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As noted by Marc Madaule (thanks!), the mono AAC decode was not functioning properly. Adjust the output stuff loop to fix.
Bugfixes to AAC and Opus codec addition
* test/run on host to null device * compilation fix
* separate malloc structures * add desync() for user when randomly seeking in the input file fix corner cases leading to exceptions when seeking * additional test when trying to crash-shake everything with a rotary encoder seeking into the currently played file * use private instead of static * remove temporary debug code
PlatformIO requires these dependencies to be explicitly mentioned. Please check the version numbers of the libraries specified are correct. However, v1.0 for each allows for successful builds. Thanks!
When reusing the AudioGeneratorMP3 object to play multiple files with errors, the error count was never reset. Also, if 3 errors are encountered, it would stop running but never clean up the buffers.
Code courtesy of Martin Laclaustra. Uses the ULP coprocessor on the ESP32 to send samples to the onboard DACs, freeing the I2S port for other uses. Connect left output to pin 25, right to pin 26.
…ned into the fs namespace in esp32 libraries. (#335) Co-authored-by: Pierre-Loup Martin <[email protected]>
* Modified AudioFileSourceFS.h to add resolution operator on items defined into the fs namespace in esp32 libraries. * Added track, set, popularimeter, compilation tags. Changed framesize loop with 32bit counter to avoid endless loops for large frames (like APIC). Co-authored-by: Pierre-Loup Martin <[email protected]>
* Fix the order of mono samples (each pair was swapped) * Remove comment on version and date (currently tracked within ESP8266Audio repo) * Added some usage instructions to the header file * Delete sample in buffer after being rendered (ULP) to avoid sound looping in the buffer at the end of playback Co-authored-by: Martin-Laclaustra <[email protected]>
* esp8266: Allows I2S-NoDAC to use only I2S-out-data GPIO avoiding conflicts with repurposed GPIO, especially conflicting with SDFS. * +message * new upstream API update * check i2s_begin's return value
Thanks to @cjwhoishe, fixes #355 . When a large enough read size was requested it is possible that for some streams multiple ICY blocks could appear. The logic doesn't support that, so only read up to 1/2 the ICY block interval to ensure there is no way to see two ICY blocks in one readInternal call.
Thanks to @d-a-v for the debug!
* replace a disappeared shoutcast url by a working one from another example * increase a buffer size (found by valgrind) * make emulated example closer to the not emulated one use `./onHost diff` to show the diff -u * add `diff` option to onHost script * add a delay to local copy of AudioOutputNull * increase another buffer size * make AudioOutputNullSlow.h compatible with arduino api (not only emulation on host)
Ensure the ICY decoder fixes make their way out.
* StreamOnHost: actually play audio * (spaces) * fix comment * update message
* Add arduino-lint CI check * Fix linting warnings
…th the fixed-point CPU of the ESP8266 (#370)
Fixes #399 rtc_clk_cal can return 0 while calibration is underway. If that happens busy-wait until it returns a valid period.
* Fix ULP on ESP32 CONFIG_IDF_TARGET_ESP32 is not yet defined before #include "AudioOutputULP.h" * AudioOutputULP: Fix DAC2 stereo in case of DAC1+DAC2 * ULP - no ESP32S2 / S3 On ESP32S2 it compiles, but no audio is coming out On ESP32S3 DAC is not supported * Astyle --------- Co-authored-by: Earle F. Philhower, III <[email protected]>
Move to Opus 1.5.1 Remove OpusFile and OGG libraries because we parse the OGG file in the object directly with way less memory and overhead. Update example for LittleFS and RP2040 Fixes #372
Only the WAV and FLAC generators can generate 8-bit values, so it is more efficient and sane to just have those generators upscale to 16-bit signed (if necessary) in generation and simplify all the other Generators and Outputs.
Move to the new I2S driver for the ESP chips, since the next revisioncl of the IDF will remove them completely. Make the I2S constructors saner and add consistent SetPinout() and SetBuffers calls to allow configuration. Remove need for separate UseMCLK or SwapClocks calls, we can figure that out from the pins selected. Fixes #758
Borrowed from BackgroundAudio PDM implementation https://github.com/earlephilhower/BackgroundAudio Requires an ESP32 model with PDM support.
Espressif changes the contents of the PDM struct depending on the chip model. Use a hopefully valid `#define` to avoid setting a member that's not there in some cases. Fixes #785
Help with testing optimization by minimizing non-decode CPU usage
* Pull untouched upstream TinySoundFont Start from the upstream TinySoundFont by @schellingb at https://github.com/schellingb/TinySoundFont and rearchitect the MIDI processing. * Convert to PROGMEM-based SoundFont structures. This makes it much faster to access the SoundFont and also removes all the static RAM usage. The samples, the regions, the instruments are all now in ROM. Only the dynamic voices and related info takes up RAM. No caching. * (Re)convert to fixed point math. TSF upstream is built for systems with real floating point units (and produces great sound with it!), but processors like the Pico, ESP8266, and several models of ESP32 don't have one so emulate FP in *extremely* slow SW. Use fixed point (integer) representations for the inner loop and wherever possible to avoid `float` math. For processors like the Pico 2 or original ESP32 with hardware FPUs this still increases performance since integer is still much faster. * Redo the example and add default SF2s Rewrite the MIDI examples to use PROGMEM for both SF2 and MID file. It simplifies life for the casual user and me. * Breaking change The MIDI object now has a `midi::SetSoundFont(tsf *)` instead of a `midi::SetSoundfont(AudioFileSource *)`. It no longer supports reading SF2 from LittleFS or SD.
By rendering into a 32b int we don't have to saturate check in the inner loop of the voice render, which is called multiple times per actual output block. That check is very expensive. In tsf_render_short make the input `short *buffer` have 2x the space as normal to allow the full 32b quantity for each sample to accumulate. After all voices have been rendered, then resample with saturation into int16_t values. About 24% faster on the Pico while still saturating properly.
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Use a numeric approximation to make powf much faster. Loss of accuracy isn't an issue in an audio output device.
The I2S write was always blocking meaning only a single loop call might have ended up playing the entire audio output. Replace the timeout of `10` with `0`.
Added section on ESP32 and Platform.IO integration issues and solutions.
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