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| author | Max Kellermann <max@duempel.org> | 2011-10-08 14:50:44 +0200 |
|---|---|---|
| committer | Max Kellermann <max@duempel.org> | 2011-10-08 14:50:44 +0200 |
| commit | b43bf4dd741f81239c7ae0c99a048ef0be6286ed (patch) | |
| tree | 3cb219018da3e22bab9b69c1bb6f94bb312b0329 /src/output | |
| parent | 14424281a0e1407ca04ef05a50541a4178f70e91 (diff) | |
| parent | 5ed0eb51d16507dcca9764771af8f05f08a078f9 (diff) | |
| download | mpd-b43bf4dd741f81239c7ae0c99a048ef0be6286ed.tar.gz mpd-b43bf4dd741f81239c7ae0c99a048ef0be6286ed.tar.xz mpd-b43bf4dd741f81239c7ae0c99a048ef0be6286ed.zip | |
Merge branch 'v0.16.x'
Diffstat (limited to '')
| -rw-r--r-- | src/output/alsa_output_plugin.c | 9 | ||||
| -rw-r--r-- | src/output/openal_output_plugin.c | 18 |
2 files changed, 15 insertions, 12 deletions
diff --git a/src/output/alsa_output_plugin.c b/src/output/alsa_output_plugin.c index 41293272d..af6d1b184 100644 --- a/src/output/alsa_output_plugin.c +++ b/src/output/alsa_output_plugin.c @@ -193,6 +193,9 @@ static snd_pcm_format_t get_bitformat(enum sample_format sample_format) { switch (sample_format) { + case SAMPLE_FORMAT_UNDEFINED: + return SND_PCM_FORMAT_UNKNOWN; + case SAMPLE_FORMAT_S8: return SND_PCM_FORMAT_S8; @@ -209,10 +212,10 @@ get_bitformat(enum sample_format sample_format) case SAMPLE_FORMAT_S32: return SND_PCM_FORMAT_S32; - - default: - return SND_PCM_FORMAT_UNKNOWN; } + + assert(false); + return SND_PCM_FORMAT_UNKNOWN; } static snd_pcm_format_t diff --git a/src/output/openal_output_plugin.c b/src/output/openal_output_plugin.c index 1473659f0..622cf559e 100644 --- a/src/output/openal_output_plugin.c +++ b/src/output/openal_output_plugin.c @@ -67,26 +67,26 @@ openal_audio_format(struct audio_format *audio_format) return AL_FORMAT_STEREO16; if (audio_format->channels == 1) return AL_FORMAT_MONO16; - break; + + /* fall back to mono */ + audio_format->channels = 1; + return openal_audio_format(audio_format); case SAMPLE_FORMAT_S8: if (audio_format->channels == 2) return AL_FORMAT_STEREO8; if (audio_format->channels == 1) return AL_FORMAT_MONO8; - break; + + /* fall back to mono */ + audio_format->channels = 1; + return openal_audio_format(audio_format); default: /* fall back to 16 bit */ audio_format->format = SAMPLE_FORMAT_S16; - if (audio_format->channels == 2) - return AL_FORMAT_STEREO16; - if (audio_format->channels == 1) - return AL_FORMAT_MONO16; - break; + return openal_audio_format(audio_format); } - - return 0; } static bool |