mirror of
https://github.com/torlando-tech/pyxis.git
synced 2026-03-29 05:19:50 +00:00
6744eb136d
- Fix use-after-free crash on hangup: set _call_state=IDLE before deleting _lxst_audio, preventing pump_call_tx() (runs without LVGL lock) from accessing freed memory - Replace single-slot _call_signal_pending with 8-element ring buffer queue to prevent signal loss when CONNECTING+ESTABLISHED arrive in rapid succession - Extract TX pump into pump_call_tx() called right after reticulum->loop() for low-latency audio TX without LVGL lock dependency (was buried at step 10) - Tune ES7210 mic gain to 21dB (was 15dB) to improve Codec2 input level without ADC clipping that occurred at 24dB - I2S capture: use APLL for accurate 8kHz clock, direct 8kHz sampling (no more 16→8kHz decimation), DMA 16x64 for encode burst headroom - Reduce Reticulum log verbosity to LOG_INFO (was LOG_TRACE) - BLE: add ble_hs_sched_reset() tiered recovery before reboot on desync, widen supervision timeout to 4.0s for WiFi coexistence - Add UDP multicast log broadcasting and OTA flash support Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
116 lines
3.8 KiB
C++
116 lines
3.8 KiB
C++
// Copyright (c) 2024 LXST contributors
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// SPDX-License-Identifier: MPL-2.0
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#pragma once
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#include <cstdint>
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#include <atomic>
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class EncodedRingBuffer;
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class VoiceFilterChain;
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class Codec2Wrapper;
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/**
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* ESP32 I2S microphone capture engine for LXST voice streaming.
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*
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* Uses I2S_NUM_1 to capture audio from the ES7210 mic array.
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* Runs a FreeRTOS task that reads I2S DMA, applies voice filters,
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* encodes with Codec2, and writes to an EncodedRingBuffer for
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* the network layer to consume.
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*
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* Audio flow:
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* I2S DMA -> accumulate to frame -> filter -> encode -> EncodedRingBuffer
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*/
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class I2SCapture {
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public:
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I2SCapture();
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~I2SCapture();
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I2SCapture(const I2SCapture&) = delete;
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I2SCapture& operator=(const I2SCapture&) = delete;
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/**
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* Initialize the I2S capture port.
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* Does NOT start capturing — call start() after init.
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* @return true on success
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*/
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bool init();
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/**
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* Configure the encoder. Must be called before start().
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* @param codec Shared Codec2Wrapper (not owned — caller manages lifecycle)
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* @param enableFilters Whether to apply HPF+LPF+AGC filter chain
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* @return true on success
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*/
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bool configureEncoder(Codec2Wrapper* codec, bool enableFilters = true);
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/** Start the capture task. Returns immediately. */
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bool start();
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/** Stop the capture task and release I2S resources. */
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void stop();
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/** Mute/unmute the microphone (sends silence when muted). */
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void setMute(bool muted) { muted_.store(muted, std::memory_order_relaxed); }
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bool isMuted() const { return muted_.load(std::memory_order_relaxed); }
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/** Check if currently capturing. */
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bool isCapturing() const { return capturing_.load(std::memory_order_relaxed); }
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/**
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* Read the next encoded packet from the ring buffer.
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* Called by the network layer.
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*
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* @param dest Output buffer for encoded packet
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* @param maxLength Size of output buffer
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* @param actualLength [out] Actual packet size
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* @return true if a packet was read
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*/
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bool readEncodedPacket(uint8_t* dest, int maxLength, int* actualLength);
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/** Number of encoded packets waiting in the ring buffer. */
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int availablePackets() const;
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/** Release capture buffers (does NOT destroy the shared codec). */
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void releaseBuffers();
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private:
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static void captureTask(void* param);
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void captureLoop();
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bool i2sInitialized_ = false;
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std::atomic<bool> capturing_{false};
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std::atomic<bool> muted_{false};
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void* taskHandle_ = nullptr;
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// Audio pipeline components
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Codec2Wrapper* codec_ = nullptr; // Shared, not owned
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VoiceFilterChain* filterChain_ = nullptr;
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EncodedRingBuffer* encodedRing_ = nullptr;
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// Accumulation buffer: I2S delivers variable bursts, we need fixed-size frames
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int16_t* accumBuffer_ = nullptr;
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int accumCount_ = 0;
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int frameSamples_ = 0; // Codec2 samples per frame (e.g., 320 for 700C, 160 for 1600/3200)
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// Pre-allocated encode output buffer
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uint8_t encodeBuf_[256];
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// Silence buffer for mute
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int16_t* silenceBuf_ = nullptr;
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bool filtersEnabled_ = true;
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static constexpr int I2S_SAMPLE_RATE = 8000; // I2S runs at 8kHz — matches Codec2 directly, no resampling needed
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static constexpr int CODEC_SAMPLE_RATE = 8000; // Codec2 expects 8kHz
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// Accumulate this many codec frames before filter+encode.
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// Matches Columba's 200ms batch (1600 samples for Codec2 3200).
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// The AGC needs large blocks for stable gain tracking.
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static constexpr int FRAMES_PER_BATCH = 10;
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static constexpr int ENCODED_RING_SLOTS = 128;
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static constexpr int ENCODED_RING_MAX_BYTES = 256;
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static constexpr int CAPTURE_TASK_STACK = 24576; // 24KB — pyxis_log→sendto uses ~4KB lwIP stack
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static constexpr int CAPTURE_TASK_PRIORITY = 5;
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static constexpr int CAPTURE_TASK_CORE = 0;
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};
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