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-rw-r--r--src/output/AlsaOutputPlugin.cxx78
1 files changed, 36 insertions, 42 deletions
diff --git a/src/output/AlsaOutputPlugin.cxx b/src/output/AlsaOutputPlugin.cxx
index d707166c5..c9bf01909 100644
--- a/src/output/AlsaOutputPlugin.cxx
+++ b/src/output/AlsaOutputPlugin.cxx
@@ -23,6 +23,8 @@
#include "MixerList.hxx"
#include "pcm/PcmExport.hxx"
#include "util/Manual.hxx"
+#include "util/Error.hxx"
+#include "util/Domain.hxx"
#include <glib.h>
#include <alsa/asoundlib.h>
@@ -115,9 +117,9 @@ struct AlsaOutput {
AlsaOutput():mode(0), writei(snd_pcm_writei) {
}
- bool Init(const config_param &param, GError **error_r) {
+ bool Init(const config_param &param, Error &error) {
return ao_base_init(&base, &alsa_output_plugin,
- param, error_r);
+ param, error);
}
void Deinit() {
@@ -125,14 +127,7 @@ struct AlsaOutput {
}
};
-/**
- * The quark used for GError.domain.
- */
-static inline GQuark
-alsa_output_quark(void)
-{
- return g_quark_from_static_string("alsa_output");
-}
+static constexpr Domain alsa_output_domain("alsa_output");
static const char *
alsa_device(const AlsaOutput *ad)
@@ -170,11 +165,11 @@ alsa_configure(AlsaOutput *ad, const config_param &param)
}
static struct audio_output *
-alsa_init(const config_param &param, GError **error_r)
+alsa_init(const config_param &param, Error &error)
{
AlsaOutput *ad = new AlsaOutput();
- if (!ad->Init(param, error_r)) {
+ if (!ad->Init(param, error)) {
delete ad;
return NULL;
}
@@ -197,7 +192,7 @@ alsa_finish(struct audio_output *ao)
}
static bool
-alsa_output_enable(struct audio_output *ao, gcc_unused GError **error_r)
+alsa_output_enable(struct audio_output *ao, gcc_unused Error &error)
{
AlsaOutput *ad = (AlsaOutput *)ao;
@@ -394,7 +389,7 @@ alsa_output_setup_format(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
*/
static bool
alsa_setup(AlsaOutput *ad, AudioFormat &audio_format,
- bool *packed_r, bool *reverse_endian_r, GError **error)
+ bool *packed_r, bool *reverse_endian_r, Error &error)
{
unsigned int sample_rate = audio_format.sample_rate;
unsigned int channels = audio_format.channels;
@@ -438,11 +433,11 @@ configure_hw:
err = alsa_output_setup_format(ad->pcm, hwparams, audio_format,
packed_r, reverse_endian_r);
if (err < 0) {
- g_set_error(error, alsa_output_quark(), err,
- "ALSA device \"%s\" does not support format %s: %s",
- alsa_device(ad),
- sample_format_to_string(audio_format.format),
- snd_strerror(-err));
+ error.Format(alsa_output_domain, err,
+ "ALSA device \"%s\" does not support format %s: %s",
+ alsa_device(ad),
+ sample_format_to_string(audio_format.format),
+ snd_strerror(-err));
return false;
}
@@ -454,10 +449,10 @@ configure_hw:
err = snd_pcm_hw_params_set_channels_near(ad->pcm, hwparams,
&channels);
if (err < 0) {
- g_set_error(error, alsa_output_quark(), err,
- "ALSA device \"%s\" does not support %i channels: %s",
- alsa_device(ad), (int)audio_format.channels,
- snd_strerror(-err));
+ error.Format(alsa_output_domain, err,
+ "ALSA device \"%s\" does not support %i channels: %s",
+ alsa_device(ad), (int)audio_format.channels,
+ snd_strerror(-err));
return false;
}
audio_format.channels = (int8_t)channels;
@@ -465,9 +460,9 @@ configure_hw:
err = snd_pcm_hw_params_set_rate_near(ad->pcm, hwparams,
&sample_rate, NULL);
if (err < 0 || sample_rate == 0) {
- g_set_error(error, alsa_output_quark(), err,
- "ALSA device \"%s\" does not support %u Hz audio",
- alsa_device(ad), audio_format.sample_rate);
+ error.Format(alsa_output_domain, err,
+ "ALSA device \"%s\" does not support %u Hz audio",
+ alsa_device(ad), audio_format.sample_rate);
return false;
}
audio_format.sample_rate = sample_rate;
@@ -594,16 +589,16 @@ configure_hw:
return true;
error:
- g_set_error(error, alsa_output_quark(), err,
- "Error opening ALSA device \"%s\" (%s): %s",
- alsa_device(ad), cmd, snd_strerror(-err));
+ error.Format(alsa_output_domain, err,
+ "Error opening ALSA device \"%s\" (%s): %s",
+ alsa_device(ad), cmd, snd_strerror(-err));
return false;
}
static bool
alsa_setup_dsd(AlsaOutput *ad, const AudioFormat audio_format,
bool *shift8_r, bool *packed_r, bool *reverse_endian_r,
- GError **error_r)
+ Error &error)
{
assert(ad->dsd_usb);
assert(audio_format.format == SampleFormat::DSD);
@@ -616,7 +611,7 @@ alsa_setup_dsd(AlsaOutput *ad, const AudioFormat audio_format,
const AudioFormat check = usb_format;
- if (!alsa_setup(ad, usb_format, packed_r, reverse_endian_r, error_r))
+ if (!alsa_setup(ad, usb_format, packed_r, reverse_endian_r, error))
return false;
/* if the device allows only 32 bit, shift all DSD-over-USB
@@ -631,9 +626,9 @@ alsa_setup_dsd(AlsaOutput *ad, const AudioFormat audio_format,
if (usb_format != check) {
/* no bit-perfect playback, which is required
for DSD over USB */
- g_set_error(error_r, alsa_output_quark(), 0,
- "Failed to configure DSD-over-USB on ALSA device \"%s\"",
- alsa_device(ad));
+ error.Format(alsa_output_domain,
+ "Failed to configure DSD-over-USB on ALSA device \"%s\"",
+ alsa_device(ad));
g_free(ad->silence);
return false;
}
@@ -643,7 +638,7 @@ alsa_setup_dsd(AlsaOutput *ad, const AudioFormat audio_format,
static bool
alsa_setup_or_dsd(AlsaOutput *ad, AudioFormat &audio_format,
- GError **error_r)
+ Error &error)
{
bool shift8 = false, packed, reverse_endian;
@@ -652,9 +647,9 @@ alsa_setup_or_dsd(AlsaOutput *ad, AudioFormat &audio_format,
const bool success = dsd_usb
? alsa_setup_dsd(ad, audio_format,
&shift8, &packed, &reverse_endian,
- error_r)
+ error)
: alsa_setup(ad, audio_format, &packed, &reverse_endian,
- error_r);
+ error);
if (!success)
return false;
@@ -665,14 +660,14 @@ alsa_setup_or_dsd(AlsaOutput *ad, AudioFormat &audio_format,
}
static bool
-alsa_open(struct audio_output *ao, AudioFormat &audio_format, GError **error)
+alsa_open(struct audio_output *ao, AudioFormat &audio_format, Error &error)
{
AlsaOutput *ad = (AlsaOutput *)ao;
int err = snd_pcm_open(&ad->pcm, alsa_device(ad),
SND_PCM_STREAM_PLAYBACK, ad->mode);
if (err < 0) {
- g_set_error(error, alsa_output_quark(), err,
+ error.Format(alsa_output_domain, err,
"Failed to open ALSA device \"%s\": %s",
alsa_device(ad), snd_strerror(err));
return false;
@@ -800,7 +795,7 @@ alsa_close(struct audio_output *ao)
static size_t
alsa_play(struct audio_output *ao, const void *chunk, size_t size,
- GError **error)
+ Error &error)
{
AlsaOutput *ad = (AlsaOutput *)ao;
@@ -824,8 +819,7 @@ alsa_play(struct audio_output *ao, const void *chunk, size_t size,
if (ret < 0 && ret != -EAGAIN && ret != -EINTR &&
alsa_recover(ad, ret) < 0) {
- g_set_error(error, alsa_output_quark(), errno,
- "%s", snd_strerror(-errno));
+ error.Set(alsa_output_domain, ret, snd_strerror(-ret));
return 0;
}
}
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-23 01:44:45 +0000