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vlc-2-2
Commit c57ef1df authored by Jean-Paul Saman's avatar Jean-Paul Saman
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access/alsa.c: Todays capture cards support more than one format for audio capturing. 

(cherry picked from commit fb46b4a117c51b4a352b739a9cc568636598bc5b)
parent 171bdb14
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Showing with 81 additions and 7 deletions
modules/access/alsa.c
View file @ c57ef1df
......@@ -44,6 +44,8 @@
#include <vlc_access.h>
#include <vlc_demux.h>
#include <vlc_input.h>
#include <vlc_fourcc.h>
#include <vlc_aout.h>
#include <unistd.h>
#include <sys/ioctl.h>
......@@ -68,6 +70,10 @@ static void DemuxClose( vlc_object_t * );
#define STEREO_LONGTEXT N_( \
"Capture the audio stream in stereo." )
#define FORMAT_TEXT N_( "Capture format (default s16l)" )
#define FORMAT_LONGTEXT N_( \
"Capture format of audio stream." )
#define SAMPLERATE_TEXT N_( "Samplerate" )
#define SAMPLERATE_LONGTEXT N_( \
"Samplerate of the captured audio stream, in Hz (eg: 11025, 22050, 44100, 48000)" )
......@@ -80,6 +86,23 @@ static void DemuxClose( vlc_object_t * );
#define ALSA_DEFAULT "hw"
#define CFG_PREFIX "alsa-"
static const char *const ppsz_fourcc[] = {
"u8", "s8", "gsm", "u16l", "s16l", "u16b", "s16b",
"u24l", "s24l", "u24b", "s24b", "u32l", "s32l",
"u32b", "s32b", "f32l", "f32b", "f64l", "f64b"
};
static const char *const ppsz_fourcc_text[] = {
N_("PCM U8"), N_("PCM S8"), N_("GSM Audio"),
N_("PCM U16 LE"), N_("PCM S16 LE"),
N_("PCM U16 BE"), N_("PCM S16 BE"),
N_("PCM U24 LE"), N_("PCM S24 LE"),
N_("PCM U24 BE"), N_("PCM S24 BE"),
N_("PCM U32 LE"), N_("PCM S32 LE"),
N_("PCM U32 BE"), N_("PCM S32 BE"),
N_("PCM F32 LE"), N_("PCM F32 BE"),
N_("PCM F64 LE"), N_("PCM F64 BE")
};
vlc_module_begin()
set_shortname( N_("ALSA") )
set_description( N_("ALSA audio capture input") )
......@@ -93,6 +116,9 @@ vlc_module_begin()
add_bool( CFG_PREFIX "stereo", true, STEREO_TEXT, STEREO_LONGTEXT,
true )
add_string( CFG_PREFIX "format", "s16l", FORMAT_TEXT,
FORMAT_LONGTEXT, true )
change_string_list( ppsz_fourcc, ppsz_fourcc_text, 0 )
add_integer( CFG_PREFIX "samplerate", 48000, SAMPLERATE_TEXT,
SAMPLERATE_LONGTEXT, true )
vlc_module_end()
......@@ -116,6 +142,7 @@ struct demux_sys_t
/* Audio */
unsigned int i_sample_rate;
bool b_stereo;
vlc_fourcc_t i_format;
size_t i_max_frame_size;
block_t *p_block;
es_out_id_t *p_es;
......@@ -245,6 +272,10 @@ static int DemuxOpen( vlc_object_t *p_this )
p_sys->p_block = NULL;
p_sys->i_next_demux_date = -1;
char *psz_format = var_InheritString( p_demux, CFG_PREFIX "format" );
p_sys->i_format = vlc_fourcc_GetCodecFromString( AUDIO_ES, psz_format );
free( psz_format );
const char *psz_device = NULL;
if( p_demux->psz_location && *p_demux->psz_location )
psz_device = p_demux->psz_location;
......@@ -434,6 +465,46 @@ static block_t* GrabAudio( demux_t *p_demux )
return p_block;
}
static snd_pcm_format_t GetAlsaPCMFormat( demux_t *p_demux, const vlc_fourcc_t i_format )
{
demux_sys_t *p_sys = p_demux->p_sys;
switch( i_format )
{
case VLC_CODEC_U8: return SND_PCM_FORMAT_U8;
case VLC_CODEC_S8: return SND_PCM_FORMAT_S8;
case VLC_CODEC_GSM: return SND_PCM_FORMAT_GSM;
case VLC_CODEC_U16L: return SND_PCM_FORMAT_U16_LE;
case VLC_CODEC_S16L: return SND_PCM_FORMAT_S16_LE;
case VLC_CODEC_U16B: return SND_PCM_FORMAT_U16_BE;
case VLC_CODEC_S16B: return SND_PCM_FORMAT_S16_BE;
case VLC_CODEC_U24L: return SND_PCM_FORMAT_U24_3LE;
case VLC_CODEC_S24L: return SND_PCM_FORMAT_S24_3LE;
case VLC_CODEC_U24B: return SND_PCM_FORMAT_U24_3BE;
case VLC_CODEC_S24B: return SND_PCM_FORMAT_S24_3BE;
case VLC_CODEC_U32L: return SND_PCM_FORMAT_U32_LE;
case VLC_CODEC_U32B: return SND_PCM_FORMAT_U32_BE;
case VLC_CODEC_S32L: return SND_PCM_FORMAT_S32_LE;
case VLC_CODEC_S32B: return SND_PCM_FORMAT_S32_BE;
case VLC_CODEC_F32L: return SND_PCM_FORMAT_FLOAT_LE;
case VLC_CODEC_F32B: return SND_PCM_FORMAT_FLOAT_BE;
case VLC_CODEC_F64L: return SND_PCM_FORMAT_FLOAT64_LE;
case VLC_CODEC_F64B: return SND_PCM_FORMAT_FLOAT64_BE;
default:
msg_Err( p_demux, "ALSA: unsupported sample format '%s' falling back to 's16l'",
(const char *)&i_format );
p_sys->i_format = VLC_CODEC_S16L;
}
return SND_PCM_FORMAT_S16_LE;
}
/*****************************************************************************
* OpenAudioDev: open and set up the audio device and probe for capabilities
*****************************************************************************/
......@@ -442,6 +513,7 @@ static int OpenAudioDevAlsa( demux_t *p_demux, const char *psz_device )
demux_sys_t *p_sys = p_demux->p_sys;
p_sys->p_alsa_pcm = NULL;
snd_pcm_hw_params_t *p_hw_params = NULL;
snd_pcm_format_t i_alsa_pcm_format;
snd_pcm_uframes_t buffer_size;
snd_pcm_uframes_t chunk_size;
......@@ -489,8 +561,9 @@ static int OpenAudioDevAlsa( demux_t *p_demux, const char *psz_device )
goto adev_fail;
}
/* Set 16 bit little endian */
if( ( i_err = snd_pcm_hw_params_set_format( p_sys->p_alsa_pcm, p_hw_params, SND_PCM_FORMAT_S16_LE ) ) < 0 )
/* Set capture format, default is signed 16 bit little endian */
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 )
{
msg_Err( p_demux, "ALSA: cannot set sample format (%s)",
snd_strerror( i_err ) );
......@@ -572,7 +645,7 @@ static int OpenAudioDevAlsa( demux_t *p_demux, const char *psz_device )
goto adev_fail;
}
int bits_per_sample = snd_pcm_format_physical_width(SND_PCM_FORMAT_S16_LE);
int bits_per_sample = snd_pcm_format_physical_width(i_alsa_pcm_format);
int bits_per_frame = bits_per_sample * channels;
p_sys->i_alsa_chunk_size = chunk_size;
......@@ -611,16 +684,17 @@ static int OpenAudioDev( demux_t *p_demux, const char *psz_device )
if( OpenAudioDevAlsa( p_demux, psz_device ) != VLC_SUCCESS )
return VLC_EGENERIC;
msg_Dbg( p_demux, "opened adev=`%s' %s %dHz",
msg_Dbg( p_demux, "opened adev=`%s' %s %dHz codec '%s'",
psz_device, p_sys->b_stereo ? "stereo" : "mono",
p_sys->i_sample_rate );
p_sys->i_sample_rate,
vlc_fourcc_GetDescription( AUDIO_ES, p_sys->i_format ) );
es_format_t fmt;
es_format_Init( &fmt, AUDIO_ES, VLC_FOURCC('a','r','a','w') );
es_format_Init( &fmt, AUDIO_ES, p_sys->i_format );
fmt.audio.i_channels = p_sys->b_stereo ? 2 : 1;
fmt.audio.i_rate = p_sys->i_sample_rate;
fmt.audio.i_bitspersample = 16;
fmt.audio.i_bitspersample = aout_BitsPerSample( p_sys->i_format );
fmt.audio.i_blockalign = fmt.audio.i_channels * fmt.audio.i_bitspersample / 8;
fmt.i_bitrate = fmt.audio.i_channels * fmt.audio.i_rate * fmt.audio.i_bitspersample;
......
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