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(-)
--
2.34.1
