In ALSA firewire stack, 8 drivers uses IEC 61883-1/6 engine for data
transmission. They have common PCM info/constraints and duplicated codes.
This commit unifies the codes into fireiwre-lib.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In recent commit for ALSA PCM core, some arrangement is done for
'struct snd_pcm_ops.ack' callback. This is called when appl_ptr is
explicitly moved in intermediate buffer for PCM frames, except for
some cases described later.
For drivers in ALSA firewire stack, usage of this callback has a merit to
reduce latency between time of PCM frame queueing and handling actual
packets in recent isochronous cycle, because no need to wait for software
IRQ context from isochronous context of OHCI 1394.
If this works well in a case that mapped page frame is used for the
intermediate buffer, user process should execute some commands for ioctl(2)
to tell the number of handled PCM frames in the intermediate buffer just
after handling them. Therefore, at present, with a combination of below
conditions, this doesn't work as expected and user process should wait for
the software IRQ context as usual:
- when ALSA PCM core judges page frame mapping is available for status
data (struct snd_pcm_mmap_status) and control data
(struct snd_pcm_mmap_control).
- user process handles PCM frames by loop just with 'snd_pcm_mmap_begin()'
and 'snd_pcm_mmap_commit()'.
- user process uses PCM hw plugin in alsa-lib to operate I/O without
'sync_ptr_ioctl' option.
Unfortunately, major use case include these three conditions.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In IEC 61883-6, AM824 is described as format of data block. In this
format, one data block consists of several data channels, which is aligned
to 32 bit. One data channel has 8 bit label field and 24 bit data field.
PCM frames are transferred in Multi Bit Linear Audio (MBLA) data channel.
This channel can include 16/20/24 bit PCM sample.
As long as I know, models which support IEC 61883-1/6 doesn't allow to
switch bit length of PCM sample in MBLA data channel. They always
transmit/receive PCM frames of 24 bit length. This can be seen for the
other models which support protocols similar to IEC 61883-1/6.
On the other hand, current drivers for these protocols supports 16 bit
length PCM sample in playback substream. In this case, PCM sample is put
into the MBLA data channel with 8 bit padding in LSB side. Although 16
bit PCM sample is major because it's in CD format, this doesn't represent
device capability as is.
This commit removes support for 16 bit PCM samples in playback substream.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some identifiers are referred just by one functions. In this case, they
can be put into the function definition. This brings two merits; readers
can easily follow codes related to the identifiers, developers are free
from name conflict.
This commit moves such identifiers to each function definition.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The counter is incremented/decremented in critical section protected with
mutex. Therefore, no need to use atomic_t.
This commit changes the type to unsigned int.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, critical section is protected by mutex in functions of
fireworks_stream.c. Callers increments/decrements substreams counter
before calling the functions. Moving mutex to the callers code allows
to change type of the substream counter from atomic_t to unsigned int.
This commit is a preparation for obsoleting usage of atomic_t for
substream counter.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The structures of type snd_bebob_clock_spec, snd_bebob_rate_spec,
snd_bebob_meter_spec, and snd_bebob_spec are never modified after they are
initialized. Make them all const.
Done with the help of Coccinelle.
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Tested-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit renames some macros just related to AM824 format. In later
commit, they're moved to AM824 layer.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Setting the format of PCM substream to AMDTP stream structure is important
to set a handler to copy actual PCM samples between buffers. The
processing should be in data block processing layer because essentially
it has no relationship to packet streaming.
This commit renames PCM format setting function to prepare for integrating
AM824 layer.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In IEC 61883-6, PCM frames are transferred in Multi Bit Linear Audio data
channel. The data channel transfers 16/20/24 bit PCM samples. Thus, PCM
substream has a constrain about it.
This commit moves codes related to the constraint from packet streaming
layer to AM824 data block processing layer.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In PCM core, when hw_params() in each driver returns error, the state of
PCM substream is kept as 'open'. In this case, current drivers for sound
units on IEEE 1394 bus doesn't decrement substream counter in hw_free()
correctly. This causes these drivers to keep streams even if not
required.
This commit fixes this bug. When snd_pcm_lib_alloc_vmalloc_buffer()
fails, hw_params function in each driver returns without incrementing the
counter.
Reported-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Behringer FCA610 transmits packets with periodic noisy PCM samples
when receiving no streams, and generates a bit noisy sound.
ALSA BeBoB driver is programmed to establish both in/out connections
when starting streaming, then transfers packets as userspace applications
requested. This means that there's a case that one of incoming/outgoing
streams is running, to save CPU and bandwidth usage. Although, it's natural
to start transferring packets in both direction.
This commit makes this driver to keeps duplex streams always.
Tested-by: Kim Tore Jensen <kim@incendio.no>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In previous commits, this driver can detect the source of clock as mush
as possible. SYT-Match mode is also available.
This commit purge the restriction.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit changes function prototype and its processing. As a result,
function caller can execute additional processing according to detected
clock source.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit is for some devices which have its own operations or quirks.
Many functionality should be implemented in user land. Then this commit adds
functionality related to stream such as sampling frequency or clock source. For
help to debug, this commit adds the functionality to get metering information
if it's available.
To help these functionalities, this commit adds some AV/C commands defined in
'AV/C Audio Subunit Specification (1394TA).
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This interface is designed for mixer/control application. By using hwdep
interface, the application can get information about firewire node, can
lock/unlock kernel streaming and can get notification at starting/stopping
kernel streaming.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds a functionality to capture/playback PCM samples.
When AMDTP stream is already running for PCM or the source of clock is not
internal, available sampling rate is limited at current one.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Sakamoto