include "output/OssOutputPlugin.hxx"
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#include "output/osx_output_plugin.h"
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#include "output/pipe_output_plugin.h"
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#include "output/pulse_output_plugin.h"
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@ -1,5 +1,5 @@
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/*
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* Copyright (C) 2003-2011 The Music Player Daemon Project
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* Copyright (C) 2003-2013 The Music Player Daemon Project
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* http://www.musicpd.org
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*
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* This program is free software; you can redistribute it and/or modify
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@ -206,11 +206,11 @@ oss_mixer_set_volume(struct mixer *mixer, unsigned volume, GError **error_r)
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}
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const struct mixer_plugin oss_mixer_plugin = {
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.init = oss_mixer_init,
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.finish = oss_mixer_finish,
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.open = oss_mixer_open,
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.close = oss_mixer_close,
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.get_volume = oss_mixer_get_volume,
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.set_volume = oss_mixer_set_volume,
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.global = true,
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oss_mixer_init,
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oss_mixer_finish,
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oss_mixer_open,
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oss_mixer_close,
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oss_mixer_get_volume,
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oss_mixer_set_volume,
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true,
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};
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@ -1,5 +1,5 @@
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/*
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* Copyright (C) 2003-2011 The Music Player Daemon Project
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* Copyright (C) 2003-2013 The Music Player Daemon Project
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* http://www.musicpd.org
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*
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* This program is free software; you can redistribute it and/or modify
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@ -18,7 +18,7 @@
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*/
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#include "config.h"
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#include "alsa_output_plugin.h"
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#include "AlsaOutputPlugin.hxx"
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#include "output_api.h"
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#include "mixer_list.h"
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#include "pcm_export.h"
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@ -26,6 +26,8 @@
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#include <glib.h>
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#include <alsa/asoundlib.h>
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#include <string>
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#undef G_LOG_DOMAIN
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#define G_LOG_DOMAIN "alsa"
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@ -43,14 +45,16 @@ enum {
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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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struct alsa_data {
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struct AlsaOutput {
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struct audio_output base;
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struct pcm_export_state export;
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struct pcm_export_state pcm_export;
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/** the configured name of the ALSA device; NULL for the
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default device */
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char *device;
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/**
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* The configured name of the ALSA device; empty for the
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* default device
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*/
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std::string device;
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/** use memory mapped I/O? */
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bool use_mmap;
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@ -101,6 +105,18 @@ struct alsa_data {
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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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AlsaOutput():mode(0), writei(snd_pcm_writei) {
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}
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bool Init(const config_param *param, GError **error_r) {
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return ao_base_init(&base, &alsa_output_plugin,
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param, error_r);
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}
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void Deinit() {
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ao_base_finish(&base);
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}
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};
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/**
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@ -113,24 +129,13 @@ alsa_output_quark(void)
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}
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static const char *
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alsa_device(const struct alsa_data *ad)
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alsa_device(const AlsaOutput *ad)
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{
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return ad->device != NULL ? ad->device : default_device;
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}
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static struct alsa_data *
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alsa_data_new(void)
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{
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struct alsa_data *ret = g_new(struct alsa_data, 1);
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ret->mode = 0;
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ret->writei = snd_pcm_writei;
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return ret;
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return ad->device.empty() ? default_device : ad->device.c_str();
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}
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static void
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alsa_configure(struct alsa_data *ad, const struct config_param *param)
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alsa_configure(AlsaOutput *ad, const struct config_param *param)
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{
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ad->device = config_dup_block_string(param, "device", NULL);
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@ -161,10 +166,10 @@ alsa_configure(struct alsa_data *ad, const struct config_param *param)
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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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{
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struct alsa_data *ad = alsa_data_new();
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AlsaOutput *ad = new AlsaOutput();
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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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if (!ad->Init(param, error_r)) {
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delete ad;
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return NULL;
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}
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@ -176,12 +181,10 @@ alsa_init(const struct config_param *param, GError **error_r)
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static void
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alsa_finish(struct audio_output *ao)
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{
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struct alsa_data *ad = (struct alsa_data *)ao;
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AlsaOutput *ad = (AlsaOutput *)ao;
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ao_base_finish(&ad->base);
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g_free(ad->device);
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g_free(ad);
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ad->Deinit();
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delete ad;
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/* free libasound's config cache */
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snd_config_update_free_global();
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@ -190,18 +193,18 @@ alsa_finish(struct audio_output *ao)
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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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AlsaOutput *ad = (AlsaOutput *)ao;
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pcm_export_init(&ad->export);
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pcm_export_init(&ad->pcm_export);
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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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AlsaOutput *ad = (AlsaOutput *)ao;
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pcm_export_deinit(&ad->export);
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pcm_export_deinit(&ad->pcm_export);
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}
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static bool
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@ -349,7 +352,8 @@ alsa_output_setup_format(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
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{
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/* try the input format first */
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int err = alsa_output_try_format(pcm, hwparams, audio_format->format,
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int err = alsa_output_try_format(pcm, hwparams,
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sample_format(audio_format->format),
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packed_r, reverse_endian_r);
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/* if unsupported by the hardware, try other formats */
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@ -383,15 +387,11 @@ alsa_output_setup_format(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
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* the configured settings and the audio format.
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*/
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static bool
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alsa_setup(struct alsa_data *ad, struct audio_format *audio_format,
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alsa_setup(AlsaOutput *ad, struct audio_format *audio_format,
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bool *packed_r, bool *reverse_endian_r, GError **error)
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{
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snd_pcm_hw_params_t *hwparams;
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snd_pcm_sw_params_t *swparams;
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unsigned int sample_rate = audio_format->sample_rate;
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unsigned int channels = audio_format->channels;
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snd_pcm_uframes_t alsa_buffer_size;
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snd_pcm_uframes_t alsa_period_size;
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int err;
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const char *cmd = NULL;
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int retry = MPD_ALSA_RETRY_NR;
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@ -401,6 +401,7 @@ alsa_setup(struct alsa_data *ad, struct audio_format *audio_format,
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period_time_ro = period_time = ad->period_time;
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configure_hw:
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/* configure HW params */
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snd_pcm_hw_params_t *hwparams;
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snd_pcm_hw_params_alloca(&hwparams);
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cmd = "snd_pcm_hw_params_any";
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err = snd_pcm_hw_params_any(ad->pcm, hwparams);
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@ -434,7 +435,7 @@ configure_hw:
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g_set_error(error, alsa_output_quark(), err,
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"ALSA device \"%s\" does not support format %s: %s",
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alsa_device(ad),
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sample_format_to_string(audio_format->format),
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sample_format_to_string(sample_format(audio_format->format)),
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snd_strerror(-err));
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return false;
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}
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@ -525,11 +526,13 @@ configure_hw:
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if (retry != MPD_ALSA_RETRY_NR)
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g_debug("ALSA period_time set to %d\n", period_time);
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snd_pcm_uframes_t alsa_buffer_size;
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cmd = "snd_pcm_hw_params_get_buffer_size";
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err = snd_pcm_hw_params_get_buffer_size(hwparams, &alsa_buffer_size);
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if (err < 0)
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goto error;
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snd_pcm_uframes_t alsa_period_size;
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cmd = "snd_pcm_hw_params_get_period_size";
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err = snd_pcm_hw_params_get_period_size(hwparams, &alsa_period_size,
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NULL);
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@ -537,6 +540,7 @@ configure_hw:
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goto error;
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/* configure SW params */
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snd_pcm_sw_params_t *swparams;
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snd_pcm_sw_params_alloca(&swparams);
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cmd = "snd_pcm_sw_params_current";
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@ -586,7 +590,7 @@ error:
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}
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static bool
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alsa_setup_dsd(struct alsa_data *ad, struct audio_format *audio_format,
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alsa_setup_dsd(AlsaOutput *ad, struct audio_format *audio_format,
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bool *shift8_r, bool *packed_r, bool *reverse_endian_r,
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GError **error_r)
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{
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@ -626,7 +630,7 @@ alsa_setup_dsd(struct alsa_data *ad, struct audio_format *audio_format,
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}
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static bool
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alsa_setup_or_dsd(struct alsa_data *ad, struct audio_format *audio_format,
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alsa_setup_or_dsd(AlsaOutput *ad, struct audio_format *audio_format,
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GError **error_r)
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{
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bool shift8 = false, packed, reverse_endian;
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@ -642,8 +646,9 @@ alsa_setup_or_dsd(struct alsa_data *ad, struct audio_format *audio_format,
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if (!success)
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return false;
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pcm_export_open(&ad->export,
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audio_format->format, audio_format->channels,
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pcm_export_open(&ad->pcm_export,
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sample_format(audio_format->format),
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audio_format->channels,
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dsd_usb, shift8, packed, reverse_endian);
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return true;
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}
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@ -651,11 +656,9 @@ alsa_setup_or_dsd(struct alsa_data *ad, struct audio_format *audio_format,
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static bool
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alsa_open(struct audio_output *ao, struct audio_format *audio_format, GError **error)
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{
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struct alsa_data *ad = (struct alsa_data *)ao;
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int err;
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bool success;
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AlsaOutput *ad = (AlsaOutput *)ao;
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err = snd_pcm_open(&ad->pcm, alsa_device(ad),
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int err = snd_pcm_open(&ad->pcm, alsa_device(ad),
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SND_PCM_STREAM_PLAYBACK, ad->mode);
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if (err < 0) {
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g_set_error(error, alsa_output_quark(), err,
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@ -667,20 +670,20 @@ alsa_open(struct audio_output *ao, struct audio_format *audio_format, GError **e
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g_debug("opened %s type=%s", snd_pcm_name(ad->pcm),
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snd_pcm_type_name(snd_pcm_type(ad->pcm)));
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success = alsa_setup_or_dsd(ad, audio_format, error);
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if (!success) {
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if (!alsa_setup_or_dsd(ad, audio_format, error)) {
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snd_pcm_close(ad->pcm);
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return false;
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}
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ad->in_frame_size = audio_format_frame_size(audio_format);
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ad->out_frame_size = pcm_export_frame_size(&ad->export, audio_format);
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ad->out_frame_size = pcm_export_frame_size(&ad->pcm_export,
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audio_format);
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return true;
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}
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static int
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alsa_recover(struct alsa_data *ad, int err)
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alsa_recover(AlsaOutput *ad, int err)
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{
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if (err == -EPIPE) {
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g_debug("Underrun on ALSA device \"%s\"\n", alsa_device(ad));
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@ -719,7 +722,7 @@ alsa_recover(struct alsa_data *ad, int err)
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static void
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alsa_drain(struct audio_output *ao)
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{
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struct alsa_data *ad = (struct alsa_data *)ao;
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AlsaOutput *ad = (AlsaOutput *)ao;
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if (snd_pcm_state(ad->pcm) != SND_PCM_STATE_RUNNING)
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return;
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@ -753,7 +756,7 @@ alsa_drain(struct audio_output *ao)
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static void
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alsa_cancel(struct audio_output *ao)
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{
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struct alsa_data *ad = (struct alsa_data *)ao;
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AlsaOutput *ad = (AlsaOutput *)ao;
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ad->period_position = 0;
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@ -763,7 +766,7 @@ alsa_cancel(struct audio_output *ao)
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static void
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alsa_close(struct audio_output *ao)
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{
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struct alsa_data *ad = (struct alsa_data *)ao;
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AlsaOutput *ad = (AlsaOutput *)ao;
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snd_pcm_close(ad->pcm);
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}
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@ -772,11 +775,11 @@ static size_t
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alsa_play(struct audio_output *ao, const void *chunk, size_t size,
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GError **error)
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{
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struct alsa_data *ad = (struct alsa_data *)ao;
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AlsaOutput *ad = (AlsaOutput *)ao;
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assert(size % ad->in_frame_size == 0);
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chunk = pcm_export(&ad->export, chunk, size, &size);
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chunk = pcm_export(&ad->pcm_export, chunk, size, &size);
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assert(size % ad->out_frame_size == 0);
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@ -789,7 +792,7 @@ alsa_play(struct audio_output *ao, const void *chunk, size_t size,
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% ad->period_frames;
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size_t bytes_written = ret * ad->out_frame_size;
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return pcm_export_source_size(&ad->export,
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return pcm_export_source_size(&ad->pcm_export,
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bytes_written);
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}
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@ -803,17 +806,20 @@ alsa_play(struct audio_output *ao, const void *chunk, size_t size,
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}
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const struct audio_output_plugin alsa_output_plugin = {
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.name = "alsa",
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.test_default_device = alsa_test_default_device,
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.init = alsa_init,
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.finish = alsa_finish,
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.enable = alsa_output_enable,
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.disable = alsa_output_disable,
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.open = alsa_open,
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.play = alsa_play,
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.drain = alsa_drain,
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.cancel = alsa_cancel,
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.close = alsa_close,
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"alsa",
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alsa_test_default_device,
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alsa_init,
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alsa_finish,
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alsa_output_enable,
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alsa_output_disable,
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alsa_open,
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alsa_close,
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nullptr,
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nullptr,
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alsa_play,
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alsa_drain,
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alsa_cancel,
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nullptr,
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.mixer_plugin = &alsa_mixer_plugin,
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&alsa_mixer_plugin,
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};
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@ -1,5 +1,5 @@
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/*
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* Copyright (C) 2003-2011 The Music Player Daemon Project
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* Copyright (C) 2003-2013 The Music Player Daemon Project
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* http://www.musicpd.org
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*
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* This program is free software; you can redistribute it and/or modify
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@ -17,8 +17,8 @@
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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*/
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#ifndef MPD_ALSA_OUTPUT_PLUGIN_H
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#define MPD_ALSA_OUTPUT_PLUGIN_H
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#ifndef MPD_ALSA_OUTPUT_PLUGIN_HXX
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#define MPD_ALSA_OUTPUT_PLUGIN_HXX
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|
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extern const struct audio_output_plugin alsa_output_plugin;
|
||||
|
||||
@ -1,5 +1,5 @@
|
||||
/*
|
||||
* Copyright (C) 2003-2011 The Music Player Daemon Project
|
||||
* Copyright (C) 2003-2013 The Music Player Daemon Project
|
||||
* http://www.musicpd.org
|
||||
*
|
||||
* This program is free software; you can redistribute it and/or modify
|
||||
@ -18,7 +18,7 @@
|
||||
*/
|
||||
|
||||
#include "config.h"
|
||||
#include "oss_output_plugin.h"
|
||||
#include "OssOutputPlugin.hxx"
|
||||
#include "output_api.h"
|
||||
#include "mixer_list.h"
|
||||
#include "fd_util.h"
|
||||
@ -60,7 +60,7 @@ struct oss_data {
|
||||
struct audio_output base;
|
||||
|
||||
#ifdef AFMT_S24_PACKED
|
||||
struct pcm_export_state export;
|
||||
struct pcm_export_state pcm_export;
|
||||
#endif
|
||||
|
||||
int fd;
|
||||
@ -163,11 +163,10 @@ oss_output_test_default_device(void)
|
||||
static struct audio_output *
|
||||
oss_open_default(GError **error)
|
||||
{
|
||||
int i;
|
||||
int err[G_N_ELEMENTS(default_devices)];
|
||||
enum oss_stat ret[G_N_ELEMENTS(default_devices)];
|
||||
|
||||
for (i = G_N_ELEMENTS(default_devices); --i >= 0; ) {
|
||||
for (int i = G_N_ELEMENTS(default_devices); --i >= 0; ) {
|
||||
ret[i] = oss_stat_device(default_devices[i], &err[i]);
|
||||
if (ret[i] == OSS_STAT_NO_ERROR) {
|
||||
struct oss_data *od = oss_data_new();
|
||||
@ -182,7 +181,7 @@ oss_open_default(GError **error)
|
||||
}
|
||||
}
|
||||
|
||||
for (i = G_N_ELEMENTS(default_devices); --i >= 0; ) {
|
||||
for (int i = G_N_ELEMENTS(default_devices); --i >= 0; ) {
|
||||
const char *dev = default_devices[i];
|
||||
switch(ret[i]) {
|
||||
case OSS_STAT_NO_ERROR:
|
||||
@ -243,7 +242,7 @@ oss_output_enable(struct audio_output *ao, G_GNUC_UNUSED GError **error_r)
|
||||
{
|
||||
struct oss_data *od = (struct oss_data *)ao;
|
||||
|
||||
pcm_export_init(&od->export);
|
||||
pcm_export_init(&od->pcm_export);
|
||||
return true;
|
||||
}
|
||||
|
||||
@ -252,7 +251,7 @@ oss_output_disable(struct audio_output *ao)
|
||||
{
|
||||
struct oss_data *od = (struct oss_data *)ao;
|
||||
|
||||
pcm_export_deinit(&od->export);
|
||||
pcm_export_deinit(&od->pcm_export);
|
||||
}
|
||||
|
||||
#endif
|
||||
@ -504,7 +503,7 @@ oss_probe_sample_format(int fd, enum sample_format sample_format,
|
||||
enum sample_format *sample_format_r,
|
||||
int *oss_format_r,
|
||||
#ifdef AFMT_S24_PACKED
|
||||
struct pcm_export_state *export,
|
||||
struct pcm_export_state *pcm_export,
|
||||
#endif
|
||||
GError **error_r)
|
||||
{
|
||||
@ -539,7 +538,7 @@ oss_probe_sample_format(int fd, enum sample_format sample_format,
|
||||
*oss_format_r = oss_format;
|
||||
|
||||
#ifdef AFMT_S24_PACKED
|
||||
pcm_export_open(export, sample_format, 0, false, false,
|
||||
pcm_export_open(pcm_export, sample_format, 0, false, false,
|
||||
oss_format == AFMT_S24_PACKED,
|
||||
oss_format == AFMT_S24_PACKED &&
|
||||
G_BYTE_ORDER != G_LITTLE_ENDIAN);
|
||||
@ -556,16 +555,16 @@ static bool
|
||||
oss_setup_sample_format(int fd, struct audio_format *audio_format,
|
||||
int *oss_format_r,
|
||||
#ifdef AFMT_S24_PACKED
|
||||
struct pcm_export_state *export,
|
||||
struct pcm_export_state *pcm_export,
|
||||
#endif
|
||||
GError **error_r)
|
||||
{
|
||||
enum sample_format mpd_format;
|
||||
enum oss_setup_result result =
|
||||
oss_probe_sample_format(fd, audio_format->format,
|
||||
oss_probe_sample_format(fd, sample_format(audio_format->format),
|
||||
&mpd_format, oss_format_r,
|
||||
#ifdef AFMT_S24_PACKED
|
||||
export,
|
||||
pcm_export,
|
||||
#endif
|
||||
error_r);
|
||||
switch (result) {
|
||||
@ -603,7 +602,7 @@ oss_setup_sample_format(int fd, struct audio_format *audio_format,
|
||||
result = oss_probe_sample_format(fd, mpd_format,
|
||||
&mpd_format, oss_format_r,
|
||||
#ifdef AFMT_S24_PACKED
|
||||
export,
|
||||
pcm_export,
|
||||
#endif
|
||||
error_r);
|
||||
switch (result) {
|
||||
@ -635,7 +634,7 @@ oss_setup(struct oss_data *od, struct audio_format *audio_format,
|
||||
oss_setup_sample_rate(od->fd, audio_format, error_r) &&
|
||||
oss_setup_sample_format(od->fd, audio_format, &od->oss_format,
|
||||
#ifdef AFMT_S24_PACKED
|
||||
&od->export,
|
||||
&od->pcm_export,
|
||||
#endif
|
||||
error_r);
|
||||
}
|
||||
@ -749,14 +748,14 @@ oss_output_play(struct audio_output *ao, const void *chunk, size_t size,
|
||||
return 0;
|
||||
|
||||
#ifdef AFMT_S24_PACKED
|
||||
chunk = pcm_export(&od->export, chunk, size, &size);
|
||||
chunk = pcm_export(&od->pcm_export, chunk, size, &size);
|
||||
#endif
|
||||
|
||||
while (true) {
|
||||
ret = write(od->fd, chunk, size);
|
||||
if (ret > 0) {
|
||||
#ifdef AFMT_S24_PACKED
|
||||
ret = pcm_export_source_size(&od->export, ret);
|
||||
ret = pcm_export_source_size(&od->pcm_export, ret);
|
||||
#endif
|
||||
return ret;
|
||||
}
|
||||
@ -771,18 +770,25 @@ oss_output_play(struct audio_output *ao, const void *chunk, size_t size,
|
||||
}
|
||||
|
||||
const struct audio_output_plugin oss_output_plugin = {
|
||||
.name = "oss",
|
||||
.test_default_device = oss_output_test_default_device,
|
||||
.init = oss_output_init,
|
||||
.finish = oss_output_finish,
|
||||
"oss",
|
||||
oss_output_test_default_device,
|
||||
oss_output_init,
|
||||
oss_output_finish,
|
||||
#ifdef AFMT_S24_PACKED
|
||||
.enable = oss_output_enable,
|
||||
.disable = oss_output_disable,
|
||||
oss_output_enable,
|
||||
oss_output_disable,
|
||||
#else
|
||||
nullptr,
|
||||
nullptr,
|
||||
#endif
|
||||
.open = oss_output_open,
|
||||
.close = oss_output_close,
|
||||
.play = oss_output_play,
|
||||
.cancel = oss_output_cancel,
|
||||
oss_output_open,
|
||||
oss_output_close,
|
||||
nullptr,
|
||||
nullptr,
|
||||
oss_output_play,
|
||||
nullptr,
|
||||
oss_output_cancel,
|
||||
nullptr,
|
||||
|
||||
.mixer_plugin = &oss_mixer_plugin,
|
||||
&oss_mixer_plugin,
|
||||
};
|
||||
@ -1,5 +1,5 @@
|
||||
/*
|
||||
* Copyright (C) 2003-2011 The Music Player Daemon Project
|
||||
* Copyright (C) 2003-2013 The Music Player Daemon Project
|
||||
* http://www.musicpd.org
|
||||
*
|
||||
* This program is free software; you can redistribute it and/or modify
|
||||
@ -17,8 +17,8 @@
|
||||
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
|
||||
*/
|
||||
|
||||
#ifndef MPD_OSS_OUTPUT_PLUGIN_H
|
||||
#define MPD_OSS_OUTPUT_PLUGIN_H
|
||||
#ifndef MPD_OSS_OUTPUT_PLUGIN_HXX
|
||||
#define MPD_OSS_OUTPUT_PLUGIN_HXX
|
||||
|
||||
extern const struct audio_output_plugin oss_output_plugin;
|
||||
|
||||
@ -87,6 +87,10 @@ struct pcm_export_state {
|
||||
uint8_t reverse_endian;
|
||||
};
|
||||
|
||||
#ifdef __cplusplus
|
||||
extern "C" {
|
||||
#endif
|
||||
|
||||
/**
|
||||
* Initialize a #pcm_export_state object.
|
||||
*/
|
||||
@ -144,4 +148,8 @@ G_GNUC_PURE
|
||||
size_t
|
||||
pcm_export_source_size(const struct pcm_export_state *state, size_t dest_size);
|
||||
|
||||
#ifdef __cplusplus
|
||||
}
|
||||
#endif
|
||||
|
||||
#endif
|
||||
|
||||
Loading…
Reference in New Issue
Block a user