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authorMax Kellermann <max@duempel.org>2009-07-19 18:18:32 +0200
committerMax Kellermann <max@duempel.org>2009-07-19 18:18:32 +0200
commitc9d43b4d713d508b6f5b67cfffba9296f9436934 (patch)
tree70100c28e365a244f5c6a361d2ecda332df2a756 /src/output
parentc5ec035fb4a533ebd9f8c69b085604560d2a5487 (diff)
parent49ede85827c095d0a6ead0ecb63e83e000a76d4f (diff)
downloadmpd-c9d43b4d713d508b6f5b67cfffba9296f9436934.tar.gz
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Merge branch 'master' of git://git.infradead.org/users/dwmw2/mpd
Conflicts: Makefile.am
Diffstat (limited to '')
-rw-r--r--src/output/alsa_plugin.c52
-rw-r--r--src/output_thread.c10
2 files changed, 56 insertions, 6 deletions
diff --git a/src/output/alsa_plugin.c b/src/output/alsa_plugin.c
index 818c83ca2..f271668b1 100644
--- a/src/output/alsa_plugin.c
+++ b/src/output/alsa_plugin.c
@@ -183,6 +183,19 @@ get_bitformat(const struct audio_format *af)
return SND_PCM_FORMAT_UNKNOWN;
}
+static snd_pcm_format_t
+byteswap_bitformat(snd_pcm_format_t fmt)
+{
+ switch(fmt) {
+ case SND_PCM_FORMAT_S16_LE: return SND_PCM_FORMAT_S16_BE;
+ case SND_PCM_FORMAT_S24_LE: return SND_PCM_FORMAT_S24_BE;
+ case SND_PCM_FORMAT_S32_LE: return SND_PCM_FORMAT_S32_BE;
+ case SND_PCM_FORMAT_S16_BE: return SND_PCM_FORMAT_S16_LE;
+ case SND_PCM_FORMAT_S24_BE: return SND_PCM_FORMAT_S24_LE;
+ case SND_PCM_FORMAT_S32_BE: return SND_PCM_FORMAT_S32_LE;
+ default: return SND_PCM_FORMAT_UNKNOWN;
+ }
+}
/**
* Set up the snd_pcm_t object which was opened by the caller. Set up
* the configured settings and the audio format.
@@ -208,7 +221,6 @@ alsa_setup(struct alsa_data *ad, struct audio_format *audio_format,
configure_hw:
/* configure HW params */
snd_pcm_hw_params_alloca(&hwparams);
-
cmd = "snd_pcm_hw_params_any";
err = snd_pcm_hw_params_any(ad->pcm, hwparams);
if (err < 0)
@@ -236,13 +248,38 @@ configure_hw:
}
err = snd_pcm_hw_params_set_format(ad->pcm, hwparams, bitformat);
+ if (err == -EINVAL &&
+ byteswap_bitformat(bitformat) != SND_PCM_FORMAT_UNKNOWN) {
+ err = snd_pcm_hw_params_set_format(ad->pcm, hwparams,
+ byteswap_bitformat(bitformat));
+ if (err == 0) {
+ g_debug("ALSA device \"%s\": converting %u bit to reverse-endian\n",
+ alsa_device(ad), audio_format->bits);
+ audio_format->reverse_endian = 1;
+ }
+ }
if (err == -EINVAL && (audio_format->bits == 24 ||
audio_format->bits == 16)) {
/* fall back to 32 bit, let pcm_convert.c do the conversion */
err = snd_pcm_hw_params_set_format(ad->pcm, hwparams,
SND_PCM_FORMAT_S32);
- if (err == 0)
+ if (err == 0) {
+ g_debug("ALSA device \"%s\": converting %u bit to 32 bit\n",
+ alsa_device(ad), audio_format->bits);
+ audio_format->bits = 32;
+ }
+ }
+ if (err == -EINVAL && (audio_format->bits == 24 ||
+ audio_format->bits == 16)) {
+ /* fall back to 32 bit, let pcm_convert.c do the conversion */
+ err = snd_pcm_hw_params_set_format(ad->pcm, hwparams,
+ byteswap_bitformat(SND_PCM_FORMAT_S32));
+ if (err == 0) {
+ g_debug("ALSA device \"%s\": converting %u bit to 32 bit backward-endian\n",
+ alsa_device(ad), audio_format->bits);
audio_format->bits = 32;
+ audio_format->reverse_endian = 1;
+ }
}
if (err == -EINVAL && audio_format->bits != 16) {
@@ -255,6 +292,17 @@ configure_hw:
audio_format->bits = 16;
}
}
+ if (err == -EINVAL && audio_format->bits != 16) {
+ /* fall back to 16 bit, let pcm_convert.c do the conversion */
+ err = snd_pcm_hw_params_set_format(ad->pcm, hwparams,
+ byteswap_bitformat(SND_PCM_FORMAT_S16));
+ if (err == 0) {
+ g_debug("ALSA device \"%s\": converting %u bit to 16 bit backward-endian\n",
+ alsa_device(ad), audio_format->bits);
+ audio_format->bits = 16;
+ audio_format->reverse_endian = 1;
+ }
+ }
if (err < 0) {
g_set_error(error, alsa_output_quark(), err,
diff --git a/src/output_thread.c b/src/output_thread.c
index 2592b2456..c7bd069b1 100644
--- a/src/output_thread.c
+++ b/src/output_thread.c
@@ -93,18 +93,20 @@ ao_open(struct audio_output *ao)
g_mutex_unlock(ao->mutex);
g_debug("opened plugin=%s name=\"%s\" "
- "audio_format=%u:%u:%u",
+ "audio_format=%u:%u:%u:%u",
ao->plugin->name, ao->name,
ao->out_audio_format.sample_rate,
ao->out_audio_format.bits,
- ao->out_audio_format.channels);
+ ao->out_audio_format.channels,
+ ao->out_audio_format.reverse_endian);
if (!audio_format_equals(&ao->in_audio_format,
&ao->out_audio_format))
- g_debug("converting from %u:%u:%u",
+ g_debug("converting from %u:%u:%u:%u",
ao->in_audio_format.sample_rate,
ao->in_audio_format.bits,
- ao->in_audio_format.channels);
+ ao->in_audio_format.channels,
+ ao->in_audio_format.reverse_endian);
}
static void