To start capture on Microchip PDMC the enable bits for each supported microphone need to be set. After this bit is set the PDMC starts to receive data from microphones and it considers this data as valid data. Thus if microphones are not ready the PDMC captures anyway data from its lines. This data is interpreted by the human ear as poc noises.
To avoid this the following software workaround need to be applied when starting capture: 1/ enable PDMC channel 2/ wait 150ms 3/ execute 16 dummy reads from RHR 4/ clear interrupts 5/ enable interrupts 6/ enable DMA channel
For this workaround to work step 6 need to be executed at the end. For step 6 was added patch 1/3 from this series. With this, driver sets its struct snd_dmaengine_pcm_config::start_dma_last = 1 and proper action is taken based on this flag when starting DAI trigger vs DMA.
The other solution that was identified for this was to extend the already existing mechanism around struct snd_soc_dai_link::stop_dma_first. The downside of this was that a potential struct snd_soc_dai_link::start_dma_last would have to be populated on sound card driver thus, had to be taken into account in all sound card drivers. At the moment, the mchp-pdmc is used only with simple-audio-card. In case of simple-audio-card a new DT binding would had to be introduced to specify this action on dai-link descriptions (as of my investigation).
Please advice what might be the best approach.
Thank you, Claudiu Beznea
Claudiu Beznea (3): ASoC: soc-generic-dmaengine-pcm: add option to start DMA after DAI ASoC: dt-bindings: sama7g5-pdmc: add microchip,startup-delay-us binding ASoC: mchp-pdmc: fix poc noise at capture startup
.../sound/microchip,sama7g5-pdmc.yaml | 6 ++ include/sound/dmaengine_pcm.h | 1 + include/sound/soc-component.h | 2 + sound/soc/atmel/mchp-pdmc.c | 55 +++++++++++++++++-- sound/soc/soc-generic-dmaengine-pcm.c | 8 ++- sound/soc/soc-pcm.c | 27 +++++++-- 6 files changed, 86 insertions(+), 13 deletions(-)