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Иван Мажукин
mpd
Commits
3dba09f3
Commit
3dba09f3
authored
Mar 21, 2012
by
Max Kellermann
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output/alsa: always receive host byte order samples
Don't use audio_format.reverse_endian.
parent
7ebf8e66
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Showing
2 changed files
with
62 additions
and
3 deletions
+62
-3
Makefile.am
Makefile.am
+1
-0
alsa_output_plugin.c
src/output/alsa_output_plugin.c
+61
-3
No files found.
Makefile.am
View file @
3dba09f3
...
...
@@ -1201,6 +1201,7 @@ test_run_output_LDADD = $(MPD_LIBS) \
$(ENCODER_LIBS)
\
libmixer_plugins.a
\
$(FILTER_LIBS)
\
libutil.a
\
$(GLIB_LIBS)
test_run_output_SOURCES
=
test
/run_output.c
\
test
/stdbin.h
\
...
...
src/output/alsa_output_plugin.c
View file @
3dba09f3
...
...
@@ -21,6 +21,8 @@
#include "alsa_output_plugin.h"
#include "output_api.h"
#include "mixer_list.h"
#include "pcm_buffer.h"
#include "pcm_byteswap.h"
#include <glib.h>
#include <alsa/asoundlib.h>
...
...
@@ -45,6 +47,13 @@ typedef snd_pcm_sframes_t alsa_writei_t(snd_pcm_t * pcm, const void *buffer,
struct
alsa_data
{
struct
audio_output
base
;
/**
* The buffer used to reverse the byte order.
*
* @see #reverse_endian
*/
struct
pcm_buffer
reverse_buffer
;
/** the configured name of the ALSA device; NULL for the
default device */
char
*
device
;
...
...
@@ -52,6 +61,21 @@ struct alsa_data {
/** use memory mapped I/O? */
bool
use_mmap
;
/**
* Does ALSA expect samples in reverse byte order? (i.e. not
* host byte order)
*
* This attribute is only valid while the device is open.
*/
bool
reverse_endian
;
/**
* Which sample format is being sent to the play() method?
*
* This attribute is only valid while the device is open.
*/
enum
sample_format
sample_format
;
/** libasound's buffer_time setting (in microseconds) */
unsigned
int
buffer_time
;
...
...
@@ -168,6 +192,23 @@ alsa_finish(struct audio_output *ao)
}
static
bool
alsa_output_enable
(
struct
audio_output
*
ao
,
G_GNUC_UNUSED
GError
**
error_r
)
{
struct
alsa_data
*
ad
=
(
struct
alsa_data
*
)
ao
;
pcm_buffer_init
(
&
ad
->
reverse_buffer
);
return
true
;
}
static
void
alsa_output_disable
(
struct
audio_output
*
ao
)
{
struct
alsa_data
*
ad
=
(
struct
alsa_data
*
)
ao
;
pcm_buffer_deinit
(
&
ad
->
reverse_buffer
);
}
static
bool
alsa_test_default_device
(
void
)
{
snd_pcm_t
*
handle
;
...
...
@@ -288,13 +329,18 @@ alsa_output_try_reverse(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
static
int
alsa_output_try_format_both
(
snd_pcm_t
*
pcm
,
snd_pcm_hw_params_t
*
hwparams
,
struct
audio_format
*
audio_format
,
bool
*
reverse_endian_r
,
enum
sample_format
sample_format
)
{
*
reverse_endian_r
=
false
;
int
err
=
alsa_output_try_format
(
pcm
,
hwparams
,
audio_format
,
sample_format
);
if
(
err
==
-
EINVAL
)
if
(
err
==
-
EINVAL
)
{
*
reverse_endian_r
=
true
;
err
=
alsa_output_try_reverse
(
pcm
,
hwparams
,
audio_format
,
sample_format
);
}
return
err
;
}
...
...
@@ -304,11 +350,13 @@ alsa_output_try_format_both(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
*/
static
int
alsa_output_setup_format
(
snd_pcm_t
*
pcm
,
snd_pcm_hw_params_t
*
hwparams
,
struct
audio_format
*
audio_format
)
struct
audio_format
*
audio_format
,
bool
*
reverse_endian_r
)
{
/* try the input format first */
int
err
=
alsa_output_try_format_both
(
pcm
,
hwparams
,
audio_format
,
reverse_endian_r
,
audio_format
->
format
);
if
(
err
!=
-
EINVAL
)
return
err
;
...
...
@@ -329,6 +377,7 @@ alsa_output_setup_format(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
continue
;
err
=
alsa_output_try_format_both
(
pcm
,
hwparams
,
audio_format
,
reverse_endian_r
,
probe_formats
[
i
]);
if
(
err
!=
-
EINVAL
)
return
err
;
...
...
@@ -387,7 +436,8 @@ configure_hw:
ad
->
writei
=
snd_pcm_writei
;
}
err
=
alsa_output_setup_format
(
ad
->
pcm
,
hwparams
,
audio_format
);
err
=
alsa_output_setup_format
(
ad
->
pcm
,
hwparams
,
audio_format
,
&
ad
->
reverse_endian
);
if
(
err
<
0
)
{
g_set_error
(
error
,
alsa_output_quark
(),
err
,
"ALSA device
\"
%s
\"
does not support format %s: %s"
,
...
...
@@ -397,6 +447,8 @@ configure_hw:
return
false
;
}
ad
->
sample_format
=
audio_format
->
format
;
err
=
snd_pcm_hw_params_set_channels_near
(
ad
->
pcm
,
hwparams
,
&
channels
);
if
(
err
<
0
)
{
...
...
@@ -660,6 +712,10 @@ alsa_play(struct audio_output *ao, const void *chunk, size_t size,
{
struct
alsa_data
*
ad
=
(
struct
alsa_data
*
)
ao
;
if
(
ad
->
reverse_endian
)
chunk
=
pcm_byteswap
(
&
ad
->
reverse_buffer
,
ad
->
sample_format
,
chunk
,
size
);
size
/=
ad
->
frame_size
;
while
(
true
)
{
...
...
@@ -684,6 +740,8 @@ const struct audio_output_plugin alsa_output_plugin = {
.
test_default_device
=
alsa_test_default_device
,
.
init
=
alsa_init
,
.
finish
=
alsa_finish
,
.
enable
=
alsa_output_enable
,
.
disable
=
alsa_output_disable
,
.
open
=
alsa_open
,
.
play
=
alsa_play
,
.
drain
=
alsa_drain
,
...
...
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