mirror of https://github.com/encounter/SDL.git
Added support for using libsamplerate to do audio resampling
This commit is contained in:
Sam Lantinga
parent
37f404fb87
commit
cbe44f7ff1
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@ -25,9 +25,14 @@
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#include "SDL_audio.h"
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#include "SDL_audio_c.h"
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#include "SDL_loadso.h"
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#include "SDL_assert.h"
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#include "../SDL_dataqueue.h"
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#ifdef HAVE_LIBSAMPLERATE
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#include "samplerate.h"
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#endif
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/* Effectively mix right and left channels into a single channel */
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static void SDLCALL
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@ -598,6 +603,9 @@ SDL_BuildAudioCVT(SDL_AudioCVT * cvt,
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return (cvt->needed);
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}
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typedef int (*SDL_ResampleAudioStreamFunc)(SDL_AudioStream *stream, const float *inbuf, const int inbuflen, float *outbuf, const int outbuflen);
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typedef void (*SDL_ResetAudioStreamResamplerFunc)(SDL_AudioStream *stream);
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typedef void (*SDL_CleanupAudioStreamResamplerFunc)(SDL_AudioStream *stream);
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struct SDL_AudioStream
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{
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@ -618,10 +626,202 @@ struct SDL_AudioStream
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int dst_rate;
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double rate_incr;
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Uint8 pre_resample_channels;
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int packetlen;
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void *resampler_state;
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SDL_ResampleAudioStreamFunc resampler_func;
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SDL_ResetAudioStreamResamplerFunc reset_resampler_func;
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SDL_CleanupAudioStreamResamplerFunc cleanup_resampler_func;
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};
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#ifdef HAVE_LIBSAMPLERATE
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typedef struct
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{
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void *SRC_lib;
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SRC_STATE* (*src_new)(int converter_type, int channels, int *error);
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int (*src_process)(SRC_STATE *state, SRC_DATA *data);
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int (*src_reset)(SRC_STATE *state);
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SRC_STATE* (*src_delete)(SRC_STATE *state);
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const char* (*src_strerror)(int error);
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SRC_STATE *SRC_state;
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} SDL_AudioStreamResamplerState_SRC;
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static SDL_bool
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LoadLibSampleRate(SDL_AudioStreamResamplerState_SRC *state)
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{
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#ifdef LIBSAMPLERATE_DYNAMIC
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state->SRC_lib = SDL_LoadObject(LIBSAMPLERATE_DYNAMIC);
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if (!state->SRC_lib) {
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return SDL_FALSE;
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}
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#endif
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state->src_new = (SRC_STATE* (*)(int converter_type, int channels, int *error))SDL_LoadFunction(state->SRC_lib, "src_new");
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state->src_process = (int (*)(SRC_STATE *state, SRC_DATA *data))SDL_LoadFunction(state->SRC_lib, "src_process");
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state->src_reset = (int(*)(SRC_STATE *state))SDL_LoadFunction(state->SRC_lib, "src_reset");
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state->src_delete = (SRC_STATE* (*)(SRC_STATE *state))SDL_LoadFunction(state->SRC_lib, "src_delete");
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state->src_strerror = (const char* (*)(int error))SDL_LoadFunction(state->SRC_lib, "src_strerror");
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if (!state->src_new || !state->src_process || !state->src_reset || !state->src_delete || !state->src_strerror) {
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return SDL_FALSE;
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}
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return SDL_TRUE;
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}
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static int
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SDL_ResampleAudioStream_SRC(SDL_AudioStream *stream, const float *inbuf, const int inbuflen, float *outbuf, const int outbuflen)
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{
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SDL_AudioStreamResamplerState_SRC *state = (SDL_AudioStreamResamplerState_SRC*)stream->resampler_state;
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SRC_DATA data;
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int result;
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data.data_in = inbuf;
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data.input_frames = inbuflen / ( sizeof(float) * stream->pre_resample_channels );
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data.input_frames_used = 0;
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data.data_out = outbuf;
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data.output_frames = outbuflen / (sizeof(float) * stream->pre_resample_channels);
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data.end_of_input = 0;
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data.src_ratio = stream->rate_incr;
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result = state->src_process(state->SRC_state, &data);
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if (result != 0) {
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SDL_SetError("src_process() failed: %s", state->src_strerror(result));
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return 0;
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}
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/* If this fails, we need to store them off somewhere */
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SDL_assert(data.input_frames_used == data.input_frames);
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return data.output_frames_gen * (sizeof(float) * stream->pre_resample_channels);
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}
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static void
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SDL_ResetAudioStreamResampler_SRC(SDL_AudioStream *stream)
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{
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SDL_AudioStreamResamplerState_SRC *state = (SDL_AudioStreamResamplerState_SRC*)stream->resampler_state;
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state->src_reset(state->SRC_state);
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}
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static void
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SDL_CleanupAudioStreamResampler_SRC(SDL_AudioStream *stream)
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{
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SDL_AudioStreamResamplerState_SRC *state = (SDL_AudioStreamResamplerState_SRC*)stream->resampler_state;
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if (state) {
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if (state->SRC_lib) {
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SDL_UnloadObject(state->SRC_lib);
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}
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state->src_delete(state->SRC_state);
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SDL_free(state);
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}
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stream->resampler_state = NULL;
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stream->resampler_func = NULL;
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stream->reset_resampler_func = NULL;
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stream->cleanup_resampler_func = NULL;
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}
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static SDL_bool
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SetupLibSampleRateResampling(SDL_AudioStream *stream)
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{
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int result;
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SDL_AudioStreamResamplerState_SRC *state = (SDL_AudioStreamResamplerState_SRC *)SDL_calloc(1, sizeof(*state));
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if (!state) {
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return SDL_FALSE;
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}
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if (!LoadLibSampleRate(state)) {
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SDL_free(state);
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return SDL_FALSE;
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}
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stream->resampler_state = state;
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stream->resampler_func = SDL_ResampleAudioStream_SRC;
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stream->reset_resampler_func = SDL_ResetAudioStreamResampler_SRC;
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stream->cleanup_resampler_func = SDL_CleanupAudioStreamResampler_SRC;
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state->SRC_state = state->src_new(SRC_SINC_FASTEST, stream->pre_resample_channels, &result);
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if (!state->SRC_state) {
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SDL_SetError("src_new() failed: %s", state->src_strerror(result));
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SDL_CleanupAudioStreamResampler_SRC(stream);
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return SDL_FALSE;
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}
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return SDL_TRUE;
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}
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#endif /* HAVE_LIBSAMPLERATE */
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typedef struct
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{
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SDL_bool resampler_seeded;
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float resampler_state[8];
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int packetlen;
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};
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} SDL_AudioStreamResamplerState;
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static int
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SDL_ResampleAudioStream(SDL_AudioStream *stream, const float *inbuf, const int inbuflen, float *outbuf, const int outbuflen)
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{
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/* !!! FIXME: this resampler sucks, but not much worse than our usual resampler. :) */ /* ... :( */
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SDL_AudioStreamResamplerState *state = (SDL_AudioStreamResamplerState*)stream->resampler_state;
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const int chans = (int)stream->pre_resample_channels;
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const int framelen = chans * sizeof(float);
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const int total = (inbuflen / framelen);
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const int finalpos = total - chans;
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const double src_incr = 1.0 / stream->rate_incr;
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double idx = 0.0;
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float *dst = outbuf;
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float last_sample[SDL_arraysize(state->resampler_state)];
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int consumed = 0;
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int i;
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SDL_assert(chans <= SDL_arraysize(last_sample));
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SDL_assert((inbuflen % framelen) == 0);
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if (!state->resampler_seeded) {
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for (i = 0; i < chans; i++) {
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state->resampler_state[i] = inbuf[i];
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}
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state->resampler_seeded = SDL_TRUE;
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}
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for (i = 0; i < chans; i++) {
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last_sample[i] = state->resampler_state[i];
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}
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while (consumed < total) {
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const int pos = ((int)idx) * chans;
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const float *src = &inbuf[(pos >= finalpos) ? finalpos : pos];
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SDL_assert(dst < (outbuf + (outbuflen / framelen)));
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for (i = 0; i < chans; i++) {
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const float val = *(src++);
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*(dst++) = (val + last_sample[i]) * 0.5f;
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last_sample[i] = val;
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}
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consumed = pos + chans;
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idx += src_incr;
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}
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for (i = 0; i < chans; i++) {
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state->resampler_state[i] = last_sample[i];
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}
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return (int)((dst - outbuf) * sizeof(float));
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}
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static void
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SDL_ResetAudioStreamResampler(SDL_AudioStream *stream)
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{
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SDL_AudioStreamResamplerState *state = (SDL_AudioStreamResamplerState*)stream->resampler_state;
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state->resampler_seeded = SDL_FALSE;
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}
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static void
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SDL_CleanupAudioStreamResampler(SDL_AudioStream *stream)
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{
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SDL_free(stream->resampler_state);
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}
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SDL_AudioStream *SDL_NewAudioStream(const SDL_AudioFormat src_format,
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const Uint8 src_channels,
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@ -661,84 +861,50 @@ SDL_AudioStream *SDL_NewAudioStream(const SDL_AudioFormat src_format,
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if (src_rate == dst_rate) {
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retval->cvt_before_resampling.needed = SDL_FALSE;
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retval->cvt_before_resampling.len_mult = 1;
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if (SDL_BuildAudioCVT(&retval->cvt_after_resampling, src_format, src_channels, dst_rate, dst_format, dst_channels, dst_rate) == -1) {
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SDL_free(retval);
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if (SDL_BuildAudioCVT(&retval->cvt_after_resampling, src_format, src_channels, dst_rate, dst_format, dst_channels, dst_rate) < 0) {
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SDL_FreeAudioStream(retval);
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return NULL; /* SDL_BuildAudioCVT should have called SDL_SetError. */
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}
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} else {
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/* Don't resample at first. Just get us to Float32 format. */
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/* !!! FIXME: convert to int32 on devices without hardware float. */
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if (SDL_BuildAudioCVT(&retval->cvt_before_resampling, src_format, src_channels, src_rate, AUDIO_F32SYS, pre_resample_channels, src_rate) == -1) {
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SDL_free(retval);
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if (SDL_BuildAudioCVT(&retval->cvt_before_resampling, src_format, src_channels, src_rate, AUDIO_F32SYS, pre_resample_channels, src_rate) < 0) {
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SDL_FreeAudioStream(retval);
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return NULL; /* SDL_BuildAudioCVT should have called SDL_SetError. */
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}
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#ifdef HAVE_LIBSAMPLERATE
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SetupLibSampleRateResampling(retval);
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#endif
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if (!retval->resampler_func) {
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retval->resampler_state = SDL_calloc(1, sizeof(SDL_AudioStreamResamplerState));
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if (!retval->resampler_state) {
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SDL_FreeAudioStream(retval);
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SDL_OutOfMemory();
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return NULL;
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}
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retval->resampler_func = SDL_ResampleAudioStream;
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retval->reset_resampler_func = SDL_ResetAudioStreamResampler;
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retval->cleanup_resampler_func = SDL_CleanupAudioStreamResampler;
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}
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/* Convert us to the final format after resampling. */
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if (SDL_BuildAudioCVT(&retval->cvt_after_resampling, AUDIO_F32SYS, pre_resample_channels, dst_rate, dst_format, dst_channels, dst_rate) == -1) {
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SDL_free(retval);
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if (SDL_BuildAudioCVT(&retval->cvt_after_resampling, AUDIO_F32SYS, pre_resample_channels, dst_rate, dst_format, dst_channels, dst_rate) < 0) {
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SDL_FreeAudioStream(retval);
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return NULL; /* SDL_BuildAudioCVT should have called SDL_SetError. */
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}
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}
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retval->queue = SDL_NewDataQueue(packetlen, packetlen * 2);
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if (!retval->queue) {
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SDL_free(retval);
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SDL_FreeAudioStream(retval);
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return NULL; /* SDL_NewDataQueue should have called SDL_SetError. */
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}
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return retval;
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}
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static int
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ResampleAudioStream(SDL_AudioStream *stream, const float *inbuf, const int inbuflen, float *outbuf, const int outbuflen)
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{
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/* !!! FIXME: this resampler sucks, but not much worse than our usual resampler. :) */ /* ... :( */
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const int chans = (int) stream->pre_resample_channels;
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const int framelen = chans * sizeof (float);
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const int total = (inbuflen / framelen);
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const int finalpos = total - chans;
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const double src_incr = 1.0 / stream->rate_incr;
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double idx = 0.0;
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float *dst = outbuf;
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float last_sample[SDL_arraysize(stream->resampler_state)];
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int consumed = 0;
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int i;
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SDL_assert(chans <= SDL_arraysize(last_sample));
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SDL_assert((inbuflen % framelen) == 0);
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if (!stream->resampler_seeded) {
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for (i = 0; i < chans; i++) {
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stream->resampler_state[i] = inbuf[i];
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}
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stream->resampler_seeded = SDL_TRUE;
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}
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for (i = 0; i < chans; i++) {
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last_sample[i] = stream->resampler_state[i];
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}
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while (consumed < total) {
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const int pos = ((int) idx) * chans;
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const float *src = &inbuf[(pos >= finalpos) ? finalpos : pos];
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SDL_assert(dst < (outbuf + (outbuflen / framelen)));
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for (i = 0; i < chans; i++) {
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const float val = *(src++);
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*(dst++) = (val + last_sample[i]) * 0.5f;
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last_sample[i] = val;
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}
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consumed = pos + chans;
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idx += src_incr;
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}
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for (i = 0; i < chans; i++) {
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stream->resampler_state[i] = last_sample[i];
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}
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return (int) ((dst - outbuf) * sizeof (float));
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}
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static Uint8 *
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EnsureBufferSize(Uint8 **buf, int *len, const int newlen)
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{
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@ -791,7 +957,7 @@ SDL_AudioStreamPut(SDL_AudioStream *stream, const void *buf, const Uint32 _bufle
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if (workbuf == NULL) {
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return -1; /* probably out of memory. */
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}
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buflen = ResampleAudioStream(stream, (float *) buf, buflen, workbuf, workbuflen);
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buflen = stream->resampler_func(stream, (float *) buf, buflen, workbuf, workbuflen);
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buf = workbuf;
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}
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@ -832,7 +998,7 @@ SDL_AudioStreamClear(SDL_AudioStream *stream)
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SDL_InvalidParamError("stream");
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} else {
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SDL_ClearDataQueue(stream->queue, stream->packetlen * 2);
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stream->resampler_seeded = SDL_FALSE;
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stream->reset_resampler_func(stream);
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}
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}
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@ -866,6 +1032,9 @@ void
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SDL_FreeAudioStream(SDL_AudioStream *stream)
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{
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if (stream) {
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if (stream->cleanup_resampler_func) {
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stream->cleanup_resampler_func(stream);
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}
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SDL_FreeDataQueue(stream->queue);
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SDL_free(stream->work_buffer);
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SDL_free(stream->resample_buffer);
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