diff options
-rw-r--r-- | Makefile.am | 8 | ||||
-rw-r--r-- | src/OutputList.cxx | 4 | ||||
-rw-r--r-- | src/mixer/OssMixerPlugin.cxx (renamed from src/mixer/oss_mixer_plugin.c) | 16 | ||||
-rw-r--r-- | src/output/AlsaOutputPlugin.cxx (renamed from src/output/alsa_output_plugin.c) | 150 | ||||
-rw-r--r-- | src/output/AlsaOutputPlugin.hxx (renamed from src/output/alsa_output_plugin.h) | 6 | ||||
-rw-r--r-- | src/output/OssOutputPlugin.cxx (renamed from src/output/oss_output_plugin.c) | 64 | ||||
-rw-r--r-- | src/output/OssOutputPlugin.hxx (renamed from src/output/oss_output_plugin.h) | 6 | ||||
-rw-r--r-- | src/pcm_export.h | 8 |
8 files changed, 142 insertions, 120 deletions
diff --git a/Makefile.am b/Makefile.am index 4a1dfea7..95986567 100644 --- a/Makefile.am +++ b/Makefile.am @@ -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 diff --git a/src/OutputList.cxx b/src/OutputList.cxx index a9a0b3d3..87e44175 100644 --- a/src/OutputList.cxx +++ b/src/OutputList.cxx @@ -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" diff --git a/src/mixer/oss_mixer_plugin.c b/src/mixer/OssMixerPlugin.cxx index 608f1f9b..490a6541 100644 --- a/src/mixer/oss_mixer_plugin.c +++ b/src/mixer/OssMixerPlugin.cxx @@ -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 @@ -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, }; diff --git a/src/output/alsa_output_plugin.c b/src/output/AlsaOutputPlugin.cxx index d8b18427..4d9f259a 100644 --- a/src/output/alsa_output_plugin.c +++ b/src/output/AlsaOutputPlugin.cxx @@ -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 "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, }; diff --git a/src/output/alsa_output_plugin.h b/src/output/AlsaOutputPlugin.hxx index daa1f361..dc7e639a 100644 --- a/src/output/alsa_output_plugin.h +++ b/src/output/AlsaOutputPlugin.hxx @@ -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_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; diff --git a/src/output/oss_output_plugin.c b/src/output/OssOutputPlugin.cxx index e366a453..5810825f 100644 --- a/src/output/oss_output_plugin.c +++ b/src/output/OssOutputPlugin.cxx @@ -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, - - .mixer_plugin = &oss_mixer_plugin, + oss_output_open, + oss_output_close, + nullptr, + nullptr, + oss_output_play, + nullptr, + oss_output_cancel, + nullptr, + + &oss_mixer_plugin, }; diff --git a/src/output/oss_output_plugin.h b/src/output/OssOutputPlugin.hxx index 2aecc2b3..6c5c9530 100644 --- a/src/output/oss_output_plugin.h +++ b/src/output/OssOutputPlugin.hxx @@ -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; diff --git a/src/pcm_export.h b/src/pcm_export.h index a7e7c3f6..005db48e 100644 --- a/src/pcm_export.h +++ b/src/pcm_export.h @@ -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 |