alsa-lib/doc/pcm.doxygen

/*! \page pcm PCM (digital audio) interface

<P>Although abbreviation PCM stands for Pulse Code Modulation, we are
understanding it as general digital audio processing with volume samples
generated in continuous time periods.</P>

<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 8kHz (telephone quality) and
48kHz (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 that 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, an 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 ADC.</P>

<P>Sample encoding limits the dynamic range of a 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 *
6dB. This means that 8 bits sampling resolution gives dynamic range of
48dB and 16 bit resolution gives 96dB.</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 8kHz/8 bits/mono but 48kHz/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 1.5Mbps 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 beyond the scope of this document. </P>

\section pcm_general_overview General overview

ALSA uses the ring buffer to store outgoing (playback) and incoming (capture,
record) samples. There are two pointers being mantained to allow
a precise communication between application and device pointing to current
processed sample by hardware and last processed sample by application.
The modern audio chips allow to program the transfer time periods.
It means that the stream of samples is divided to small chunks. Device
acknowledges to application when the transfer of a chunk is complete.

\section pcm_transfer Transfer methods in unix environments

In the unix environment, data chunk acknowledges are received via standard I/O
calls or event waiting routines (poll or select function). To accomplish
this list, the asynchronous notification of acknowledges should be listed
here. The ALSA implementation for these methods is described in
the \ref alsa_transfers section.

\subsection pcm_transfer_io Standard I/O transfers

The standard I/O transfers are using the read (see 'man 2 read') and write
(see 'man 2 write') C functions. There are two basic behaviours of these
functions - blocked and non-blocked (see the O_NONBLOCK flag for the
standard C open function - see 'man 2 open'). In non-blocked behaviour,
these I/O functions never stops, they return -EAGAIN error code, when no
data can be transferred (the ring buffer is full in our case). In blocked
behaviour, these I/O functions stop and wait until there is a room in the
ring buffer (playback) or until there are a new samples (capture). The ALSA
implementation can be found in the \ref alsa_pcm_rw section.

\subsection pcm_transfer_event Event waiting routines

The poll or select functions (see 'man 2 poll' or 'man 2 select' for further
details) allows to receive the acknowledges from the device while
application can wait to events from other sources (like keyboard, screen,
network etc.), too. The select function is old and deprecated in modern
applications, so the ALSA library does not support it. The implemented
transfer routines can be found in the \ref alsa_transfers section.

\subsection pcm_transfer_async Asynchronous notification

ALSA driver and library knows to handle the asynchronous notifications over
the SIGIO signal. This signal allows to interrupt application and transfer
data in the signal handler. For further details see the sigaction function
('man 2 sigaction'). The section \ref pcm_async describes the ALSA API for
this extension. The implemented transfer routines can be found in the
\ref alsa_transfers section.

\section pcm_open_behaviour Blocked and non-blocked open

The ALSA PCM API uses a different behaviour when the device is opened
with blocked or non-blocked mode. The mode can be specified with
\a mode argument in \link ::snd_pcm_open() \endlink function.
The blocked mode is the default (without \link ::SND_PCM_NONBLOCK \endlink mode).
In this mode, the behaviour is that if the resources have already used
with another application, then it blocks the caller, until resources are
free. The non-blocked behaviour (with \link ::SND_PCM_NONBLOCK \endlink)
doesn't block the caller in any way and returns -EBUSY error when the
resources are not available. Note that the mode also determines the
behaviour of standard I/O calls, returning -EAGAIN when non-blocked mode is
used and the ring buffer is full (playback) or empty (capture).
The operation mode for I/O calls can be changed later with
the \link snd_pcm_nonblock() \endlink function.

\section pcm_async Asynchronous mode

There is also possibility to receive asynchronous notification after
specified time periods. You may see the \link ::SND_PCM_ASYNC \endlink
mode for \link ::snd_pcm_open() \endlink function and
\link ::snd_async_add_pcm_handler() \endlink function for further details.

\section pcm_handshake Handshake between application and library

The ALSA PCM API design uses the states to determine the communication
phase between application and library. The actual state can be determined
using \link ::snd_pcm_state() \endlink call. There are these states:

\par SND_PCM_STATE_OPEN
The PCM device is in the open state. After the \link ::snd_pcm_open() \endlink open call,
the device is in this state. Also, when \link ::snd_pcm_hw_params() \endlink call fails,
then this state is entered to force application calling 
\link ::snd_pcm_hw_params() \endlink function to set right communication
parameters.

\par SND_PCM_STATE_SETUP
The PCM device has accepted communication parameters and it is waiting
for \link ::snd_pcm_prepare() \endlink call to prepare the hardware for
selected operation (playback or capture).

\par SND_PCM_STATE_PREPARE
The PCM device is prepared for operation. Application can use
\link ::snd_pcm_start() \endlink call, write or read data to start
the operation.

\par SND_PCM_STATE_RUNNING
The PCM device is running. It processes the samples. The stream can
be stopped using the \link ::snd_pcm_drop() \endlink or
\link ::snd_pcm_drain \endlink calls.

\par SND_PCM_STATE_XRUN
The PCM device reached overrun (capture) or underrun (playback).
You can use the -EPIPE return code from I/O functions
(\link ::snd_pcm_writei() \endlink, \link ::snd_pcm_writen() \endlink,
 \link ::snd_pcm_readi() \endlink, \link ::snd_pcm_readi() \endlink)
to determine this state without checking
the actual state via \link ::snd_pcm_state() \endlink call. You can recover from
this state with \link ::snd_pcm_prepare() \endlink,
\link ::snd_pcm_drop() \endlink or \link ::snd_pcm_drain() \endlink calls.

\par SND_PCM_STATE_DRAINING
The device is in this state when application using the capture mode
called \link ::snd_pcm_drain() \endlink function. Until all data are
read from the internal ring buffer using I/O routines
(\link ::snd_pcm_readi() \endlink, \link ::snd_pcm_readn() \endlink),
then the device stays in this state.

\par SND_PCM_STATE_PAUSED
The device is in this state when application called
the \link ::snd_pcm_pause() \endlink function until the pause is released.
Not all hardware supports this feature. Application should check the
capability with the \link ::snd_pcm_hw_params_can_pause() \endlink.

\par SND_PCM_STATE_SUSPENDED
The device is in the suspend state provoked with the power management
system. The stream can be resumed using \link ::snd_pcm_resume() \endlink
call, but not all hardware supports this feature. Application should check
the capability with the \link ::snd_pcm_hw_params_can_resume() \endlink.
In other case, the calls \link ::snd_pcm_prepare() \endlink,
\link ::snd_pcm_drop() \endlink, \link ::snd_pcm_drain() \endlink can be used
to leave this state.

\section pcm_formats PCM formats

The full list of formats present the \link ::snd_pcm_format_t \endlink type.
The 24-bit linear samples uses 32-bit physical space, but the sample is
stored in low three bits. Some hardware does not support processing of full
range, thus you may get the significative bits for linear samples via
\link ::snd_pcm_hw_params_get_sbits \endlink function. The example: ICE1712
chips support 32-bit sample processing, but low byte is ignored (playback)
or zero (capture). The function \link ::snd_pcm_hw_params_get_sbits() \endlink
returns 24 in the case.

\section alsa_transfers ALSA transfers

There are two methods to transfer samples in application. The first method
is the standard read / write one. The second method, uses the direct audio
buffer to communicate with the device while ALSA library manages this space
itself.

\subsection alsa_pcm_rw Read / Write transfer

There are two versions of read / write routines. The first expects the
interleaved samples at input, and the second one expects non-interleaved
(samples in separated buffers) at input. There are these functions for
interleaved transfers: \link ::snd_pcm_writei \endlink,
\link ::snd_pcm_readi \endlink. For non-interleaved transfers, there are
these functions: \link ::snd_pcm_writen \endlink and \link ::snd_pcm_readn
\endlink.

\subsection alsa_mmap_rw Direct Read / Write transfer (via mmaped areas)

There are two functions for this kind of transfer. Application can get an
access to memory areas via \link ::snd_pcm_mmap_begin \endlink function.
This functions returns the areas (single area is equal to a channel)
containing the direct pointers to memory and sample position description
in \link ::snd_pcm_channel_area_t \endlink structure. After application
transfers the data in the memory areas, then it must be acknowledged
the end of transfer via \link ::snd_pcm_mmap_commit() \endlink function
to allow the ALSA library update the pointers to ring buffer. This sort of
communication is also called "zero-copy", because the device does not require
to copy the samples from application to another place in system memory.

*/