[alsa-devel] Trouble understanding ALSA's DMA buffers
I'm writing an ALSA SOC driver for an I2S-based device, and I'm having a really hard time understanding how ALSA uses the DMA buffers. And yes, I've read the documentation and studied some sample source code.
I used to write audio drivers for a living, but that was many years ago, and it wasn't for Linux. Perhaps the concepts in my head are outdated, but I just don't see enough explanation as to how DMA buffers are supposed to work.
Back then, audio drivers used "ping pong" DMA buffers. A single DMA buffer is allocated, and the audio hardware is programmed to read from that buffer in a loop. That is, it would automatically restart reading from the beginning of the buffer without any reprogramming. The hardware would also be programmed to issue an interrupt when it got to the end of the buffer, and when it got to the half-way point.
To start playback, you first filled the whole buffer with audio data, and then told the hardware to start playing. After the hardware got to the half-way point, it would issue an interrupt. You would then tell the OS you need more data, and you'd get it. You then copy that data into the first half of the buffer *while* the hardware was playing the second half. Later, the hardware would interrupt you when it got to the end of the buffer. You'd then copy more data to the 2nd half while the hardware is playing the first half.
And so - hardware plays one half while you copy data to the other half. Hence, "ping pong".
So how do I implement this in ALSA? The "Writing an ALSA Driver" document doesn't even contain the words "ping" or "pong".
Hi,
Now ALSA (audio ) buffer is divided into periods, i.e. a chain of small packets.
periods size is configurable. Data transfer to the codec starts only after reaching start_threshold point (start_threshold is in periods), this time DMA trigger is called.
This trigger onwards application will get notification from the kernel saying that period buffer is empty you can write into it.Till the end of music file.
Nobin Mathew
On 6/12/07, Timur Tabi timur@freescale.com wrote:
I'm writing an ALSA SOC driver for an I2S-based device, and I'm having a really hard time understanding how ALSA uses the DMA buffers. And yes, I've read the documentation and studied some sample source code.
I used to write audio drivers for a living, but that was many years ago, and it wasn't for Linux. Perhaps the concepts in my head are outdated, but I just don't see enough explanation as to how DMA buffers are supposed to work.
Back then, audio drivers used "ping pong" DMA buffers. A single DMA buffer is allocated, and the audio hardware is programmed to read from that buffer in a loop. That is, it would automatically restart reading from the beginning of the buffer without any reprogramming. The hardware would also be programmed to issue an interrupt when it got to the end of the buffer, and when it got to the half-way point.
To start playback, you first filled the whole buffer with audio data, and then told the hardware to start playing. After the hardware got to the half-way point, it would issue an interrupt. You would then tell the OS you need more data, and you'd get it. You then copy that data into the first half of the buffer *while* the hardware was playing the second half. Later, the hardware would interrupt you when it got to the end of the buffer. You'd then copy more data to the 2nd half while the hardware is playing the first half.
And so - hardware plays one half while you copy data to the other half. Hence, "ping pong".
So how do I implement this in ALSA? The "Writing an ALSA Driver" document doesn't even contain the words "ping" or "pong".
-- Timur Tabi Linux Kernel Developer @ Freescale _______________________________________________ Alsa-devel mailing list Alsa-devel@alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
At Tue, 12 Jun 2007 01:59:45 +0530, Nobin Mathew wrote:
Hi,
Now ALSA (audio ) buffer is divided into periods, i.e. a chain of small packets.
periods size is configurable. Data transfer to the codec starts only after reaching start_threshold point (start_threshold is in periods), this time DMA trigger is called.
This trigger onwards application will get notification from the kernel saying that period buffer is empty you can write into it.Till the end of music file.
Yes. And the "ping-poing" is the case that you have two periods in a ring buffer.
Most hardwares support more periods practically. That's why "periods" (corresponds to "fragments" in OSS) was introduced, as more generic abstraction.
Takashi
Nobin Mathew
On 6/12/07, Timur Tabi timur@freescale.com wrote:
I'm writing an ALSA SOC driver for an I2S-based device, and I'm having a really hard time understanding how ALSA uses the DMA buffers. And yes, I've read the documentation and studied some sample source code.
I used to write audio drivers for a living, but that was many years ago, and it wasn't for Linux. Perhaps the concepts in my head are outdated, but I just don't see enough explanation as to how DMA buffers are supposed to work.
Back then, audio drivers used "ping pong" DMA buffers. A single DMA buffer is allocated, and the audio hardware is programmed to read from that buffer in a loop. That is, it would automatically restart reading from the beginning of the buffer without any reprogramming. The hardware would also be programmed to issue an interrupt when it got to the end of the buffer, and when it got to the half-way point.
To start playback, you first filled the whole buffer with audio data, and then told the hardware to start playing. After the hardware got to the half-way point, it would issue an interrupt. You would then tell the OS you need more data, and you'd get it. You then copy that data into the first half of the buffer *while* the hardware was playing the second half. Later, the hardware would interrupt you when it got to the end of the buffer. You'd then copy more data to the 2nd half while the hardware is playing the first half.
And so - hardware plays one half while you copy data to the other half. Hence, "ping pong".
So how do I implement this in ALSA? The "Writing an ALSA Driver" document doesn't even contain the words "ping" or "pong".
-- Timur Tabi Linux Kernel Developer @ Freescale _______________________________________________ Alsa-devel mailing list Alsa-devel@alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
Alsa-devel mailing list Alsa-devel@alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
On 6/12/07, Takashi Iwai tiwai@suse.de wrote:
At Tue, 12 Jun 2007 01:59:45 +0530, Nobin Mathew wrote:
Hi,
Now ALSA (audio ) buffer is divided into periods, i.e. a chain of small
packets.
periods size is configurable. Data transfer to the codec starts only after reaching start_threshold point (start_threshold is in periods), this time DMA trigger is called.
This trigger onwards application will get notification from the kernel saying that period buffer is empty you can write into it.Till the end of music file.
Yes. And the "ping-poing" is the case that you have two periods in a ring buffer.
Most hardwares support more periods practically. That's why "periods" (corresponds to "fragments" in OSS) was introduced, as more generic abstraction.
Takashi
Nobin Mathew
On 6/12/07, Timur Tabi timur@freescale.com wrote:
I'm writing an ALSA SOC driver for an I2S-based device, and I'm having
a really hard time
understanding how ALSA uses the DMA buffers. And yes, I've read the
documentation and
studied some sample source code.
I used to write audio drivers for a living, but that was many years
ago, and it wasn't for
Linux. Perhaps the concepts in my head are outdated, but I just don't
see enough
explanation as to how DMA buffers are supposed to work.
Back then, audio drivers used "ping pong" DMA buffers. A single DMA
buffer is allocated,
and the audio hardware is programmed to read from that buffer in a
loop. That is, it
would automatically restart reading from the beginning of the buffer
without any
reprogramming. The hardware would also be programmed to issue an
interrupt when it got to
the end of the buffer, and when it got to the half-way point.
To start playback, you first filled the whole buffer with audio data,
and then told the
hardware to start playing. After the hardware got to the half-way
point, it would issue
an interrupt. You would then tell the OS you need more data, and
you'd get it. You then
copy that data into the first half of the buffer *while* the hardware
was playing the
second half. Later, the hardware would interrupt you when it got to
the end of the
buffer. You'd then copy more data to the 2nd half while the hardware
is playing the first
half.
And so - hardware plays one half while you copy data to the other
half. Hence, "ping pong".
So how do I implement this in ALSA? The "Writing an ALSA Driver"
document doesn't even
contain the words "ping" or "pong".
-- Timur Tabi Linux Kernel Developer @ Freescale _______________________________________________ Alsa-devel mailing list Alsa-devel@alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
Alsa-devel mailing list Alsa-devel@alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
Alsa-devel mailing list Alsa-devel@alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
Hi
yet another newbie question about periods here:
1. AFAIK, the period size is closely dependent on the h/w, but after reading some docs I collected that, they can be given values depending on how much we care about the latency. Does it mean that, I can vary it without paying any attention to what h/w manual says just because I want low or high latency? I hope this question is clear.
2. As periods correspond to fragment size from OSS world, what the other periods related fields correspond to i.e. what do the following fields mean?
period_bytes_min, period_bytes_max, periods_min, periods_max,
I know what they mean after looking at them but I want to know the relationship between various fields.
For e.g.
I have,
buffer_bytes_max = 8192 * 8 i.e. = AUDIO_FRAGSIZE_DEFAULT * AUDIO_NBFRAGS_DEFAULT
here AUDIO_FRAGSIZE_DEFAULT is size of period right? Then to get the max buffer size we should multiply it by number of periods, is this correct? Also, these are default values of the period and no of periods, then do I need to see the h/w manual to decide the periods_min/periods_max and period_bytes_min/period_bytes_max fields?
I found the following doc, it talks about periods in depth with a figure.
http://delivery.acm.org/10.1145/1020000/1017977/6735.html?key1=1017977&k...
-pharaoh.
On 6/12/07, Pharaoh . pharaoh137@gmail.com wrote:
On 6/12/07, Takashi Iwai tiwai@suse.de wrote:
At Tue, 12 Jun 2007 01:59:45 +0530, Nobin Mathew wrote:
Hi,
Now ALSA (audio ) buffer is divided into periods, i.e. a chain of small
packets.
periods size is configurable. Data transfer to the codec starts only after reaching start_threshold point (start_threshold is in periods), this time DMA trigger is called.
This trigger onwards application will get notification from the kernel saying that period buffer is empty you can write into it.Till the end of music file.
Yes. And the "ping-poing" is the case that you have two periods in a ring buffer.
Most hardwares support more periods practically. That's why "periods" (corresponds to "fragments" in OSS) was introduced, as more generic abstraction.
Takashi
Nobin Mathew
On 6/12/07, Timur Tabi timur@freescale.com wrote:
I'm writing an ALSA SOC driver for an I2S-based device, and I'm having
a really hard time
understanding how ALSA uses the DMA buffers. And yes, I've read the
documentation and
studied some sample source code.
I used to write audio drivers for a living, but that was many years
ago, and it wasn't for
Linux. Perhaps the concepts in my head are outdated, but I just don't
see enough
explanation as to how DMA buffers are supposed to work.
Back then, audio drivers used "ping pong" DMA buffers. A single DMA
buffer is allocated,
and the audio hardware is programmed to read from that buffer in a
loop. That is, it
would automatically restart reading from the beginning of the buffer
without any
reprogramming. The hardware would also be programmed to issue an
interrupt when it got to
the end of the buffer, and when it got to the half-way point.
To start playback, you first filled the whole buffer with audio data,
and then told the
hardware to start playing. After the hardware got to the half-way
point, it would issue
an interrupt. You would then tell the OS you need more data, and
you'd get it. You then
copy that data into the first half of the buffer *while* the hardware
was playing the
second half. Later, the hardware would interrupt you when it got to
the end of the
buffer. You'd then copy more data to the 2nd half while the hardware
is playing the first
half.
And so - hardware plays one half while you copy data to the other
half. Hence, "ping pong".
So how do I implement this in ALSA? The "Writing an ALSA Driver"
document doesn't even
contain the words "ping" or "pong".
-- Timur Tabi Linux Kernel Developer @ Freescale _______________________________________________ Alsa-devel mailing list Alsa-devel@alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
Alsa-devel mailing list Alsa-devel@alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
Alsa-devel mailing list Alsa-devel@alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
Hi
yet another newbie question about periods here:
- AFAIK, the period size is closely dependent on the h/w, but after reading
some docs I collected that, they can be given values depending on how much we care about the latency. Does it mean that, I can vary it without paying any attention to what h/w manual says just because I want low or high latency? I hope this question is clear.
- As periods correspond to fragment size from OSS world, what the other
periods related fields correspond to i.e. what do the following fields mean?
period_bytes_min, period_bytes_max, periods_min, periods_max,
I know what they mean after looking at them but I want to know the relationship between various fields.
For e.g.
I have,
buffer_bytes_max = 8192 * 8 i.e. = AUDIO_FRAGSIZE_DEFAULT * AUDIO_NBFRAGS_DEFAULT
here AUDIO_FRAGSIZE_DEFAULT is size of period right? Then to get the max buffer size we should multiply it by number of periods, is this correct? Also, these are default values of the period and no of periods, then do I need to see the h/w manual to decide the periods_min/periods_max and period_bytes_min/period_bytes_max fields?
Takashi Iwai wrote:
Yes. And the "ping-poing" is the case that you have two periods in a ring buffer.
Ok, I understand the 'periods' concept now, thanks.
So when/where does ALSA copy data to the DMA buffer? Isn't there supposed to be some kind of callback where ALSA calls the driver and says, "here's some data, please copy it to your DMA buffer?"
Or does ALSA do all the copying itself whenever the driver calls snd_pcm_period_elapsed()?
At Tue, 12 Jun 2007 11:36:33 -0500, Timur Tabi wrote:
Takashi Iwai wrote:
Yes. And the "ping-poing" is the case that you have two periods in a ring buffer.
Ok, I understand the 'periods' concept now, thanks.
So when/where does ALSA copy data to the DMA buffer? Isn't there supposed to be some kind of callback where ALSA calls the driver and says, "here's some data, please copy it to your DMA buffer?"
Or does ALSA do all the copying itself whenever the driver calls snd_pcm_period_elapsed()?
It depends on the implementation, but usually the latter case. The whole data-transfer task is done by ALSA PCM middle layer, and each driver has to take care of the DMA setup -- as long as your hardware can have a buffer on RAM. If you need to copy the data to hardware buffer manually in some way, the things become complicated...
Takashi
participants (4)
-
Nobin Mathew -
Pharaoh . -
Takashi Iwai -
Timur Tabi