On Mon, Aug 19, 2024 at 3:42 PM Pierre-Louis Bossart pierre-louis.bossart@linux.intel.com wrote:
On 8/16/24 12:42, Shengjiu Wang wrote:
Implement the ASRC memory to memory function using the compress framework, user can use this function with compress ioctl interface.
Define below private metadata key value for output format, output rate and ratio modifier configuration. ASRC_OUTPUT_FORMAT 0x80000001 ASRC_OUTPUT_RATE 0x80000002 ASRC_RATIO_MOD 0x80000003
Can the output format/rate change at run-time?
Seldom I think.
If no, then these parameters should be moved somewhere else - e.g. hw_params or something.
This means I will do some changes in compress_params.h, add output format and output rate definition, follow Jaroslav's example right?
I am still not very clear on the expanding the SET_METADATA ioctl to deal with the ratio changes. This isn't linked to the control layer as suggested before, and there's no precedent of calling it multiple times during streaming.
Which control layer? if you means the snd_kcontrol_new? it is bound with sound card, but in my case, I need to the control bind with the snd_compr_stream to support multi streams/instances.
I also wonder how it was tested since tinycompress does not support this?
I wrote a unit test to test these ASRC M2M functions.
+static int fsl_asrc_m2m_fill_codec_caps(struct fsl_asrc *asrc,
struct snd_compr_codec_caps *codec)
+{
struct fsl_asrc_m2m_cap cap;
__u32 rates[MAX_NUM_BITRATES];
snd_pcm_format_t k;
int i = 0, j = 0;
int ret;
ret = asrc->m2m_get_cap(&cap);
if (ret)
return -EINVAL;
if (cap.rate_in & SNDRV_PCM_RATE_5512)
rates[i++] = snd_pcm_rate_bit_to_rate(SNDRV_PCM_RATE_5512);
this doesn't sound compatible with the patch2 definitions?
cap->rate_in = SNDRV_PCM_RATE_8000_768000;
This ASRC M2M driver is designed for two kinds of hw ASRC modules.
one cap is : cap->rate_in = SNDRV_PCM_RATE_8000_192000 | SNDRV_PCM_RATE_5512; another is : cap->rate_in = SNDRV_PCM_RATE_8000_768000; they are in patch2 and patch3
if (cap.rate_in & SNDRV_PCM_RATE_8000)
rates[i++] = snd_pcm_rate_bit_to_rate(SNDRV_PCM_RATE_8000);
if (cap.rate_in & SNDRV_PCM_RATE_11025)
rates[i++] = snd_pcm_rate_bit_to_rate(SNDRV_PCM_RATE_11025);
if (cap.rate_in & SNDRV_PCM_RATE_16000)
rates[i++] = snd_pcm_rate_bit_to_rate(SNDRV_PCM_RATE_16000);
if (cap.rate_in & SNDRV_PCM_RATE_22050)
rates[i++] = snd_pcm_rate_bit_to_rate(SNDRV_PCM_RATE_22050);
missing 24 kHz
There is no SNDRV_PCM_RATE_24000 in ALSA.
if (cap.rate_in & SNDRV_PCM_RATE_32000)
rates[i++] = snd_pcm_rate_bit_to_rate(SNDRV_PCM_RATE_32000);
if (cap.rate_in & SNDRV_PCM_RATE_44100)
rates[i++] = snd_pcm_rate_bit_to_rate(SNDRV_PCM_RATE_44100);
if (cap.rate_in & SNDRV_PCM_RATE_48000)
rates[i++] = snd_pcm_rate_bit_to_rate(SNDRV_PCM_RATE_48000);
missing 64kHz
Yes, will add it.
Best regards Shengjiu Wang
if (cap.rate_in & SNDRV_PCM_RATE_88200)
rates[i++] = snd_pcm_rate_bit_to_rate(SNDRV_PCM_RATE_88200);
if (cap.rate_in & SNDRV_PCM_RATE_96000)
rates[i++] = snd_pcm_rate_bit_to_rate(SNDRV_PCM_RATE_96000);
if (cap.rate_in & SNDRV_PCM_RATE_176400)
rates[i++] = snd_pcm_rate_bit_to_rate(SNDRV_PCM_RATE_176400);
if (cap.rate_in & SNDRV_PCM_RATE_192000)
rates[i++] = snd_pcm_rate_bit_to_rate(SNDRV_PCM_RATE_192000);
if (cap.rate_in & SNDRV_PCM_RATE_352800)
rates[i++] = snd_pcm_rate_bit_to_rate(SNDRV_PCM_RATE_352800);
if (cap.rate_in & SNDRV_PCM_RATE_384000)
rates[i++] = snd_pcm_rate_bit_to_rate(SNDRV_PCM_RATE_384000);
if (cap.rate_in & SNDRV_PCM_RATE_705600)
rates[i++] = snd_pcm_rate_bit_to_rate(SNDRV_PCM_RATE_705600);
if (cap.rate_in & SNDRV_PCM_RATE_768000)
rates[i++] = snd_pcm_rate_bit_to_rate(SNDRV_PCM_RATE_768000);
pcm_for_each_format(k) {
if (pcm_format_to_bits(k) & cap.fmt_in) {
codec->descriptor[j].max_ch = cap.chan_max;
memcpy(codec->descriptor[j].sample_rates, rates, i * sizeof(__u32));
codec->descriptor[j].num_sample_rates = i;
codec->descriptor[j].formats = k;
j++;
}
}
codec->codec = SND_AUDIOCODEC_PCM;
codec->num_descriptors = j;
return 0;