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Initial revision
1998-08-13 15:42:56 +00:00
<HTML>
<HEAD>
<TITLE>Advanced Linux Sound Architecture - Library API: Digital Audio (PCM) Interface</TITLE>
</HEAD>
<BODY>
<A HREF="soundapi-4.html">Previous</A>
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<A HREF="soundapi.html#toc5">Table of Contents</A>
<HR>
<H2><A NAME="s5">5. Digital Audio (PCM) Interface</A></H2>
<P>Digital audio is the most commonly used method of representing sound inside
a computer. In this method sound is stored as a sequence of samples taken
from the audio signal using constant time intervals. A sample represents
volume of the signal at the moment when it was measured. In uncompressed
digital audio each sample require one or more bytes of storage. The number of
bytes required depends on number of channels (mono, stereo) and sample
format (8 or 16 bits, mu-Law, etc.). The length of this interval determines
the sampling rate. Commonly used sampling rates are between 8 kHz (telephone
quality) and 48 kHz (DAT tapes).</P>
<P>The physical devices used in digital audio are called the ADC (Analog to
Digital Converter) and DAC (Digital to Analog Converter). A device containing
both ADC and DAC is commonly known as a codec. The codec device used in
a Sound Blaster cards is called a DSP which is somewhat misleading since DSP
also stands for Digital Signal Processor (the SB DSP chip is very limited
when compared to "true" DSP chips). </P>
<P>Sampling parameters affect the quality of sound which can be reproduced from
the recorded signal. The most fundamental parameter is sampling rate which
limits the highest frequency than can be stored. It is well known (Nyquist's
Sampling Theorem) that the highest frequency that can be stored in a sampled
signal is at most 1/2 of the sampling frequency. For example, a 8 kHz sampling
rate permits the recording of a signal in which the highest frequency is less
than 4 kHz. Higher frequency signals must be filtered out before feeding them
to DAC. </P>
<P>Sample encoding limits the dynamic range of recorded signal (difference between
the faintest and the loudest signal that can be recorded). In theory the
maximum dynamic range of signal is number_of_bits * 6 dB . This means that
8 bits sampling resolution gives dynamic range of 48 dB and 16 bit resolution
gives 96 dB. </P>
<P>Quality has price. The number of bytes required to store an audio sequence
depends on sampling rate, number of channels and sampling resolution. For
example just 8000 bytes of memory is required to store one second of sound
using 8 kHz/8 bits/mono but 48 kHz/16bit/stereo takes 192 kilobytes. A 64 kbps
ISDN channel is required to transfer a 8kHz/8bit/mono audio stream in real
time, and about 1.5 Mbps is required for DAT quality (48kHz/16bit/stereo).
On the other hand it is possible to store just 5.46 seconds of sound in
a megabyte of memory when using 48kHz/16bit/stereo sampling. With
8kHz/8bits/mono it is possible to store 131 seconds of sound using the same
amount of memory. It is possible to reduce memory and communication costs by
compressing the recorded signal but this is out of the scope of this document. </P>
<H2><A NAME="ss5.1">5.1 Low-Level Layer</A></H2>
<P>Audio devices are opened exclusively for a selected direction. This doesn't
allow open from more than one processes for the same audio device in the
same direction, but does allow one open call to each playback direction and
second open call to record direction independently. Audio devices return
EBUSY error to applications when other applications have already opened the
requested direction.</P>
<P>Low-Level layer supports these formats:
<BLOCKQUOTE><CODE>
<HR>
<PRE>
#define SND_PCM_SFMT_MU_LAW 0
#define SND_PCM_SFMT_A_LAW 1
#define SND_PCM_SFMT_IMA_ADPCM 2
#define SND_PCM_SFMT_U8 3
#define SND_PCM_SFMT_S16_LE 4
#define SND_PCM_SFMT_S16_BE 5
#define SND_PCM_SFMT_S8 6
#define SND_PCM_SFMT_U16_LE 7
#define SND_PCM_SFMT_U16_BE 8
#define SND_PCM_SFMT_MPEG 9
#define SND_PCM_SFMT_GSM 10
#define SND_PCM_FMT_MU_LAW (1 &lt;&lt; SND_PCM_SFMT_MU_LAW)
#define SND_PCM_FMT_A_LAW (1 &lt;&lt; SND_PCM_SFMT_A_LAW)
#define SND_PCM_FMT_IMA_ADPCM (1 &lt;&lt; SND_PCM_SFMT_IMA_ADPCM)
#define SND_PCM_FMT_U8 (1 &lt;&lt; SND_PCM_SFMT_U8)
#define SND_PCM_FMT_S16_LE (1 &lt;&lt; SND_PCM_SFMT_S16_LE)
#define SND_PCM_FMT_S16_BE (1 &lt;&lt; SND_PCM_SFMT_S16_BE)
#define SND_PCM_FMT_S8 (1 &lt;&lt; SND_PCM_SFMT_S8)
#define SND_PCM_FMT_U16_LE (1 &lt;&lt; SND_PCM_SFMT_U16_LE)
#define SND_PCM_FMT_U16_BE (1 &lt;&lt; SND_PCM_SFMT_U16_BE)
#define SND_PCM_FMT_MPEG (1 &lt;&lt; SND_PCM_SFMT_MPEG)
#define SND_PCM_FMT_GSM (1 &lt;&lt; SND_PCM_SFMT_GSM)
</PRE>
<HR>
</CODE></BLOCKQUOTE>
Constants with prefix <B>SND_PCM_FMT_</B> are used in info structures
and constants with prefix <B>SND_PCM_SFMT_</B> are used in format structures.</P>
<H3>int snd_pcm_open( void **handle, int card, int device, int mode ) </H3>
<P>Creates a new handle and opens a connection to kernel sound
audio interface for soundcard number <I>card</I> (0-N) and audio
device number <I>device</I>. Function also checks if protocol is
compatible to prevent use of old programs with a new kernel API. Function
returns zero if successful,ful otherwise it returns an error code.
Error code -EBUSY is returned when some process ownes the selected direction.</P>
<P>Default format after opening is mono <I>mu-Law</I> at 8000Hz. This device
can be used directly for playback of standard .au (Sparc) files.</P>
<P>The following modes should be used for the <I>mode</I> argument:
<HR>
<PRE>
#define SND_PCM_OPEN_PLAYBACK (O_WRONLY)
#define SND_PCM_OPEN_RECORD (O_RDONLY)
#define SND_PCM_OPEN_DUPLEX (O_RDWR)
</PRE>
<HR>
</P>
<H3>int snd_pcm_close( void *handle ) </H3>
<P>Frees all resources allocated with audio handle and
closes the connection to the kernel sound audio interface. Function
returns zero if successful, otherwise it returns an error code.</P>
<H3>int snd_pcm_file_descriptor( void *handle ) </H3>
<P>Returns the file descriptor of the connection to the kernel sound
audio interface. Function returns an error code if an
error was encountered. </P>
<P>The file descriptor should be used for the <I>select</I> synchronous
multiplexer function for setting the read direction. Application should
call <I>snd_pcm_read</I> or <I>snd_pcm_write</I> functions if some
data is waiting for reading or a write can be performed. Calling this
function is highly recomended, as it leaves a place for the API to things
like data conversions, if needed.</P>
<H3>int snd_pcm_block_mode( void *handle, int enable ) </H3>
<P>Sets up block (default) or nonblock mode for a handle. Block mode suspends
execution of a program when <I>snd_pcm_read</I> or <I>snd_pcm_write</I>
is called for the time which is needed for the actual playback or record
over of the entire buffer. In nonblock mode, programs aren't suspended and
the above functions returns immediately with the count of bytes which were
read or written by the driver. When used in this way, don't try to use the
entire buffer after the call, but instead process the number of bytes
returned, and call the function again.</P>
<H3>int snd_pcm_info( void *handle, snd_pcm_info_t *info ) </H3>
<P>Fills the *info structure with data about the PCM device selected by
*handle. Function returns zero if successful, otherwise it returns
an error code.
<HR>
<PRE>
#define SND_PCM_INFO_CODEC 0x00000001
#define SND_PCM_INFO_DSP SND_PCM_INFO_CODEC
#define SND_PCM_INFO_MMAP 0x00000002 /* reserved */
#define SND_PCM_INFO_PLAYBACK 0x00000100
#define SND_PCM_INFO_RECORD 0x00000200
#define SND_PCM_INFO_DUPLEX 0x00000400
#define SND_PCM_INFO_DUPLEX_LIMIT 0x00000800 /* rate for playback & record are same */
struct snd_pcm_info {
unsigned int type; /* soundcard type */
unsigned int flags; /* see SND_PCM_INFO_XXXX */
unsigned char id[32]; /* ID of this PCM device */
unsigned char name[80]; /* name of this device */
unsigned char reserved[64]; /* reserved for future use */
};
</PRE>
<HR>
<DL>
<DT><B>SND_PCM_INFO_MMAP</B><DD><P>This flag is reserved and should be never used. It remains for
compatibility with Open Sound System driver.</P>
<DT><B>SND_PCM_INFO_DUPLEX_LIMIT</B><DD><P>If this bit is set, rate must be same for playback and record direction.</P>
</DL>
</P>
<H3>int snd_pcm_playback_info( void *handle, snd_pcm_playback_info_t *info ) </H3>
<P>Fills the *info structure with data about PCM playback. Function returns
zero if successful, otherwise it returns an error code.
<HR>
<PRE>
#define SND_PCM_PINFO_BATCH 0x00000001
#define SND_PCM_PINFO_8BITONLY 0x00000002
#define SND_PCM_PINFO_16BITONLY 0x00000004
struct snd_pcm_playback_info {
unsigned int flags; /* see SND_PCM_PINFO_XXXX */
unsigned int formats; /* supported formats */
unsigned int min_rate; /* min rate (in Hz) */
unsigned int max_rate; /* max rate (in Hz) */
unsigned int min_channels; /* min channels (probably always 1) */
unsigned int max_channels; /* max channels */
unsigned int buffer_size; /* playback buffer size */
unsigned int min_fragment_size; /* min fragment size in bytes */
unsigned int max_fragment_size; /* max fragment size in bytes */
unsigned int fragment_align; /* align fragment value */
unsigned char reserved[64]; /* reserved for future use */
};
</PRE>
<HR>
<DL>
<DT><B>SND_PCM_PINFO_BATCH</B><DD><P>Driver implements double buffering with this device. This means that
the chip used for data processing has its own memory, and output should be
more delayed than if a traditional codec chip is used.</P>
<DT><B>SND_PCM_PINFO_8BITONLY</B><DD><P>If this bit is set, the driver uses 8-bit format for 16-bit samples and
does software conversion. This bit is set on broken SoundBlaster 16/AWE
soundcards which can't do full 16-bit duplex. If this bit is set
application or highter digital audio layer should do the conversion from
16-bit samples to 8-bit samples rather than making the driver to do it in
the kernel.</P>
<DT><B>SND_PCM_PINFO_16BITONLY</B><DD><P>If this bit is set, driver uses 16-bit format for 8-bit samples and
does software conversion. This bit is set on broken SoundBlaster 16/AWE
soundcards which can't do full 8-bit duplex. If this bit is set the
application or highter digital audio layer should do conversion from
8-bit samples to 16-bit samples rather than making the driver to do it in
the kernel.</P>
</DL>
</P>
<H3>int snd_pcm_record_info( void *handle, snd_pcm_record_info_t *info ) </H3>
<P>Fills the *info structure. Returns zero if successful, otherwise it returns
an error code.
<HR>
<PRE>
#define SND_PCM_RINFO_BATCH 0x00000001
#define SND_PCM_RINFO_8BITONLY 0x00000002
#define SND_PCM_RINFO_16BITONLY 0x00000004
struct snd_pcm_record_info {
unsigned int flags; /* see to SND_PCM_RINFO_XXXX */
unsigned int formats; /* supported formats */
unsigned int min_rate; /* min rate (in Hz) */
unsigned int max_rate; /* max rate (in Hz) */
unsigned int min_channels; /* min channels (probably always 1) */
unsigned int max_channels; /* max channels */
unsigned int buffer_size; /* record buffer size */
unsigned int min_fragment_size; /* min fragment size in bytes */
unsigned int max_fragment_size; /* max fragment size in bytes */
unsigned int fragment_align; /* align fragment value */
unsigned char reserved[64]; /* reserved for future... */
};
</PRE>
<HR>
<DL>
<DT><B>SND_PCM_PINFO_BATCH</B><DD><P>Driver implements buffering for this device. This means that
the chip used for data processing has its own memory and output should be
more delayed than if a traditional codec chip is used.</P>
<DT><B>SND_PCM_PINFO_8BITONLY</B><DD><P>If this bit is set, the device uses 8-bit format for 16-bit samples and
does software conversion. This bit is set on broken SoundBlaster 16/AWE
soundcards which can't do full 16-bit duplex. If this bit is set the
application or highter digital audio layer should do conversion from
16-bit samples to 8-bit samples rather than making the driver to do it in
the kernel.</P>
<DT><B>SND_PCM_PINFO_16BITONLY</B><DD><P>If this bit is set, the device uses a 16-bit format for 8-bit samples and
does software conversion. This bit is set on broken SoundBlaster 16/AWE
soundcards which can't do full 8-bit duplex. If this bit is set the
application or highter digital audio layer should do the conversion from
8-bit samples to 16-bit samples rather than making the driver to do it in
the kernel.</P>
</DL>
</P>
<H3>int snd_pcm_playback_format( void *handle, snd_pcm_format_t *format ) </H3>
<P>Sets up format, rate (in Hz) and number of channels for playback, in the
desired direction. Function returns zero if successful, otherwise it
returns an error code.
<HR>
<PRE>
struct snd_pcm_format {
unsigned int format; /* SND_PCM_SFMT_XXXX */
unsigned int rate; /* rate in Hz */
unsigned int channels; /* channels (voices) */
unsigned char reserved[16];
};
</PRE>
<HR>
</P>
<H3>int snd_pcm_record_format( void *handle, snd_pcm_format_t *format ) </H3>
<P>Sets up format, rate (in Hz) and number of channels for used for recording
in the specified direction. Function returns zero if successful, otherwise
it returns an error code.
<HR>
<PRE>
struct snd_pcm_format {
unsigned int format; /* SND_PCM_SFMT_XXXX */
unsigned int rate; /* rate in Hz */
unsigned int channels; /* channels (voices) */
unsigned char reserved[16];
};
</PRE>
<HR>
</P>
<H3>int snd_pcm_playback_params( void *handle, snd_pcm_playback_params_t *params ) </H3>
<P>Sets various parameters for playback direction. Function returns zero if
successful, otherwise it returns an error code.
<HR>
<PRE>
struct snd_pcm_playback_params {
int fragment_size;
int fragments_max;
int fragments_room;
unsigned char reserved[16]; /* must be filled with zero */
};
</PRE>
<HR>
<DL>
<DT><B>fragment_size</B><DD><P>Requested size of fragment. This value should be aligned for current
format (for example to 4 if stereo 16-bit samples are used) or with the
<I>fragment_align</I> variable from <I>snd_pcm_playback_info_t</I>
structure. Its range can be from <I>min_fragment_size</I> to
<I>max_fragment_size</I>.</P>
<DT><B>fragments_max</B><DD><P>Maximum number of fragments in queue for wakeup. This number doesn't
counts partly used fragment. If current count of filled playback fragments
is greater than this value driver block application or return immediately
back if nonblock mode is active.</P>
<DT><B>fragments_room</B><DD><P>Minumum number of fragments writeable for wakeup. This value should be
in most cases 1 which means return back to application if at least
one fragment is free for playback. This value includes partly used fragments,
too.</P>
</DL>
</P>
<H3>int snd_pcm_record_params( void *handle, snd_pcm_record_params_t *params ) </H3>
<P>Function sets various parameters for the recording direction. Function returns
zero if successful, otherwise it returns an error code.
<HR>
<PRE>
struct snd_pcm_record_params {
int fragment_size;
int fragments_min;
unsigned char reserved[16];
};
</PRE>
<HR>
<DL>
<DT><B>fragment_size</B><DD><P>Requested size of fragment. This value should be aligned for current
format (for example to 4 if stereo 16-bit samples are used) or set to the
<I>fragment_align</I> variable from <I>snd_pcm_playback_info_t</I>
structure. Its range can be from <I>min_fragment_size</I> to
<I>max_fragment_size</I>.</P>
<DT><B>fragments_min</B><DD><P>Minimum filled fragments for wakeup. Driver blocks the application (if
block mode is selected) until it isn't filled with number of fragments
specified with this value.</P>
</DL>
</P>
<H3>int snd_pcm_playback_status( void *handle, snd_pcm_playback_status_t *status ) </H3>
<P>Fills the *status structure. Function returns zero if successful, otherwise
it returns an error code.
<HR>
<PRE>
struct snd_pcm_playback_status {
unsigned int rate;
int fragments;
int fragment_size;
int count;
int queue;
int underrun;
struct timeval time;
struct timeval stime;
unsigned char reserved[16];
};
</PRE>
<HR>
<DL>
<DT><B>rate</B><DD><P>Real playback rate. This value reflects hardware limitations.</P>
<DT><B>fragments</B><DD><P>Currently allocated fragments by the driver for playback direction.</P>
<DT><B>fragment_size</B><DD><P>Current fragment size used by driver for the playback direction.</P>
<DT><B>count</B><DD><P>Count of bytes writeable without blocking.</P>
<DT><B>queue</B><DD><P>Count of bytes in queue. Note: <I>(fragments * fragment_size) - queue</I>
should not be equal to <I>count</I>.</P>
<DT><B>underrun</B><DD><P>This value tells the application the number of underruns since the ast call
of <I>snd_pcm_playback_status</I>.</P>
<DT><B>time</B><DD><P>Delay till played of the first sample from next write. This value should
be used for time synchronization. Returned value is in the same format as
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returned from the standard C function <I>gettimeofday( &time, NULL )</I>.
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This variable contains right value only if playback time mode is enabled
(look to <I>snd_pcm_playback_time</I> function).</P>
<DT><B>stime</B><DD><P>Time when playback was started.
This variable contains right value only if playback time mode is enabled
(look to <I>snd_pcm_playback_time</I> function).</P>
</DL>
</P>
<H3>int snd_pcm_record_status( void *handle, snd_pcm_record_status_t *status ) </H3>
<P>Fills the *status structure. Function returns zero if successful, otherwise
it returns an error code.
<HR>
<PRE>
struct snd_pcm_record_status {
unsigned int rate;
int fragments;
int fragment_size;
int count;
int free;
int overrun;
struct timeval time;
unsigned char reserved[16];
};
</PRE>
<HR>
<DL>
<DT><B>rate</B><DD><P>Real record rate. This value reflects hardware limitations.</P>
<DT><B>fragments</B><DD><P>Currently allocated fragments by driver for the record direction.</P>
<DT><B>fragment_size</B><DD><P>Current fragment size used by driver for the record direction.</P>
<DT><B>count</B><DD><P>Count of bytes readable without blocking.</P>
<DT><B>free</B><DD><P>Count of bytes in buffer still free. Note: <I>(fragments * fragment_size) - free</I>
should not be equal to <I>count</I>.</P>
<DT><B>overrun</B><DD><P>This value tells application the count of overruns since the last call
to <I>snd_pcm_record_status</I>.</P>
<DT><B>time</B><DD><P>Lag since the next sample read was recorded. This value should be used for time
synchronization. Returned value is in the same format as returned by the
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from standard C function <I>gettimeofday( &time, NULL )</I>. This
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variable contains right value only if record time mode is enabled (look to
<I>snd_pcm_record_time</I> function).</P>
<DT><B>stime</B><DD><P>Time when record was started. This variable contains right value only if
record time mode is enabled (look to <I>snd_pcm_record_time</I> function).</P>
</DL>
</P>
<H3>int snd_pcm_drain_playback( void *handle ) </H3>
<P>This function drain playback buffers immediately. Function returns zero
if successful, otherwise it returns an error code. </P>
<H3>int snd_pcm_flush_playback( void *handle ) </H3>
<P>This function flushes the playback buffers. It blocks the program while the
all the waiting samples in kernel playback buffers are processed. Function
returns zero if successful, otherwise it returns an error code.</P>
<H3>int snd_pcm_flush_record( void *handle ) </H3>
<P>This function flushes (destroyes) record buffers. Function returns zero
if successful, otherwise it returns an error code. </P>
<H3>int snd_pcm_playback_time( void *handle, int enable ) </H3>
<P>This function enables or disables time mode for playback direction. Time mode
allows to application better time synchronization. Function returns zero
if successful, otherwise it returns an error code.</P>
<H3>int snd_pcm_record_time( void *handle, int enable ) </H3>
<P>This function enables or disables time mode for record direction. Time mode
allows to application better time synchronization. Function returns zero
if successful, otherwise it returns an error code.</P>
<H3>ssize_t snd_pcm_write( void *handle, const void *buffer, size_t size ) </H3>
<P>Writes samples to the device which must be in the proper format
specified by the <I>snd_pcm_playback_format</I> function. Function
returns zero or positive value if playback was successful (value represents
count of bytes which was successfuly written to device) or an
error value if error occured. Function should suspend process if
block mode is active.</P>
<H3>ssize_t snd_pcm_read( void *handle, void *buffer, size_t size ) </H3>
<P>Function reads samples from driver. Samples are in format specified
by <I>snd_pcm_record_format</I> function. Function returns zero
or positive value if record was success (value represents count of bytes
which was successfuly read from device) or negative error value if
error occured. Function should suspend process if block mode is active.</P>
<H2><A NAME="ss5.2">5.2 Examples</A></H2>
<P>The following example shows how to play the first 512kB from the
/tmp/test.au file with soundcard #0 and PCM device #0:</P>
<P>
<BLOCKQUOTE><CODE>
<HR>
<PRE>
int card = 0, device = 0, err, fd, count, size, idx;
void *handle;
snd_pcm_format_t format;
char *buffer;
buffer = (char *)malloc( 512 * 1024 );
if ( !buffer ) return;
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if ( (err = snd_pcm_open( &handle, card, device, SND_PCM_OPEN_PLAYBACK )) &lt; 0 ) {
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fprintf( stderr, &quot;open failed: %s\n&quot;, snd_strerror( err ) );
return;
}
format.format = SND_PCM_SFMT_MU_LAW;
format.rate = 8000;
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format.channels = 1;
if ( (err = snd_pcm_playback_format( handle, &format )) &lt; 0 ) {
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fprintf( stderr, &quot;format setup failed: %s\n&quot;, snd_strerror( err ) );
snd_pcm_close( handle );
return;
}
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fd = open( &quot;/tmp/test.au&quot;, O_RDONLY );
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if ( fd &lt; 0 ) {
perror( &quot;open file&quot; );
snd_pcm_close( handle );
return;
}
idx = 0;
count = read( fd, buffer, 512 * 1024 );
if ( count &lt;= 0 ) {
perror( &quot;read from file&quot; );
snd_pcm_close( handle );
return;
}
close( fd );
if ( !memcmp( buffer, &quot;.snd&quot;, 4 ) ) {
idx = (buffer[4]&lt;&lt;24)|(buffer[5]&lt;&lt;16)|(buffer[6]&lt;&lt;8)|(buffer[7]);
if ( idx &gt; 128 ) idx = 128;
if ( idx &gt; count ) idx = count;
}
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size = snd_pcm_write( handle, &buffer[ idx ], count - idx );
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printf( &quot;Bytes written %i from %i...\n&quot;, size, count - idx );
snd_pcm_close( handle );
free( buffer );
</PRE>
<HR>
</CODE></BLOCKQUOTE>
</P>
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