access/alsa.c: Make fourcc available for access alsa input.

Make fourcc available for access alsa input without dragging in
vlc_aout.h.

modules/access/alsa.c

@@ -45,6 +45,8 @@
 #include <vlc_demux.h>
 #include <vlc_input.h>
 
+#include <vlc_fourcc.h>
+
 #include <unistd.h>
 #include <sys/ioctl.h>
 #include <sys/mman.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_( "Format" )
+#define FORMAT_LONGTEXT N_( \
+    "Sample format FOURCC (eg: araw, s8, s16l, s24l, s32l, f64l)" )
+
 #define SAMPLERATE_TEXT N_( "Samplerate" )
 #define SAMPLERATE_LONGTEXT N_( \
     "Samplerate of the captured audio stream, in Hz (eg: 11025, 22050, 44100, 48000)" )
@@ -98,6 +104,8 @@ vlc_module_begin()
 
     add_bool( CFG_PREFIX "stereo", true, NULL, STEREO_TEXT, STEREO_LONGTEXT,
                 true )
+    add_string( CFG_PREFIX "format", "araw", NULL, FORMAT_TEXT,
+                FORMAT_LONGTEXT, true )
     add_integer( CFG_PREFIX "samplerate", 48000, NULL, SAMPLERATE_TEXT,
                 SAMPLERATE_LONGTEXT, true )
     add_integer( CFG_PREFIX "caching", DEFAULT_PTS_DELAY / 1000, NULL,
@@ -137,6 +145,7 @@ struct demux_sys_t
     int i_cache;
     unsigned int i_sample_rate;
     bool b_stereo;
+    vlc_fourcc_t i_codec;
     size_t i_max_frame_size;
     block_t *p_block;
     es_out_id_t *p_es;
@@ -151,6 +160,43 @@ struct demux_sys_t
     audio_grab_t *p_grab;
 };
 
+static unsigned int BitsPerSample( vlc_fourcc_t i_fourcc )
+{
+    switch( vlc_fourcc_GetCodec( AUDIO_ES, i_fourcc ))
+    {
+        case VLC_CODEC_U8:
+        case VLC_CODEC_S8:
+            return 8;
+
+        case VLC_CODEC_U16L:
+        case VLC_CODEC_S16L:
+        case VLC_CODEC_U16B:
+        case VLC_CODEC_S16B:
+            return 16;
+
+        case VLC_CODEC_U24L:
+        case VLC_CODEC_S24L:
+        case VLC_CODEC_U24B:
+        case VLC_CODEC_S24B:
+            return 24;
+
+        case VLC_CODEC_S32L:
+        case VLC_CODEC_S32B:
+        case VLC_CODEC_F32L:
+        case VLC_CODEC_F32B:
+        case VLC_CODEC_FI32:
+            return 32;
+
+        case VLC_CODEC_F64L:
+        case VLC_CODEC_F64B:
+            return 64;
+
+        default: /* fall through */
+           break;
+    }
+    return 0;
+}
+ 
 static int FindMainDevice( demux_t *p_demux, const char *psz_device )
 {
     if( psz_device )
@@ -265,6 +311,11 @@ static int DemuxOpen( vlc_object_t *p_this )
     p_sys->i_sample_rate = var_CreateGetInteger( p_demux, CFG_PREFIX "samplerate" );
     p_sys->b_stereo = var_CreateGetBool( p_demux, CFG_PREFIX "stereo" );
     p_sys->i_cache = var_CreateGetInteger( p_demux, CFG_PREFIX "caching" );
+
+    char *psz_format = var_CreateGetString( p_demux, CFG_PREFIX "format" );
+    p_sys->i_codec = vlc_fourcc_GetCodecFromString( AUDIO_ES, psz_format );
+    free( psz_format );
+
     p_sys->p_es = NULL;
     p_sys->p_block = NULL;
     p_sys->i_next_demux_date = -1;
@@ -576,19 +627,20 @@ static int OpenAudioDevAlsa( demux_t *p_demux, const char *psz_device )
     int i_err;
 
     if( ( i_err = snd_pcm_open( &p_sys->p_alsa_pcm, psz_device,
-        SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK ) ) < 0)
+        SND_PCM_STREAM_CAPTURE, 0 /* SND_PCM_NONBLOCK */ ) ) < 0)
     {
         msg_Err( p_demux, "Cannot open ALSA audio device %s (%s)",
                  psz_device, snd_strerror( i_err ) );
         goto adev_fail;
     }
-
+#if 0
     if( ( i_err = snd_pcm_nonblock( p_sys->p_alsa_pcm, 1 ) ) < 0)
     {
         msg_Err( p_demux, "Cannot set ALSA nonblock (%s)",
                  snd_strerror( i_err ) );
         goto adev_fail;
     }
+#endif
 
     /* Begin setting hardware parameters */
 
@@ -617,7 +669,20 @@ static int OpenAudioDevAlsa( demux_t *p_demux, const char *psz_device )
     }
 
     /* 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 )
+    snd_pcm_format_t i_alsa_pcm_format = SND_PCM_FORMAT_S16_LE;
+    switch( BitsPerSample( p_sys->i_codec ) )
+    {
+        case  8: i_alsa_pcm_format = SND_PCM_FORMAT_S8; break;
+        case 16: i_alsa_pcm_format = SND_PCM_FORMAT_S16_LE; break;
+        case 24: i_alsa_pcm_format = SND_PCM_FORMAT_S24_LE; break;
+        case 32: i_alsa_pcm_format = SND_PCM_FORMAT_S32_LE; break;
+        case 64: i_alsa_pcm_format = SND_PCM_FORMAT_FLOAT64_LE; break;
+        default:
+            msg_Err( p_demux, "ALSA: unknown sample format (%s)", (char *)&p_sys->i_codec );
+            goto adev_fail;
+    }
+
+    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 ) );
@@ -699,7 +764,7 @@ 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;
@@ -749,11 +814,11 @@ static int OpenAudioDev( demux_t *p_demux, const char *psz_device )
              p_sys->i_sample_rate );
 
     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_codec );
 
     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 = BitsPerSample( p_sys->i_codec );
     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;