micheleandClaude Sonnet 5
e8f79c4b79
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
66 lines
1.9 KiB
C++
66 lines
1.9 KiB
C++
/**
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* @file phone_stream.cpp
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* @brief net::PhoneStreamSink implementation over the shared I2S sink.
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*
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* DigiRadio firmware — https://github.com/manvalan/DigiRadio
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*
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* Copyright 2026 Michele Bigi
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* SPDX-License-Identifier: Apache-2.0
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*/
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#include "phone_stream.hpp"
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#include "esp32_i2s_sink.hpp"
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#include <array>
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#include <cstdint>
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namespace phone_stream {
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namespace {
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/** Convert one chunk of interleaved 16-bit stereo PCM to the ADAU's
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* 32-bit-slot I2S format and hand it to the shared sink in one write,
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* same rationale as web_radio_stream.cpp's writeFrame(): one
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* i2s_channel_write() call per sample would be a needless source of
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* jitter under sustained streaming. */
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bool writePcm16Stereo(const std::int16_t* interleaved,
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std::size_t frameCount)
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{
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constexpr std::size_t kMaxFramesPerCall = 1024U;
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static std::int32_t out[kMaxFramesPerCall * 2U];
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std::size_t offset = 0U;
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while (offset < frameCount) {
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const std::size_t batch =
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(frameCount - offset) < kMaxFramesPerCall
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? (frameCount - offset)
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: kMaxFramesPerCall;
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for (std::size_t i = 0U; i < batch; ++i) {
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const std::int16_t left = interleaved[(offset + i) * 2U];
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const std::int16_t right = interleaved[(offset + i) * 2U + 1U];
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out[i * 2U] = static_cast<std::int32_t>(left) << 16;
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out[i * 2U + 1U] = static_cast<std::int32_t>(right) << 16;
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}
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if (!esp32_i2s_sink::writeSamples(out, batch * 2U)) {
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return false;
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}
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offset += batch;
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}
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return true;
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}
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} // namespace
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net::PhoneStreamSink& sink() noexcept
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{
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static net::PhoneStreamSink instance{
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.tryAcquire = &esp32_i2s_sink::tryAcquire,
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.release = &esp32_i2s_sink::release,
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.writePcm16Stereo = &writePcm16Stereo,
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};
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return instance;
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}
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} // namespace phone_stream
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