Skip to content
Projects
Groups
Snippets
Help
This project
Loading...
Sign in / Register
Toggle navigation
M
mpd
Project
Project
Details
Activity
Cycle Analytics
Repository
Repository
Files
Commits
Branches
Tags
Contributors
Graph
Compare
Charts
Issues
0
Issues
0
List
Board
Labels
Milestones
Merge Requests
0
Merge Requests
0
CI / CD
CI / CD
Pipelines
Jobs
Schedules
Charts
Registry
Registry
Wiki
Wiki
Snippets
Snippets
Members
Members
Collapse sidebar
Close sidebar
Activity
Graph
Charts
Create a new issue
Jobs
Commits
Issue Boards
Open sidebar
Иван Мажукин
mpd
Commits
8004ae34
Commit
8004ae34
authored
Mar 05, 2005
by
Warren Dukes
Browse files
Options
Browse Files
Download
Email Patches
Plain Diff
more alsa work
git-svn-id:
https://svn.musicpd.org/mpd/trunk@3019
09075e82-0dd4-0310-85a5-a0d7c8717e4f
parent
b94fa9c9
Hide whitespace changes
Inline
Side-by-side
Showing
1 changed file
with
57 additions
and
41 deletions
+57
-41
audioOutput_alsa.c
src/audioOutputs/audioOutput_alsa.c
+57
-41
No files found.
src/audioOutputs/audioOutput_alsa.c
View file @
8004ae34
...
...
@@ -43,19 +43,21 @@ typedef snd_pcm_sframes_t alsa_writei_t(snd_pcm_t *pcm, const void *buffer,
typedef
struct
_AlsaData
{
char
*
device
;
snd_pcm_t
*
pcm_handle
;
int
mmap
;
snd_pcm_t
*
pcmHandle
;
alsa_writei_t
*
writei
;
int
sampleSize
;
int
useMmap
;
int
canPause
;
int
canResume
;
}
AlsaData
;
static
AlsaData
*
newAlsaData
()
{
AlsaData
*
ret
=
malloc
(
sizeof
(
AlsaData
));
ret
->
device
=
NULL
;
ret
->
pcm
_h
andle
=
NULL
;
ret
->
pcm
H
andle
=
NULL
;
ret
->
writei
=
snd_pcm_writei
;
ret
->
m
map
=
0
;
ret
->
useM
map
=
0
;
return
ret
;
}
...
...
@@ -116,43 +118,43 @@ static int alsa_openDevice(AudioOutput * audioOutput)
return
-
1
;
}
err
=
snd_pcm_open
(
&
ad
->
pcm
_h
andle
,
ad
->
device
,
err
=
snd_pcm_open
(
&
ad
->
pcm
H
andle
,
ad
->
device
,
SND_PCM_STREAM_PLAYBACK
,
SND_PCM_NONBLOCK
);
if
(
err
<
0
)
{
ad
->
pcm
_h
andle
=
NULL
;
ad
->
pcm
H
andle
=
NULL
;
goto
error
;
}
err
=
snd_pcm_nonblock
(
ad
->
pcm
_h
andle
,
0
);
err
=
snd_pcm_nonblock
(
ad
->
pcm
H
andle
,
0
);
if
(
err
<
0
)
goto
error
;
/* configure HW params */
snd_pcm_hw_params_alloca
(
&
hwparams
);
err
=
snd_pcm_hw_params_any
(
ad
->
pcm
_h
andle
,
hwparams
);
err
=
snd_pcm_hw_params_any
(
ad
->
pcm
H
andle
,
hwparams
);
if
(
err
<
0
)
goto
error
;
if
(
ad
->
m
map
)
{
err
=
snd_pcm_hw_params_set_access
(
ad
->
pcm
_h
andle
,
hwparams
,
if
(
ad
->
useM
map
)
{
err
=
snd_pcm_hw_params_set_access
(
ad
->
pcm
H
andle
,
hwparams
,
SND_PCM_ACCESS_MMAP_INTERLEAVED
);
if
(
err
<
0
)
{
ERROR
(
"Cannot set mmap'ed mode on alsa device
\"
%s
\"
: "
" %s
\n
"
,
ad
->
device
,
snd_strerror
(
-
err
));
ERROR
(
"Falling back to direct write mode
\n
"
);
ad
->
m
map
=
0
;
ad
->
useM
map
=
0
;
}
else
ad
->
writei
=
snd_pcm_mmap_writei
;
}
if
(
!
ad
->
m
map
)
{
err
=
snd_pcm_hw_params_set_access
(
ad
->
pcm
_h
andle
,
hwparams
,
if
(
!
ad
->
useM
map
)
{
err
=
snd_pcm_hw_params_set_access
(
ad
->
pcm
H
andle
,
hwparams
,
SND_PCM_ACCESS_RW_INTERLEAVED
);
if
(
err
<
0
)
goto
error
;
ad
->
writei
=
snd_pcm_writei
;
}
err
=
snd_pcm_hw_params_set_format
(
ad
->
pcm
_h
andle
,
hwparams
,
bitformat
);
err
=
snd_pcm_hw_params_set_format
(
ad
->
pcm
H
andle
,
hwparams
,
bitformat
);
if
(
err
<
0
)
{
ERROR
(
"Alsa device
\"
%s
\"
does not support %i bit audio: "
"%s
\n
"
,
ad
->
device
,
(
int
)
bitformat
,
...
...
@@ -160,7 +162,7 @@ static int alsa_openDevice(AudioOutput * audioOutput)
goto
fail
;
}
err
=
snd_pcm_hw_params_set_channels
(
ad
->
pcm
_h
andle
,
hwparams
,
err
=
snd_pcm_hw_params_set_channels
(
ad
->
pcm
H
andle
,
hwparams
,
audioFormat
->
channels
);
if
(
err
<
0
)
{
ERROR
(
"Alsa device
\"
%s
\"
does not support %i channels: "
...
...
@@ -169,7 +171,7 @@ static int alsa_openDevice(AudioOutput * audioOutput)
goto
fail
;
}
err
=
snd_pcm_hw_params_set_rate_near
(
ad
->
pcm
_h
andle
,
hwparams
,
err
=
snd_pcm_hw_params_set_rate_near
(
ad
->
pcm
H
andle
,
hwparams
,
&
sampleRate
,
0
);
if
(
err
<
0
||
sampleRate
==
0
)
{
ERROR
(
"Alsa device
\"
%s
\"
does not support %i Hz audio
\n
"
,
...
...
@@ -177,15 +179,15 @@ static int alsa_openDevice(AudioOutput * audioOutput)
goto
fail
;
}
err
=
snd_pcm_hw_params_set_buffer_time_near
(
ad
->
pcm
_h
andle
,
hwparams
,
err
=
snd_pcm_hw_params_set_buffer_time_near
(
ad
->
pcm
H
andle
,
hwparams
,
&
alsa_buffer_time
,
0
);
if
(
err
<
0
)
goto
error
;
err
=
snd_pcm_hw_params_set_period_time_near
(
ad
->
pcm
_h
andle
,
hwparams
,
err
=
snd_pcm_hw_params_set_period_time_near
(
ad
->
pcm
H
andle
,
hwparams
,
&
alsa_period_time
,
0
);
if
(
err
<
0
)
goto
error
;
err
=
snd_pcm_hw_params
(
ad
->
pcm
_h
andle
,
hwparams
);
err
=
snd_pcm_hw_params
(
ad
->
pcm
H
andle
,
hwparams
);
if
(
err
<
0
)
goto
error
;
err
=
snd_pcm_hw_params_get_buffer_size
(
hwparams
,
&
alsa_buffer_size
);
...
...
@@ -194,15 +196,18 @@ static int alsa_openDevice(AudioOutput * audioOutput)
err
=
snd_pcm_hw_params_get_period_size
(
hwparams
,
&
alsa_period_size
,
0
);
if
(
err
<
0
)
goto
error
;
ad
->
canPause
=
snd_pcm_hw_params_can_pause
(
hwparams
);
ad
->
canResume
=
snd_pcm_hw_params_can_resume
(
hwparams
);
/* configure SW params */
snd_pcm_sw_params_alloca
(
&
swparams
);
snd_pcm_sw_params_current
(
ad
->
pcm
_h
andle
,
swparams
);
snd_pcm_sw_params_current
(
ad
->
pcm
H
andle
,
swparams
);
err
=
snd_pcm_sw_params_set_start_threshold
(
ad
->
pcm
_h
andle
,
swparams
,
err
=
snd_pcm_sw_params_set_start_threshold
(
ad
->
pcm
H
andle
,
swparams
,
alsa_buffer_size
-
alsa_period_size
);
if
(
err
<
0
)
goto
error
;
err
=
snd_pcm_sw_params
(
ad
->
pcm
_h
andle
,
swparams
);
err
=
snd_pcm_sw_params
(
ad
->
pcm
H
andle
,
swparams
);
if
(
err
<
0
)
goto
error
;
ad
->
sampleSize
=
(
audioFormat
->
bits
/
8
)
*
audioFormat
->
channels
;
...
...
@@ -215,8 +220,8 @@ error:
ERROR
(
"Error opening alsa device
\"
%s
\"
: %s
\n
"
,
ad
->
device
,
snd_strerror
(
-
err
));
fail:
if
(
ad
->
pcm
_handle
)
snd_pcm_close
(
ad
->
pcm_h
andle
);
ad
->
pcm
_h
andle
=
NULL
;
if
(
ad
->
pcm
Handle
)
snd_pcm_close
(
ad
->
pcmH
andle
);
ad
->
pcm
H
andle
=
NULL
;
audioOutput
->
open
=
0
;
return
-
1
;
}
...
...
@@ -224,8 +229,8 @@ fail:
static
void
alsa_dropBufferedAudio
(
AudioOutput
*
audioOutput
)
{
AlsaData
*
ad
=
audioOutput
->
data
;
snd_pcm_drop
(
ad
->
pcm
_h
andle
);
snd_pcm_prepare
(
ad
->
pcm
_h
andle
);
snd_pcm_drop
(
ad
->
pcm
H
andle
);
snd_pcm_prepare
(
ad
->
pcm
H
andle
);
}
inline
static
int
alsa_errorRecovery
(
AlsaData
*
ad
,
int
err
)
{
...
...
@@ -236,12 +241,19 @@ inline static int alsa_errorRecovery(AlsaData * ad, int err) {
DEBUG
(
"alsa device
\"
%s
\"
was suspended
\n
"
,
ad
->
device
);
}
switch
(
snd_pcm_state
(
ad
->
pcm_handle
))
{
switch
(
snd_pcm_state
(
ad
->
pcmHandle
))
{
case
SND_PCM_STATE_PAUSED
:
err
=
snd_pcm_pause
(
ad
->
pcmHandle
,
/* disable */
0
);
break
;
case
SND_PCM_STATE_SUSPENDED
:
err
=
ad
->
canResume
?
snd_pcm_resume
(
ad
->
pcmHandle
)
:
snd_pcm_prepare
(
ad
->
pcmHandle
);
break
;
case
SND_PCM_STATE_SETUP
:
case
SND_PCM_STATE_XRUN
:
err
=
snd_pcm_prepare
(
ad
->
pcm_handle
);
if
(
err
<
0
)
return
-
1
;
return
0
;
err
=
snd_pcm_prepare
(
ad
->
pcmHandle
);
break
;
default:
/* unknown state, do nothing */
break
;
...
...
@@ -253,10 +265,10 @@ inline static int alsa_errorRecovery(AlsaData * ad, int err) {
static
void
alsa_closeDevice
(
AudioOutput
*
audioOutput
)
{
AlsaData
*
ad
=
audioOutput
->
data
;
if
(
ad
->
pcm
_h
andle
)
{
snd_pcm_drain
(
ad
->
pcm
_h
andle
);
snd_pcm_close
(
ad
->
pcm
_h
andle
);
ad
->
pcm
_h
andle
=
NULL
;
if
(
ad
->
pcm
H
andle
)
{
snd_pcm_drain
(
ad
->
pcm
H
andle
);
snd_pcm_close
(
ad
->
pcm
H
andle
);
ad
->
pcm
H
andle
=
NULL
;
}
audioOutput
->
open
=
0
;
...
...
@@ -271,17 +283,21 @@ static int alsa_playAudio(AudioOutput * audioOutput, char * playChunk,
size
/=
ad
->
sampleSize
;
while
(
size
>
0
)
{
ret
=
ad
->
writei
(
ad
->
pcm
_h
andle
,
playChunk
,
size
);
ret
=
ad
->
writei
(
ad
->
pcm
H
andle
,
playChunk
,
size
);
if
(
ret
==
-
EAGAIN
)
continue
;
if
(
ret
<
0
&&
alsa_errorRecovery
(
ad
,
ret
)
<
0
)
{
ERROR
(
"closing alsa device
\"
%s
\"
due to write error:"
" %s
\n
"
,
ad
->
device
,
snd_strerror
(
-
errno
));
alsa_closeDevice
(
audioOutput
);
return
-
1
;
if
(
ret
<
0
)
{
if
(
alsa_errorRecovery
(
ad
,
ret
)
<
0
)
{
ERROR
(
"closing alsa device
\"
%s
\"
due to write "
"error: %s
\n
"
,
ad
->
device
,
snd_strerror
(
-
errno
));
alsa_closeDevice
(
audioOutput
);
return
-
1
;
}
continue
;
}
playChunk
+=
ret
*
ad
->
sampleSize
;
size
-=
ret
;
}
...
...
Write
Preview
Markdown
is supported
0%
Try again
or
attach a new file
Attach a file
Cancel
You are about to add
0
people
to the discussion. Proceed with caution.
Finish editing this message first!
Cancel
Please
register
or
sign in
to comment