Merge branch 'master' of git://git.infradead.org/users/dwmw2/mpd

Conflicts:

	Makefile.am

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