DSP/Audio: Add documenation, cleanup, fix a weird crinkly audio bug, fix truncated ADPCM decoding

src/core/CMakeLists.txt

@@ -11,7 +11,7 @@ set(SRCS
             arm/skyeye_common/vfp/vfpdouble.cpp
             arm/skyeye_common/vfp/vfpinstr.cpp
             arm/skyeye_common/vfp/vfpsingle.cpp
-            audio/stream.cpp
+            audio/audio.cpp
             core.cpp
             core_timing.cpp
             file_sys/archive_backend.cpp
@@ -138,7 +138,7 @@ set(HEADERS
             arm/skyeye_common/vfp/asm_vfp.h
             arm/skyeye_common/vfp/vfp.h
             arm/skyeye_common/vfp/vfp_helper.h
-            audio/stream.h
+            audio/audio.h
             core.h
             core_timing.h
             file_sys/archive_backend.h

src/core/audio/audio.cpp

@@ -0,0 +1,339 @@
+
+#include "AL/al.h"
+#include "AL/alc.h"
+#include "AL/alext.h"
+
+#include "common/logging/log.h"
+
+#include "core/audio/audio.h"
+
+#include <algorithm>
+#include <array>
+#include <queue>
+
+namespace Audio {
+
+static const int BASE_SAMPLE_RATE = 22050;
+
+struct Buffer {
+    u16 id; ///< buffer_id that userland gives us
+    ALuint buffer;
+    bool is_looping;
+
+    bool operator < (const Buffer& other) const {
+        // We want things with lower id to appear first, unless we have wraparound.
+        // priority_queue puts a before b when b < a.
+        // Should perhaps be a instead.
+        if ((other.id - id) > 1000) return true;
+        if ((id - other.id) > 1000) return false;
+        return id > other.id;
+    }
+};
+
+struct AdpcmState {
+    // Two historical samples from previous processed buffer
+    s16 yn1; ///< y[n-1]
+    s16 yn2; ///< y[n-2]
+};
+
+// GC-ADPCM with scale factor and variable coefficients.
+// Frames are 8 bytes long containing 14 samples each.
+// Samples are 4 bits (one nybble) long.
+std::vector<s16> DecodeADPCM(const u8 * const data, const size_t sample_count, const std::array<s16, 16>& adpcm_coeff, AdpcmState& state) {
+    const size_t FRAME_LEN = 8;
+    const size_t SAMPLES_PER_FRAME = 14;
+    const static int SIGNED_NYBBLES[16] = { 0,1,2,3,4,5,6,7,-8,-7,-6,-5,-4,-3,-2,-1 };
+
+    std::vector<s16> ret(sample_count);
+    int yn1 = 0, yn2 = 0;// state.yn1, yn2 = state.yn2;
+
+    const int NUM_FRAMES = (sample_count + (SAMPLES_PER_FRAME-1)) / SAMPLES_PER_FRAME; // Round up.
+    for (int frameno = 0; frameno < NUM_FRAMES; frameno++) {
+        int frame_header = data[frameno * FRAME_LEN];
+        int scale = 1 << (frame_header & 0xF);
+        int idx = (frame_header >> 4) & 0x7;
+
+        // Coefficients are fixed point with 11 bits fractional part.
+        int coef1 = adpcm_coeff[idx * 2 + 0];
+        int coef2 = adpcm_coeff[idx * 2 + 1];
+
+        auto process_nybble = [&](int nybble) -> s16 {
+            int xn = nybble * scale;
+            // We first transform everything into 11 bit fixed point, perform the second order digital filter, then transform back.
+            // 0x400 == 0.5 in 11 bit fixed point.
+            // Filter: y[n] = x[n] + 0.5 + c1 * y[n-1] + c2 * y[n-2]
+            int val = ((xn << 11) + 0x400 + coef1 * yn1 + coef2 * yn2) >> 11;
+            // Clamp to output range.
+            if (val >= 32767) val = 32767;
+            if (val <= -32768) val = -32768;
+            // Advance output feedback.
+            yn2 = yn1;
+            yn1 = val;
+            return (s16)val;
+        };
+
+        int outputi = frameno * SAMPLES_PER_FRAME;
+        int datai = frameno * FRAME_LEN + 1;
+        for (int i = 0; i < SAMPLES_PER_FRAME && outputi < sample_count; i += 2) {
+            ret[outputi++] = process_nybble(SIGNED_NYBBLES[data[datai] & 0xF]);
+            ret[outputi++] = process_nybble(SIGNED_NYBBLES[data[datai] >> 4]);
+            datai++;
+        }
+    }
+
+    state.yn1 = yn1;
+    state.yn2 = yn2;
+
+    return ret;
+}
+
+struct OutputChannel {
+    ALuint source;                     ///< Each channel has it's own output, we lean on OpenAL to do our mixing.
+
+    // Configuration
+    int mono_or_stereo;                ///< Value from userland. 1 == mono, 2 == stereo, other == ???
+    Format format;
+    bool enabled;                      ///< Userland wants us to remind them we have enabled this channel.
+
+    // Buffer management
+    std::priority_queue<Buffer> queue; ///< Things we have gotten from userland we haven't queued onto `source` yet.
+    std::queue<Buffer> playing;        ///< Things we have queued onto `source`.
+    u16 last_bufid;                    ///< Userland wants us to report back what was the thing we last played.
+
+    // For ADPCM decoding use.
+    std::array<s16, 16> adpcm_coeffs;
+    AdpcmState adpcm_state;
+};
+
+OutputChannel chans[24];
+
+int InitAL() {
+    ALCdevice *device = alcOpenDevice(nullptr);
+    if (!device) {
+        LOG_CRITICAL(Audio, "Could not open a device!");
+        return 1;
+    }
+
+    ALCcontext *ctx = alcCreateContext(device, nullptr);
+    if (ctx == nullptr || alcMakeContextCurrent(ctx) == ALC_FALSE) {
+        if (ctx != nullptr) {
+            alcDestroyContext(ctx);
+        }
+        alcCloseDevice(device);
+        LOG_CRITICAL(Audio, "Could not set a context!");
+        return 1;
+    }
+
+    LOG_INFO(Audio, "Audio output is on \"%s\"", alcGetString(device, ALC_DEVICE_SPECIFIER));
+    return 0;
+}
+
+ALCint dev_rate;                ///< Native sample rate of our output device
+std::array<u8, 10000> silence;  ///< Some silence, used if an audio error occurs
+
+void Init() {
+    InitAL();
+
+    ALCdevice *device = alcGetContextsDevice(alcGetCurrentContext());
+    alcGetIntegerv(device, ALC_FREQUENCY, 1, &dev_rate);
+    if (alcGetError(device) != ALC_NO_ERROR) {
+        LOG_CRITICAL(Audio, "Failed to get device sample rate");
+    }
+    LOG_INFO(Audio, "Device Frequency: %i", dev_rate);
+
+    for (int i = 0; i < 24; i++) {
+        alGenSources(1, &chans[i].source);
+        if (alGetError() != AL_NO_ERROR) {
+            LOG_CRITICAL(Audio, "Channel %i: Failed to setup sound source", i);
+        }
+    }
+
+    silence.fill(0);
+}
+
+void Shutdown() {
+    ALCcontext *ctx = alcGetCurrentContext();
+    if (ctx == nullptr) {
+        return;
+    }
+
+    ALCdevice* dev = alcGetContextsDevice(ctx);
+
+    for (int i = 0; i < 24; i++) {
+        alDeleteSources(1, &chans[i].source);
+        while (!chans[i].queue.empty()) {
+            alDeleteBuffers(1, &chans[i].queue.top().buffer);
+            chans[i].queue.pop();
+        }
+        while (!chans[i].playing.empty()) {
+            alDeleteBuffers(1, &chans[i].playing.front().buffer);
+            chans[i].playing.pop();
+        }
+    }
+
+    alcMakeContextCurrent(nullptr);
+    alcDestroyContext(ctx);
+    alcCloseDevice(dev);
+}
+
+void UpdateFormat(int chanid, int mono_or_stereo, Format format) {
+    chans[chanid].mono_or_stereo = mono_or_stereo;
+    chans[chanid].format = format;
+}
+
+void UpdateAdpcm(int chanid, s16 coeffs[16]) {
+    LOG_DEBUG(Audio, "Channel %i: ADPCM Coeffs updated", chanid);
+    std::copy(coeffs, coeffs+16, std::begin(chans[chanid].adpcm_coeffs));
+}
+
+void EnqueueBuffer(int chanid, u16 buffer_id, void* data, int sample_count, bool is_looping) {
+    LOG_DEBUG(Audio, "Channel %i: Buffer %i: Enqueued (size %i)", chanid, buffer_id, sample_count);
+
+    if (is_looping) {
+        LOG_WARNING(Audio, "Channel %i: Buffer %i: Looped buffers are unimplemented", chanid, buffer_id);
+    }
+
+    ALuint b;
+    alGenBuffers(1, &b);
+
+    switch(chans[chanid].format) {
+    case FORMAT_PCM16:
+        switch (chans[chanid].mono_or_stereo) {
+        case 2:
+            alBufferData(b, AL_FORMAT_STEREO16, data, sample_count * 4, BASE_SAMPLE_RATE);
+            break;
+        case 1:
+        default:
+            alBufferData(b, AL_FORMAT_MONO16, data, sample_count * 2, BASE_SAMPLE_RATE);
+            break;
+        }
+
+        if (alGetError() != AL_NO_ERROR) goto do_silence;
+
+        break;
+    case FORMAT_PCM8:
+        switch (chans[chanid].mono_or_stereo) {
+        case 2:
+            alBufferData(b, AL_FORMAT_STEREO8, data, sample_count * 2, BASE_SAMPLE_RATE);
+            break;
+        case 1:
+        default:
+            alBufferData(b, AL_FORMAT_MONO8, data, sample_count * 1, BASE_SAMPLE_RATE);
+            break;
+        }
+
+        if (alGetError() != AL_NO_ERROR) goto do_silence;
+
+        break;
+    case FORMAT_ADPCM: {
+        if (chans[chanid].mono_or_stereo != 1) {
+            LOG_ERROR(Audio, "Channel %i: Buffer %i: Being fed non-mono ADPCM (size: %i samples)", chanid, buffer_id, sample_count);
+        }
+
+        std::vector<s16> decoded = DecodeADPCM((u8*)data, sample_count, chans[chanid].adpcm_coeffs, chans[chanid].adpcm_state);
+        alBufferData(b, AL_FORMAT_STEREO16, decoded.data(), decoded.size() * 2, BASE_SAMPLE_RATE);
+
+        if (alGetError() != AL_NO_ERROR) goto do_silence;
+
+        break;
+    }
+    default:
+        LOG_ERROR(Audio, "Channel %i: Buffer %i: Unrecognised audio format (size: %i samples)", chanid, buffer_id, sample_count);
+    do_silence:
+        if (alGetError() != AL_NO_ERROR) {
+            LOG_CRITICAL(Audio, "Channel %i: Buffer %i: OpenAL says \"%s\"", chanid, buffer_id, alGetString(alGetError()));
+        }
+        alBufferData(b, AL_FORMAT_MONO8, silence.data(), silence.size(), BASE_SAMPLE_RATE);
+        if (alGetError() != AL_NO_ERROR) {
+            LOG_CRITICAL(Audio, "Channel %i: Failed to init silence buffer!!! (%s)", chanid, alGetString(alGetError()));
+        }
+        break;
+    }
+
+    chans[chanid].queue.emplace( Buffer { buffer_id, b, is_looping });
+
+    if (chans[chanid].queue.size() > 10) {
+        LOG_ERROR(Audio, "We have far far too many buffers enqueued on channel %i (%i of them)", chanid, chans[chanid].queue.size());
+    }
+}
+
+void Play(int chanid, bool play) {
+    if (play) {
+        LOG_INFO(Audio, "Channel %i: Enabled", chanid);
+    } else {
+        LOG_INFO(Audio, "Channel %i: Disabled", chanid);
+    }
+    chans[chanid].enabled = play;
+}
+
+void Tick(int chanid) {
+    auto& c = chans[chanid];
+
+    if (!c.queue.empty()) {
+        while (!c.queue.empty()) {
+            alSourceQueueBuffers(c.source, 1, &c.queue.top().buffer);
+            if (alGetError() != AL_NO_ERROR) {
+                alDeleteBuffers(1, &c.queue.top().buffer);
+                LOG_CRITICAL(Audio, "Channel %i: Buffer %i: Failed to enqueue : %s", chanid, c.queue.top().id, alGetString(alGetError()));
+                c.queue.pop();
+                continue;
+            }
+            c.playing.emplace(c.queue.top());
+            c.queue.pop();
+        }
+        if (c.enabled) {
+            ALint state;
+            alGetSourcei(c.source, AL_SOURCE_STATE, &state);
+            if (state != AL_PLAYING) {
+                alSourcePlay(c.source);
+            }
+        }
+    }
+
+    if (chans[chanid].playing.size() > 10) {
+        LOG_ERROR(Audio, "Channel %i: We have far far too many buffers enqueued (%i of them)", chanid, chans[chanid].playing.size());
+    }
+
+    ALint processed;
+    alGetSourcei(c.source, AL_BUFFERS_PROCESSED, &processed);
+    while (processed > 0) {
+        ALuint buf;
+        alSourceUnqueueBuffers(c.source, 1, &buf);
+        processed--;
+
+        if (!c.playing.empty()) {
+            if (c.playing.front().buffer != buf) {
+                LOG_CRITICAL(Audio, "Channel %i: Play queue desynced with OpenAL queue. (buf???)", chanid);
+            } else {
+                LOG_DEBUG(Audio, "Channel %i: Buffer %i: Finished playing", chanid, c.playing.front().id);
+            }
+            c.last_bufid = c.playing.front().id;
+            c.playing.pop();
+        } else {
+            LOG_CRITICAL(Audio, "Channel %i: Play queue desynced with OpenAL queue. (empty)", chanid);
+        }
+
+        alDeleteBuffers(1, &buf);
+    }
+
+    if (!c.playing.empty()) {
+        c.last_bufid = c.playing.front().id;
+    }
+}
+
+std::tuple<bool, u16, u32> GetStatus(int chanid) {
+    auto& c = chans[chanid];
+
+    bool isplaying = c.enabled;
+    u16 bufid = c.last_bufid;
+    u32 pos;
+
+    ALint state, samples;
+    alGetSourcei(c.source, AL_SOURCE_STATE, &state);
+    alGetSourcei(c.source, AL_SAMPLE_OFFSET, &samples);
+    pos = samples;
+
+    return std::make_tuple(isplaying, bufid, pos);
+}
+
+};

src/core/audio/audio.h

@@ -0,0 +1,37 @@
+#pragma once
+
+#include "AL/al.h"
+#include "AL/alc.h"
+#include "AL/alext.h"
+
+#include "common/common_types.h"
+
+#include <tuple>
+
+namespace Audio {
+
+void Init();
+void Shutdown();
+
+enum Format : u16 {
+    FORMAT_PCM8 = 0,
+    FORMAT_PCM16 = 1,
+    FORMAT_ADPCM = 2
+};
+
+void UpdateFormat(int chanid, int mono_or_stereo, Format format);
+void UpdateAdpcm(int chanid, s16 coeffs[16]);
+
+void Play(int chanid, bool play);
+
+void EnqueueBuffer(int chanid, u16 buffer_id, void* data, int sample_count, bool is_looping);
+
+void Tick(int chanid);
+
+// Return values:
+// <1>: is_enabled
+// <2>: prev buffer_id
+// <3>: current sample position in current buffer
+std::tuple<bool, u16, u32> GetStatus(int chanid);
+
+};

src/core/audio/stream.cpp

@@ -1,288 +0,0 @@
-
-#include "AL/al.h"
-#include "AL/alc.h"
-#include "AL/alext.h"
-
-#include "common/logging/log.h"
-
-#include "core/audio/stream.h"
-
-#include <algorithm>
-#include <array>
-#include <queue>
-
-namespace Audio {
-    std::vector<s16> DecodeADPCM(u8* data, size_t sample_count, bool has_adpcm, u16 adpcm_ps, s16* adpcm_yn, const std::array<s16, 16>& adpcm_coeff);
-
-    static const int BASE_SAMPLE_RATE = 22050;
-
-    struct Buffer {
-        u16 id;
-        ALuint buffer;
-        bool is_looping;
-
-        bool operator < (const Buffer& other) const {
-            if ((other.id - id) > 1000) return true;
-            if ((id - other.id) > 1000) return false;
-            return id > other.id;
-        }
-    };
-
-    struct AdpcmState {
-        u16 ps;
-        s16 yn0;
-        s16 yn1;
-    };
-
-    std::vector<s16> DecodeADPCM(u8* data, size_t sample_count, bool has_adpcm, u16 adpcm_ps, s16 adpcm_yn[2], const std::array<s16, 16>& adpcm_coeff, AdpcmState& state) {
-        std::vector<s16> ret(sample_count);
-
-        int yn0 = state.yn0, yn1 = state.yn1;
-
-        if (sample_count % 14 != 0) {
-            LOG_ERROR(Audio, "Audio stream has incomplete frames");
-        }
-
-        const static int signed_nybbles[16] = { 0,1,2,3,4,5,6,7,-8,-7,-6,-5,-4,-3,-2,-1 };
-
-        const int num_frames = sample_count / 14;
-        for (int frameno = 0; frameno < num_frames; frameno++) {
-            int frame_header = data[frameno * 8];
-
-            int scale = 1 << (frame_header & 0xF);
-            int idx = (frame_header >> 4) & 0x7;
-
-            int coef0 = (s16)adpcm_coeff[idx * 2 + 0];
-            int coef1 = (s16)adpcm_coeff[idx * 2 + 1];
-
-            auto next_nybble = [&](int nybble) -> s16 {
-                int val = (((nybble * scale) << 11) + 0x400 + coef0 * yn0 + coef1 * yn1) >> 11;
-                if (val >= 32767) val = 32767;
-                if (val <= -32768) val = -32768;
-                yn1 = yn0;
-                yn0 = val;
-                return (s16)val;
-            };
-
-            for (int i = frameno * 14, datai = frameno * 8 + 1, samplecount = 0; samplecount < 14; i += 2, datai++, samplecount += 2) {
-                ret[i + 0] = next_nybble(signed_nybbles[data[datai] & 0xF]);
-                ret[i + 1] = next_nybble(signed_nybbles[data[datai] >> 4]);
-            }
-        }
-
-        state.yn0 = yn0;
-        state.yn1 = yn1;
-
-        return ret;
-    }
-
-    struct OutputChannel {
-        ALuint source;
-
-        int mono_or_stereo;
-        Format format;
-        int format_rest;
-
-        std::priority_queue<Buffer> queue;
-        std::queue<Buffer> playing;
-        u16 last_bufid;
-
-        bool enabled;
-
-        std::array<s16, 16> adpcm_coeffs;
-        AdpcmState adpcm_state;
-    };
-
-    OutputChannel chans[24];
-
-    int InitAL(void)
-    {
-        ALCdevice *device;
-        ALCcontext *ctx;
-
-        /* Open and initialize a device with default settings */
-        device = alcOpenDevice(NULL);
-        if (!device)
-        {
-            LOG_CRITICAL(Audio, "Could not open a device!");
-            return 1;
-        }
-
-        ctx = alcCreateContext(device, NULL);
-        if (ctx == NULL || alcMakeContextCurrent(ctx) == ALC_FALSE)
-        {
-            if (ctx != NULL)
-                alcDestroyContext(ctx);
-            alcCloseDevice(device);
-            LOG_CRITICAL(Audio, "Could not set a context!");
-            return 1;
-        }
-
-        LOG_INFO(Audio, "Opened \"%s\"", alcGetString(device, ALC_DEVICE_SPECIFIER));
-        return 0;
-    }
-
-    ALuint silencebuffer;
-    ALCint dev_rate;
-    std::array<u8, 10000> silence;
-
-    void Init() {
-        InitAL();
-
-        {
-            ALCdevice *device = alcGetContextsDevice(alcGetCurrentContext());
-            alcGetIntegerv(device, ALC_FREQUENCY, 1, &dev_rate);
-            if (alcGetError(device) != ALC_NO_ERROR) LOG_CRITICAL(Audio, "Failed to get device sample rate");
-            LOG_INFO(Audio, "Device Frequency: %i", dev_rate);
-        }
-
-        for (int i = 0; i < 24; i++) {
-            alGenSources(1, &chans[i].source);
-            if (alGetError() != AL_NO_ERROR) LOG_CRITICAL(Audio, "Failed to setup sound source");
-        }
-
-        silence.fill(0);
-    }
-
-    void Shutdown() {}
-
-    void UpdateFormat(int chanid, int mono_or_stereo, Format format, int rest) {
-        chans[chanid].mono_or_stereo = mono_or_stereo;
-        chans[chanid].format = format;
-        chans[chanid].format_rest = rest;
-    }
-
-    void UpdateAdpcm(int chanid, s16 coeffs[16]) {
-        LOG_INFO(Audio, "ADPCM Coeffs updated for channel %i", chanid);
-        std::copy(coeffs, coeffs+16, std::begin(chans[chanid].adpcm_coeffs));
-    }
-
-    void EnqueueBuffer(int chanid, u16 buffer_id,
-            void* data, int sample_count,
-            bool has_adpcm, u16 adpcm_ps, s16 adpcm_yn[2],
-            bool is_looping) {
-        LOG_INFO(Audio, "enqueu for %i", chanid);
-
-        if (is_looping) {
-            LOG_WARNING(Audio, "Looped buffers are unimplemented");
-        }
-
-        ALuint b;
-        alGenBuffers(1, &b);
-
-        if (chans[chanid].format == FORMAT_PCM16) {
-            switch (chans[chanid].mono_or_stereo) {
-            case 2:
-                alBufferData(b, AL_FORMAT_STEREO16, data, sample_count * 4, BASE_SAMPLE_RATE);
-                break;
-            case 1:
-            default:
-                alBufferData(b, AL_FORMAT_MONO16, data, sample_count * 2, BASE_SAMPLE_RATE);
-                break;
-            }
-            if (alGetError() != AL_NO_ERROR) LOG_CRITICAL(Audio, "Failed to init buffer");
-        } else if (chans[chanid].format == FORMAT_PCM8) {
-            switch (chans[chanid].mono_or_stereo) {
-            case 2:
-                alBufferData(b, AL_FORMAT_STEREO8, data, sample_count * 2, BASE_SAMPLE_RATE);
-                break;
-            case 1:
-            default:
-                alBufferData(b, AL_FORMAT_MONO8, data, sample_count * 1, BASE_SAMPLE_RATE);
-                break;
-            }
-            if (alGetError() != AL_NO_ERROR) LOG_CRITICAL(Audio, "Failed to init buffer");
-        } else if (chans[chanid].format == FORMAT_ADPCM) {
-            if (chans[chanid].mono_or_stereo != 1) {
-                LOG_ERROR(Audio, "Being fed non-mono ADPCM");
-            }
-            std::vector<s16> decoded = DecodeADPCM((u8*)data, sample_count, has_adpcm, adpcm_ps, adpcm_yn, chans[chanid].adpcm_coeffs, chans[chanid].adpcm_state);
-            alBufferData(b, AL_FORMAT_STEREO16, decoded.data(), decoded.size()*2, BASE_SAMPLE_RATE);
-            if (alGetError() != AL_NO_ERROR) LOG_CRITICAL(Audio, "Failed to init buffer");
-        } else {
-            LOG_ERROR(Audio, "Unrecognised audio format in buffer 0x%04x (size: %i samples)", buffer_id, sample_count);
-            alBufferData(b, AL_FORMAT_MONO8, silence.data(), silence.size(), BASE_SAMPLE_RATE);
-            if (alGetError() != AL_NO_ERROR) LOG_CRITICAL(Audio, "Failed to init buffer");
-        }
-
-        chans[chanid].queue.emplace( Buffer { buffer_id, b, is_looping });
-    }
-
-    void Play(int chanid, bool play) {
-        LOG_INFO(Audio, "Play(%i,%i)", chanid, play);
-            chans[chanid].enabled = play;
-    }
-
-    void Tick(int chanid) {
-        auto& c = chans[chanid];
-
-        if (!c.queue.empty()) {
-            while (!c.queue.empty()) {
-                alSourceQueueBuffers(c.source, 1, &c.queue.top().buffer);
-                if (alGetError() != AL_NO_ERROR) {
-                    LOG_CRITICAL(Audio, "Failed to enqueue buffer");
-                    c.queue.pop();
-                    continue;
-                }
-                c.playing.emplace(c.queue.top());
-                LOG_DEBUG(Audio, "Enqueued buffer id 0x%04x", c.queue.top().id);
-                c.queue.pop();
-            }
-            if (c.enabled) {
-                ALint state;
-                alGetSourcei(c.source, AL_SOURCE_STATE, &state);
-                if (state != AL_PLAYING) {
-                    alSourcePlay(c.source);
-                }
-            }
-        }
-
-        if (!c.playing.empty()) {
-            c.last_bufid = c.playing.front().id;
-        }
-
-        ALint processed;
-        alGetSourcei(c.source, AL_BUFFERS_PROCESSED, &processed);
-        while (processed > 0) {
-            ALuint buf;
-            alSourceUnqueueBuffers(c.source, 1, &buf);
-            processed--;
-
-            LOG_DEBUG(Audio, "Finished buffer id 0x%04x", c.playing.front().id);
-
-            if (!c.playing.empty()) {
-                if (c.playing.front().buffer != buf) LOG_CRITICAL(Audio, "Audio is extremely funky. Should abort. (Desynced queue.)");
-
-                c.last_bufid = c.playing.front().id;
-                c.playing.pop();
-            } else {
-                LOG_CRITICAL(Audio, "Audio is extremely funky. Should abort. (Empty queue.)");
-            }
-
-            alDeleteBuffers(1, &buf);
-        }
-
-        if (!c.playing.empty()) {
-            c.last_bufid = c.playing.front().id;
-        }
-    }
-
-    std::tuple<bool, u16, u32> GetStatus(int chanid) {
-        auto& c = chans[chanid];
-
-        bool isplaying = c.enabled;
-        u16 bufid = 0;
-        u32 pos = 0;
-
-        ALint state, samples;
-        alGetSourcei(c.source, AL_SOURCE_STATE, &state);
-        alGetSourcei(c.source, AL_SAMPLE_OFFSET, &samples);
-
-        bufid = c.last_bufid;
-
-        pos = samples;
-
-        return std::make_tuple(isplaying, bufid, pos);
-    }
-
-};

src/core/audio/stream.h

@@ -1,34 +0,0 @@
-#pragma once
-
-#include "AL/al.h"
-#include "AL/alc.h"
-#include "AL/alext.h"
-
-#include "common/common_types.h"
-
-#include <tuple>
-
-namespace Audio {
-    void Init();
-    void Shutdown();
-
-    enum Format : u16 {
-        FORMAT_PCM8 = 0,
-        FORMAT_PCM16 = 1,
-        FORMAT_ADPCM = 2
-    };
-
-    void UpdateFormat(int chanid, int mono_or_stereo, Format format, int rest);
-    void UpdateAdpcm(int chanid, s16 coeffs[16]);
-
-    void Play(int chanid, bool play);
-
-    void EnqueueBuffer(int chanid, u16 buffer_id,
-        void* data, int sample_count,
-        bool has_adpcm, u16 adpcm_ps, s16 adpcm_yn[2],
-        bool is_looping);
-
-    void Tick(int chanid);
-
-    std::tuple<bool, u16, u32> GetStatus(int chanid);
-};

src/core/hle/service/dsp_dsp.cpp

@@ -5,7 +5,7 @@
 #include "common/bit_field.h"
 #include "common/logging/log.h"
 
-#include "core/audio/stream.h"
+#include "core/audio/audio.h"
 #include "core/core_timing.h"
 #include "core/hle/hle.h"
 #include "core/hle/kernel/event.h"
@@ -36,51 +36,51 @@ static std::unordered_map<std::pair<u32, u32>, Kernel::SharedPtr<Kernel::Event>,
 static const u64 frame_tick = 1310252ull;
 static int tick_event;
 
-// Addresses of various things
-static const VAddr BASE_ADDR_0 = Memory::DSP_RAM_VADDR + 0x40000;
-static const VAddr BASE_ADDR_1 = Memory::DSP_RAM_VADDR + 0x60000;
-static constexpr VAddr DspAddrToVAddr(VAddr base, u32 dsp_addr) {
-    return (VAddr(dsp_addr) << 1) + base;
-}
-static const u32 DSPADDR0 = 0xBFFF; // Frame Counter
-static const u32 DSPADDR1 = 0x9E92; // Channel Context (x24)
-static const u32 DSPADDR2 = 0x8680; // Channel Status (x24)
-static const u32 DSPADDR3 = 0xA792; // ADPCM Coefficients (x24)
-static const u32 DSPADDR4 = 0x9430; // Context
-static const u32 DSPADDR5 = 0x8400; // Status
-static const u32 DSPADDR6 = 0x8540; // Loopback Samples
-static const u32 DSPADDR7 = 0x9494;
-static const u32 DSPADDR8 = 0x8710;
-static const u32 DSPADDR9 = 0x8410; // ???
-static const u32 DSPADDR10 = 0xA912;
-static const u32 DSPADDR11 = 0xAA12;
-static const u32 DSPADDR12 = 0xAAD2;
-static const u32 DSPADDR13 = 0xAC52;
-static const u32 DSPADDR14 = 0xAC5C;
-static const u32 DSPADDR_frame_counter = DSPADDR0;
-
 static const int NUM_CHANNELS = 24;
 
+// DSP Addresses
+
+static const VAddr BASE_ADDR_0 = Memory::DSP_RAM_VADDR + 0x40000;
+static const VAddr BASE_ADDR_1 = Memory::DSP_RAM_VADDR + 0x60000;
+
+enum DspRegion {
+    DSPADDR0 = 0xBFFF, // Frame Counter
+    DSPADDR1 = 0x9E92, // Channel Context (x24)
+    DSPADDR2 = 0x8680, // Channel Status (x24)
+    DSPADDR3 = 0xA792, // ADPCM Coefficients (x24)
+    DSPADDR4 = 0x9430, // Context
+    DSPADDR5 = 0x8400, // Status
+    DSPADDR6 = 0x8540, // Loopback Samples
+    DSPADDR7 = 0x9494,
+    DSPADDR8 = 0x8710,
+    DSPADDR9 = 0x8410, // ???
+    DSPADDR10 = 0xA912,
+    DSPADDR11 = 0xAA12,
+    DSPADDR12 = 0xAAD2,
+    DSPADDR13 = 0xAC52,
+    DSPADDR14 = 0xAC5C
+};
+
+static constexpr VAddr DspAddrToVAddr(VAddr base, DspRegion dsp_addr) {
+    return (VAddr(dsp_addr) << 1) + base;
+}
+
 /**
- * DSP_DSP::DspEndian
+ * dsp_u32:
- *     Care must be taken when reading/writing 32-bit values. The DSP has a 16-bit wordsize and is big-endian.
+ *     Care must be taken when reading/writing 32-bit values as the words are not in the expected order.
- *     The bytes in each word when viewed from the ARM11, however, are in little-endian.
- *     Thus we have what appears to be a middle-endian encoding.
- *
- *     The below function is its own inverse.
  */
 struct dsp_u32 {
+    operator u32() {
+        return Convert(storage);
+    }
+    void operator=(u32 newvalue) {
+        storage = Convert(newvalue);
+    }
+private:
     static constexpr u32 Convert(u32 value) {
         return ((value & 0x0000FFFF) << 16) | ((value & 0xFFFF0000) >> 16);
     }
-    operator u32() {
+    u32 storage;
-        return Convert(value);
-    }
-    void operator=(u32 newvalue) {
-        value = Convert(newvalue);
-    }
-private:
-    u32 value;
 };
 
 #define INSERT_PADDING_DSPWORDS(num_words) u16 CONCAT2(pad, __LINE__)[(num_words)]
@@ -88,21 +88,16 @@ private:
     static_assert(std::is_standard_layout<name>::value, "Structure doesn't use standard layout"); \
     static_assert(sizeof(name) == (size), "Unexpected struct size")
 
-/*
- * ADPCM seems to be the usual Nintendo format.
- * ps = predictor / scaler
- * yn[0,1] = sample history
- * Coefficients are found at DSPADDR3
- */
-
 struct Buffer {
     dsp_u32 physical_address;
     dsp_u32 sample_count;
-    u16 adpcm_ps;
+
-    s16 adpcm_yn[2];
+    INSERT_PADDING_DSPWORDS(3);
-    u8 has_adpcm;
+    INSERT_PADDING_BYTES(1);
+
     u8 is_looping;
     u16 buffer_id;
+
     INSERT_PADDING_DSPWORDS(1);
 };
 
@@ -114,36 +109,39 @@ struct ChannelContext {
     float mix[12];
     float rate;
     u8 rim[2];
-    u16 iirFilterType;
+    u16 iirfilter_type;
-    u16 iirFilter_mono[2];
+    u16 iirfilter_mono[2];
-    u16 iirFilter_biquad[5];
+    u16 iirfilter_biquad[5];
 
     // Buffer Queue
     u16 buffers_dirty;                //< Which of those queued buffers is dirty (bit i == buffers[i])
     Buffer buffers[4];                //< Queued Buffers
 
     INSERT_PADDING_DSPWORDS(2);
+
     u16 is_active;                    //< Lower 8 bits == 0x01 if true.
     u16 sync;
+
     INSERT_PADDING_DSPWORDS(4);
 
-    // Current Buffer
+    // Embedded Buffer
     dsp_u32 physical_address;
     dsp_u32 sample_count;
+
     union {
         u16 flags1_raw;
         BitField<0, 2, u16> mono_or_stereo;
         BitField<2, 2, Audio::Format> format;
-        BitField<4, 12, u16> rest;
     };
-    u16 adpcm_ps;
+
-    s16 adpcm_yn[2];
+    INSERT_PADDING_DSPWORDS(3);
+
     union {
         u16 flags2_raw;
         BitField<0, 1, u16> has_adpcm;
         BitField<1, 1, u16> is_looping;
-        BitField<2, 14, u16> rest2;
     };
+
     u16 buffer_id;
 };
 ASSERT_STRUCT(ChannelContext, 192);
@@ -154,7 +152,7 @@ struct ChannelStatus {
     u16 sync;
     dsp_u32 buffer_position;
     u16 current_buffer_id;
-    u16 previous_buffer_id;
+    INSERT_PADDING_DSPWORDS(1);
 };
 ASSERT_STRUCT(ChannelStatus, 12);
 
@@ -175,8 +173,9 @@ static void AudioTick(u64, int cycles_late) {
     VAddr current_base;
 
     {
-        int id0 = (int)Memory::Read16(DspAddrToVAddr(BASE_ADDR_0, DSPADDR_frame_counter));
+        // Frame IDs.
-        int id1 = (int)Memory::Read16(DspAddrToVAddr(BASE_ADDR_1, DSPADDR_frame_counter));
+        int id0 = (int)Memory::Read16(DspAddrToVAddr(BASE_ADDR_0, DSPADDR0));
+        int id1 = (int)Memory::Read16(DspAddrToVAddr(BASE_ADDR_1, DSPADDR0));
 
         // The frame id increments once per audio frame, with wraparound at 65,535.
         // I am uncertain whether the real DSP actually does something like this,
@@ -207,7 +206,7 @@ static void AudioTick(u64, int cycles_late) {
         if (ctx.dirty) {
             if (TestAndUnsetBit(ctx.dirty, 29)) {
                 // First time init
-                LOG_WARNING(Service_DSP, "Unimplemented dirty bit 29");
+                LOG_DEBUG(Service_DSP, "Channel %i: First Time Init", chanid);
             }
 
             if (TestAndUnsetBit(ctx.dirty, 2)) {
@@ -217,50 +216,42 @@ static void AudioTick(u64, int cycles_late) {
 
             if (TestAndUnsetBit(ctx.dirty, 17)) {
                 // Interpolation type
-                LOG_WARNING(Service_DSP, "Unimplemented dirty bit 17");
+                LOG_WARNING(Service_DSP, "Channel %i: Unimplemented dirty bit 17", chanid);
             }
 
             if (TestAndUnsetBit(ctx.dirty, 18)) {
                 // Rate
-                LOG_WARNING(Service_DSP, "Unimplemented dirty bit 18");
+                LOG_WARNING(Service_DSP, "Channel %i: Unimplemented Rate %f", chanid, ctx.rate);
-                LOG_INFO(Service_DSP, "Rate %f", ctx.rate);
             }
 
             if (TestAndUnsetBit(ctx.dirty, 22)) {
                 // IIR
-                LOG_WARNING(Service_DSP, "Unimplemented dirty bit 22");
+                LOG_WARNING(Service_DSP, "Channel %i: Unimplemented IIR %x", chanid, ctx.iirfilter_type);
-                LOG_INFO(Service_DSP, "IIR %x", ctx.iirFilterType);
             }
 
             if (TestAndUnsetBit(ctx.dirty, 28)) {
                 // Sync count
-                LOG_DEBUG(Service_DSP, "Update Sync Count");
+                LOG_DEBUG(Service_DSP, "Channel %i: Update Sync Count");
-
                 status0.sync = ctx.sync;
                 status1.sync = ctx.sync;
             }
 
             if (TestAndUnsetBit(ctx.dirty, 25) | TestAndUnsetBit(ctx.dirty, 26) | TestAndUnsetBit(ctx.dirty, 27)) {
                 // Mix
-                LOG_WARNING(Service_DSP, "Unimplemented dirty bit 25/26/27");
                 for (int i = 0; i < 12; i++)
-                    LOG_INFO(Service_DSP, "mix[%i] %f", i, ctx.mix[i]);
+                    LOG_DEBUG(Service_DSP, "Channel %i: mix[%i] %f", chanid, i, ctx.mix[i]);
             }
 
             if (TestAndUnsetBit(ctx.dirty, 4) | TestAndUnsetBit(ctx.dirty, 21) | TestAndUnsetBit(ctx.dirty, 30)) {
                 // TODO(merry): One of these bits might merely signify an update to the format. Verify this.
-                // Embedded Buffer Changed
+
-                Audio::UpdateFormat(chanid, ctx.mono_or_stereo, ctx.format, ctx.rest);
+                // Format updated
+                Audio::UpdateFormat(chanid, ctx.mono_or_stereo, ctx.format);
                 channel_contex0[chanid].flags1_raw = channel_contex1[chanid].flags1_raw = ctx.flags1_raw;
                 channel_contex0[chanid].flags2_raw = channel_contex1[chanid].flags2_raw = ctx.flags2_raw;
-                if (ctx.rest || ctx.rest2) {
+
-                    LOG_ERROR(Service_DSP, "chan %i rest %04x rest2 %04x", chanid, ctx.rest, ctx.rest2);
+                // Embedded Buffer Changed
-                }
+                Audio::EnqueueBuffer(chanid, ctx.buffer_id, Memory::GetPhysicalPointer(ctx.physical_address), ctx.sample_count, ctx.is_looping);
-                Audio::UpdateAdpcm(chanid, channel_adpcm_coeffs[chanid].coeff);
-                Audio::EnqueueBuffer(chanid, ctx.buffer_id,
-                        Memory::GetPhysicalPointer(ctx.physical_address), ctx.sample_count,
-                        ctx.has_adpcm, ctx.adpcm_ps, ctx.adpcm_yn,
-                        ctx.is_looping);
 
                 status0.is_playing |= 0x100; // TODO: This is supposed to flicker on then turn off.
             }
@@ -270,15 +261,12 @@ static void AudioTick(u64, int cycles_late) {
                 for (int i = 0; i < 4; i++) {
                     if (TestAndUnsetBit(ctx.buffers_dirty, i)) {
                         auto& b = ctx.buffers[i];
-                        Audio::EnqueueBuffer(chanid, b.buffer_id,
+                        Audio::EnqueueBuffer(chanid, b.buffer_id, Memory::GetPhysicalPointer(b.physical_address), b.sample_count, b.is_looping);
-                                Memory::GetPhysicalPointer(b.physical_address), b.sample_count,
-                                b.has_adpcm, b.adpcm_ps, b.adpcm_yn,
-                                b.is_looping);
                     }
                 }
 
                 if (ctx.buffers_dirty) {
-                    LOG_ERROR(Service_DSP, "Unknown channel buffer dirty bits: 0x%04x", ctx.buffers_dirty);
+                    LOG_ERROR(Service_DSP, "Channel %i: Unknown channel buffer dirty bits: 0x%04x", chanid, ctx.buffers_dirty);
                 }
 
                 ctx.buffers_dirty = 0;
@@ -292,16 +280,14 @@ static void AudioTick(u64, int cycles_late) {
             }
 
             if (ctx.dirty) {
-                LOG_ERROR(Service_DSP, "Unknown channel dirty bits: 0x%08x", ctx.dirty);
+                LOG_ERROR(Service_DSP, "Channel %i: Unknown channel dirty bits: 0x%08x", chanid, ctx.dirty);
-                LOG_ERROR(Service_DSP, "%i Rim %i %i", chanid, ctx.rim[0], ctx.rim[1]);
-                LOG_ERROR(Service_DSP, "%i IIR-type %i", chanid, ctx.iirFilterType);
-                LOG_ERROR(Service_DSP, "%i Mono %f %f", chanid, ctx.iirFilter_mono[0], ctx.iirFilter_mono[1]);
-                LOG_ERROR(Service_DSP, "%i Biquad %f %f %f %f %f", chanid, ctx.iirFilter_biquad[0], ctx.iirFilter_biquad[1], ctx.iirFilter_biquad[2], ctx.iirFilter_biquad[3], ctx.iirFilter_biquad[4]);
             }
 
             ctx.dirty = 0;
         }
 
+        // TODO: Detect any change to the structures without a dirty flag update to identify what the other bits do.
+
         Audio::Tick(chanid);
 
         // Update channel status
@@ -335,7 +321,7 @@ static void ConvertProcessAddressFromDspDram(Service::Interface* self) {
     u32 addr = cmd_buff[1];
 
     cmd_buff[1] = 0; // No error
-    cmd_buff[2] = DspAddrToVAddr(BASE_ADDR_0, addr);
+    cmd_buff[2] = DspAddrToVAddr(BASE_ADDR_0, (DspRegion)addr);
 }
 
 /**

src/core/system.cpp

@@ -2,7 +2,7 @@
 // Licensed under GPLv2 or any later version
 // Refer to the license.txt file included.
 
-#include "core/audio/stream.h"
+#include "core/audio/audio.h"
 #include "core/core.h"
 #include "core/core_timing.h"
 #include "core/system.h"