From c412d6251e9cd3abe735b7622af4003502e54f72 Mon Sep 17 00:00:00 2001
From: Max Kellermann <max@duempel.org>
Date: Tue, 10 Nov 2009 17:11:34 +0100
Subject: 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.
---
src/output/alsa_plugin.c | 73 +++++++++++++++++++++++---------------
src/output/ao_plugin.c | 23 ++++++++----
src/output/jack_output_plugin.c | 11 +++---
src/output/mvp_plugin.c | 12 +++----
src/output/openal_plugin.c | 20 ++++++-----
src/output/oss_plugin.c | 9 ++---
src/output/osx_plugin.c | 19 +++++++---
src/output/pulse_output_plugin.c | 2 +-
src/output/solaris_output_plugin.c | 4 +--
9 files changed, 108 insertions(+), 65 deletions(-)
(limited to 'src/output')
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);
--
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