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README.md
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README.md
@@ -3,17 +3,17 @@
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### Synchronous Multiroom audio streaming client for [Snapcast](https://github.com/badaix/snapcast) ported to ESP32
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## Feature list
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- Opus, FLAC and PCM decoding currently supported
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- FLAC decoding currently supported
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- Wifi setup from menuconfig or through espressif Android App "SoftAP Prov"
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- Auto connect to snapcast server on network
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- Buffers up to 800ms on Wroom modules
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- Buffers up to 1000ms on Wroom modules (tested with 44100:16:2)
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- Buffers more then enough on Wrover modules
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- Multiroom sync delay controlled from Snapcast server (user has to ensure not to set this too high on the server)
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## Description
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I have continued the work from @badaix, @bridadan and @jorgenkraghjakobsen towards a ESP32 Snapcast
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client. Currently it support basic features like multiroom sync, network
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controlled volume and mute. For now it only support Opus, FLAC and PCM 16bit
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controlled volume and mute. For now it only support FLAC 16bit
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audio streams with sample rates up to 48Khz maybe more, I didn't test.
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Please check out the task list and feel free to fill in.
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@@ -30,7 +30,7 @@ The codebase is split into components and build on ESP-IDF v4.3.1. I still
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have some refactoring on the todo list as the concept has started to settle and
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allow for new features can be added in a structured manner. In the code you
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will find parts that are only partly related features and still not on the task
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list.
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list. Also there is a lot of code clean up needed.
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Components
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- audio-board : taken from ADF, stripped down to strictly necessary parts for usage with Lyrat v4.3
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@@ -41,7 +41,7 @@ Components
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- dsp_processor : Audio Processor, low pass filters, effects, etc.
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- esp-dsp : Submodule to the ESP-ADF done by David Douard
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- esp-peripherals : taken from ADF, stripped down to strictly necessary parts for usage with Lyrat v4.3
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- flac : flac audio cider/decoder full submodule
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- flac : flac audio encoder/decoder full submodule
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- libmedian: Median Filter implementation. Many thanks to @accabog https://github.com/accabog/MedianFilter
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- libbuffer : Generic buffer abstraction
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- lightsnapcast :
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@@ -57,7 +57,7 @@ Components
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- wifi_interface : wifi provisoning and init code for wifi module and AP connection
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The snapclient functionanlity are implemented in a task included in main - but
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will be refactored to a component in near future.
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should be refactored to a component at some point.
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I did my own syncing implementation which is different than @jorgenkraghjakobsen's
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approach in the original repository, at least regarding syncing itself. I tried to
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@@ -84,7 +84,7 @@ In this implementation I have separated the sync task to a backend on the other
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end of a freeRTOS queue. Now the front end just needs to pass on the decoded audio
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data to the queue with the server timestamp and chunk size. The backend reads
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timestamps and waits until the audio chunk has the correct playback-delay
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to be written to the DAC amplifer speaker pipeline. When the backend pipeline
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to be written to the DAC amplifer speaker through i2s DMA. When the backend pipeline
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is in sync, any offset get rolled in by micro tuning the APLL on the ESP. No
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sample manipulation needed.
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@@ -165,18 +165,22 @@ Then on every `git commit`, a few sanity/formatting checks will be performed.
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## Task list
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- [ok] Fix to alinge with above
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- [ ] put kconfig to better locations in tree
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- [ok] put kconfig to better locations in tree
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* add codec description
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- [ok] Integrate ESP wifi provision
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- [ok] Find and connect to Avahi broadcasted Snapcast server name
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- [ ] Add a client command interface layer like volume/mute control
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- [ ] add missing codec's (ogg, etc.)
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- [ ] test esp-dsp functionality after ADF drop
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- [ ] Check compatibility with different HW than Lyrat v4.3
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- [ ] rework dsp_processor and test. At the moment only dspfStereo and dspfBassBoost will work. Also ensure/test we got enough RAM on WROVER modules
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- [ ] add missing codec's (ogg, pcm, opus)
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- [ok] test esp-dsp functionality after ADF drop
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- [ok] Check compatibility with different HW than Lyrat v4.3
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- [ok] rework dsp_processor and test. At the moment only dspfStereo and dspfBassBoost will work. Also ensure/test we got enough RAM on WROVER modules
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- [ ] reduce dsp_processor memory footprint
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- [ ] dsp_processor: add equalizer
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* Control interface for equalizer
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-[ ] clean and polish code (remove all unused variables etc.)
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## Minor task
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- [ ] soft mute - play sample in buffer with decreasing volume
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- [ok] soft mute - play sample in buffer with decreasing volume
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- [ok] hard mute - using ADF's HAL
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- [ok] Startup: do not start parsing on samples to codec before sample ring buffer hits requested buffer size.
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- [ok] Start from empty buffer
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