Alex - thanks. That was helpful. One follow up question.

if you use the areas-related parameters correctly, the plugin will

also work with planar or complex layouts.

I am not sure if I'm using snd_pcm_channel_area_t correctly. I don't really understand all the fields. For 1 channel setup I stole the following function from the speex plugin

static inline void *area_addr(const snd_pcm_channel_area_t *area, snd_pcm_uframes_t offset) { unsigned int bitofs = area->first + area->step * offset; return (char *) area->addr + bitofs / 8; }

This works for 1 channel but I don't understand how I would use area for interleaved data. My thoughts are just take the returned address from area_addr then use a for loop separating out odd and even shorts. Is there a better way?

Will consider your points about PulseAudio too. This plugin is targeting embedded arm-linux systems so I'm not totally sure if Pulse would even be used.

-----Original Message----- From: Alexander E. Patrakov [mailto:patrakov@gmail.com] Sent: Tuesday, July 01, 2014 11:45 PM To: Brandon Yates Cc: ALSA Development Mailing List Subject: Re: [alsa-devel] PCM Plugin Channel question

02.07.2014 04:59, Brandon Yates wrote:

Hello, I have developed a pcm ext plugin that applies some processing to the audio stream in the transfer callback.

Yes, this is how extplug is supposed to be used.

My plugin works for a mono (1 channel) stream. I'd now like to expand it to support stereo data in the form of 2 channels interleaved. I can't find much information on how this works. I have a couple questions that I'd greatly appreciate if anyone could help with.

1. Is there any way in the 'SND_PCM_PLUGIN_DEFINE_FUNC' function that

I can determine if the user application has opened the stream with 1 channel vs 2 channel?

There certainly is such information, but not directly in SND_PCM_PLUGIN_DEFINE_FUNC. The code should be placed in the init callback. The argument of that callback has a parameter of type snd_pcm_extplug_t*, and that structure has a "channels" field.

You can use https://gitorious.org/dtsenc/dtsenc/source/alsaplugin.c as a model (but it needs to be updated to support channel maps). All actual processing is done in the dcaenc_convert_s32 function which is called by the transfer callback.

2. I assume in the case of two channels my transfer function would

need to: a)deinterleave the data from source so my processing routine can access it in a linear buffer and b) reinterleave the data before I copy it to destination. Is this correct? Is there more to it or things I am overlooking?

This is more-or-less correct. Just be sure to use the areas-related parameters to do deinterleaving correctly, because they can describe not only a simple interleaved format.

3. For IO-plugin I see there is a `SND_PCM_IOPLUG_HW_ACCESS` value I

could use to restrict the stream type to `SND_PCM_ACCESS_RW_INTERLEAVED`. Does there exist a similar function for External PCM Plugin?

You just don't have to apply this restriction.

SND_PCM_ACCESS_MMAP_INTERLEAVED will also work without any effort on your side, due to your transfer function being called on snd_pcm_mmap_commit. And, if you use the areas-related parameters correctly, the plugin will also work with planar or complex layouts.

<troll> But... why are you writing an ALSA plugin? This means that your code will not be usable with PulseAudio out of the box, because PulseAudio can talk to custom PCMs only through module-alsa-sink, which means there will be no jack detection and similar card-related features.

Both ALSA and PulseAudio can wrap LADSPA plugins. Write that. And PulseAudio knows how to use N copies of a mono plugin to process an N-channel stream, so you don't have to deal with that yourself. </troll>

-- Alexander E. Patrakov

Excellent. The fact that the areas are arrays in the case of multiple channels certainly was the one piece I was missing. Thanks for the great explanation, you just made my day easier!

-----Original Message----- From: Alexander E. Patrakov [mailto:patrakov@gmail.com] Sent: Wednesday, July 02, 2014 9:51 AM To: Brandon Yates Cc: alsa-devel@alsa-project.org Subject: Re: [alsa-devel] PCM Plugin Channel question

02.07.2014 19:25, Brandon Yates пишет:

Alex - thanks. That was helpful. One follow up question.

...

if you use the areas-related parameters correctly, the plugin will

also work with planar or complex layouts.

I am not sure if I'm using snd_pcm_channel_area_t correctly. I don't really understand all the fields. For 1 channel setup I stole the following function from the speex plugin

static inline void *area_addr(const snd_pcm_channel_area_t *area, snd_pcm_uframes_t offset) { unsigned int bitofs = area->first + area->step * offset; return (char *) area->addr + bitofs / 8; }

This works for 1 channel but I don't understand how I would use area for interleaved data. My thoughts are just take the returned address from area_addr then use a for loop separating out odd and even shorts. Is there a better way?

For stereo data, the key thought that you are missing is that there will be in fact _two_ areas for stereo data: area[0] and area[1]. Similarly for 4-channel data: 4 areas. One area per channel.

So, assuming interleaved stereo S16 audio, you will get the following two areas:

In area[0], addr points to the 0th left sample, first == 0, step == 32. In area[1], either addr points to the 0th left sample and first == 16, or addr points to 0th right sample and first == 0. In both cases, step == 32.

Step == 32 means that you have to move right by 32 bits (4 bytes) to go to the next sample in this channel.

For non-interleaved audio, for each area, addr points to the beginning of the 0th sample in the corresponding channel, first == 0 and step == 16.

You just have to pass the area corresponding to the correct channel, and the sample index to the function you quoted, and get the address of that sample.

So the things are in fact very simple :)

-- Alexander E. Patrakov