o copy component audio_board from ADF and create custom component from it
o copy component audio_hal from ADF and create custom component from it
o copy component audio_sal from ADF and create custom component from it
o copy component esp_peripherals from ADF and create custom component from it
- add fLaC support through xiph's original repository as a git module
- run player_task on core 1 with max priority
- create/destroy chunk queue as required automatically
- use mutex to access snapcast config and communicate this change by queue with size 1
- increase buffers for WIFI and LWIP in sdkconfig for NO_SPIRAM config
- add some more debug output
- detect snapcast configuration and init everything accordingly, e.g sample rate, chunk duration, ...
o calculate apll predefines in dependence of sample rate
o communicate these settings to interested parties
- remove typos
- minimize RAM usage of all components
- use both IRAM and DRAM in player component so we can buffer up to 1s on modules without SPI RAM
- support fragemented pcm chunks so we can use all available RAM if there isn't a big enough block available but still enough HEAP
- reinclude all components from jorgen's master branch
- add custom i2s driver to get a precise timing of initial sync
- change wrong usage of esp_timer for latency measurement of snapcast protocol
- add player component
- change syncing to be more predictable using I2S_EVENT_TX_DONE
o also increase DMA length so i2s won't eat up so much processing time
o ensure at least one chunk is in DMA buffer
- drop ADF pipeline stuff related to playback, this introduced too much non deterministic delay
- add fast median calculation component
- increase LWIP buffers
- try to improve synced playback using APLL adjustments
- reduce diffBuf size to 1, bigger values seem to be problematic if packet loss is happening, maybe we need to detect this and clear diffBuf
- do early time syncs on boot to fill diffBuf
- playing with pipline ringbuf sizes and I2S DMA buffer sizes
- use espressif ADF, remove external opus rep
o uses audio pipelines now
- change code to use flac decoder
- remove mersus code
- add first try of audio synchronization
o needed to sync timeofday to server on reception of server settings to avoid overflows in timeval calculations (int32_t on esp32 SDK)
o still a lot of TODO's in the code, but it's almost in sync, although there is quite some chunk skipping which I am currently working on
