[alsa-devel] ALSA/ASOC new device and routing
Hi
I'm a newbie to ALSA. I'm targeting to use ALSA or ASoC solution in order to route capture/playback streams of a USB headset (that uses ALSA) to/from my own DSP device. I saw DPCM which is a part of ASoC manages dynamically connections between FE and BE streams. In my product, I just need them to be connected. No need for the "dynamically" part.
My questions: 1. Can someone please recommend a step by step guide, or an example how to make my DSP "look like" a BE? Do I need to write an ALSA driver for it or a ASOC driver? 2. After I have it as a BE, how can I connect the streams? I want the transfers to be done in kernel level, without need for application interventions for each buffer. 3. I'm using kernel 2.6.10 (quite old, I know). I don't want to port my entire kernel to a new one. As my requirement is not very complex, I don't think I'd encounter too many issues. However, if someone can point the important parts I need to back port, I'd appreciate it. Alternatively, if someone sees a showstopper in this approach, I'd be grateful to be aware of it.
Thanks, Best regards,
Eli
________________________________
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Reflowed the message.
On Thu, 2014-11-27 at 09:21 +0000, Eli Britstein wrote:
Hi
I'm a newbie to ALSA. I'm targeting to use ALSA or ASoC solution in order to route capture/playback streams of a USB headset (that uses ALSA) to/from my own DSP device. I saw DPCM which is a part of ASoC manages dynamically connections between FE and BE streams. In my product, I just need them to be connected. No need for the "dynamically" part.
Ok, so you have standard static routing from source to sink. ? or does your DSP allow routing from PCM 0 to HW port 0 .. N ?
Best to also describe your HW in more detail so that people on the list can provide better answers.
My questions:
- Can someone please recommend a step by step guide, or an example
how to make my DSP "look like" a BE? Do I need to write an ALSA driver for it or a ASOC driver?
If the routing is static then there is no need to create FE and BE DAI links for your DSP. Just create standard PCM and DAI link. Otherwise have a look at existing DPCM based drivers like Intel Haswell/Broadwell and OMAP4.
- After I have it as a BE, how can I connect the streams? I want the
transfers to be done in kernel level, without need for application interventions for each buffer.
The connection depends on your DSP routing topology. Again look at the existing examples. I think you are also asking about DMA here ? if so, the kernel takes care of DMA configuration, however your application will probably still have to perform IO on your audio buffers.
- I'm using kernel 2.6.10 (quite old, I know). I don't want to port
my entire kernel to a new one. As my requirement is not very complex, I don't think I'd encounter too many issues. However, if someone can point the important parts I need to back port, I'd appreciate it. Alternatively, if someone sees a showstopper in this approach, I'd be grateful to be aware of it.
2.6.10 is really ancient and there have been many improvements since. I would strongly recommend upgrading to the latest release.
Liam
Thanks, Best regards,
Eli
Hi
I'd like to clarify a bit. Also more details below. My system is IP phone, that has a DSP device (proprietary of Audiocodes). Currently, I did things in user space. I read a buffer from ALSA (capture), and write it to the DSP. I read a buffer from the DSP, and write it to the ALSA. The problem is that when I establish a call, there are many things happening, so probably my user space threads don't run on time, so I get under runs and over runs, from which I couldn't recover.
I wanted to move to kernel to assure CPU for the ALSA I/O. I want a very simple thing. I don't mind even changing the my USB ALSA driver to call my DSP_TX whenever it has a capture buffer ready, and to be able to do something like snd_pcm_writei (or mmap_writei) from my DSP driver. I just don't know where to hack to it...
I still don't understand if I need ALSA driver (in which there is no DAI), or an ASOC driver. BTW, I don't need any "mixer" capabilities.
Thanks, Eli.
My system is IP phone. -----Original Message----- From: Liam Girdwood [mailto:liam.r.girdwood@linux.intel.com] Sent: Thursday, November 27, 2014 15:12 To: Eli Britstein Cc: alsa-devel@alsa-project.org; Mark Brown Subject: Re: [alsa-devel] ALSA/ASOC new device and routing
Reflowed the message.
On Thu, 2014-11-27 at 09:21 +0000, Eli Britstein wrote:
Hi
I'm a newbie to ALSA. I'm targeting to use ALSA or ASoC solution in order to route capture/playback streams of a USB headset (that uses ALSA) to/from my own DSP device. I saw DPCM which is a part of ASoC manages dynamically connections between FE and BE streams. In my product, I just need them to be connected. No need for the "dynamically" part.
Ok, so you have standard static routing from source to sink. ? or does your DSP allow routing from PCM 0 to HW port 0 .. N ? [EliB] my DSP doesn't have any relationship with ALSA, so I guess the answer for that is no.
Best to also describe your HW in more detail so that people on the list can provide better answers. [EliB] see above.
My questions:
- Can someone please recommend a step by step guide, or an example
how to make my DSP "look like" a BE? Do I need to write an ALSA driver for it or a ASOC driver?
If the routing is static then there is no need to create FE and BE DAI links for your DSP. Just create standard PCM and DAI link. Otherwise have a look at existing DPCM based drivers like Intel Haswell/Broadwell and OMAP4. [EliB] the routing I want is static.
- After I have it as a BE, how can I connect the streams? I want the
transfers to be done in kernel level, without need for application interventions for each buffer.
The connection depends on your DSP routing topology. Again look at the existing examples. I think you are also asking about DMA here ? if so, the kernel takes care of DMA configuration, however your application will probably still have to perform IO on your audio buffers.
- I'm using kernel 2.6.10 (quite old, I know). I don't want to port
my entire kernel to a new one. As my requirement is not very complex, I don't think I'd encounter too many issues. However, if someone can point the important parts I need to back port, I'd appreciate it. Alternatively, if someone sees a showstopper in this approach, I'd be grateful to be aware of it.
2.6.10 is really ancient and there have been many improvements since. I would strongly recommend upgrading to the latest release.
[EliB] I understand, but I cannot. Such upgrade is quite massive. As I think my requirements are quite simple, I think I can live without it.
Liam
Thanks, Best regards,
Eli
________________________________
This email and any files transmitted with it are confidential material. They are intended solely for the use of the designated individual or entity to whom they are addressed. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, use, distribution or copying of this communication is strictly prohibited and may be unlawful.
If you have received this email in error please immediately notify the sender and delete or destroy any copy of this message
On Thu, 2014-11-27 at 13:56 +0000, Eli Britstein wrote:
Hi
I'd like to clarify a bit. Also more details below. My system is IP phone, that has a DSP device (proprietary of Audiocodes).
Ok, but I meant what CPU and codec are you using ? You will see that a lot of CPU platforms and codecs are already supported.
Currently, I did things in user space. I read a buffer from ALSA (capture), and write it to the DSP. I read a buffer from the DSP, and write it to the ALSA. The problem is that when I establish a call, there are many things happening, so probably my user space threads don't run on time, so I get under runs and over runs, from which I couldn't recover.
aplay and pulseaudio show good examples of how to use the alsa userspace APIs.
I wanted to move to kernel to assure CPU for the ALSA I/O. I want a very simple thing. I don't mind even changing the my USB ALSA driver to call my DSP_TX whenever it has a capture buffer ready, and to be able to do something like snd_pcm_writei (or mmap_writei) from my DSP driver. I just don't know where to hack to it...
I would suggest looking at the documentation available on writing an alsa driver. This will explain what goes where in your driver.
http://www.alsa-project.org/~tiwai/writing-an-alsa-driver/
I still don't understand if I need ALSA driver (in which there is no DAI), or an ASOC driver. BTW, I don't need any "mixer" capabilities.
If you are playing or capturing audio then you will need an ALSA/ASoC driver. If your sound hardware is SoC based then use the ASoC APIs, you will probably find that some of your HW is already supported under ASoC. ASoC extends the ALSA APIs for SoC based devices with multiple audio components (probably like your IP phone HW).
Liam
Thanks, Eli.
My system is IP phone. -----Original Message----- From: Liam Girdwood [mailto:liam.r.girdwood@linux.intel.com] Sent: Thursday, November 27, 2014 15:12 To: Eli Britstein Cc: alsa-devel@alsa-project.org; Mark Brown Subject: Re: [alsa-devel] ALSA/ASOC new device and routing
Reflowed the message.
On Thu, 2014-11-27 at 09:21 +0000, Eli Britstein wrote:
Hi
I'm a newbie to ALSA. I'm targeting to use ALSA or ASoC solution in order to route capture/playback streams of a USB headset (that uses ALSA) to/from my own DSP device. I saw DPCM which is a part of ASoC manages dynamically connections between FE and BE streams. In my product, I just need them to be connected. No need for the "dynamically" part.
Ok, so you have standard static routing from source to sink. ? or does your DSP allow routing from PCM 0 to HW port 0 .. N ? [EliB] my DSP doesn't have any relationship with ALSA, so I guess the answer for that is no.
Best to also describe your HW in more detail so that people on the list can provide better answers. [EliB] see above.
My questions:
- Can someone please recommend a step by step guide, or an example
how to make my DSP "look like" a BE? Do I need to write an ALSA driver for it or a ASOC driver?
If the routing is static then there is no need to create FE and BE DAI links for your DSP. Just create standard PCM and DAI link. Otherwise have a look at existing DPCM based drivers like Intel Haswell/Broadwell and OMAP4. [EliB] the routing I want is static.
- After I have it as a BE, how can I connect the streams? I want the
transfers to be done in kernel level, without need for application interventions for each buffer.
The connection depends on your DSP routing topology. Again look at the existing examples. I think you are also asking about DMA here ? if so, the kernel takes care of DMA configuration, however your application will probably still have to perform IO on your audio buffers.
- I'm using kernel 2.6.10 (quite old, I know). I don't want to port
my entire kernel to a new one. As my requirement is not very complex, I don't think I'd encounter too many issues. However, if someone can point the important parts I need to back port, I'd appreciate it. Alternatively, if someone sees a showstopper in this approach, I'd be grateful to be aware of it.
2.6.10 is really ancient and there have been many improvements since. I would strongly recommend upgrading to the latest release.
[EliB] I understand, but I cannot. Such upgrade is quite massive. As I think my requirements are quite simple, I think I can live without it.
Liam
Thanks, Best regards,
Eli
This email and any files transmitted with it are confidential material. They are intended solely for the use of the designated individual or entity to whom they are addressed. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, use, distribution or copying of this communication is strictly prohibited and may be unlawful.
If you have received this email in error please immediately notify the sender and delete or destroy any copy of this message
Below.
Thanks, Best regards,
Eli Britstein
-----Original Message----- From: Liam Girdwood [mailto:liam.r.girdwood@linux.intel.com] Sent: Thursday, November 27, 2014 21:06 To: Eli Britstein Cc: alsa-devel@alsa-project.org; Mark Brown Subject: Re: [alsa-devel] ALSA/ASOC new device and routing
On Thu, 2014-11-27 at 13:56 +0000, Eli Britstein wrote:
Hi
I'd like to clarify a bit. Also more details below. My system is IP phone, that has a DSP device (proprietary of Audiocodes).
Ok, but I meant what CPU and codec are you using ? You will see that a lot of CPU platforms and codecs are already supported. [EliB] My CPU is mips. The buffers I'd like to use are simple linear 16bit samples. However, the DSP is proprietary. It doesn't have any standard interface in Linux, but I have my own driver to communicate with it.
Currently, I did things in user space. I read a buffer from ALSA (capture), and write it to the DSP. I read a buffer from the DSP, and write it to the ALSA. The problem is that when I establish a call, there are many things happening, so probably my user space threads don't run on time, so I get under runs and over runs, from which I couldn't recover.
aplay and pulseaudio show good examples of how to use the alsa userspace APIs. [EliB] I will take a look. Thanks.
I wanted to move to kernel to assure CPU for the ALSA I/O. I want a very simple thing. I don't mind even changing the my USB ALSA driver to call my DSP_TX whenever it has a capture buffer ready, and to be able to do something like snd_pcm_writei (or mmap_writei) from my DSP driver. I just don't know where to hack to it...
I would suggest looking at the documentation available on writing an alsa driver. This will explain what goes where in your driver.
http://www.alsa-project.org/~tiwai/writing-an-alsa-driver/ [EliB] OK, I will write an ALSA driver. However, suppose I do, or even better, suppose already 2 existing sound cards. One is the PC speakers, and the other one is a USB headset. Is it possible to "connect" them? For example so I would talk into the USB microphone and hear myself in the PC's speakers? I mean some way in kernel mode, not writing a userspace application to do it.
I still don't understand if I need ALSA driver (in which there is no DAI), or an ASOC driver. BTW, I don't need any "mixer" capabilities.
If you are playing or capturing audio then you will need an ALSA/ASoC driver. If your sound hardware is SoC based then use the ASoC APIs, you will probably find that some of your HW is already supported under ASoC. ASoC extends the ALSA APIs for SoC based devices with multiple audio components (probably like your IP phone HW). [EliB] I think I need just an ALSA driver, as I don't think I need any extensions.
Liam
Thanks, Eli.
My system is IP phone. -----Original Message----- From: Liam Girdwood [mailto:liam.r.girdwood@linux.intel.com] Sent: Thursday, November 27, 2014 15:12 To: Eli Britstein Cc: alsa-devel@alsa-project.org; Mark Brown Subject: Re: [alsa-devel] ALSA/ASOC new device and routing
Reflowed the message.
On Thu, 2014-11-27 at 09:21 +0000, Eli Britstein wrote:
Hi
I'm a newbie to ALSA. I'm targeting to use ALSA or ASoC solution in order to route capture/playback streams of a USB headset (that uses ALSA) to/from my own DSP device. I saw DPCM which is a part of ASoC manages dynamically connections between FE and BE streams. In my product, I just need them to be connected. No need for the "dynamically" part.
Ok, so you have standard static routing from source to sink. ? or does your DSP allow routing from PCM 0 to HW port 0 .. N ? [EliB] my DSP doesn't have any relationship with ALSA, so I guess the answer for that is no.
Best to also describe your HW in more detail so that people on the list can provide better answers. [EliB] see above.
My questions:
- Can someone please recommend a step by step guide, or an example
how to make my DSP "look like" a BE? Do I need to write an ALSA driver for it or a ASOC driver?
If the routing is static then there is no need to create FE and BE DAI links for your DSP. Just create standard PCM and DAI link. Otherwise have a look at existing DPCM based drivers like Intel Haswell/Broadwell and OMAP4. [EliB] the routing I want is static.
- After I have it as a BE, how can I connect the streams? I want
the transfers to be done in kernel level, without need for application interventions for each buffer.
The connection depends on your DSP routing topology. Again look at the existing examples. I think you are also asking about DMA here ? if so, the kernel takes care of DMA configuration, however your application will probably still have to perform IO on your audio buffers.
- I'm using kernel 2.6.10 (quite old, I know). I don't want to port
my entire kernel to a new one. As my requirement is not very complex, I don't think I'd encounter too many issues. However, if someone can point the important parts I need to back port, I'd appreciate it. Alternatively, if someone sees a showstopper in this approach, I'd be grateful to be aware of it.
2.6.10 is really ancient and there have been many improvements since. I would strongly recommend upgrading to the latest release.
[EliB] I understand, but I cannot. Such upgrade is quite massive. As I think my requirements are quite simple, I think I can live without it.
Liam
Thanks, Best regards,
Eli
This email and any files transmitted with it are confidential material. They are intended solely for the use of the designated individual or entity to whom they are addressed. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, use, distribution or copying of this communication is strictly prohibited and may be unlawful.
If you have received this email in error please immediately notify the sender and delete or destroy any copy of this message
________________________________
This email and any files transmitted with it are confidential material. They are intended solely for the use of the designated individual or entity to whom they are addressed. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, use, distribution or copying of this communication is strictly prohibited and may be unlawful.
If you have received this email in error please immediately notify the sender and delete or destroy any copy of this message
participants (2)
-
Eli Britstein -
Liam Girdwood