Add PUT /api/stream/phone: raw PCM audio push from a phone app

New endpoint accepts chunked, header-less, 16-bit LE stereo PCM @ 48 kHz in the request body and writes it straight to the shared I2S sink for as long as the connection stays open. Chosen deliberately unencoded (no MP3/AAC/Opus decode) to keep this path as simple and low-risk as possible — the companion app controls the encoding on its side. Extracted the I2S TX channel that used to be owned outright by web_radio_stream.cpp into main/esp32_i2s_sink.cpp, shared by both producers with a simple tryAcquire()/release() exclusivity guard — web radio streaming and a phone PCM stream would otherwise fight over the same physical wire. web_radio_stream.cpp now acquires/releases around each streamWhileEnabled() cycle instead of owning the channel itself. net::PhoneStreamSink is a plain function-pointer struct (not a class hierarchy) threaded through NetBootstrap::start() -> SetupWebServer::start() -> HttpRouteContext, so components/net stays free of I2S driver headers; main/phone_stream.cpp supplies the concrete functions (bound to esp32_i2s_sink) and does the int16->ADAU 32-bit-slot conversion, batched per chunk rather than per sample for the same reason as the web radio stutter fix (6974095/7e65394 lineage). Verified by build only — not confirmed live yet (board disconnected this session); the actual phone app that will exercise this endpoint doesn't exist yet either. Co-Authored-By: Claude Sonnet 5 <noreply@anthropic.com> Claude-Session: https://claude.ai/code/session_0178rASQ6ZETPMUamvpoR2KR
2026-08-18 08:03:39 +02:00
parent 39984a97d1
commit e8f79c4b79
11 changed files with 394 additions and 70 deletions
@@ -0,0 +1,65 @@
+/**
+ * @file    phone_stream.cpp
+ * @brief   net::PhoneStreamSink implementation over the shared I2S sink.
+ *
+ * DigiRadio firmware — https://github.com/manvalan/DigiRadio
+ *
+ * Copyright 2026 Michele Bigi
+ * SPDX-License-Identifier: Apache-2.0
+ */
+
+#include "phone_stream.hpp"
+
+#include "esp32_i2s_sink.hpp"
+
+#include <array>
+#include <cstdint>
+
+namespace phone_stream {
+
+namespace {
+
+/** Convert one chunk of interleaved 16-bit stereo PCM to the ADAU's
+ *  32-bit-slot I2S format and hand it to the shared sink in one write,
+ *  same rationale as web_radio_stream.cpp's writeFrame(): one
+ *  i2s_channel_write() call per sample would be a needless source of
+ *  jitter under sustained streaming. */
+bool writePcm16Stereo(const std::int16_t* interleaved,
+                      std::size_t frameCount)
+{
+    constexpr std::size_t kMaxFramesPerCall = 1024U;
+    static std::int32_t out[kMaxFramesPerCall * 2U];
+
+    std::size_t offset = 0U;
+    while (offset < frameCount) {
+        const std::size_t batch =
+            (frameCount - offset) < kMaxFramesPerCall
+                ? (frameCount - offset)
+                : kMaxFramesPerCall;
+        for (std::size_t i = 0U; i < batch; ++i) {
+            const std::int16_t left = interleaved[(offset + i) * 2U];
+            const std::int16_t right = interleaved[(offset + i) * 2U + 1U];
+            out[i * 2U] = static_cast<std::int32_t>(left) << 16;
+            out[i * 2U + 1U] = static_cast<std::int32_t>(right) << 16;
+        }
+        if (!esp32_i2s_sink::writeSamples(out, batch * 2U)) {
+            return false;
+        }
+        offset += batch;
+    }
+    return true;
+}
+
+} // namespace
+
+net::PhoneStreamSink& sink() noexcept
+{
+    static net::PhoneStreamSink instance{
+        .tryAcquire = &esp32_i2s_sink::tryAcquire,
+        .release = &esp32_i2s_sink::release,
+        .writePcm16Stereo = &writePcm16Stereo,
+    };
+    return instance;
+}
+
+} // namespace phone_stream