Commit bb9c9396 authored by Jean-Paul Saman's avatar Jean-Paul Saman

access/alsa.c: Cleanup

No functional changes.
parent fb46b4a1
/***************************************************************************** /*****************************************************************************
* alsa.c : Alsa input module for vlc * alsa.c : Alsa input module for vlc
***************************************************************************** *****************************************************************************
* Copyright (C) 2002-2009 the VideoLAN team * Copyright (C) 2002-2011 the VideoLAN team
* $Id$ * $Id$
* *
* Authors: Benjamin Pracht <bigben at videolan dot org> * Authors: Benjamin Pracht <bigben at videolan dot org>
...@@ -369,7 +369,6 @@ static int Demux( demux_t *p_demux ) ...@@ -369,7 +369,6 @@ static int Demux( demux_t *p_demux )
return 1; return 1;
} }
/***************************************************************************** /*****************************************************************************
* GrabAudio: Grab an audio frame * GrabAudio: Grab an audio frame
*****************************************************************************/ *****************************************************************************/
...@@ -391,7 +390,8 @@ static block_t* GrabAudio( demux_t *p_demux ) ...@@ -391,7 +390,8 @@ static block_t* GrabAudio( demux_t *p_demux )
p_sys->p_block = p_block; p_sys->p_block = p_block;
/* ALSA */ /* ALSA */
i_read = snd_pcm_readi( p_sys->p_alsa_pcm, p_block->p_buffer, p_sys->i_alsa_chunk_size ); i_read = snd_pcm_readi( p_sys->p_alsa_pcm, p_block->p_buffer,
p_sys->i_alsa_chunk_size );
if( i_read == -EAGAIN ) if( i_read == -EAGAIN )
{ {
snd_pcm_wait( p_sys->p_alsa_pcm, 10 ); /* See poll() comment in oss.c */ snd_pcm_wait( p_sys->p_alsa_pcm, 10 ); /* See poll() comment in oss.c */
...@@ -410,11 +410,13 @@ static block_t* GrabAudio( demux_t *p_demux ) ...@@ -410,11 +410,13 @@ static block_t* GrabAudio( demux_t *p_demux )
{ {
/* suspend */ /* suspend */
int i_resume = snd_pcm_resume( p_sys->p_alsa_pcm ); int i_resume = snd_pcm_resume( p_sys->p_alsa_pcm );
if( i_resume < 0 && i_resume != -EAGAIN ) snd_pcm_prepare( p_sys->p_alsa_pcm ); if( i_resume < 0 && i_resume != -EAGAIN )
snd_pcm_prepare( p_sys->p_alsa_pcm );
break; break;
} }
default: default:
msg_Err( p_demux, "Failed to read alsa frame (%s)", snd_strerror( i_read ) ); msg_Err( p_demux, "Failed to read alsa frame (%s)",
snd_strerror( i_read ) );
return NULL; return NULL;
} }
} }
...@@ -549,7 +551,8 @@ static int OpenAudioDevAlsa( demux_t *p_demux, const char *psz_device ) ...@@ -549,7 +551,8 @@ static int OpenAudioDevAlsa( demux_t *p_demux, const char *psz_device )
} }
/* Set Interleaved access */ /* Set Interleaved access */
if( ( i_err = snd_pcm_hw_params_set_access( p_sys->p_alsa_pcm, p_hw_params, SND_PCM_ACCESS_RW_INTERLEAVED ) ) < 0 ) if( ( i_err = snd_pcm_hw_params_set_access( p_sys->p_alsa_pcm, p_hw_params,
SND_PCM_ACCESS_RW_INTERLEAVED ) ) < 0 )
{ {
msg_Err( p_demux, "ALSA: cannot set access type (%s)", msg_Err( p_demux, "ALSA: cannot set access type (%s)",
snd_strerror( i_err ) ); snd_strerror( i_err ) );
...@@ -558,7 +561,8 @@ static int OpenAudioDevAlsa( demux_t *p_demux, const char *psz_device ) ...@@ -558,7 +561,8 @@ static int OpenAudioDevAlsa( demux_t *p_demux, const char *psz_device )
/* Set capture format, default is signed 16 bit little endian */ /* Set capture format, default is signed 16 bit little endian */
i_alsa_pcm_format = GetAlsaPCMFormat( p_demux, p_sys->i_format ); i_alsa_pcm_format = GetAlsaPCMFormat( p_demux, p_sys->i_format );
if( ( i_err = snd_pcm_hw_params_set_format( p_sys->p_alsa_pcm, p_hw_params, i_alsa_pcm_format ) ) < 0 ) if( ( i_err = snd_pcm_hw_params_set_format( p_sys->p_alsa_pcm, p_hw_params,
i_alsa_pcm_format ) ) < 0 )
{ {
msg_Err( p_demux, "ALSA: cannot set sample format (%s)", msg_Err( p_demux, "ALSA: cannot set sample format (%s)",
snd_strerror( i_err ) ); snd_strerror( i_err ) );
...@@ -566,7 +570,8 @@ static int OpenAudioDevAlsa( demux_t *p_demux, const char *psz_device ) ...@@ -566,7 +570,8 @@ static int OpenAudioDevAlsa( demux_t *p_demux, const char *psz_device )
} }
/* Set sample rate */ /* Set sample rate */
i_err = snd_pcm_hw_params_set_rate_near( p_sys->p_alsa_pcm, p_hw_params, &p_sys->i_sample_rate, NULL ); i_err = snd_pcm_hw_params_set_rate_near( p_sys->p_alsa_pcm, p_hw_params,
&p_sys->i_sample_rate, NULL );
if( i_err < 0 ) if( i_err < 0 )
{ {
msg_Err( p_demux, "ALSA: cannot set sample rate (%s)", msg_Err( p_demux, "ALSA: cannot set sample rate (%s)",
...@@ -576,14 +581,16 @@ static int OpenAudioDevAlsa( demux_t *p_demux, const char *psz_device ) ...@@ -576,14 +581,16 @@ static int OpenAudioDevAlsa( demux_t *p_demux, const char *psz_device )
/* Set channels */ /* Set channels */
unsigned int channels = p_sys->b_stereo ? 2 : 1; unsigned int channels = p_sys->b_stereo ? 2 : 1;
if( ( i_err = snd_pcm_hw_params_set_channels( p_sys->p_alsa_pcm, p_hw_params, channels ) ) < 0 ) if( ( i_err = snd_pcm_hw_params_set_channels( p_sys->p_alsa_pcm, p_hw_params,
channels ) ) < 0 )
{ {
channels = ( channels==1 ) ? 2 : 1; channels = ( channels==1 ) ? 2 : 1;
msg_Warn( p_demux, "ALSA: cannot set channel count (%s). " msg_Warn( p_demux, "ALSA: cannot set channel count (%s). "
"Trying with channels=%d", "Trying with channels=%d",
snd_strerror( i_err ), snd_strerror( i_err ),
channels ); channels );
if( ( i_err = snd_pcm_hw_params_set_channels( p_sys->p_alsa_pcm, p_hw_params, channels ) ) < 0 ) if( ( i_err = snd_pcm_hw_params_set_channels( p_sys->p_alsa_pcm, p_hw_params,
channels ) ) < 0 )
{ {
msg_Err( p_demux, "ALSA: cannot set channel count (%s)", msg_Err( p_demux, "ALSA: cannot set channel count (%s)",
snd_strerror( i_err ) ); snd_strerror( i_err ) );
...@@ -604,7 +611,8 @@ static int OpenAudioDevAlsa( demux_t *p_demux, const char *psz_device ) ...@@ -604,7 +611,8 @@ static int OpenAudioDevAlsa( demux_t *p_demux, const char *psz_device )
/* Set period time */ /* Set period time */
unsigned int period_time = buffer_time / 4; unsigned int period_time = buffer_time / 4;
i_err = snd_pcm_hw_params_set_period_time_near( p_sys->p_alsa_pcm, p_hw_params, &period_time, 0 ); i_err = snd_pcm_hw_params_set_period_time_near( p_sys->p_alsa_pcm, p_hw_params,
&period_time, 0 );
if( i_err < 0 ) if( i_err < 0 )
{ {
msg_Err( p_demux, "ALSA: cannot set period time (%s)", msg_Err( p_demux, "ALSA: cannot set period time (%s)",
...@@ -613,7 +621,8 @@ static int OpenAudioDevAlsa( demux_t *p_demux, const char *psz_device ) ...@@ -613,7 +621,8 @@ static int OpenAudioDevAlsa( demux_t *p_demux, const char *psz_device )
} }
/* Set buffer time */ /* Set buffer time */
i_err = snd_pcm_hw_params_set_buffer_time_near( p_sys->p_alsa_pcm, p_hw_params, &buffer_time, 0 ); i_err = snd_pcm_hw_params_set_buffer_time_near( p_sys->p_alsa_pcm, p_hw_params,
&buffer_time, 0 );
if( i_err < 0 ) if( i_err < 0 )
{ {
msg_Err( p_demux, "ALSA: cannot set buffer time (%s)", msg_Err( p_demux, "ALSA: cannot set buffer time (%s)",
...@@ -670,7 +679,6 @@ static int OpenAudioDevAlsa( demux_t *p_demux, const char *psz_device ) ...@@ -670,7 +679,6 @@ static int OpenAudioDevAlsa( demux_t *p_demux, const char *psz_device )
p_sys->p_alsa_pcm = NULL; p_sys->p_alsa_pcm = NULL;
return VLC_EGENERIC; return VLC_EGENERIC;
} }
static int OpenAudioDev( demux_t *p_demux, const char *psz_device ) static int OpenAudioDev( demux_t *p_demux, const char *psz_device )
......
Markdown is supported
0%
or
You are about to add 0 people to the discussion. Proceed with caution.
Finish editing this message first!
Please register or to comment