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Commit 26a9ce7b authored by Max Kellermann's avatar Max Kellermann
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output/{alsa,oss}: convert to C++

parent 76417d44
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Showing with 142 additions and 120 deletions
Makefile.am
View file @ 26a9ce7b
......@@ -809,7 +809,8 @@ libmixer_plugins_a_CPPFLAGS = $(AM_CPPFLAGS) \
if HAVE_ALSA
liboutput_plugins_a_SOURCES += \
src/output/alsa_output_plugin.c src/output/alsa_output_plugin.h
src/output/AlsaOutputPlugin.cxx \
src/output/AlsaOutputPlugin.hxx
libmixer_plugins_a_SOURCES += src/mixer/AlsaMixerPlugin.cxx
endif
......@@ -851,8 +852,9 @@ endif
if HAVE_OSS
liboutput_plugins_a_SOURCES += \
src/output/oss_output_plugin.c src/output/oss_output_plugin.h
libmixer_plugins_a_SOURCES += src/mixer/oss_mixer_plugin.c
src/output/OssOutputPlugin.cxx \
src/output/OssOutputPlugin.hxx
libmixer_plugins_a_SOURCES += src/mixer/OssMixerPlugin.cxx
endif
if HAVE_OPENAL
......
src/OutputList.cxx
View file @ 26a9ce7b
......@@ -20,7 +20,7 @@
#include "config.h"
#include "OutputList.hxx"
#include "output_api.h"
#include "output/alsa_output_plugin.h"
#include "output/AlsaOutputPlugin.hxx"
#include "output/ao_output_plugin.h"
#include "output/ffado_output_plugin.h"
#include "output/fifo_output_plugin.h"
......@@ -29,7 +29,7 @@
#include "output/mvp_output_plugin.h"
#include "output/null_output_plugin.h"
#include "output/openal_output_plugin.h"
#include "output/oss_output_plugin.h"
#include "output/OssOutputPlugin.hxx"
#include "output/osx_output_plugin.h"
#include "output/pipe_output_plugin.h"
#include "output/pulse_output_plugin.h"
......
src/mixer/oss_mixer_plugin.c → src/mixer/OssMixerPlugin.cxx
View file @ 26a9ce7b
/*
* 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
......@@ -206,11 +206,11 @@ oss_mixer_set_volume(struct mixer *mixer, unsigned volume, GError **error_r)
}
const struct mixer_plugin oss_mixer_plugin = {
.init = oss_mixer_init,
.finish = oss_mixer_finish,
.open = oss_mixer_open,
.close = oss_mixer_close,
.get_volume = oss_mixer_get_volume,
.set_volume = oss_mixer_set_volume,
.global = true,
oss_mixer_init,
oss_mixer_finish,
oss_mixer_open,
oss_mixer_close,
oss_mixer_get_volume,
oss_mixer_set_volume,
true,
};
src/output/alsa_output_plugin.c → src/output/AlsaOutputPlugin.cxx
View file @ 26a9ce7b
/*
* 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 "alsa_output_plugin.h"
#include "AlsaOutputPlugin.hxx"
#include "output_api.h"
#include "mixer_list.h"
#include "pcm_export.h"
......@@ -26,6 +26,8 @@
#include <glib.h>
#include <alsa/asoundlib.h>
#include <string>
#undef G_LOG_DOMAIN
#define G_LOG_DOMAIN "alsa"
......@@ -43,14 +45,16 @@ enum {
typedef snd_pcm_sframes_t alsa_writei_t(snd_pcm_t * pcm, const void *buffer,
snd_pcm_uframes_t size);
struct alsa_data {
struct AlsaOutput {
struct audio_output base;
struct pcm_export_state export;
struct pcm_export_state pcm_export;
/** the configured name of the ALSA device; NULL for the
default device */
char *device;
/**
* The configured name of the ALSA device; empty for the
* default device
*/
std::string device;
/** use memory mapped I/O? */
bool use_mmap;
......@@ -101,6 +105,18 @@ struct alsa_data {
* The number of frames written in the current period.
*/
snd_pcm_uframes_t period_position;
AlsaOutput():mode(0), writei(snd_pcm_writei) {
}
bool Init(const config_param *param, GError **error_r) {
return ao_base_init(&base, &alsa_output_plugin,
param, error_r);
}
void Deinit() {
ao_base_finish(&base);
}
};
/**
......@@ -113,24 +129,13 @@ alsa_output_quark(void)
}
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)
alsa_device(const AlsaOutput *ad)
{
struct alsa_data *ret = g_new(struct alsa_data, 1);
ret->mode = 0;
ret->writei = snd_pcm_writei;
return ret;
return ad->device.empty() ? default_device : ad->device.c_str();
}
static void
alsa_configure(struct alsa_data *ad, const struct config_param *param)
alsa_configure(AlsaOutput *ad, const struct config_param *param)
{
ad->device = config_dup_block_string(param, "device", NULL);
......@@ -161,10 +166,10 @@ alsa_configure(struct alsa_data *ad, const struct config_param *param)
static struct audio_output *
alsa_init(const struct config_param *param, GError **error_r)
{
struct alsa_data *ad = alsa_data_new();
AlsaOutput *ad = new AlsaOutput();
if (!ao_base_init(&ad->base, &alsa_output_plugin, param, error_r)) {
g_free(ad);
if (!ad->Init(param, error_r)) {
delete ad;
return NULL;
}
......@@ -176,12 +181,10 @@ alsa_init(const struct config_param *param, GError **error_r)
static void
alsa_finish(struct audio_output *ao)
{
struct alsa_data *ad = (struct alsa_data *)ao;
ao_base_finish(&ad->base);
AlsaOutput *ad = (AlsaOutput *)ao;
g_free(ad->device);
g_free(ad);
ad->Deinit();
delete ad;
/* free libasound's config cache */
snd_config_update_free_global();
......@@ -190,18 +193,18 @@ alsa_finish(struct audio_output *ao)
static bool
alsa_output_enable(struct audio_output *ao, G_GNUC_UNUSED GError **error_r)
{
struct alsa_data *ad = (struct alsa_data *)ao;
AlsaOutput *ad = (AlsaOutput *)ao;
pcm_export_init(&ad->export);
pcm_export_init(&ad->pcm_export);
return true;
}
static void
alsa_output_disable(struct audio_output *ao)
{
struct alsa_data *ad = (struct alsa_data *)ao;
AlsaOutput *ad = (AlsaOutput *)ao;
pcm_export_deinit(&ad->export);
pcm_export_deinit(&ad->pcm_export);
}
static bool
......@@ -349,7 +352,8 @@ alsa_output_setup_format(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
{
/* try the input format first */
int err = alsa_output_try_format(pcm, hwparams, audio_format->format,
int err = alsa_output_try_format(pcm, hwparams,
sample_format(audio_format->format),
packed_r, reverse_endian_r);
/* if unsupported by the hardware, try other formats */
......@@ -383,15 +387,11 @@ alsa_output_setup_format(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
* the configured settings and the audio format.
*/
static bool
alsa_setup(struct alsa_data *ad, struct audio_format *audio_format,
alsa_setup(AlsaOutput *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;
......@@ -401,6 +401,7 @@ alsa_setup(struct alsa_data *ad, struct audio_format *audio_format,
period_time_ro = period_time = ad->period_time;
configure_hw:
/* configure HW params */
snd_pcm_hw_params_t *hwparams;
snd_pcm_hw_params_alloca(&hwparams);
cmd = "snd_pcm_hw_params_any";
err = snd_pcm_hw_params_any(ad->pcm, hwparams);
......@@ -434,7 +435,7 @@ configure_hw:
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),
sample_format_to_string(sample_format(audio_format->format)),
snd_strerror(-err));
return false;
}
......@@ -525,11 +526,13 @@ configure_hw:
if (retry != MPD_ALSA_RETRY_NR)
g_debug("ALSA period_time set to %d\n", period_time);
snd_pcm_uframes_t alsa_buffer_size;
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;
snd_pcm_uframes_t alsa_period_size;
cmd = "snd_pcm_hw_params_get_period_size";
err = snd_pcm_hw_params_get_period_size(hwparams, &alsa_period_size,
NULL);
......@@ -537,6 +540,7 @@ configure_hw:
goto error;
/* configure SW params */
snd_pcm_sw_params_t *swparams;
snd_pcm_sw_params_alloca(&swparams);
cmd = "snd_pcm_sw_params_current";
......@@ -586,7 +590,7 @@ error:
}
static bool
alsa_setup_dsd(struct alsa_data *ad, struct audio_format *audio_format,
alsa_setup_dsd(AlsaOutput *ad, struct audio_format *audio_format,
bool *shift8_r, bool *packed_r, bool *reverse_endian_r,
GError **error_r)
{
......@@ -626,7 +630,7 @@ alsa_setup_dsd(struct alsa_data *ad, struct audio_format *audio_format,
}
static bool
alsa_setup_or_dsd(struct alsa_data *ad, struct audio_format *audio_format,
alsa_setup_or_dsd(AlsaOutput *ad, struct audio_format *audio_format,
GError **error_r)
{
bool shift8 = false, packed, reverse_endian;
......@@ -642,8 +646,9 @@ alsa_setup_or_dsd(struct alsa_data *ad, struct audio_format *audio_format,
if (!success)
return false;
pcm_export_open(&ad->export,
audio_format->format, audio_format->channels,
pcm_export_open(&ad->pcm_export,
sample_format(audio_format->format),
audio_format->channels,
dsd_usb, shift8, packed, reverse_endian);
return true;
}
......@@ -651,12 +656,10 @@ alsa_setup_or_dsd(struct alsa_data *ad, struct audio_format *audio_format,
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;
AlsaOutput *ad = (AlsaOutput *)ao;
err = snd_pcm_open(&ad->pcm, alsa_device(ad),
SND_PCM_STREAM_PLAYBACK, ad->mode);
int 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",
......@@ -667,20 +670,20 @@ alsa_open(struct audio_output *ao, struct audio_format *audio_format, GError **e
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) {
if (!alsa_setup_or_dsd(ad, audio_format, error)) {
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);
ad->out_frame_size = pcm_export_frame_size(&ad->pcm_export,
audio_format);
return true;
}
static int
alsa_recover(struct alsa_data *ad, int err)
alsa_recover(AlsaOutput *ad, int err)
{
if (err == -EPIPE) {
g_debug("Underrun on ALSA device \"%s\"\n", alsa_device(ad));
......@@ -719,7 +722,7 @@ alsa_recover(struct alsa_data *ad, int err)
static void
alsa_drain(struct audio_output *ao)
{
struct alsa_data *ad = (struct alsa_data *)ao;
AlsaOutput *ad = (AlsaOutput *)ao;
if (snd_pcm_state(ad->pcm) != SND_PCM_STATE_RUNNING)
return;
......@@ -753,7 +756,7 @@ alsa_drain(struct audio_output *ao)
static void
alsa_cancel(struct audio_output *ao)
{
struct alsa_data *ad = (struct alsa_data *)ao;
AlsaOutput *ad = (AlsaOutput *)ao;
ad->period_position = 0;
......@@ -763,7 +766,7 @@ alsa_cancel(struct audio_output *ao)
static void
alsa_close(struct audio_output *ao)
{
struct alsa_data *ad = (struct alsa_data *)ao;
AlsaOutput *ad = (AlsaOutput *)ao;
snd_pcm_close(ad->pcm);
}
......@@ -772,11 +775,11 @@ 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;
AlsaOutput *ad = (AlsaOutput *)ao;
assert(size % ad->in_frame_size == 0);
chunk = pcm_export(&ad->export, chunk, size, &size);
chunk = pcm_export(&ad->pcm_export, chunk, size, &size);
assert(size % ad->out_frame_size == 0);
......@@ -789,7 +792,7 @@ alsa_play(struct audio_output *ao, const void *chunk, size_t size,
% ad->period_frames;
size_t bytes_written = ret * ad->out_frame_size;
return pcm_export_source_size(&ad->export,
return pcm_export_source_size(&ad->pcm_export,
bytes_written);
}
......@@ -803,17 +806,20 @@ alsa_play(struct audio_output *ao, const void *chunk, size_t size,
}
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,
"alsa",
alsa_test_default_device,
alsa_init,
alsa_finish,
alsa_output_enable,
alsa_output_disable,
alsa_open,
alsa_close,
nullptr,
nullptr,
alsa_play,
alsa_drain,
alsa_cancel,
nullptr,
&alsa_mixer_plugin,
};
src/output/alsa_output_plugin.h → src/output/AlsaOutputPlugin.hxx
View file @ 26a9ce7b
/*
* 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_ALSA_OUTPUT_PLUGIN_H
#define MPD_ALSA_OUTPUT_PLUGIN_H
#ifndef MPD_ALSA_OUTPUT_PLUGIN_HXX
#define MPD_ALSA_OUTPUT_PLUGIN_HXX
extern const struct audio_output_plugin alsa_output_plugin;
......
src/output/oss_output_plugin.c → src/output/OssOutputPlugin.cxx
View file @ 26a9ce7b
/*
* 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,
.mixer_plugin = &oss_mixer_plugin,
oss_output_open,
oss_output_close,
nullptr,
nullptr,
oss_output_play,
nullptr,
oss_output_cancel,
nullptr,
&oss_mixer_plugin,
};
src/output/oss_output_plugin.h → src/output/OssOutputPlugin.hxx
View file @ 26a9ce7b
/*
* 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;
......
src/pcm_export.h
View file @ 26a9ce7b
......@@ -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
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