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output/alsa: moved code to alsa_output_setup_format()

This commit is contained in:
Max Kellermann 2010-01-16 17:14:30 +01:00
parent 87c861cae3
commit 79848e3414
1 changed files with 82 additions and 74 deletions
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156
src/output/alsa_plugin.c
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@ -216,13 +216,92 @@ byteswap_bitformat(snd_pcm_format_t fmt)
default: return SND_PCM_FORMAT_UNKNOWN;
}
}
/**
* Configure a sample format, and probe other formats if that fails.
*/
static int
alsa_output_setup_format(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
struct audio_format *audio_format)
{
snd_pcm_format_t bitformat = get_bitformat(audio_format);
if (bitformat == SND_PCM_FORMAT_UNKNOWN) {
/* sample format is not supported by this plugin -
fall back to 16 bit samples */
audio_format->format = SAMPLE_FORMAT_S16;
bitformat = SND_PCM_FORMAT_S16;
}
int err = snd_pcm_hw_params_set_format(pcm, hwparams, bitformat);
if (err == -EINVAL &&
byteswap_bitformat(bitformat) != SND_PCM_FORMAT_UNKNOWN) {
err = snd_pcm_hw_params_set_format(pcm, hwparams,
byteswap_bitformat(bitformat));
if (err == 0) {
g_debug("converting format %s to reverse-endian",
sample_format_to_string(audio_format->format));
audio_format->reverse_endian = 1;
}
}
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(pcm, hwparams,
SND_PCM_FORMAT_S32);
if (err == 0) {
g_debug("converting format %s to 32 bit\n",
sample_format_to_string(audio_format->format));
audio_format->format = SAMPLE_FORMAT_S32;
}
}
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(pcm, hwparams,
byteswap_bitformat(SND_PCM_FORMAT_S32));
if (err == 0) {
g_debug("converting format %s to 32 bit backward-endian\n",
sample_format_to_string(audio_format->format));
audio_format->format = SAMPLE_FORMAT_S32;
audio_format->reverse_endian = 1;
}
}
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(pcm, hwparams,
SND_PCM_FORMAT_S16);
if (err == 0) {
g_debug("converting format %s to 16 bit\n",
sample_format_to_string(audio_format->format));
audio_format->format = SAMPLE_FORMAT_S16;
}
}
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(pcm, hwparams,
byteswap_bitformat(SND_PCM_FORMAT_S16));
if (err == 0) {
g_debug("converting format %s to 16 bit backward-endian\n",
sample_format_to_string(audio_format->format));
audio_format->format = SAMPLE_FORMAT_S16;
audio_format->reverse_endian = 1;
}
}
return err;
}
/**
* Set up the snd_pcm_t object which was opened by the caller. Set up
* the configured settings and the audio format.
*/
static bool
alsa_setup(struct alsa_data *ad, struct audio_format *audio_format,
snd_pcm_format_t bitformat,
GError **error)
{
snd_pcm_hw_params_t *hwparams;
@ -267,68 +346,7 @@ configure_hw:
ad->writei = snd_pcm_writei;
}
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 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->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 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->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 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->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 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->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 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;
}
}
err = alsa_output_setup_format(ad->pcm, hwparams, audio_format);
if (err < 0) {
g_set_error(error, alsa_output_quark(), err,
"ALSA device \"%s\" does not support format %s: %s",
@ -455,19 +473,9 @@ static bool
alsa_open(void *data, struct audio_format *audio_format, GError **error)
{
struct alsa_data *ad = data;
snd_pcm_format_t bitformat;
int err;
bool success;
bitformat = get_bitformat(audio_format);
if (bitformat == SND_PCM_FORMAT_UNKNOWN) {
/* sample format is not supported by this plugin -
fall back to 16 bit samples */
audio_format->format = SAMPLE_FORMAT_S16;
bitformat = SND_PCM_FORMAT_S16;
}
err = snd_pcm_open(&ad->pcm, alsa_device(ad),
SND_PCM_STREAM_PLAYBACK, ad->mode);
if (err < 0) {
@ -477,7 +485,7 @@ alsa_open(void *data, struct audio_format *audio_format, GError **error)
return false;
}
success = alsa_setup(ad, audio_format, bitformat, error);
success = alsa_setup(ad, audio_format, error);
if (!success) {
snd_pcm_close(ad->pcm);
return false;