audio_format: changed "bits" to "enum sample_format"

This patch prepares support for floating point samples (and probably
other formats).  It changes the meaning of the "bits" attribute from a
bit count to a symbolic value.

9 files changed, 108 insertions, 65 deletions

diff --git a/src/output/alsa_plugin.c b/src/output/alsa_plugin.c
index 2c642015d..b7325de07 100644
--- a/src/output/alsa_plugin.c
+++ b/src/output/alsa_plugin.c
@@ -185,13 +185,22 @@ alsa_test_default_device(void)
 static snd_pcm_format_t
 get_bitformat(const struct audio_format *af)
 {
-	switch (af->bits) {
-	case 8: return SND_PCM_FORMAT_S8;
-	case 16: return SND_PCM_FORMAT_S16;
-	case 24: return SND_PCM_FORMAT_S24;
-	case 32: return SND_PCM_FORMAT_S32;
+	switch (af->format) {
+	case SAMPLE_FORMAT_S8:
+		return SND_PCM_FORMAT_S8;
+
+	case SAMPLE_FORMAT_S16:
+		return SND_PCM_FORMAT_S16;
+
+	case SAMPLE_FORMAT_S24_P32:
+		return SND_PCM_FORMAT_S24;
+
+	case SAMPLE_FORMAT_S32:
+		return SND_PCM_FORMAT_S32;
+
+	default:
+		return SND_PCM_FORMAT_UNKNOWN;
 	}
-	return SND_PCM_FORMAT_UNKNOWN;
 }
 
 static snd_pcm_format_t
@@ -264,61 +273,67 @@ configure_hw:
 		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);
+			g_debug("ALSA device \"%s\": converting format %s to reverse-endian",
+				alsa_device(ad),
+				sample_format_to_string(audio_format->format));
 			audio_format->reverse_endian = 1;
 		}
 	}
-	if (err == -EINVAL && (audio_format->bits == 24 ||
-			       audio_format->bits == 16)) {
+	if (err == -EINVAL && (audio_format->format == SAMPLE_FORMAT_S24_P32 ||
+			       audio_format->format == SAMPLE_FORMAT_S16)) {
 		/* 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) {
-			g_debug("ALSA device \"%s\": converting %u bit to 32 bit\n",
-				alsa_device(ad), audio_format->bits);
-			audio_format->bits = 32;
+			g_debug("ALSA device \"%s\": converting format %s to 32 bit\n",
+				alsa_device(ad),
+				sample_format_to_string(audio_format->format));
+			audio_format->format = SAMPLE_FORMAT_S32;
 		}
 	}
-	if (err == -EINVAL && (audio_format->bits == 24 ||
-			       audio_format->bits == 16)) {
+	if (err == -EINVAL && (audio_format->format == SAMPLE_FORMAT_S24_P32 ||
+			       audio_format->format == SAMPLE_FORMAT_S16)) {
 		/* 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;
+			g_debug("ALSA device \"%s\": converting format %s to 32 bit backward-endian\n",
+				alsa_device(ad),
+				sample_format_to_string(audio_format->format));
+			audio_format->format = SAMPLE_FORMAT_S32;
 			audio_format->reverse_endian = 1;
 		}
 	}
 
-	if (err == -EINVAL && audio_format->bits != 16) {
+	if (err == -EINVAL && audio_format->format != SAMPLE_FORMAT_S16) {
 		/* fall back to 16 bit, let pcm_convert.c do the conversion */
 		err = snd_pcm_hw_params_set_format(ad->pcm, hwparams,
 						   SND_PCM_FORMAT_S16);
 		if (err == 0) {
-			g_debug("ALSA device \"%s\": converting %u bit to 16 bit\n",
-				alsa_device(ad), audio_format->bits);
-			audio_format->bits = 16;
+			g_debug("ALSA device \"%s\": converting format %s to 16 bit\n",
+				alsa_device(ad),
+				sample_format_to_string(audio_format->format));
+			audio_format->format = SAMPLE_FORMAT_S16;
 		}
 	}
-	if (err == -EINVAL && audio_format->bits != 16) {
+	if (err == -EINVAL && audio_format->format != SAMPLE_FORMAT_S16) {
 		/* 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;
+			g_debug("ALSA device \"%s\": converting format %s to 16 bit backward-endian\n",
+				alsa_device(ad),
+				sample_format_to_string(audio_format->format));
+			audio_format->format = SAMPLE_FORMAT_S16;
 			audio_format->reverse_endian = 1;
 		}
 	}
 
 	if (err < 0) {
 		g_set_error(error, alsa_output_quark(), err,
-			    "ALSA device \"%s\" does not support %u bit audio: %s",
-			    alsa_device(ad), audio_format->bits,
+			    "ALSA device \"%s\" does not support format %s: %s",
+			    alsa_device(ad),
+			    sample_format_to_string(audio_format->format),
 			    snd_strerror(-err));
 		return false;
 	}
@@ -449,7 +464,7 @@ alsa_open(void *data, struct audio_format *audio_format, GError **error)
 		/* sample format is not supported by this plugin -
 		   fall back to 16 bit samples */
 
-		audio_format->bits = 16;
+		audio_format->format = SAMPLE_FORMAT_S16;
 		bitformat = SND_PCM_FORMAT_S16;
 	}
 
diff --git a/src/output/ao_plugin.c b/src/output/ao_plugin.c
index d69175272..7afca0db2 100644
--- a/src/output/ao_plugin.c
+++ b/src/output/ao_plugin.c
@@ -170,13 +170,24 @@ ao_output_open(void *data, struct audio_format *audio_format,
 	ao_sample_format format;
 	struct ao_data *ad = (struct ao_data *)data;
 
-	/* support for 24 bit samples in libao is currently dubious,
-	   and until we have sorted that out, resample everything to
-	   16 bit */
-	if (audio_format->bits > 16)
-		audio_format->bits = 16;
+	switch (audio_format->format) {
+	case SAMPLE_FORMAT_S8:
+		format.bits = 8;
+		break;
+
+	case SAMPLE_FORMAT_S16:
+		format.bits = 16;
+		break;
+
+	default:
+		/* support for 24 bit samples in libao is currently
+		   dubious, and until we have sorted that out,
+		   convert everything to 16 bit */
+		audio_format->format = SAMPLE_FORMAT_S16;
+		format.bits = 16;
+		break;
+	}
 
-	format.bits = audio_format->bits;
 	format.rate = audio_format->sample_rate;
 	format.byte_format = AO_FMT_NATIVE;
 	format.channels = audio_format->channels;
diff --git a/src/output/jack_output_plugin.c b/src/output/jack_output_plugin.c
index 7e5a52993..f50bc37d0 100644
--- a/src/output/jack_output_plugin.c
+++ b/src/output/jack_output_plugin.c
@@ -157,8 +157,9 @@ set_audioformat(struct jack_data *jd, struct audio_format *audio_format)
 	else if (audio_format->channels > jd->num_source_ports)
 		audio_format->channels = 2;
 
-	if (audio_format->bits != 16 && audio_format->bits != 24)
-		audio_format->bits = 24;
+	if (audio_format->format != SAMPLE_FORMAT_S16 &&
+	    audio_format->format != SAMPLE_FORMAT_S24_P32)
+		audio_format->format = SAMPLE_FORMAT_S24_P32;
 }
 
 static void
@@ -606,13 +607,13 @@ static void
 mpd_jack_write_samples(struct jack_data *jd, const void *src,
 		       unsigned num_samples)
 {
-	switch (jd->audio_format.bits) {
-	case 16:
+	switch (jd->audio_format.format) {
+	case SAMPLE_FORMAT_S16:
 		mpd_jack_write_samples_16(jd, (const int16_t*)src,
 					  num_samples);
 		break;
 
-	case 24:
+	case SAMPLE_FORMAT_S24_P32:
 		mpd_jack_write_samples_24(jd, (const int32_t*)src,
 					  num_samples);
 		break;
diff --git a/src/output/mvp_plugin.c b/src/output/mvp_plugin.c
index 5a9a9b48b..86f147e5a 100644
--- a/src/output/mvp_plugin.c
+++ b/src/output/mvp_plugin.c
@@ -172,19 +172,19 @@ mvp_set_pcm_params(struct mvp_data *md, struct audio_format *audio_format,
 	}
 
 	/* 0,1=24bit(24) , 2,3=16bit */
-	switch (audio_format->bits) {
-	case 16:
+	switch (audio_format->format) {
+	case SAMPLE_FORMAT_S16:
 		mix[1] = 2;
 		break;
 
-	case 24:
+	case SAMPLE_FORMAT_S24_P32:
 		mix[1] = 0;
 		break;
 
 	default:
-		g_debug("unsupported sample format %u - falling back to stereo",
-			audio_format->bits);
-		audio_format->bits = 16;
+		g_debug("unsupported sample format %s - falling back to 16 bit",
+			sample_format_to_string(audio_format->format));
+		audio_format->format = SAMPLE_FORMAT_S16;
 		mix[1] = 2;
 		break;
 	}
diff --git a/src/output/openal_plugin.c b/src/output/openal_plugin.c
index 8fda110e1..0aded4d9a 100644
--- a/src/output/openal_plugin.c
+++ b/src/output/openal_plugin.c
@@ -58,25 +58,29 @@ openal_output_quark(void)
 static ALenum
 openal_audio_format(struct audio_format *audio_format)
 {
-	/* Only 8 and 16 bit samples are supported */
-	if (audio_format->bits != 16 && audio_format->bits != 8)
-		audio_format->bits = 16;
-
-	switch (audio_format->bits)
-	{
-	case 16:
+	switch (audio_format->format) {
+	case SAMPLE_FORMAT_S16:
 		if (audio_format->channels == 2)
 			return AL_FORMAT_STEREO16;
 		if (audio_format->channels == 1)
 			return AL_FORMAT_MONO16;
 		break;
 
-	case 8:
+	case SAMPLE_FORMAT_S8:
 		if (audio_format->channels == 2)
 			return AL_FORMAT_STEREO8;
 		if (audio_format->channels == 1)
 			return AL_FORMAT_MONO8;
 		break;
+
+	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 0;
diff --git a/src/output/oss_plugin.c b/src/output/oss_plugin.c
index b02d7d62e..f16374e39 100644
--- a/src/output/oss_plugin.c
+++ b/src/output/oss_plugin.c
@@ -490,17 +490,18 @@ oss_setup(struct oss_data *od, GError **error)
 	}
 	od->audio_format.sample_rate = tmp;
 
-	switch (od->audio_format.bits) {
-	case 8:
+	switch (od->audio_format.format) {
+	case SAMPLE_FORMAT_S8:
 		tmp = AFMT_S8;
 		break;
-	case 16:
+
+	case SAMPLE_FORMAT_S16:
 		tmp = AFMT_S16_MPD;
 		break;
 
 	default:
 		/* not supported by OSS - fall back to 16 bit */
-		od->audio_format.bits = 16;
+		od->audio_format.format = SAMPLE_FORMAT_S16;
 		tmp = AFMT_S16_MPD;
 		break;
 	}
diff --git a/src/output/osx_plugin.c b/src/output/osx_plugin.c
index afcd143b3..22b742ee5 100644
--- a/src/output/osx_plugin.c
+++ b/src/output/osx_plugin.c
@@ -166,9 +166,6 @@ osx_output_open(void *data, struct audio_format *audio_format, GError **error)
 	OSStatus status;
 	ComponentResult result;
 
-	if (audio_format->bits > 16)
-		audio_format->bits = 16;
-
 	desc.componentType = kAudioUnitType_Output;
 	desc.componentSubType = kAudioUnitSubType_DefaultOutput;
 	desc.componentManufacturer = kAudioUnitManufacturer_Apple;
@@ -226,7 +223,21 @@ osx_output_open(void *data, struct audio_format *audio_format, GError **error)
 	stream_description.mFramesPerPacket = 1;
 	stream_description.mBytesPerFrame = stream_description.mBytesPerPacket;
 	stream_description.mChannelsPerFrame = audio_format->channels;
-	stream_description.mBitsPerChannel = audio_format->bits;
+
+	switch (audio_format->format) {
+	case SAMPLE_FORMAT_S8:
+		stream_description.mBitsPerChannel = 8;
+		break;
+
+	case SAMPLE_FORMAT_S16:
+		stream_description.mBitsPerChannel = 16;
+		break;
+
+	default:
+		audio_format->format = SAMPLE_FORMAT_S16;
+		stream_description.mBitsPerChannel = 16;
+		break;
+	}
 
 	result = AudioUnitSetProperty(od->au, kAudioUnitProperty_StreamFormat,
 				      kAudioUnitScope_Input, 0,
diff --git a/src/output/pulse_output_plugin.c b/src/output/pulse_output_plugin.c
index 3da1b3593..a64157920 100644
--- a/src/output/pulse_output_plugin.c
+++ b/src/output/pulse_output_plugin.c
@@ -467,7 +467,7 @@ pulse_output_open(void *data, struct audio_format *audio_format,
 
 	/* MPD doesn't support the other pulseaudio sample formats, so
 	   we just force MPD to send us everything as 16 bit */
-	audio_format->bits = 16;
+	audio_format->format = SAMPLE_FORMAT_S16;
 
 	ss.format = PA_SAMPLE_S16NE;
 	ss.rate = audio_format->sample_rate;
diff --git a/src/output/solaris_output_plugin.c b/src/output/solaris_output_plugin.c
index b187630ee..fe84068f1 100644
--- a/src/output/solaris_output_plugin.c
+++ b/src/output/solaris_output_plugin.c
@@ -89,7 +89,7 @@ solaris_output_open(void *data, struct audio_format *audio_format,
 
 	/* support only 16 bit mono/stereo for now; nothing else has
 	   been tested */
-	audio_format->bits = 16;
+	audio_format->format = SAMPLE_FORMAT_S16;
 
 	/* open the device in non-blocking mode */
 
@@ -119,7 +119,7 @@ solaris_output_open(void *data, struct audio_format *audio_format,
 
 	info.play.sample_rate = audio_format->sample_rate;
 	info.play.channels = audio_format->channels;
-	info.play.precision = audio_format->bits;
+	info.play.precision = 16;
 	info.play.encoding = AUDIO_ENCODING_LINEAR;
 
 	ret = ioctl(so->fd, AUDIO_SETINFO, &info);