could anybody help me please with ASRC support for audio playback in i.MX6q-based board.
I use LTIB Linux kernel as a reference and wrote a driver for DAC, which is connected to SSI2. The code is derived from sgtl5000 and cs42888 drivers. I've added 2 dai links with names: "HiFi" and "HiFi_ASRC". The first link assumes to play the stream with sample rate it has, and the second - should use ASRC. "HiFi" configuration works, when I start the following command:
$ aplay -D "hw:0,0" my.wav
I see a data at oscilloscope and hear sound. But when I try to use "HiFi_ASRC":
$ aplay -D "hw:0,1" my.wav
I get an I/O error after some timeout. Moreover I see about 4-5 SDMA interrupts only in 'cat /proc/interrupts' after each execution, while for the first case there are about 100.
In my base driver implementation I used the following ASRC configuration:
config.input_word_width = get_asrc_input_width(params);
config.output_word_width = iprtd->p2p->p2p_width;
config.pair = iprtd->asrc_index;
config.channel_num = channel;
config.input_sample_rate = rate;
config.output_sample_rate = iprtd->p2p->p2p_rate;
config.inclk = INCLK_NONE;
config.outclk = OUTCLK_SSI2_TX;
But also I tried to use different combinations of clocks for 'inclk' and 'outclk', but result was the same.
So I have several question and any hints are highly appreciated:
1. How ASRC works, does it require both clocks: input and output? According to the RM, it uses input clock to process samples, but how the output clock is involved to this process?
2. How "asrc_clk" and "asrc_serial_clk" are used, should I set some magic rate for them?
3. Lets say I have an input stream with sample rate 44.1KHz and I want to play it over SSI2 at 48KHz rate. Which clocks should I use? Should I manually setup input clock rate to: F = 44100 * WORD_SIZE * CHANNELS Hz?
With best regards,