all repos — mgba @ 38cb3eeaa857e744917d3ed57263f7585bf74233

mGBA Game Boy Advance Emulator

Merge branch 'feature/blip-buf'
Jeffrey Pfau jeffrey@endrift.com
Mon, 29 Dec 2014 02:38:38 -0800
commit

38cb3eeaa857e744917d3ed57263f7585bf74233

parent

4fa94019712a9eb6177f65962ea032ed784ed019

M CHANGESCHANGES

@@ -3,13 +3,13 @@ Features:

- Support for gamepad axes, e.g. analog sticks or triggers - Add scale presets for up to 6x - Debugger: Add CLI "frame", frame advance command - - Better audio resampling via FFmpeg - Settings window - Bilinear resampling option - Add option to skip BIOS start screen - List of recently opened games - Support for games using the Solar Sensor - Debugger: Add CLI functions for writing to memory + - Better audio resampling via blip-buf Bugfixes: - Qt: Fix issue with set frame sizes being the wrong height - Qt: Fix emulator crashing when full screen if a game is not running
M CMakeLists.txtCMakeLists.txt

@@ -8,6 +8,7 @@ set(USE_FFMPEG ON CACHE BOOL "Whether or not to enable FFmpeg support")

set(USE_PNG ON CACHE BOOL "Whether or not to enable PNG support") set(USE_LIBZIP ON CACHE BOOL "Whether or not to enable ZIP support") set(USE_MAGICK ON CACHE BOOL "Whether or not to enable ImageMagick support") +set(USE_BLIP ON CACHE BOOL "Whether or not to enable blip_buf support") set(BUILD_QT ON CACHE BOOL "Build Qt frontend") set(BUILD_SDL ON CACHE BOOL "Build SDL frontend") set(BUILD_PERF OFF CACHE BOOL "Build performance profiling tool")

@@ -156,7 +157,6 @@ endif()

include_directories(AFTER ${LIBAVCODEC_INCLUDE_DIRS} ${LIBAVFORMAT_INCLUDE_DIRS} ${LIBAVRESAMPLE_INCLUDE_DIRS} ${LIBAVUTIL_INCLUDE_DIRS} ${LIBSWSCALE_INCLUDE_DIRS}) link_directories(${LIBAVCODEC_LIBRARY_DIRS} ${LIBAVFORMAT_LIBRARY_DIRS} ${LIBAVRESAMPLE_LIBRARY_DIRS} ${LIBAVUTIL_LIBRARY_DIRS} ${LIBSWSCALE_LIBRARY_DIRS}) list(APPEND UTIL_SRC "${CMAKE_SOURCE_DIR}/src/platform/ffmpeg/ffmpeg-encoder.c") - list(APPEND UTIL_SRC "${CMAKE_SOURCE_DIR}/src/platform/ffmpeg/ffmpeg-resample.c") string(REGEX MATCH "^[0-9]+" LIBAVCODEC_VERSION_MAJOR ${libavcodec_VERSION}) string(REGEX MATCH "^[0-9]+" LIBAVFORMAT_VERSION_MAJOR ${libavformat_VERSION}) string(REGEX MATCH "^[0-9]+" LIBAVRESAMPLE_VERSION_MAJOR ${libavresample_VERSION})

@@ -165,6 +165,13 @@ string(REGEX MATCH "^[0-9]+" LIBSWSCALE_VERSION_MAJOR ${libswscale_VERSION})

list(APPEND DEPENDENCY_LIB ${LIBAVCODEC_LIBRARIES} ${LIBAVFORMAT_LIBRARIES} ${LIBAVRESAMPLE_LIBRARIES} ${LIBAVUTIL_LIBRARIES} ${LIBSWSCALE_LIBRARIES}) set(CPACK_DEBIAN_PACKAGE_DEPENDS "${CPACK_DEBIAN_PACKAGE_DEPENDS},libavcodec${LIBAVCODEC_VERSION_MAJOR},libavformat${LIBAVFORMAT_VERSION_MAJOR},libavresample${LIBAVRESAMPLE_VERSION_MAJOR},libavutil${LIBAVUTIL_VERSION_MAJOR},libswscale${LIBSWSCALE_VERSION_MAJOR}") set(CPACK_DEBIAN_PACKAGE_RECOMMENDS "libavcodec-extra") +endif() + +if(USE_BLIP) + list(APPEND UTIL_SRC "${CMAKE_SOURCE_DIR}/src/third-party/blip_buf/blip_buf.c") + add_definitions(-DRESAMPLE_LIBRARY=RESAMPLE_BLIP_BUF) +else() + add_definitions(-DRESAMPLE_LIBRARY=RESAMPLE_NN) endif() if(USE_MAGICK)

@@ -256,7 +263,7 @@ message(STATUS " Video recording: ${USE_FFMPEG}")

message(STATUS " GIF recording: ${USE_MAGICK}") message(STATUS " Screenshot/advanced savestate support: ${USE_PNG}") message(STATUS " ZIP support: ${USE_LIBZIP}") -message(STATUS " Better audio resampling: ${USE_FFMPEG}") +message(STATUS " Better audio resampling: ${USE_BLIP}") message(STATUS "Frontend summary:") message(STATUS " Qt: ${BUILD_QT}") message(STATUS " SDL (${SDL_VERSION}): ${BUILD_SDL}")
M README.mdREADME.md

@@ -114,4 +114,4 @@ ---------

mGBA is Copyright © 2013 – 2014 Jeffrey Pfau. It is distributed under the [Mozilla Public License version 2.0](https://www.mozilla.org/MPL/2.0/). A copy of the license is available in the distributed LICENSE file. -mGBA contains [inih](https://code.google.com/p/inih/), which is copyright © 2009 Brush Technology and used under a BSD 3-clause license. +mGBA contains [inih](https://code.google.com/p/inih/), which is copyright © 2009 Brush Technology and used under a BSD 3-clause license; and [blip-buf](https://code.google.com/p/blip-buf/), which is copyright © 2003 – 2009 Shay Green and used under a Lesser GNU Public License.
M src/gba/gba-audio.csrc/gba/gba-audio.c

@@ -12,6 +12,7 @@ #include "gba-thread.h"

#include "gba-video.h" const unsigned GBA_AUDIO_SAMPLES = 2048; +const unsigned BLIP_BUFFER_SIZE = 0x4000; const unsigned GBA_AUDIO_FIFO_SIZE = 8 * sizeof(int32_t); #define SWEEP_CYCLES (GBA_ARM7TDMI_FREQUENCY / 128)

@@ -27,8 +28,17 @@ static int _applyBias(struct GBAAudio* audio, int sample);

static void _sample(struct GBAAudio* audio); void GBAAudioInit(struct GBAAudio* audio, size_t samples) { + audio->samples = samples; +#if RESAMPLE_LIBRARY != RESAMPLE_BLIP_BUF CircleBufferInit(&audio->left, samples * sizeof(int16_t)); CircleBufferInit(&audio->right, samples * sizeof(int16_t)); +#else + audio->left = blip_new(BLIP_BUFFER_SIZE); + audio->right = blip_new(BLIP_BUFFER_SIZE); + // Guess too large; we hang producing extra samples if we guess too low + blip_set_rates(audio->left, GBA_ARM7TDMI_FREQUENCY, 96000); + blip_set_rates(audio->right, GBA_ARM7TDMI_FREQUENCY, 96000); +#endif CircleBufferInit(&audio->chA.fifo, GBA_AUDIO_FIFO_SIZE); CircleBufferInit(&audio->chB.fifo, GBA_AUDIO_FIFO_SIZE); }

@@ -77,21 +87,34 @@ audio->playingCh4 = false;

audio->enable = false; audio->sampleInterval = GBA_ARM7TDMI_FREQUENCY / audio->sampleRate; +#if RESAMPLE_LIBRARY != RESAMPLE_BLIP_BUF CircleBufferClear(&audio->left); CircleBufferClear(&audio->right); +#else + blip_clear(audio->left); + blip_clear(audio->right); + audio->clock = 0; +#endif CircleBufferClear(&audio->chA.fifo); CircleBufferClear(&audio->chB.fifo); } void GBAAudioDeinit(struct GBAAudio* audio) { +#if RESAMPLE_LIBRARY != RESAMPLE_BLIP_BUF CircleBufferDeinit(&audio->left); CircleBufferDeinit(&audio->right); +#else + blip_delete(audio->left); + blip_delete(audio->right); +#endif CircleBufferDeinit(&audio->chA.fifo); CircleBufferDeinit(&audio->chB.fifo); } void GBAAudioResizeBuffer(struct GBAAudio* audio, size_t samples) { GBASyncLockAudio(audio->p->sync); + audio->samples = samples; +#if RESAMPLE_LIBRARY != RESAMPLE_BLIP_BUF size_t oldCapacity = audio->left.capacity; int16_t* buffer = malloc(oldCapacity); int16_t dummy;

@@ -119,6 +142,11 @@ }

} free(buffer); +#else + blip_clear(audio->left); + blip_clear(audio->right); + audio->clock = 0; +#endif GBASyncConsumeAudio(audio->p->sync); }

@@ -424,6 +452,13 @@ }

void GBAAudioWriteSOUNDBIAS(struct GBAAudio* audio, uint16_t value) { audio->soundbias = value; +#if RESAMPLE_LIBRARY == RESAMPLE_BLIP_BUF + unsigned newSampleRate = 0x8000 << GBARegisterSOUNDBIASGetResolution(audio->soundbias); + if (audio->sampleRate != newSampleRate) { + audio->sampleRate = newSampleRate; + audio->sampleInterval = GBA_ARM7TDMI_FREQUENCY / audio->sampleRate; + } +#endif } void GBAAudioWriteWaveRAM(struct GBAAudio* audio, int address, uint32_t value) {

@@ -495,6 +530,7 @@ }

CircleBufferRead8(&channel->fifo, &channel->sample); } +#if RESAMPLE_LIBRARY != RESAMPLE_BLIP_BUF unsigned GBAAudioCopy(struct GBAAudio* audio, void* left, void* right, unsigned nSamples) { GBASyncLockAudio(audio->p->sync); unsigned read = 0;

@@ -553,6 +589,7 @@ }

} return totalRead; } +#endif bool _writeEnvelope(struct GBAAudioEnvelope* envelope, uint16_t value) { envelope->length = GBAAudioRegisterEnvelopeGetLength(value);

@@ -771,14 +808,32 @@ sampleLeft = _applyBias(audio, sampleLeft);

sampleRight = _applyBias(audio, sampleRight); GBASyncLockAudio(audio->p->sync); + unsigned produced; +#if RESAMPLE_LIBRARY != RESAMPLE_BLIP_BUF CircleBufferWrite16(&audio->left, sampleLeft); CircleBufferWrite16(&audio->right, sampleRight); - unsigned produced = CircleBufferSize(&audio->left); + produced = CircleBufferSize(&audio->left) / 2; +#else + if ((size_t) blip_samples_avail(audio->left) < audio->samples) { + blip_add_delta(audio->left, audio->clock, sampleLeft - audio->lastLeft); + blip_add_delta(audio->right, audio->clock, sampleRight - audio->lastRight); + audio->lastLeft = sampleLeft; + audio->lastRight = sampleRight; + audio->clock += audio->sampleInterval; + int clockNeeded = blip_clocks_needed(audio->left, audio->samples / 32); + if (audio->clock >= clockNeeded) { + blip_end_frame(audio->left, audio->clock); + blip_end_frame(audio->right, audio->clock); + audio->clock -= clockNeeded; + } + } + produced = blip_samples_avail(audio->left); +#endif struct GBAThread* thread = GBAThreadGetContext(); if (thread && thread->stream) { thread->stream->postAudioFrame(thread->stream, sampleLeft, sampleRight); } - GBASyncProduceAudio(audio->p->sync, produced >= CircleBufferCapacity(&audio->left)); + GBASyncProduceAudio(audio->p->sync, produced >= audio->samples); } void GBAAudioSerialize(const struct GBAAudio* audio, struct GBASerializedState* state) {
M src/gba/gba-audio.hsrc/gba/gba-audio.h

@@ -11,9 +11,16 @@ #include "macros.h"

#include "util/circle-buffer.h" +#if RESAMPLE_LIBRARY == RESAMPLE_BLIP_BUF +#include "third-party/blip_buf/blip_buf.h" +#endif + struct GBADMA; extern const unsigned GBA_AUDIO_SAMPLES; + +#define RESAMPLE_NN 0 +#define RESAMPLE_BLIP_BUF 2 DECL_BITFIELD(GBAAudioRegisterEnvelope, uint16_t); DECL_BITS(GBAAudioRegisterEnvelope, Length, 0, 6);

@@ -177,8 +184,16 @@

struct GBAAudioFIFO chA; struct GBAAudioFIFO chB; +#if RESAMPLE_LIBRARY != RESAMPLE_BLIP_BUF struct CircleBuffer left; struct CircleBuffer right; +#else + blip_t* left; + blip_t* right; + int16_t lastLeft; + int16_t lastRight; + int clock; +#endif uint8_t volumeRight; uint8_t volumeLeft;

@@ -207,6 +222,7 @@ bool playingCh3;

bool playingCh4; bool enable; + size_t samples; unsigned sampleRate; GBARegisterSOUNDBIAS soundbias;

@@ -256,8 +272,10 @@ void GBAAudioWriteFIFO16(struct GBAAudio* audio, int address, uint16_t value);

void GBAAudioWriteFIFO(struct GBAAudio* audio, int address, uint32_t value); void GBAAudioSampleFIFO(struct GBAAudio* audio, int fifoId, int32_t cycles); +#if RESAMPLE_LIBRARY != RESAMPLE_BLIP_BUF unsigned GBAAudioCopy(struct GBAAudio* audio, void* left, void* right, unsigned nSamples); unsigned GBAAudioResampleNN(struct GBAAudio*, float ratio, float* drift, struct GBAStereoSample* output, unsigned nSamples); +#endif struct GBASerializedState; void GBAAudioSerialize(const struct GBAAudio* audio, struct GBASerializedState* state);
D src/platform/ffmpeg/ffmpeg-resample.c

@@ -1,64 +0,0 @@

-/* Copyright (c) 2013-2014 Jeffrey Pfau - * - * This Source Code Form is subject to the terms of the Mozilla Public - * License, v. 2.0. If a copy of the MPL was not distributed with this - * file, You can obtain one at http://mozilla.org/MPL/2.0/. */ -#include "ffmpeg-resample.h" - -#include "gba-audio.h" - -#include <libavresample/avresample.h> -#include <libavutil/opt.h> - -struct AVAudioResampleContext* GBAAudioOpenLAVR(unsigned inputRate, unsigned outputRate) { - AVAudioResampleContext *avr = avresample_alloc_context(); - av_opt_set_int(avr, "in_channel_layout", AV_CH_LAYOUT_STEREO, 0); - av_opt_set_int(avr, "out_channel_layout", AV_CH_LAYOUT_STEREO, 0); - av_opt_set_int(avr, "in_sample_rate", inputRate, 0); - av_opt_set_int(avr, "out_sample_rate", outputRate, 0); - av_opt_set_int(avr, "in_sample_fmt", AV_SAMPLE_FMT_S16P, 0); - av_opt_set_int(avr, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0); - if (avresample_open(avr)) { - avresample_free(&avr); - return 0; - } - return avr; -} - -struct AVAudioResampleContext* GBAAudioReopenLAVR(struct AVAudioResampleContext* avr, unsigned inputRate, unsigned outputRate) { - avresample_close(avr); - av_opt_set_int(avr, "in_sample_rate", inputRate, 0); - av_opt_set_int(avr, "out_sample_rate", outputRate, 0); - if (avresample_open(avr)) { - avresample_free(&avr); - return 0; - } - return avr; -} - -unsigned GBAAudioResampleLAVR(struct GBAAudio* audio, struct AVAudioResampleContext* avr, struct GBAStereoSample* output, unsigned nSamples) { - int16_t left[GBA_AUDIO_SAMPLES]; - int16_t right[GBA_AUDIO_SAMPLES]; - int16_t* samples[2] = { left, right }; - - size_t totalRead = 0; - size_t available = avresample_available(avr); - if (available) { - totalRead = avresample_read(avr, (uint8_t**) &output, nSamples); - nSamples -= totalRead; - output += totalRead; - } - while (nSamples) { - unsigned read = GBAAudioCopy(audio, left, right, GBA_AUDIO_SAMPLES); - if (read == 0) { - memset(output, 0, nSamples * sizeof(struct GBAStereoSample)); - break; - } - - size_t currentRead = avresample_convert(avr, (uint8_t**) &output, nSamples * sizeof(struct GBAStereoSample), nSamples, (uint8_t**) samples, sizeof(left), read); - nSamples -= currentRead; - output += currentRead; - totalRead += currentRead; - } - return totalRead; -}
D src/platform/ffmpeg/ffmpeg-resample.h

@@ -1,17 +0,0 @@

-/* Copyright (c) 2013-2014 Jeffrey Pfau - * - * This Source Code Form is subject to the terms of the Mozilla Public - * License, v. 2.0. If a copy of the MPL was not distributed with this - * file, You can obtain one at http://mozilla.org/MPL/2.0/. */ -#ifndef FFMPEG_RESAMPLE -#define FFMPEG_RESAMPLE - -struct AVAudioResampleContext; -struct GBAAudio; -struct GBAStereoSample; - -struct AVAudioResampleContext* GBAAudioOpenLAVR(unsigned inputRate, unsigned outputRate); -struct AVAudioResampleContext* GBAAudioReopenLAVR(struct AVAudioResampleContext* avr, unsigned inputRate, unsigned outputRate); -unsigned GBAAudioResampleLAVR(struct GBAAudio* audio, struct AVAudioResampleContext* avr, struct GBAStereoSample* output, unsigned nSamples); - -#endif
M src/platform/qt/GameController.cppsrc/platform/qt/GameController.cpp

@@ -369,6 +369,7 @@ m_threadContext.activeKeys = activeKeys;

} void GameController::redoSamples(int samples) { +#if RESAMPLE_LIBRARY != RESAMPLE_BLIP_BUF float sampleRate = 0x8000; float ratio; if (m_threadContext.gba) {

@@ -376,6 +377,9 @@ sampleRate = m_threadContext.gba->audio.sampleRate;

} ratio = GBAAudioCalculateRatio(sampleRate, m_threadContext.fpsTarget, 44100); m_threadContext.audioBuffers = ceil(samples / ratio); +#else + m_threadContext.audioBuffers = samples; +#endif if (m_threadContext.gba) { GBAAudioResizeBuffer(&m_threadContext.gba->audio, m_threadContext.audioBuffers); }
M src/platform/sdl/sdl-audio.csrc/platform/sdl/sdl-audio.c

@@ -8,9 +8,8 @@

#include "gba.h" #include "gba-thread.h" -#ifdef USE_FFMPEG -#include "platform/ffmpeg/ffmpeg-resample.h" -#include <libavresample/avresample.h> +#if RESAMPLE_LIBRARY == RESAMPLE_BLIP_BUF +#include "third-party/blip_buf/blip_buf.h" #endif #define BUFFER_SIZE (GBA_AUDIO_SAMPLES >> 2)

@@ -29,7 +28,7 @@ context->desiredSpec.channels = 2;

context->desiredSpec.samples = context->samples; context->desiredSpec.callback = _GBASDLAudioCallback; context->desiredSpec.userdata = context; -#ifndef USE_FFMPEG +#if RESAMPLE_LIBRARY == RESAMPLE_NN context->drift = 0.f; #endif if (SDL_OpenAudio(&context->desiredSpec, &context->obtainedSpec) < 0) {

@@ -44,10 +43,6 @@ if (context->samples > threadContext->audioBuffers) {

threadContext->audioBuffers = context->samples * 2; } -#ifdef USE_FFMPEG - context->avr = 0; -#endif - SDL_PauseAudio(0); return true; }

@@ -57,9 +52,6 @@ UNUSED(context);

SDL_PauseAudio(1); SDL_CloseAudio(); SDL_QuitSubSystem(SDL_INIT_AUDIO); -#ifdef USE_FFMPEG - avresample_free(&context->avr); -#endif } void GBASDLPauseAudio(struct GBASDLAudio* context) {

@@ -78,7 +70,7 @@ if (!context || !audioContext->thread || !audioContext->thread->gba) {

memset(data, 0, len); return; } -#ifndef USE_FFMPEG +#if RESAMPLE_LIBRARY == RESAMPLE_NN audioContext->ratio = GBAAudioCalculateRatio(audioContext->thread->gba->audio.sampleRate, audioContext->thread->fpsTarget, audioContext->obtainedSpec.freq); if (audioContext->ratio == INFINITY) { memset(data, 0, len);

@@ -89,21 +81,23 @@ len /= 2 * audioContext->obtainedSpec.channels;

if (audioContext->obtainedSpec.channels == 2) { GBAAudioResampleNN(&audioContext->thread->gba->audio, audioContext->ratio, &audioContext->drift, ssamples, len); } -#else - float ratio = GBAAudioCalculateRatio(audioContext->thread->gba->audio.sampleRate, audioContext->thread->fpsTarget, audioContext->thread->gba->audio.sampleRate); - if (!audioContext->avr) { - if (!audioContext->thread->gba->audio.sampleRate) { - memset(data, 0, len); - return; - } - audioContext->ratio = ratio; - audioContext->avr = GBAAudioOpenLAVR(audioContext->thread->gba->audio.sampleRate / ratio, audioContext->obtainedSpec.freq); - } else if (ratio != audioContext->ratio) { - audioContext->ratio = ratio; - audioContext->avr = GBAAudioReopenLAVR(audioContext->avr, audioContext->thread->gba->audio.sampleRate / ratio, audioContext->obtainedSpec.freq); - } - struct GBAStereoSample* ssamples = (struct GBAStereoSample*) data; +#elif RESAMPLE_LIBRARY == RESAMPLE_BLIP_BUF + double fauxClock = GBAAudioCalculateRatio(1, audioContext->thread->fpsTarget, 1); + GBASyncLockAudio(&audioContext->thread->sync); + blip_set_rates(audioContext->thread->gba->audio.left, GBA_ARM7TDMI_FREQUENCY, audioContext->obtainedSpec.freq * fauxClock); + blip_set_rates(audioContext->thread->gba->audio.right, GBA_ARM7TDMI_FREQUENCY, audioContext->obtainedSpec.freq * fauxClock); len /= 2 * audioContext->obtainedSpec.channels; - GBAAudioResampleLAVR(&audioContext->thread->gba->audio, audioContext->avr, ssamples, len); + int available = blip_samples_avail(audioContext->thread->gba->audio.left); + if (available > len) { + available = len; + } + blip_read_samples(audioContext->thread->gba->audio.left, (short*) data, available, audioContext->obtainedSpec.channels == 2); + if (audioContext->obtainedSpec.channels == 2) { + blip_read_samples(audioContext->thread->gba->audio.right, ((short*) data) + 1, available, 1); + } + GBASyncConsumeAudio(&audioContext->thread->sync); + if (available < len) { + memset(((short*) data) + audioContext->obtainedSpec.channels * available, 0, len - available); + } #endif }
M src/platform/sdl/sdl-audio.hsrc/platform/sdl/sdl-audio.h

@@ -10,6 +10,8 @@ #include "util/common.h"

#include <SDL.h> +#include "gba-audio.h" + struct GBASDLAudio { // Input size_t samples;

@@ -17,11 +19,11 @@

// State SDL_AudioSpec desiredSpec; SDL_AudioSpec obtainedSpec; +#if RESAMPLE_LIBRARY != RESAMPLE_BLIP_BUF float ratio; -#ifndef USE_FFMPEG +#endif +#if RESAMPLE_LIBRARY == RESAMPLE_NN float drift; -#else - struct AVAudioResampleContext* avr; #endif struct GBAThread* thread;
A src/third-party/blip_buf/blip_buf.c

@@ -0,0 +1,344 @@

+/* blip_buf 1.1.0. http://www.slack.net/~ant/ */ + +#include "blip_buf.h" + +#include <assert.h> +#include <limits.h> +#include <string.h> +#include <stdlib.h> + +/* Library Copyright (C) 2003-2009 Shay Green. This library is free software; +you can redistribute it and/or modify it under the terms of the GNU Lesser +General Public License as published by the Free Software Foundation; either +version 2.1 of the License, or (at your option) any later version. This +library is distributed in the hope that it will be useful, but WITHOUT ANY +WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR +A PARTICULAR PURPOSE. See the GNU Lesser General Public License for more +details. You should have received a copy of the GNU Lesser General Public +License along with this module; if not, write to the Free Software Foundation, +Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ + +#if defined (BLARGG_TEST) && BLARGG_TEST + #include "blargg_test.h" +#endif + +/* Equivalent to ULONG_MAX >= 0xFFFFFFFF00000000. +Avoids constants that don't fit in 32 bits. */ +#if ULONG_MAX/0xFFFFFFFF > 0xFFFFFFFF + typedef unsigned long fixed_t; + enum { pre_shift = 32 }; + +#elif defined(ULLONG_MAX) + typedef unsigned long long fixed_t; + enum { pre_shift = 32 }; + +#else + typedef unsigned fixed_t; + enum { pre_shift = 0 }; + +#endif + +enum { time_bits = pre_shift + 20 }; + +static fixed_t const time_unit = (fixed_t) 1 << time_bits; + +enum { bass_shift = 9 }; /* affects high-pass filter breakpoint frequency */ +enum { end_frame_extra = 2 }; /* allows deltas slightly after frame length */ + +enum { half_width = 8 }; +enum { buf_extra = half_width*2 + end_frame_extra }; +enum { phase_bits = 5 }; +enum { phase_count = 1 << phase_bits }; +enum { delta_bits = 15 }; +enum { delta_unit = 1 << delta_bits }; +enum { frac_bits = time_bits - pre_shift }; + +/* We could eliminate avail and encode whole samples in offset, but that would +limit the total buffered samples to blip_max_frame. That could only be +increased by decreasing time_bits, which would reduce resample ratio accuracy. +*/ + +/** Sample buffer that resamples to output rate and accumulates samples +until they're read out */ +struct blip_t +{ + fixed_t factor; + fixed_t offset; + int avail; + int size; + int integrator; +}; + +typedef int buf_t; + +/* probably not totally portable */ +#define SAMPLES( buf ) ((buf_t*) ((buf) + 1)) + +/* Arithmetic (sign-preserving) right shift */ +#define ARITH_SHIFT( n, shift ) \ + ((n) >> (shift)) + +enum { max_sample = +32767 }; +enum { min_sample = -32768 }; + +#define CLAMP( n ) \ + {\ + if ( (short) n != n )\ + n = ARITH_SHIFT( n, 16 ) ^ max_sample;\ + } + +static void check_assumptions( void ) +{ + int n; + + #if INT_MAX < 0x7FFFFFFF || UINT_MAX < 0xFFFFFFFF + #error "int must be at least 32 bits" + #endif + + assert( (-3 >> 1) == -2 ); /* right shift must preserve sign */ + + n = max_sample * 2; + CLAMP( n ); + assert( n == max_sample ); + + n = min_sample * 2; + CLAMP( n ); + assert( n == min_sample ); + + assert( blip_max_ratio <= time_unit ); + assert( blip_max_frame <= (fixed_t) -1 >> time_bits ); +} + +blip_t* blip_new( int size ) +{ + blip_t* m; + assert( size >= 0 ); + + m = (blip_t*) malloc( sizeof *m + (size + buf_extra) * sizeof (buf_t) ); + if ( m ) + { + m->factor = time_unit / blip_max_ratio; + m->size = size; + blip_clear( m ); + check_assumptions(); + } + return m; +} + +void blip_delete( blip_t* m ) +{ + if ( m != NULL ) + { + /* Clear fields in case user tries to use after freeing */ + memset( m, 0, sizeof *m ); + free( m ); + } +} + +void blip_set_rates( blip_t* m, double clock_rate, double sample_rate ) +{ + double factor = time_unit * sample_rate / clock_rate; + m->factor = (fixed_t) factor; + + /* Fails if clock_rate exceeds maximum, relative to sample_rate */ + assert( 0 <= factor - m->factor && factor - m->factor < 1 ); + + /* Avoid requiring math.h. Equivalent to + m->factor = (int) ceil( factor ) */ + if ( m->factor < factor ) + m->factor++; + + /* At this point, factor is most likely rounded up, but could still + have been rounded down in the floating-point calculation. */ +} + +void blip_clear( blip_t* m ) +{ + /* We could set offset to 0, factor/2, or factor-1. 0 is suitable if + factor is rounded up. factor-1 is suitable if factor is rounded down. + Since we don't know rounding direction, factor/2 accommodates either, + with the slight loss of showing an error in half the time. Since for + a 64-bit factor this is years, the halving isn't a problem. */ + + m->offset = m->factor / 2; + m->avail = 0; + m->integrator = 0; + memset( SAMPLES( m ), 0, (m->size + buf_extra) * sizeof (buf_t) ); +} + +int blip_clocks_needed( const blip_t* m, int samples ) +{ + fixed_t needed; + + /* Fails if buffer can't hold that many more samples */ + assert( samples >= 0 && m->avail + samples <= m->size ); + + needed = (fixed_t) samples * time_unit; + if ( needed < m->offset ) + return 0; + + return (needed - m->offset + m->factor - 1) / m->factor; +} + +void blip_end_frame( blip_t* m, unsigned t ) +{ + fixed_t off = t * m->factor + m->offset; + m->avail += off >> time_bits; + m->offset = off & (time_unit - 1); + + /* Fails if buffer size was exceeded */ + assert( m->avail <= m->size ); +} + +int blip_samples_avail( const blip_t* m ) +{ + return m->avail; +} + +static void remove_samples( blip_t* m, int count ) +{ + buf_t* buf = SAMPLES( m ); + int remain = m->avail + buf_extra - count; + m->avail -= count; + + memmove( &buf [0], &buf [count], remain * sizeof buf [0] ); + memset( &buf [remain], 0, count * sizeof buf [0] ); +} + +int blip_read_samples( blip_t* m, short out [], int count, int stereo ) +{ + assert( count >= 0 ); + + if ( count > m->avail ) + count = m->avail; + + if ( count ) + { + int const step = stereo ? 2 : 1; + buf_t const* in = SAMPLES( m ); + buf_t const* end = in + count; + int sum = m->integrator; + do + { + /* Eliminate fraction */ + int s = ARITH_SHIFT( sum, delta_bits ); + + sum += *in++; + + CLAMP( s ); + + *out = s; + out += step; + + /* High-pass filter */ + sum -= s << (delta_bits - bass_shift); + } + while ( in != end ); + m->integrator = sum; + + remove_samples( m, count ); + } + + return count; +} + +/* Things that didn't help performance on x86: + __attribute__((aligned(128))) + #define short int + restrict +*/ + +/* Sinc_Generator( 0.9, 0.55, 4.5 ) */ +static short const bl_step [phase_count + 1] [half_width] = +{ +{ 43, -115, 350, -488, 1136, -914, 5861,21022}, +{ 44, -118, 348, -473, 1076, -799, 5274,21001}, +{ 45, -121, 344, -454, 1011, -677, 4706,20936}, +{ 46, -122, 336, -431, 942, -549, 4156,20829}, +{ 47, -123, 327, -404, 868, -418, 3629,20679}, +{ 47, -122, 316, -375, 792, -285, 3124,20488}, +{ 47, -120, 303, -344, 714, -151, 2644,20256}, +{ 46, -117, 289, -310, 634, -17, 2188,19985}, +{ 46, -114, 273, -275, 553, 117, 1758,19675}, +{ 44, -108, 255, -237, 471, 247, 1356,19327}, +{ 43, -103, 237, -199, 390, 373, 981,18944}, +{ 42, -98, 218, -160, 310, 495, 633,18527}, +{ 40, -91, 198, -121, 231, 611, 314,18078}, +{ 38, -84, 178, -81, 153, 722, 22,17599}, +{ 36, -76, 157, -43, 80, 824, -241,17092}, +{ 34, -68, 135, -3, 8, 919, -476,16558}, +{ 32, -61, 115, 34, -60, 1006, -683,16001}, +{ 29, -52, 94, 70, -123, 1083, -862,15422}, +{ 27, -44, 73, 106, -184, 1152,-1015,14824}, +{ 25, -36, 53, 139, -239, 1211,-1142,14210}, +{ 22, -27, 34, 170, -290, 1261,-1244,13582}, +{ 20, -20, 16, 199, -335, 1301,-1322,12942}, +{ 18, -12, -3, 226, -375, 1331,-1376,12293}, +{ 15, -4, -19, 250, -410, 1351,-1408,11638}, +{ 13, 3, -35, 272, -439, 1361,-1419,10979}, +{ 11, 9, -49, 292, -464, 1362,-1410,10319}, +{ 9, 16, -63, 309, -483, 1354,-1383, 9660}, +{ 7, 22, -75, 322, -496, 1337,-1339, 9005}, +{ 6, 26, -85, 333, -504, 1312,-1280, 8355}, +{ 4, 31, -94, 341, -507, 1278,-1205, 7713}, +{ 3, 35, -102, 347, -506, 1238,-1119, 7082}, +{ 1, 40, -110, 350, -499, 1190,-1021, 6464}, +{ 0, 43, -115, 350, -488, 1136, -914, 5861} +}; + +/* Shifting by pre_shift allows calculation using unsigned int rather than +possibly-wider fixed_t. On 32-bit platforms, this is likely more efficient. +And by having pre_shift 32, a 32-bit platform can easily do the shift by +simply ignoring the low half. */ + +void blip_add_delta( blip_t* m, unsigned time, int delta ) +{ + unsigned fixed = (unsigned) ((time * m->factor + m->offset) >> pre_shift); + buf_t* out = SAMPLES( m ) + m->avail + (fixed >> frac_bits); + + int const phase_shift = frac_bits - phase_bits; + int phase = fixed >> phase_shift & (phase_count - 1); + short const* in = bl_step [phase]; + short const* rev = bl_step [phase_count - phase]; + + int interp = fixed >> (phase_shift - delta_bits) & (delta_unit - 1); + int delta2 = (delta * interp) >> delta_bits; + delta -= delta2; + + /* Fails if buffer size was exceeded */ + assert( out <= &SAMPLES( m ) [m->size + end_frame_extra] ); + + out [0] += in[0]*delta + in[half_width+0]*delta2; + out [1] += in[1]*delta + in[half_width+1]*delta2; + out [2] += in[2]*delta + in[half_width+2]*delta2; + out [3] += in[3]*delta + in[half_width+3]*delta2; + out [4] += in[4]*delta + in[half_width+4]*delta2; + out [5] += in[5]*delta + in[half_width+5]*delta2; + out [6] += in[6]*delta + in[half_width+6]*delta2; + out [7] += in[7]*delta + in[half_width+7]*delta2; + + in = rev; + out [ 8] += in[7]*delta + in[7-half_width]*delta2; + out [ 9] += in[6]*delta + in[6-half_width]*delta2; + out [10] += in[5]*delta + in[5-half_width]*delta2; + out [11] += in[4]*delta + in[4-half_width]*delta2; + out [12] += in[3]*delta + in[3-half_width]*delta2; + out [13] += in[2]*delta + in[2-half_width]*delta2; + out [14] += in[1]*delta + in[1-half_width]*delta2; + out [15] += in[0]*delta + in[0-half_width]*delta2; +} + +void blip_add_delta_fast( blip_t* m, unsigned time, int delta ) +{ + unsigned fixed = (unsigned) ((time * m->factor + m->offset) >> pre_shift); + buf_t* out = SAMPLES( m ) + m->avail + (fixed >> frac_bits); + + int interp = fixed >> (frac_bits - delta_bits) & (delta_unit - 1); + int delta2 = delta * interp; + + /* Fails if buffer size was exceeded */ + assert( out <= &SAMPLES( m ) [m->size + end_frame_extra] ); + + out [7] += delta * delta_unit - delta2; + out [8] += delta2; +}
A src/third-party/blip_buf/blip_buf.h

@@ -0,0 +1,72 @@

+/** \file +Sample buffer that resamples from input clock rate to output sample rate */ + +/* blip_buf 1.1.0 */ +#ifndef BLIP_BUF_H +#define BLIP_BUF_H + +#ifdef __cplusplus + extern "C" { +#endif + +/** First parameter of most functions is blip_t*, or const blip_t* if nothing +is changed. */ +typedef struct blip_t blip_t; + +/** Creates new buffer that can hold at most sample_count samples. Sets rates +so that there are blip_max_ratio clocks per sample. Returns pointer to new +buffer, or NULL if insufficient memory. */ +blip_t* blip_new( int sample_count ); + +/** Sets approximate input clock rate and output sample rate. For every +clock_rate input clocks, approximately sample_rate samples are generated. */ +void blip_set_rates( blip_t*, double clock_rate, double sample_rate ); + +enum { /** Maximum clock_rate/sample_rate ratio. For a given sample_rate, +clock_rate must not be greater than sample_rate*blip_max_ratio. */ +blip_max_ratio = 1 << 20 }; + +/** Clears entire buffer. Afterwards, blip_samples_avail() == 0. */ +void blip_clear( blip_t* ); + +/** Adds positive/negative delta into buffer at specified clock time. */ +void blip_add_delta( blip_t*, unsigned int clock_time, int delta ); + +/** Same as blip_add_delta(), but uses faster, lower-quality synthesis. */ +void blip_add_delta_fast( blip_t*, unsigned int clock_time, int delta ); + +/** Length of time frame, in clocks, needed to make sample_count additional +samples available. */ +int blip_clocks_needed( const blip_t*, int sample_count ); + +enum { /** Maximum number of samples that can be generated from one time frame. */ +blip_max_frame = 4000 }; + +/** Makes input clocks before clock_duration available for reading as output +samples. Also begins new time frame at clock_duration, so that clock time 0 in +the new time frame specifies the same clock as clock_duration in the old time +frame specified. Deltas can have been added slightly past clock_duration (up to +however many clocks there are in two output samples). */ +void blip_end_frame( blip_t*, unsigned int clock_duration ); + +/** Number of buffered samples available for reading. */ +int blip_samples_avail( const blip_t* ); + +/** Reads and removes at most 'count' samples and writes them to 'out'. If +'stereo' is true, writes output to every other element of 'out', allowing easy +interleaving of two buffers into a stereo sample stream. Outputs 16-bit signed +samples. Returns number of samples actually read. */ +int blip_read_samples( blip_t*, short out [], int count, int stereo ); + +/** Frees buffer. No effect if NULL is passed. */ +void blip_delete( blip_t* ); + + +/* Deprecated */ +typedef blip_t blip_buffer_t; + +#ifdef __cplusplus + } +#endif + +#endif
A src/third-party/blip_buf/license.txt

@@ -0,0 +1,504 @@

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