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audio: libsamplerate loading now happens once at init time.

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This commit is contained in:
Ryan C. Gordon 2017-01-08 14:18:03 -05:00
parent 98cc9d10d3
commit 19e937fc2e
3 changed files with 100 additions and 59 deletions
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74
src/audio/SDL_audio.c
View file

@ -107,6 +107,72 @@ static const AudioBootStrap *const bootstrap[] = {
NULL
};
#ifdef HAVE_LIBSAMPLERATE_H
#ifdef SDL_LIBSAMPLERATE_DYNAMIC
static void *SRC_lib = NULL;
#endif
SDL_bool SRC_available = SDL_FALSE;
SRC_STATE* (*SRC_src_new)(int converter_type, int channels, int *error) = NULL;
int (*SRC_src_process)(SRC_STATE *state, SRC_DATA *data) = NULL;
int (*SRC_src_reset)(SRC_STATE *state) = NULL;
SRC_STATE* (*SRC_src_delete)(SRC_STATE *state) = NULL;
const char* (*SRC_src_strerror)(int error) = NULL;
static SDL_bool
LoadLibSampleRate(void)
{
SRC_available = SDL_FALSE;
if (!SDL_GetHintBoolean("SDL_AUDIO_ALLOW_LIBRESAMPLE", SDL_TRUE)) {
return SDL_FALSE;
}
#ifdef SDL_LIBSAMPLERATE_DYNAMIC
SDL_assert(SRC_lib == NULL);
SRC_lib = SDL_LoadObject(SDL_LIBSAMPLERATE_DYNAMIC);
if (!SRC_lib) {
return SDL_FALSE;
}
#endif
SRC_src_new = (SRC_STATE* (*)(int converter_type, int channels, int *error))SDL_LoadFunction(state->SRC_lib, "src_new");
SRC_src_process = (int (*)(SRC_STATE *state, SRC_DATA *data))SDL_LoadFunction(state->SRC_lib, "src_process");
SRC_src_reset = (int(*)(SRC_STATE *state))SDL_LoadFunction(state->SRC_lib, "src_reset");
SRC_src_delete = (SRC_STATE* (*)(SRC_STATE *state))SDL_LoadFunction(state->SRC_lib, "src_delete");
SRC_src_strerror = (const char* (*)(int error))SDL_LoadFunction(state->SRC_lib, "src_strerror");
if (!SRC_src_new || !SRC_src_process || !SRC_src_reset || !SRC_src_delete || !SRC_src_strerror) {
#ifdef SDL_LIBSAMPLERATE_DYNAMIC
SDL_UnloadObject(SRC_lib);
SRC_lib = NULL;
#endif
return SDL_FALSE;
}
SRC_available = SDL_TRUE;
return SDL_TRUE;
}
static void
UnloadLibSampleRate(void)
{
#ifdef SDL_LIBSAMPLERATE_DYNAMIC
if (SRC_lib != NULL) {
SDL_UnloadObject(SRC_lib);
}
SRC_lib = NULL;
#endif
SRC_available = SDL_FALSE;
SRC_src_new = NULL;
SRC_src_process = NULL;
SRC_src_reset = NULL;
SRC_src_delete = NULL;
SRC_src_strerror = NULL;
}
#endif
static SDL_AudioDevice *
get_audio_device(SDL_AudioDeviceID id)
{
@ -828,6 +894,10 @@ SDL_AudioInit(const char *driver_name)
/* Make sure we have a list of devices available at startup. */
current_audio.impl.DetectDevices();
#ifdef HAVE_LIBSAMPLERATE_H
LoadLibSampleRate();
#endif
return 0;
}
@ -1427,6 +1497,10 @@ SDL_AudioQuit(void)
SDL_zero(current_audio);
SDL_zero(open_devices);
#ifdef HAVE_LIBSAMPLERATE_H
UnloadLibSampleRate();
#endif
}
#define NUM_FORMATS 10

10
src/audio/SDL_audio_c.h
View file

@ -36,6 +36,16 @@
/* Functions and variables exported from SDL_audio.c for SDL_sysaudio.c */
#ifdef HAVE_LIBSAMPLERATE_H
extern SDL_bool SRC_available;
typedef struct SRC_STATE SRC_STATE;
extern SRC_STATE* (*SRC_src_new)(int converter_type, int channels, int *error);
extern int (*SRC_src_process)(SRC_STATE *state, SRC_DATA *data);
extern int (*SRC_src_reset)(SRC_STATE *state);
extern SRC_STATE* (*SRC_src_delete)(SRC_STATE *state);
extern const char* (*SRC_src_strerror)(int error);
#endif
/* Functions to get a list of "close" audio formats */
extern SDL_AudioFormat SDL_FirstAudioFormat(SDL_AudioFormat format);
extern SDL_AudioFormat SDL_NextAudioFormat(void);

75
src/audio/SDL_audiocvt.c
View file

@ -634,45 +634,10 @@ struct SDL_AudioStream
};
#ifdef HAVE_LIBSAMPLERATE_H
typedef struct
{
void *SRC_lib;
SRC_STATE* (*src_new)(int converter_type, int channels, int *error);
int (*src_process)(SRC_STATE *state, SRC_DATA *data);
int (*src_reset)(SRC_STATE *state);
SRC_STATE* (*src_delete)(SRC_STATE *state);
const char* (*src_strerror)(int error);
SRC_STATE *SRC_state;
} SDL_AudioStreamResamplerState_SRC;
static SDL_bool
LoadLibSampleRate(SDL_AudioStreamResamplerState_SRC *state)
{
#ifdef SDL_LIBSAMPLERATE_DYNAMIC
state->SRC_lib = SDL_LoadObject(SDL_LIBSAMPLERATE_DYNAMIC);
if (!state->SRC_lib) {
return SDL_FALSE;
}
#endif
state->src_new = (SRC_STATE* (*)(int converter_type, int channels, int *error))SDL_LoadFunction(state->SRC_lib, "src_new");
state->src_process = (int (*)(SRC_STATE *state, SRC_DATA *data))SDL_LoadFunction(state->SRC_lib, "src_process");
state->src_reset = (int(*)(SRC_STATE *state))SDL_LoadFunction(state->SRC_lib, "src_reset");
state->src_delete = (SRC_STATE* (*)(SRC_STATE *state))SDL_LoadFunction(state->SRC_lib, "src_delete");
state->src_strerror = (const char* (*)(int error))SDL_LoadFunction(state->SRC_lib, "src_strerror");
if (!state->src_new || !state->src_process || !state->src_reset || !state->src_delete || !state->src_strerror) {
return SDL_FALSE;
}
return SDL_TRUE;
}
static int
SDL_ResampleAudioStream_SRC(SDL_AudioStream *stream, const float *inbuf, const int inbuflen, float *outbuf, const int outbuflen)
{
SDL_AudioStreamResamplerState_SRC *state = (SDL_AudioStreamResamplerState_SRC*)stream->resampler_state;
SRC_STATE *state = (SRC_STATE *)stream->resampler_state;
SRC_DATA data;
int result;
@ -686,9 +651,9 @@ SDL_ResampleAudioStream_SRC(SDL_AudioStream *stream, const float *inbuf, const i
data.end_of_input = 0;
data.src_ratio = stream->rate_incr;
result = state->src_process(state->SRC_state, &data);
result = SRC_src_process(state, &data);
if (result != 0) {
SDL_SetError("src_process() failed: %s", state->src_strerror(result));
SDL_SetError("src_process() failed: %s", SRC_src_strerror(result));
return 0;
}
@ -701,20 +666,15 @@ SDL_ResampleAudioStream_SRC(SDL_AudioStream *stream, const float *inbuf, const i
static void
SDL_ResetAudioStreamResampler_SRC(SDL_AudioStream *stream)
{
SDL_AudioStreamResamplerState_SRC *state = (SDL_AudioStreamResamplerState_SRC*)stream->resampler_state;
state->src_reset(state->SRC_state);
SRC_src_reset((SRC_STATE *)stream->resampler_state);
}
static void
SDL_CleanupAudioStreamResampler_SRC(SDL_AudioStream *stream)
{
SDL_AudioStreamResamplerState_SRC *state = (SDL_AudioStreamResamplerState_SRC*)stream->resampler_state;
SRC_STATE *state = (SRC_STATE *)stream->resampler_state;
if (state) {
if (state->SRC_lib) {
SDL_UnloadObject(state->SRC_lib);
}
state->src_delete(state->SRC_state);
SDL_free(state);
SRC_src_delete(state);
}
stream->resampler_state = NULL;
@ -726,15 +686,18 @@ SDL_CleanupAudioStreamResampler_SRC(SDL_AudioStream *stream)
static SDL_bool
SetupLibSampleRateResampling(SDL_AudioStream *stream)
{
int result;
int result = 0;
SRC_STATE *state = NULL;
SDL_AudioStreamResamplerState_SRC *state = (SDL_AudioStreamResamplerState_SRC *)SDL_calloc(1, sizeof(*state));
if (!state) {
return SDL_FALSE;
if (SRC_available) {
state = SRC_src_new(SRC_SINC_FASTEST, stream->pre_resample_channels, &result);
if (!state) {
SDL_SetError("src_new() failed: %s", SRC_src_strerror(result));
}
}
if (!LoadLibSampleRate(state)) {
SDL_free(state);
if (!state) {
SDL_CleanupAudioStreamResampler_SRC(stream);
return SDL_FALSE;
}
@ -743,17 +706,11 @@ SetupLibSampleRateResampling(SDL_AudioStream *stream)
stream->reset_resampler_func = SDL_ResetAudioStreamResampler_SRC;
stream->cleanup_resampler_func = SDL_CleanupAudioStreamResampler_SRC;
state->SRC_state = state->src_new(SRC_SINC_FASTEST, stream->pre_resample_channels, &result);
if (!state->SRC_state) {
SDL_SetError("src_new() failed: %s", state->src_strerror(result));
SDL_CleanupAudioStreamResampler_SRC(stream);
return SDL_FALSE;
}
return SDL_TRUE;
}
#endif /* HAVE_LIBSAMPLERATE_H */
typedef struct
{
SDL_bool resampler_seeded;