Add MP3 audio stream support with hardware/software decoder backends
New ASSET_LOADER_TYPE_MP3 (hand-rolled MPEG-1/2/2.5 Layer III header parser - no third-party dependency needed just for metadata, since PSP's hardware path doesn't need one at all) plus a shared audiostreammp3.c stream layer mirroring audiostreampcm.c's shape. Generalized the stream dispatch (hoisted sampleRate/channels onto audiostream_t, added audioStreamGetTotalFrames()/Seek()/Read()) so all three platform audio backends keep working unchanged, just calling the generic names instead of PCM-specific ones. Two decoder backends behind one interface: PSP uses the real sceMp3 hardware decoder (firmware-offloaded, lazily initialized on first use); Linux and Dolphin share one minimp3-based software decoder (public domain, vendored via CMake FetchContent) - libogc's own MP3Player wraps libmad (GPL) and drives its own output pipeline, not a fit for the ansnd-based architecture already in place, so skipped in favor of the shared minimp3 path. Fixed three real bugs found via hardware/runtime testing along the way: - LAME's Xing header counts its own placeholder frame in the declared total, which made playback stall permanently one frame short of the declared end (looked like "never loops") - fixed by subtracting it. - sceMp3Decode() can return more PCM than one MPEG frame's worth in a single call (PSP's pcmBuf is provisioned for 2x), overflowing the shared per-frame decode buffer with no bound check - very intermittent corruption/clicking on real hardware. Widened the buffer to the real worst case and added an assertion. - sceMp3ResetPlayPosition()'s exact internal reset semantics aren't documented precisely enough to trust for looping - occasionally disagreed with the fresh stream position fed right after, clicking at the loop boundary about 1 in 3-4 loops. Rewind now fully tears down and recreates the decoder instead, the same path already proven correct at first Init. Also widened the PSP ring buffer to absorb that now-heavier operation, capping each top-up call's own work so the bigger buffer doesn't turn into one long blocking decode burst instead. Co-Authored-By: Claude Sonnet 5 <[email protected]>
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@@ -9,3 +9,21 @@ target_sources(${DUSK_LIBRARY_TARGET_NAME}
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audiodolphin.c
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audiostreamdolphin.c
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)
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# No hardware MP3 decoder used on Dolphin (see audiostreammp3decoder.h's
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# own comment on why libogc's MP3Player wrapper isn't a fit) - use the
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# same shared minimp3-based software backend as Linux, sourced directly
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# from src/dusk/audio rather than through its own unconditional
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# CMakeLists.txt so platforms with a hardware decoder (PSP) never pull
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# minimp3 in at all.
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if(NOT minimp3_FOUND)
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find_package(minimp3 REQUIRED)
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endif()
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# PRIVATE to match this toolchain's own convention (see cmake/targets/
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# dolphin.cmake) of avoiding PUBLIC library visibility, which has tripped
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# up the PPC linker for other dependencies in the past.
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target_link_libraries(${DUSK_LIBRARY_TARGET_NAME} PRIVATE minimp3)
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target_sources(${DUSK_LIBRARY_TARGET_NAME}
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PUBLIC
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"${DUSK_SOURCES_DIR}/dusk/audio/audiostreammp3decodersw.c"
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)
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@@ -47,8 +47,8 @@ errorret_t audioStreamDolphinDispose(audiostream_t *stream) {
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errorret_t audioStreamDolphinBuffer(audiostream_t *stream) {
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assertNotNull(stream, "Stream cannot be NULL.");
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const size_t frameSize = stream->pcm.channels * sizeof(int16_t);
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const size_t totalFrames = audioStreamPcmGetTotalFrames(stream);
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const size_t frameSize = stream->channels * sizeof(int16_t);
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const size_t totalFrames = audioStreamGetTotalFrames(stream);
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// loopEndFrame/loopToFrame define the loop segment [loopToFrame,
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// loopEndFrame) the DSP wraps within, once looping is enabled -
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@@ -56,10 +56,10 @@ errorret_t audioStreamDolphinBuffer(audiostream_t *stream) {
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// behaviour is unchanged when no explicit loop points are configured.
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// Same math as PSP/Linux - see their own comments.
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const size_t loopEndFrame = stream->loopStart >= 0
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? mathMin((size_t) (stream->loopStart * stream->pcm.sampleRate), totalFrames)
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? mathMin((size_t) (stream->loopStart * stream->sampleRate), totalFrames)
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: totalFrames;
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const size_t loopToFrame = mathMin(
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(size_t) (stream->loopTo * stream->pcm.sampleRate), loopEndFrame
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(size_t) (stream->loopTo * stream->sampleRate), loopEndFrame
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);
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const size_t startFrame = mathMin(stream->startFrame, totalFrames);
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@@ -90,10 +90,10 @@ errorret_t audioStreamDolphinBuffer(audiostream_t *stream) {
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stream->platform.buffer = NULL;
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}
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errorChain(audioStreamPcmSeek(stream, 0));
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errorChain(audioStreamSeek(stream, 0));
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uint8_t *buffer = (uint8_t *) memoryAllocate(bufferFrames * frameSize);
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size_t framesRead = 0;
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errorret_t readRet = audioStreamPcmRead(
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errorret_t readRet = audioStreamRead(
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stream, (int16_t *) buffer, bufferFrames, &framesRead
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);
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if(errorIsNotOk(readRet)) {
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@@ -111,9 +111,9 @@ errorret_t audioStreamDolphinBuffer(audiostream_t *stream) {
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ansnd_pcm_voice_config_t config;
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memoryZero(&config, sizeof(ansnd_pcm_voice_config_t));
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config.samplerate = stream->pcm.sampleRate;
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config.samplerate = stream->sampleRate;
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config.format = ANSND_VOICE_PCM_FORMAT_SIGNED_16_PCM;
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config.channels = stream->pcm.channels;
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config.channels = stream->channels;
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config.pitch = 1.0f;
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config.left_volume = baseVolume * leftFactor;
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config.right_volume = baseVolume * rightFactor;
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@@ -0,0 +1,16 @@
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/**
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* Copyright (c) 2026 Dominic Masters
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*
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* This software is released under the MIT License.
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* https://opensource.org/licenses/MIT
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*/
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#pragma once
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// GameCube/Wii have no usable hardware MP3 decoder for this engine's
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// architecture (libogc's MP3Player wraps libmad but drives its own
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// internal audio output, bypassing the ansnd-based pipeline
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// audiostreamdolphin.c already owns - not a fit here), so Dolphin uses the
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// same shared minimp3-based software backend as Linux - see that header's
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// own documentation for the actual struct/interface.
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#include "audio/audiostreammp3decodersw.h"
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