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| author | Max Kellermann <max@duempel.org> | 2009-03-26 18:23:23 +0100 |
|---|---|---|
| committer | Max Kellermann <max@duempel.org> | 2009-03-26 18:23:23 +0100 |
| commit | 66a2c5669e1fcd709a00bd4de8ddfb0d3a0c2a58 (patch) | |
| tree | c138f37dfc13af01df8606ed7a8f882964a32f27 /src/output/alsa_plugin.c | |
| parent | 209c8a540c80faffb6b04ac6e7fed64a03c679ac (diff) | |
| download | mpd-66a2c5669e1fcd709a00bd4de8ddfb0d3a0c2a58.tar.gz mpd-66a2c5669e1fcd709a00bd4de8ddfb0d3a0c2a58.tar.xz mpd-66a2c5669e1fcd709a00bd4de8ddfb0d3a0c2a58.zip | |
output_plugin: replaced output_plugin.get_mixer() with mixer_plugin
The mixer core library is now responsible for creating and managing
the mixer object. This removes duplicated code from the output
plugins.
Diffstat (limited to 'src/output/alsa_plugin.c')
| -rw-r--r-- | src/output/alsa_plugin.c | 20 |
1 files changed, 1 insertions, 19 deletions
diff --git a/src/output/alsa_plugin.c b/src/output/alsa_plugin.c index 69f4d500f..0295742e2 100644 --- a/src/output/alsa_plugin.c +++ b/src/output/alsa_plugin.c @@ -19,7 +19,6 @@ #include "../output_api.h" #include "mixer_list.h" -#include "mixer_control.h" #include <glib.h> #include <alsa/asoundlib.h> @@ -70,9 +69,6 @@ struct alsa_data { /** the size of one audio frame */ size_t frame_size; - - /** the mixer object associated with this output */ - struct mixer *mixer; }; /** @@ -105,7 +101,6 @@ static void alsa_data_free(struct alsa_data *ad) { g_free(ad->device); - mixer_free(ad->mixer); g_free(ad); } @@ -151,7 +146,6 @@ alsa_init(G_GNUC_UNUSED const struct audio_format *audio_format, } alsa_configure(ad, param); - ad->mixer = mixer_new(&alsa_mixer, param); return ad; } @@ -164,14 +158,6 @@ alsa_finish(void *data) alsa_data_free(ad); } -static struct mixer * -alsa_get_mixer(void *data) -{ - struct alsa_data *ad = data; - - return ad->mixer; -} - static bool alsa_test_default_device(void) { @@ -400,8 +386,6 @@ alsa_open(void *data, struct audio_format *audio_format, GError **error) int err; bool success; - mixer_open(ad->mixer); - bitformat = get_bitformat(audio_format); if (bitformat == SND_PCM_FORMAT_UNKNOWN) { /* sample format is not supported by this plugin - @@ -484,8 +468,6 @@ alsa_close(void *data) snd_pcm_drain(ad->pcm); snd_pcm_close(ad->pcm); - - mixer_close(ad->mixer); } static size_t @@ -514,9 +496,9 @@ const struct audio_output_plugin alsaPlugin = { .test_default_device = alsa_test_default_device, .init = alsa_init, .finish = alsa_finish, - .get_mixer = alsa_get_mixer, .open = alsa_open, .play = alsa_play, .cancel = alsa_cancel, .close = alsa_close, + .mixer_plugin = &alsa_mixer, }; |