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Use libsamplerate to handle pitch for MojoAL

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This commit is contained in:
Benau 2021-05-19 11:29:48 +08:00
parent be7b46f7d9
commit 33459e852c
2 changed files with 78 additions and 65 deletions
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11
CMakeLists.txt
View File

@ -370,6 +370,15 @@ endif()
if(NOT SERVER_ONLY)
# OpenAL
if (USE_MOJOAL)
find_library(LIBSAMPLERATE_LIBRARY NAMES samplerate libsamplerate)
find_path(LIBSAMPLERATE_INCLUDEDIR NAMES samplerate.h PATHS)
if (NOT LIBSAMPLERATE_LIBRARY OR NOT LIBSAMPLERATE_INCLUDEDIR)
message(FATAL_ERROR "libsamplerate not found. "
"libsamplerate is required to handle pitch for MojoAL.")
else()
include_directories("${LIBSAMPLERATE_INCLUDEDIR}")
MESSAGE(STATUS "Use system libsamplerate: ${LIBSAMPLERATE_LIBRARY}")
endif()
add_definitions(-DAL_LIBTYPE_STATIC)
add_subdirectory("${PROJECT_SOURCE_DIR}/lib/mojoal")
include_directories(BEFORE "${PROJECT_SOURCE_DIR}/lib/mojoal")
@ -642,7 +651,7 @@ endif()
if(NOT SERVER_ONLY)
if (USE_MOJOAL)
target_link_libraries(supertuxkart mojoal)
target_link_libraries(supertuxkart mojoal ${LIBSAMPLERATE_LIBRARY})
else()
target_link_libraries(supertuxkart ${OPENAL_LIBRARY})
endif()

132
lib/mojoal/mojoal.c
View File

@ -22,6 +22,7 @@
#include "AL/al.h"
#include "AL/alc.h"
#include "SDL.h"
#include "samplerate.h"
#ifdef __SSE__ /* if you are on x86 or x86-64, we assume you have SSE1 by now. */
#define NEED_SCALAR_FALLBACK 0
@ -444,7 +445,7 @@ SIMDALIGNEDSTRUCT ALsource
ALfloat cone_outer_angle;
ALfloat cone_outer_gain;
ALbuffer *buffer;
SDL_AudioStream *stream; /* for resampling. */
SRC_STATE *src_state; /* for resampling. */
BufferQueue buffer_queue;
BufferQueue buffer_queue_processed;
ALsizei offset; /* offset in bytes for converted stream! */
@ -1171,33 +1172,58 @@ static ALboolean mix_source_buffer(ALCcontext *ctx, ALsource *src, BufferQueueIt
const int bufferframesize = (int) (buffer->channels * sizeof (float));
const int deviceframesize = ctx->device->framesize;
const int framesneeded = *len / deviceframesize;
const float pitch = src->pitch;
SDL_assert(src->offset < buffer->len);
if (src->stream) { /* resampling? */
int mixframes, mixlen, remainingmixframes;
while ( (((mixlen = SDL_AudioStreamAvailable(src->stream)) / bufferframesize) < framesneeded) && (src->offset < buffer->len) ) {
const int framesput = (buffer->len - src->offset) / bufferframesize;
const int bytesput = SDL_min(framesput, 1024) * bufferframesize;
FIXME("dynamically adjust frames here?"); /* we hardcode 1024 samples when opening the audio device, too. */
SDL_AudioStreamPut(src->stream, data, bytesput);
src->offset += bytesput;
data += bytesput / sizeof (float);
}
mixframes = SDL_min(mixlen / bufferframesize, framesneeded);
remainingmixframes = mixframes;
while (remainingmixframes > 0) {
float mixbuf[256];
const int mixbuflen = sizeof (mixbuf);
const int mixbufframes = mixbuflen / bufferframesize;
const int getframes = SDL_min(remainingmixframes, mixbufframes);
SDL_AudioStreamGet(src->stream, mixbuf, getframes * bufferframesize);
mix_buffer(buffer, src->panning, mixbuf, *stream, getframes);
*len -= getframes * deviceframesize;
*stream += getframes * ctx->device->channels;
remainingmixframes -= getframes;
}
/*
SRC_SINC_BEST_QUALITY
SRC_SINC_MEDIUM_QUALITY
SRC_SINC_FASTEST
SRC_ZERO_ORDER_HOLD
SRC_LINEAR
*/
const int converter = SRC_LINEAR;
int src_error = 0;
if ((pitch != 1.0f || ctx->device->frequency != buffer->frequency) && src->src_state == NULL) {
src->src_state = src_new(converter, buffer->channels, &src_error);
if (src->src_state == NULL)
fprintf(stderr, "src_new error: %s\n", src_strerror(src_error));
}
else if (pitch == 1.0f && ctx->device->frequency == buffer->frequency && src->src_state != NULL) {
src_delete(src->src_state);
src->src_state = NULL;
}
else if (src->src_state != NULL && src_get_channels(src->src_state) != buffer->channels) {
src_delete(src->src_state);
src->src_state = src_new(converter, buffer->channels, &src_error);
if (src->src_state == NULL)
fprintf(stderr, "src_new error: %s\n", src_strerror(src_error));
}
if (src->src_state) { /* resampling? */
int used_frame = 0;
do {
SRC_DATA src_data = {};
src_data.input_frames = (buffer->len - src->offset) / buffer->channels / sizeof (float);
src_data.output_frames = framesneeded - used_frame;
if (src_data.output_frames <= 0 || src->offset >= buffer->len)
break;
src_data.data_in = data;
float *data_out = (float *) alloca((framesneeded - used_frame) * buffer->channels * sizeof (float));
src_data.data_out = data_out;
src_data.src_ratio = (double) ctx->device->frequency / (double) buffer->frequency / pitch;
int ret = src_process(src->src_state, &src_data);
if (ret != 0) {
fprintf(stderr, "src_process error: %s\n", src_strerror(ret));
break;
}
mix_buffer(buffer, src->panning, src_data.data_out, *stream, src_data.output_frames_gen);
*len -= src_data.output_frames_gen * deviceframesize;
*stream += src_data.output_frames_gen * ctx->device->channels;
src->offset += src_data.input_frames_used * buffer->channels * sizeof (float);
used_frame += src_data.output_frames_gen;
data += src_data.input_frames_used * buffer->channels;
} while (*len > 0);
} else {
const int framesavail = (buffer->len - src->offset) / bufferframesize;
const int mixframes = SDL_min(framesneeded, framesavail);
@ -2035,7 +2061,8 @@ static void _alcDestroyContext(ALCcontext *ctx)
continue;
}
SDL_FreeAudioStream(src->stream);
if (src->src_state)
src_delete(src->src_state);
source_release_buffer_queue(ctx, src);
if (--sb->used == 0) {
break;
@ -3415,9 +3442,9 @@ static void _alDeleteSources(const ALsizei n, const ALuint *names)
(void) SDL_AtomicDecRef(&source->buffer->refcount);
source->buffer = NULL;
}
if (source->stream) {
SDL_FreeAudioStream(source->stream);
source->stream = NULL;
if (source->src_state) {
src_delete(source->src_state);
source->src_state = NULL;
}
block->used--;
}
@ -3448,7 +3475,16 @@ static void _alSourcefv(const ALuint name, const ALenum param, const ALfloat *va
case AL_REFERENCE_DISTANCE: src->reference_distance = *values; break;
case AL_ROLLOFF_FACTOR: src->rolloff_factor = *values; break;
case AL_MAX_DISTANCE: src->max_distance = *values; break;
case AL_PITCH: src->pitch = *values; break;
case AL_PITCH:
{
float pitch = *values;
if (pitch < 0.5f)
pitch = 0.5f;
else if (pitch > 2.0f)
pitch = 2.0f;
src->pitch = pitch;
break;
}
case AL_CONE_INNER_ANGLE: src->cone_inner_angle = *values; break;
case AL_CONE_OUTER_ANGLE: src->cone_outer_angle = *values; break;
case AL_CONE_OUTER_GAIN: src->cone_outer_gain = *values; break;
@ -3517,18 +3553,6 @@ static void set_source_static_buffer(ALCcontext *ctx, ALsource *src, const ALuin
set_al_error(ctx, AL_INVALID_VALUE);
} else {
const ALboolean must_lock = SDL_AtomicGet(&src->mixer_accessible) ? AL_TRUE : AL_FALSE;
SDL_AudioStream *stream = NULL;
SDL_AudioStream *freestream = NULL;
/* We only use the stream for resampling, not for channel conversion. */
FIXME("keep the existing stream if formats match?");
if (buffer && (ctx->device->frequency != buffer->frequency)) {
stream = SDL_NewAudioStream(AUDIO_F32SYS, buffer->channels, buffer->frequency, AUDIO_F32SYS, buffer->channels, ctx->device->frequency);
if (!stream) {
set_al_error(ctx, AL_OUT_OF_MEMORY);
return;
}
FIXME("need a way to prealloc space in the stream, so the mixer doesn't have to malloc");
}
/* this can happen if you alSource(AL_BUFFER) while the exact source is in the middle of mixing */
FIXME("Double-check this lock; we shouldn't be able to reach this if the source is playing.");
@ -3552,18 +3576,13 @@ static void set_source_static_buffer(ALCcontext *ctx, ALsource *src, const ALuin
source_release_buffer_queue(ctx, src);
if (src->stream != stream) {
freestream = src->stream; /* free this after unlocking. */
src->stream = stream;
if (src->src_state) {
src_reset(src->src_state);
}
if (must_lock) {
SDL_UnlockMutex(ctx->source_lock);
}
if (freestream) {
SDL_FreeAudioStream(freestream);
}
}
}
}
@ -4052,7 +4071,6 @@ static void _alSourceQueueBuffers(const ALuint name, const ALsizei nb, const ALu
ALint queue_channels = 0;
ALsizei queue_frequency = 0;
ALboolean failed = AL_FALSE;
SDL_AudioStream *stream = NULL;
if (!src) {
return;
@ -4129,16 +4147,6 @@ static void _alSourceQueueBuffers(const ALuint name, const ALsizei nb, const ALu
if (!src->queue_frequency) {
SDL_assert(!src->queue_channels);
SDL_assert(!src->stream);
/* We only use the stream for resampling, not for channel conversion. */
if (ctx->device->frequency != queue_frequency) {
stream = SDL_NewAudioStream(AUDIO_F32SYS, queue_channels, queue_frequency, AUDIO_F32SYS, queue_channels, ctx->device->frequency);
if (!stream) {
set_al_error(ctx, AL_OUT_OF_MEMORY);
failed = AL_TRUE;
}
FIXME("need a way to prealloc space in the stream, so the mixer doesn't have to malloc");
}
}
if (failed) {
@ -4155,9 +4163,6 @@ static void _alSourceQueueBuffers(const ALuint name, const ALsizei nb, const ALu
queueend->next = ctx->device->playback.buffer_queue_pool;
ctx->device->playback.buffer_queue_pool = queue;
}
if (stream) {
SDL_FreeAudioStream(stream);
}
return;
}
@ -4169,7 +4174,6 @@ static void _alSourceQueueBuffers(const ALuint name, const ALsizei nb, const ALu
if (!src->queue_channels) {
src->queue_channels = queue_channels;
src->queue_frequency = queue_frequency;
src->stream = stream;
}
/* so we're going to put these on a linked list called just_queued,