diff options
Diffstat (limited to 'Source/3rdParty/SDL2/src/audio/SDL_audiocvt.c')
| -rw-r--r-- | Source/3rdParty/SDL2/src/audio/SDL_audiocvt.c | 1673 |
1 files changed, 0 insertions, 1673 deletions
diff --git a/Source/3rdParty/SDL2/src/audio/SDL_audiocvt.c b/Source/3rdParty/SDL2/src/audio/SDL_audiocvt.c deleted file mode 100644 index ee0ba32..0000000 --- a/Source/3rdParty/SDL2/src/audio/SDL_audiocvt.c +++ /dev/null @@ -1,1673 +0,0 @@ -/* - Simple DirectMedia Layer - Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org> - - This software is provided 'as-is', without any express or implied - warranty. In no event will the authors be held liable for any damages - arising from the use of this software. - - Permission is granted to anyone to use this software for any purpose, - including commercial applications, and to alter it and redistribute it - freely, subject to the following restrictions: - - 1. The origin of this software must not be misrepresented; you must not - claim that you wrote the original software. If you use this software - in a product, an acknowledgment in the product documentation would be - appreciated but is not required. - 2. Altered source versions must be plainly marked as such, and must not be - misrepresented as being the original software. - 3. This notice may not be removed or altered from any source distribution. -*/ -#include "../SDL_internal.h" - -/* Functions for audio drivers to perform runtime conversion of audio format */ - -/* FIXME: Channel weights when converting from more channels to fewer may need to be adjusted, see https://msdn.microsoft.com/en-us/library/windows/desktop/ff819070(v=vs.85).aspx -*/ - -#include "SDL.h" -#include "SDL_audio.h" -#include "SDL_audio_c.h" - -#include "SDL_loadso.h" -#include "SDL_assert.h" -#include "../SDL_dataqueue.h" -#include "SDL_cpuinfo.h" - -#define DEBUG_AUDIOSTREAM 0 - -#ifdef __SSE3__ -#define HAVE_SSE3_INTRINSICS 1 -#endif - -#if HAVE_SSE3_INTRINSICS -/* Convert from stereo to mono. Average left and right. */ -static void SDLCALL -SDL_ConvertStereoToMono_SSE3(SDL_AudioCVT * cvt, SDL_AudioFormat format) -{ - float *dst = (float *) cvt->buf; - const float *src = dst; - int i = cvt->len_cvt / 8; - - LOG_DEBUG_CONVERT("stereo", "mono (using SSE3)"); - SDL_assert(format == AUDIO_F32SYS); - - /* We can only do this if dst is aligned to 16 bytes; since src is the - same pointer and it moves by 2, it can't be forcibly aligned. */ - if ((((size_t) dst) & 15) == 0) { - /* Aligned! Do SSE blocks as long as we have 16 bytes available. */ - const __m128 divby2 = _mm_set1_ps(0.5f); - while (i >= 4) { /* 4 * float32 */ - _mm_store_ps(dst, _mm_mul_ps(_mm_hadd_ps(_mm_load_ps(src), _mm_load_ps(src+4)), divby2)); - i -= 4; src += 8; dst += 4; - } - } - - /* Finish off any leftovers with scalar operations. */ - while (i) { - *dst = (src[0] + src[1]) * 0.5f; - dst++; i--; src += 2; - } - - cvt->len_cvt /= 2; - if (cvt->filters[++cvt->filter_index]) { - cvt->filters[cvt->filter_index] (cvt, format); - } -} -#endif - -/* Convert from stereo to mono. Average left and right. */ -static void SDLCALL -SDL_ConvertStereoToMono(SDL_AudioCVT * cvt, SDL_AudioFormat format) -{ - float *dst = (float *) cvt->buf; - const float *src = dst; - int i; - - LOG_DEBUG_CONVERT("stereo", "mono"); - SDL_assert(format == AUDIO_F32SYS); - - for (i = cvt->len_cvt / 8; i; --i, src += 2) { - *(dst++) = (src[0] + src[1]) * 0.5f; - } - - cvt->len_cvt /= 2; - if (cvt->filters[++cvt->filter_index]) { - cvt->filters[cvt->filter_index] (cvt, format); - } -} - - -/* Convert from 5.1 to stereo. Average left and right, distribute center, discard LFE. */ -static void SDLCALL -SDL_Convert51ToStereo(SDL_AudioCVT * cvt, SDL_AudioFormat format) -{ - float *dst = (float *) cvt->buf; - const float *src = dst; - int i; - - LOG_DEBUG_CONVERT("5.1", "stereo"); - SDL_assert(format == AUDIO_F32SYS); - - /* SDL's 5.1 layout: FL+FR+FC+LFE+BL+BR */ - for (i = cvt->len_cvt / (sizeof (float) * 6); i; --i, src += 6, dst += 2) { - const float front_center_distributed = src[2] * 0.5f; - dst[0] = (src[0] + front_center_distributed + src[4]) / 2.5f; /* left */ - dst[1] = (src[1] + front_center_distributed + src[5]) / 2.5f; /* right */ - } - - cvt->len_cvt /= 3; - if (cvt->filters[++cvt->filter_index]) { - cvt->filters[cvt->filter_index] (cvt, format); - } -} - - -/* Convert from quad to stereo. Average left and right. */ -static void SDLCALL -SDL_ConvertQuadToStereo(SDL_AudioCVT * cvt, SDL_AudioFormat format) -{ - float *dst = (float *) cvt->buf; - const float *src = dst; - int i; - - LOG_DEBUG_CONVERT("quad", "stereo"); - SDL_assert(format == AUDIO_F32SYS); - - for (i = cvt->len_cvt / (sizeof (float) * 4); i; --i, src += 4, dst += 2) { - dst[0] = (src[0] + src[2]) * 0.5f; /* left */ - dst[1] = (src[1] + src[3]) * 0.5f; /* right */ - } - - cvt->len_cvt /= 2; - if (cvt->filters[++cvt->filter_index]) { - cvt->filters[cvt->filter_index] (cvt, format); - } -} - - -/* Convert from 7.1 to 5.1. Distribute sides across front and back. */ -static void SDLCALL -SDL_Convert71To51(SDL_AudioCVT * cvt, SDL_AudioFormat format) -{ - float *dst = (float *) cvt->buf; - const float *src = dst; - int i; - - LOG_DEBUG_CONVERT("7.1", "5.1"); - SDL_assert(format == AUDIO_F32SYS); - - for (i = cvt->len_cvt / (sizeof (float) * 8); i; --i, src += 8, dst += 6) { - const float surround_left_distributed = src[6] * 0.5f; - const float surround_right_distributed = src[7] * 0.5f; - dst[0] = (src[0] + surround_left_distributed) / 1.5f; /* FL */ - dst[1] = (src[1] + surround_right_distributed) / 1.5f; /* FR */ - dst[2] = src[2] / 1.5f; /* CC */ - dst[3] = src[3] / 1.5f; /* LFE */ - dst[4] = (src[4] + surround_left_distributed) / 1.5f; /* BL */ - dst[5] = (src[5] + surround_right_distributed) / 1.5f; /* BR */ - } - - cvt->len_cvt /= 8; - cvt->len_cvt *= 6; - if (cvt->filters[++cvt->filter_index]) { - cvt->filters[cvt->filter_index] (cvt, format); - } -} - - -/* Convert from 5.1 to quad. Distribute center across front, discard LFE. */ -static void SDLCALL -SDL_Convert51ToQuad(SDL_AudioCVT * cvt, SDL_AudioFormat format) -{ - float *dst = (float *) cvt->buf; - const float *src = dst; - int i; - - LOG_DEBUG_CONVERT("5.1", "quad"); - SDL_assert(format == AUDIO_F32SYS); - - /* SDL's 4.0 layout: FL+FR+BL+BR */ - /* SDL's 5.1 layout: FL+FR+FC+LFE+BL+BR */ - for (i = cvt->len_cvt / (sizeof (float) * 6); i; --i, src += 6, dst += 4) { - const float front_center_distributed = src[2] * 0.5f; - dst[0] = (src[0] + front_center_distributed) / 1.5f; /* FL */ - dst[1] = (src[1] + front_center_distributed) / 1.5f; /* FR */ - dst[2] = src[4] / 1.5f; /* BL */ - dst[3] = src[5] / 1.5f; /* BR */ - } - - cvt->len_cvt /= 6; - cvt->len_cvt *= 4; - if (cvt->filters[++cvt->filter_index]) { - cvt->filters[cvt->filter_index] (cvt, format); - } -} - - -/* Upmix mono to stereo (by duplication) */ -static void SDLCALL -SDL_ConvertMonoToStereo(SDL_AudioCVT * cvt, SDL_AudioFormat format) -{ - const float *src = (const float *) (cvt->buf + cvt->len_cvt); - float *dst = (float *) (cvt->buf + cvt->len_cvt * 2); - int i; - - LOG_DEBUG_CONVERT("mono", "stereo"); - SDL_assert(format == AUDIO_F32SYS); - - for (i = cvt->len_cvt / sizeof (float); i; --i) { - src--; - dst -= 2; - dst[0] = dst[1] = *src; - } - - cvt->len_cvt *= 2; - if (cvt->filters[++cvt->filter_index]) { - cvt->filters[cvt->filter_index] (cvt, format); - } -} - - -/* Upmix stereo to a pseudo-5.1 stream */ -static void SDLCALL -SDL_ConvertStereoTo51(SDL_AudioCVT * cvt, SDL_AudioFormat format) -{ - int i; - float lf, rf, ce; - const float *src = (const float *) (cvt->buf + cvt->len_cvt); - float *dst = (float *) (cvt->buf + cvt->len_cvt * 3); - - LOG_DEBUG_CONVERT("stereo", "5.1"); - SDL_assert(format == AUDIO_F32SYS); - - for (i = cvt->len_cvt / (sizeof(float) * 2); i; --i) { - dst -= 6; - src -= 2; - lf = src[0]; - rf = src[1]; - ce = (lf + rf) * 0.5f; - /* !!! FIXME: FL and FR may clip */ - dst[0] = lf + (lf - ce); /* FL */ - dst[1] = rf + (rf - ce); /* FR */ - dst[2] = ce; /* FC */ - dst[3] = 0; /* LFE (only meant for special LFE effects) */ - dst[4] = lf; /* BL */ - dst[5] = rf; /* BR */ - } - - cvt->len_cvt *= 3; - if (cvt->filters[++cvt->filter_index]) { - cvt->filters[cvt->filter_index] (cvt, format); - } -} - - -/* Upmix quad to a pseudo-5.1 stream */ -static void SDLCALL -SDL_ConvertQuadTo51(SDL_AudioCVT * cvt, SDL_AudioFormat format) -{ - int i; - float lf, rf, lb, rb, ce; - const float *src = (const float *) (cvt->buf + cvt->len_cvt); - float *dst = (float *) (cvt->buf + cvt->len_cvt * 3 / 2); - - LOG_DEBUG_CONVERT("quad", "5.1"); - SDL_assert(format == AUDIO_F32SYS); - SDL_assert(cvt->len_cvt % (sizeof(float) * 4) == 0); - - for (i = cvt->len_cvt / (sizeof(float) * 4); i; --i) { - dst -= 6; - src -= 4; - lf = src[0]; - rf = src[1]; - lb = src[2]; - rb = src[3]; - ce = (lf + rf) * 0.5f; - /* !!! FIXME: FL and FR may clip */ - dst[0] = lf + (lf - ce); /* FL */ - dst[1] = rf + (rf - ce); /* FR */ - dst[2] = ce; /* FC */ - dst[3] = 0; /* LFE (only meant for special LFE effects) */ - dst[4] = lb; /* BL */ - dst[5] = rb; /* BR */ - } - - cvt->len_cvt = cvt->len_cvt * 3 / 2; - if (cvt->filters[++cvt->filter_index]) { - cvt->filters[cvt->filter_index] (cvt, format); - } -} - - -/* Upmix stereo to a pseudo-4.0 stream (by duplication) */ -static void SDLCALL -SDL_ConvertStereoToQuad(SDL_AudioCVT * cvt, SDL_AudioFormat format) -{ - const float *src = (const float *) (cvt->buf + cvt->len_cvt); - float *dst = (float *) (cvt->buf + cvt->len_cvt * 2); - float lf, rf; - int i; - - LOG_DEBUG_CONVERT("stereo", "quad"); - SDL_assert(format == AUDIO_F32SYS); - - for (i = cvt->len_cvt / (sizeof(float) * 2); i; --i) { - dst -= 4; - src -= 2; - lf = src[0]; - rf = src[1]; - dst[0] = lf; /* FL */ - dst[1] = rf; /* FR */ - dst[2] = lf; /* BL */ - dst[3] = rf; /* BR */ - } - - cvt->len_cvt *= 2; - if (cvt->filters[++cvt->filter_index]) { - cvt->filters[cvt->filter_index] (cvt, format); - } -} - - -/* Upmix 5.1 to 7.1 */ -static void SDLCALL -SDL_Convert51To71(SDL_AudioCVT * cvt, SDL_AudioFormat format) -{ - float lf, rf, lb, rb, ls, rs; - int i; - const float *src = (const float *) (cvt->buf + cvt->len_cvt); - float *dst = (float *) (cvt->buf + cvt->len_cvt * 4 / 3); - - LOG_DEBUG_CONVERT("5.1", "7.1"); - SDL_assert(format == AUDIO_F32SYS); - SDL_assert(cvt->len_cvt % (sizeof(float) * 6) == 0); - - for (i = cvt->len_cvt / (sizeof(float) * 6); i; --i) { - dst -= 8; - src -= 6; - lf = src[0]; - rf = src[1]; - lb = src[4]; - rb = src[5]; - ls = (lf + lb) * 0.5f; - rs = (rf + rb) * 0.5f; - /* !!! FIXME: these four may clip */ - lf += lf - ls; - rf += rf - ls; - lb += lb - ls; - rb += rb - ls; - dst[3] = src[3]; /* LFE */ - dst[2] = src[2]; /* FC */ - dst[7] = rs; /* SR */ - dst[6] = ls; /* SL */ - dst[5] = rb; /* BR */ - dst[4] = lb; /* BL */ - dst[1] = rf; /* FR */ - dst[0] = lf; /* FL */ - } - - cvt->len_cvt = cvt->len_cvt * 4 / 3; - - if (cvt->filters[++cvt->filter_index]) { - cvt->filters[cvt->filter_index] (cvt, format); - } -} - -/* SDL's resampler uses a "bandlimited interpolation" algorithm: - https://ccrma.stanford.edu/~jos/resample/ */ - -#define RESAMPLER_ZERO_CROSSINGS 5 -#define RESAMPLER_BITS_PER_SAMPLE 16 -#define RESAMPLER_SAMPLES_PER_ZERO_CROSSING (1 << ((RESAMPLER_BITS_PER_SAMPLE / 2) + 1)) -#define RESAMPLER_FILTER_SIZE ((RESAMPLER_SAMPLES_PER_ZERO_CROSSING * RESAMPLER_ZERO_CROSSINGS) + 1) - -/* This is a "modified" bessel function, so you can't use POSIX j0() */ -static double -bessel(const double x) -{ - const double xdiv2 = x / 2.0; - double i0 = 1.0f; - double f = 1.0f; - int i = 1; - - while (SDL_TRUE) { - const double diff = SDL_pow(xdiv2, i * 2) / SDL_pow(f, 2); - if (diff < 1.0e-21f) { - break; - } - i0 += diff; - i++; - f *= (double) i; - } - - return i0; -} - -/* build kaiser table with cardinal sine applied to it, and array of differences between elements. */ -static void -kaiser_and_sinc(float *table, float *diffs, const int tablelen, const double beta) -{ - const int lenm1 = tablelen - 1; - const int lenm1div2 = lenm1 / 2; - int i; - - table[0] = 1.0f; - for (i = 1; i < tablelen; i++) { - const double kaiser = bessel(beta * SDL_sqrt(1.0 - SDL_pow(((i - lenm1) / 2.0) / lenm1div2, 2.0))) / bessel(beta); - table[tablelen - i] = (float) kaiser; - } - - for (i = 1; i < tablelen; i++) { - const float x = (((float) i) / ((float) RESAMPLER_SAMPLES_PER_ZERO_CROSSING)) * ((float) M_PI); - table[i] *= SDL_sinf(x) / x; - diffs[i - 1] = table[i] - table[i - 1]; - } - diffs[lenm1] = 0.0f; -} - - -static SDL_SpinLock ResampleFilterSpinlock = 0; -static float *ResamplerFilter = NULL; -static float *ResamplerFilterDifference = NULL; - -int -SDL_PrepareResampleFilter(void) -{ - SDL_AtomicLock(&ResampleFilterSpinlock); - if (!ResamplerFilter) { - /* if dB > 50, beta=(0.1102 * (dB - 8.7)), according to Matlab. */ - const double dB = 80.0; - const double beta = 0.1102 * (dB - 8.7); - const size_t alloclen = RESAMPLER_FILTER_SIZE * sizeof (float); - - ResamplerFilter = (float *) SDL_malloc(alloclen); - if (!ResamplerFilter) { - SDL_AtomicUnlock(&ResampleFilterSpinlock); - return SDL_OutOfMemory(); - } - - ResamplerFilterDifference = (float *) SDL_malloc(alloclen); - if (!ResamplerFilterDifference) { - SDL_free(ResamplerFilter); - ResamplerFilter = NULL; - SDL_AtomicUnlock(&ResampleFilterSpinlock); - return SDL_OutOfMemory(); - } - kaiser_and_sinc(ResamplerFilter, ResamplerFilterDifference, RESAMPLER_FILTER_SIZE, beta); - } - SDL_AtomicUnlock(&ResampleFilterSpinlock); - return 0; -} - -void -SDL_FreeResampleFilter(void) -{ - SDL_free(ResamplerFilter); - SDL_free(ResamplerFilterDifference); - ResamplerFilter = NULL; - ResamplerFilterDifference = NULL; -} - -static int -ResamplerPadding(const int inrate, const int outrate) -{ - if (inrate == outrate) { - return 0; - } else if (inrate > outrate) { - return (int) SDL_ceil(((float) (RESAMPLER_SAMPLES_PER_ZERO_CROSSING * inrate) / ((float) outrate))); - } - return RESAMPLER_SAMPLES_PER_ZERO_CROSSING; -} - -/* lpadding and rpadding are expected to be buffers of (ResamplePadding(inrate, outrate) * chans * sizeof (float)) bytes. */ -static int -SDL_ResampleAudio(const int chans, const int inrate, const int outrate, - const float *lpadding, const float *rpadding, - const float *inbuf, const int inbuflen, - float *outbuf, const int outbuflen) -{ - const double finrate = (double) inrate; - const double outtimeincr = 1.0 / ((float) outrate); - const double ratio = ((float) outrate) / ((float) inrate); - const int paddinglen = ResamplerPadding(inrate, outrate); - const int framelen = chans * (int)sizeof (float); - const int inframes = inbuflen / framelen; - const int wantedoutframes = (int) ((inbuflen / framelen) * ratio); /* outbuflen isn't total to write, it's total available. */ - const int maxoutframes = outbuflen / framelen; - const int outframes = SDL_min(wantedoutframes, maxoutframes); - float *dst = outbuf; - double outtime = 0.0; - int i, j, chan; - - for (i = 0; i < outframes; i++) { - const int srcindex = (int) (outtime * inrate); - const double intime = ((double) srcindex) / finrate; - const double innexttime = ((double) (srcindex + 1)) / finrate; - const double interpolation1 = 1.0 - ((innexttime - outtime) / (innexttime - intime)); - const int filterindex1 = (int) (interpolation1 * RESAMPLER_SAMPLES_PER_ZERO_CROSSING); - const double interpolation2 = 1.0 - interpolation1; - const int filterindex2 = (int) (interpolation2 * RESAMPLER_SAMPLES_PER_ZERO_CROSSING); - - for (chan = 0; chan < chans; chan++) { - float outsample = 0.0f; - - /* do this twice to calculate the sample, once for the "left wing" and then same for the right. */ - /* !!! FIXME: do both wings in one loop */ - for (j = 0; (filterindex1 + (j * RESAMPLER_SAMPLES_PER_ZERO_CROSSING)) < RESAMPLER_FILTER_SIZE; j++) { - const int srcframe = srcindex - j; - /* !!! FIXME: we can bubble this conditional out of here by doing a pre loop. */ - const float insample = (srcframe < 0) ? lpadding[((paddinglen + srcframe) * chans) + chan] : inbuf[(srcframe * chans) + chan]; - outsample += (float)(insample * (ResamplerFilter[filterindex1 + (j * RESAMPLER_SAMPLES_PER_ZERO_CROSSING)] + (interpolation1 * ResamplerFilterDifference[filterindex1 + (j * RESAMPLER_SAMPLES_PER_ZERO_CROSSING)]))); - } - - for (j = 0; (filterindex2 + (j * RESAMPLER_SAMPLES_PER_ZERO_CROSSING)) < RESAMPLER_FILTER_SIZE; j++) { - const int srcframe = srcindex + 1 + j; - /* !!! FIXME: we can bubble this conditional out of here by doing a post loop. */ - const float insample = (srcframe >= inframes) ? rpadding[((srcframe - inframes) * chans) + chan] : inbuf[(srcframe * chans) + chan]; - outsample += (float)(insample * (ResamplerFilter[filterindex2 + (j * RESAMPLER_SAMPLES_PER_ZERO_CROSSING)] + (interpolation2 * ResamplerFilterDifference[filterindex2 + (j * RESAMPLER_SAMPLES_PER_ZERO_CROSSING)]))); - } - *(dst++) = outsample; - } - - outtime += outtimeincr; - } - - return outframes * chans * sizeof (float); -} - -int -SDL_ConvertAudio(SDL_AudioCVT * cvt) -{ - /* !!! FIXME: (cvt) should be const; stack-copy it here. */ - /* !!! FIXME: (actually, we can't...len_cvt needs to be updated. Grr.) */ - - /* Make sure there's data to convert */ - if (cvt->buf == NULL) { - return SDL_SetError("No buffer allocated for conversion"); - } - - /* Return okay if no conversion is necessary */ - cvt->len_cvt = cvt->len; - if (cvt->filters[0] == NULL) { - return 0; - } - - /* Set up the conversion and go! */ - cvt->filter_index = 0; - cvt->filters[0] (cvt, cvt->src_format); - return 0; -} - -static void SDLCALL -SDL_Convert_Byteswap(SDL_AudioCVT *cvt, SDL_AudioFormat format) -{ -#if DEBUG_CONVERT - printf("Converting byte order\n"); -#endif - - switch (SDL_AUDIO_BITSIZE(format)) { - #define CASESWAP(b) \ - case b: { \ - Uint##b *ptr = (Uint##b *) cvt->buf; \ - int i; \ - for (i = cvt->len_cvt / sizeof (*ptr); i; --i, ++ptr) { \ - *ptr = SDL_Swap##b(*ptr); \ - } \ - break; \ - } - - CASESWAP(16); - CASESWAP(32); - CASESWAP(64); - - #undef CASESWAP - - default: SDL_assert(!"unhandled byteswap datatype!"); break; - } - - if (cvt->filters[++cvt->filter_index]) { - /* flip endian flag for data. */ - if (format & SDL_AUDIO_MASK_ENDIAN) { - format &= ~SDL_AUDIO_MASK_ENDIAN; - } else { - format |= SDL_AUDIO_MASK_ENDIAN; - } - cvt->filters[cvt->filter_index](cvt, format); - } -} - -static int -SDL_AddAudioCVTFilter(SDL_AudioCVT *cvt, const SDL_AudioFilter filter) -{ - if (cvt->filter_index >= SDL_AUDIOCVT_MAX_FILTERS) { - return SDL_SetError("Too many filters needed for conversion, exceeded maximum of %d", SDL_AUDIOCVT_MAX_FILTERS); - } - if (filter == NULL) { - return SDL_SetError("Audio filter pointer is NULL"); - } - cvt->filters[cvt->filter_index++] = filter; - cvt->filters[cvt->filter_index] = NULL; /* Moving terminator */ - return 0; -} - -static int -SDL_BuildAudioTypeCVTToFloat(SDL_AudioCVT *cvt, const SDL_AudioFormat src_fmt) -{ - int retval = 0; /* 0 == no conversion necessary. */ - - if ((SDL_AUDIO_ISBIGENDIAN(src_fmt) != 0) == (SDL_BYTEORDER == SDL_LIL_ENDIAN)) { - if (SDL_AddAudioCVTFilter(cvt, SDL_Convert_Byteswap) < 0) { - return -1; - } - retval = 1; /* added a converter. */ - } - - if (!SDL_AUDIO_ISFLOAT(src_fmt)) { - const Uint16 src_bitsize = SDL_AUDIO_BITSIZE(src_fmt); - const Uint16 dst_bitsize = 32; - SDL_AudioFilter filter = NULL; - - switch (src_fmt & ~SDL_AUDIO_MASK_ENDIAN) { - case AUDIO_S8: filter = SDL_Convert_S8_to_F32; break; - case AUDIO_U8: filter = SDL_Convert_U8_to_F32; break; - case AUDIO_S16: filter = SDL_Convert_S16_to_F32; break; - case AUDIO_U16: filter = SDL_Convert_U16_to_F32; break; - case AUDIO_S32: filter = SDL_Convert_S32_to_F32; break; - default: SDL_assert(!"Unexpected audio format!"); break; - } - - if (!filter) { - return SDL_SetError("No conversion from source format to float available"); - } - - if (SDL_AddAudioCVTFilter(cvt, filter) < 0) { - return -1; - } - if (src_bitsize < dst_bitsize) { - const int mult = (dst_bitsize / src_bitsize); - cvt->len_mult *= mult; - cvt->len_ratio *= mult; - } else if (src_bitsize > dst_bitsize) { - cvt->len_ratio /= (src_bitsize / dst_bitsize); - } - - retval = 1; /* added a converter. */ - } - - return retval; -} - -static int -SDL_BuildAudioTypeCVTFromFloat(SDL_AudioCVT *cvt, const SDL_AudioFormat dst_fmt) -{ - int retval = 0; /* 0 == no conversion necessary. */ - - if (!SDL_AUDIO_ISFLOAT(dst_fmt)) { - const Uint16 dst_bitsize = SDL_AUDIO_BITSIZE(dst_fmt); - const Uint16 src_bitsize = 32; - SDL_AudioFilter filter = NULL; - switch (dst_fmt & ~SDL_AUDIO_MASK_ENDIAN) { - case AUDIO_S8: filter = SDL_Convert_F32_to_S8; break; - case AUDIO_U8: filter = SDL_Convert_F32_to_U8; break; - case AUDIO_S16: filter = SDL_Convert_F32_to_S16; break; - case AUDIO_U16: filter = SDL_Convert_F32_to_U16; break; - case AUDIO_S32: filter = SDL_Convert_F32_to_S32; break; - default: SDL_assert(!"Unexpected audio format!"); break; - } - - if (!filter) { - return SDL_SetError("No conversion from float to destination format available"); - } - - if (SDL_AddAudioCVTFilter(cvt, filter) < 0) { - return -1; - } - if (src_bitsize < dst_bitsize) { - const int mult = (dst_bitsize / src_bitsize); - cvt->len_mult *= mult; - cvt->len_ratio *= mult; - } else if (src_bitsize > dst_bitsize) { - cvt->len_ratio /= (src_bitsize / dst_bitsize); - } - retval = 1; /* added a converter. */ - } - - if ((SDL_AUDIO_ISBIGENDIAN(dst_fmt) != 0) == (SDL_BYTEORDER == SDL_LIL_ENDIAN)) { - if (SDL_AddAudioCVTFilter(cvt, SDL_Convert_Byteswap) < 0) { - return -1; - } - retval = 1; /* added a converter. */ - } - - return retval; -} - -static void -SDL_ResampleCVT(SDL_AudioCVT *cvt, const int chans, const SDL_AudioFormat format) -{ - /* !!! FIXME in 2.1: there are ten slots in the filter list, and the theoretical maximum we use is six (seven with NULL terminator). - !!! FIXME in 2.1: We need to store data for this resampler, because the cvt structure doesn't store the original sample rates, - !!! FIXME in 2.1: so we steal the ninth and tenth slot. :( */ - const int inrate = (int) (size_t) cvt->filters[SDL_AUDIOCVT_MAX_FILTERS-1]; - const int outrate = (int) (size_t) cvt->filters[SDL_AUDIOCVT_MAX_FILTERS]; - const float *src = (const float *) cvt->buf; - const int srclen = cvt->len_cvt; - /*float *dst = (float *) cvt->buf; - const int dstlen = (cvt->len * cvt->len_mult);*/ - /* !!! FIXME: remove this if we can get the resampler to work in-place again. */ - float *dst = (float *) (cvt->buf + srclen); - const int dstlen = (cvt->len * cvt->len_mult) - srclen; - const int paddingsamples = (ResamplerPadding(inrate, outrate) * chans); - float *padding; - - SDL_assert(format == AUDIO_F32SYS); - - /* we keep no streaming state here, so pad with silence on both ends. */ - padding = (float *) SDL_calloc(paddingsamples ? paddingsamples : 1, sizeof (float)); - if (!padding) { - SDL_OutOfMemory(); - return; - } - - cvt->len_cvt = SDL_ResampleAudio(chans, inrate, outrate, padding, padding, src, srclen, dst, dstlen); - - SDL_free(padding); - - SDL_memmove(cvt->buf, dst, cvt->len_cvt); /* !!! FIXME: remove this if we can get the resampler to work in-place again. */ - - if (cvt->filters[++cvt->filter_index]) { - cvt->filters[cvt->filter_index](cvt, format); - } -} - -/* !!! FIXME: We only have this macro salsa because SDL_AudioCVT doesn't - !!! FIXME: store channel info, so we have to have function entry - !!! FIXME: points for each supported channel count and multiple - !!! FIXME: vs arbitrary. When we rev the ABI, clean this up. */ -#define RESAMPLER_FUNCS(chans) \ - static void SDLCALL \ - SDL_ResampleCVT_c##chans(SDL_AudioCVT *cvt, SDL_AudioFormat format) { \ - SDL_ResampleCVT(cvt, chans, format); \ - } -RESAMPLER_FUNCS(1) -RESAMPLER_FUNCS(2) -RESAMPLER_FUNCS(4) -RESAMPLER_FUNCS(6) -RESAMPLER_FUNCS(8) -#undef RESAMPLER_FUNCS - -static SDL_AudioFilter -ChooseCVTResampler(const int dst_channels) -{ - switch (dst_channels) { - case 1: return SDL_ResampleCVT_c1; - case 2: return SDL_ResampleCVT_c2; - case 4: return SDL_ResampleCVT_c4; - case 6: return SDL_ResampleCVT_c6; - case 8: return SDL_ResampleCVT_c8; - default: break; - } - - return NULL; -} - -static int -SDL_BuildAudioResampleCVT(SDL_AudioCVT * cvt, const int dst_channels, - const int src_rate, const int dst_rate) -{ - SDL_AudioFilter filter; - - if (src_rate == dst_rate) { - return 0; /* no conversion necessary. */ - } - - filter = ChooseCVTResampler(dst_channels); - if (filter == NULL) { - return SDL_SetError("No conversion available for these rates"); - } - - if (SDL_PrepareResampleFilter() < 0) { - return -1; - } - - /* Update (cvt) with filter details... */ - if (SDL_AddAudioCVTFilter(cvt, filter) < 0) { - return -1; - } - - /* !!! FIXME in 2.1: there are ten slots in the filter list, and the theoretical maximum we use is six (seven with NULL terminator). - !!! FIXME in 2.1: We need to store data for this resampler, because the cvt structure doesn't store the original sample rates, - !!! FIXME in 2.1: so we steal the ninth and tenth slot. :( */ - if (cvt->filter_index >= (SDL_AUDIOCVT_MAX_FILTERS-2)) { - return SDL_SetError("Too many filters needed for conversion, exceeded maximum of %d", SDL_AUDIOCVT_MAX_FILTERS-2); - } - cvt->filters[SDL_AUDIOCVT_MAX_FILTERS-1] = (SDL_AudioFilter) (size_t) src_rate; - cvt->filters[SDL_AUDIOCVT_MAX_FILTERS] = (SDL_AudioFilter) (size_t) dst_rate; - - if (src_rate < dst_rate) { - const double mult = ((double) dst_rate) / ((double) src_rate); - cvt->len_mult *= (int) SDL_ceil(mult); - cvt->len_ratio *= mult; - } else { - cvt->len_ratio /= ((double) src_rate) / ((double) dst_rate); - } - - /* !!! FIXME: remove this if we can get the resampler to work in-place again. */ - /* the buffer is big enough to hold the destination now, but - we need it large enough to hold a separate scratch buffer. */ - cvt->len_mult *= 2; - - return 1; /* added a converter. */ -} - -static SDL_bool -SDL_SupportedAudioFormat(const SDL_AudioFormat fmt) -{ - switch (fmt) { - case AUDIO_U8: - case AUDIO_S8: - case AUDIO_U16LSB: - case AUDIO_S16LSB: - case AUDIO_U16MSB: - case AUDIO_S16MSB: - case AUDIO_S32LSB: - case AUDIO_S32MSB: - case AUDIO_F32LSB: - case AUDIO_F32MSB: - return SDL_TRUE; /* supported. */ - - default: - break; - } - - return SDL_FALSE; /* unsupported. */ -} - -static SDL_bool -SDL_SupportedChannelCount(const int channels) -{ - switch (channels) { - case 1: /* mono */ - case 2: /* stereo */ - case 4: /* quad */ - case 6: /* 5.1 */ - case 8: /* 7.1 */ - return SDL_TRUE; /* supported. */ - - default: - break; - } - - return SDL_FALSE; /* unsupported. */ -} - - -/* Creates a set of audio filters to convert from one format to another. - Returns 0 if no conversion is needed, 1 if the audio filter is set up, - or -1 if an error like invalid parameter, unsupported format, etc. occurred. -*/ - -int -SDL_BuildAudioCVT(SDL_AudioCVT * cvt, - SDL_AudioFormat src_fmt, Uint8 src_channels, int src_rate, - SDL_AudioFormat dst_fmt, Uint8 dst_channels, int dst_rate) -{ - /* Sanity check target pointer */ - if (cvt == NULL) { - return SDL_InvalidParamError("cvt"); - } - - /* Make sure we zero out the audio conversion before error checking */ - SDL_zerop(cvt); - - if (!SDL_SupportedAudioFormat(src_fmt)) { - return SDL_SetError("Invalid source format"); - } else if (!SDL_SupportedAudioFormat(dst_fmt)) { - return SDL_SetError("Invalid destination format"); - } else if (!SDL_SupportedChannelCount(src_channels)) { - return SDL_SetError("Invalid source channels"); - } else if (!SDL_SupportedChannelCount(dst_channels)) { - return SDL_SetError("Invalid destination channels"); - } else if (src_rate == 0) { - return SDL_SetError("Source rate is zero"); - } else if (dst_rate == 0) { - return SDL_SetError("Destination rate is zero"); - } - -#if DEBUG_CONVERT - printf("Build format %04x->%04x, channels %u->%u, rate %d->%d\n", - src_fmt, dst_fmt, src_channels, dst_channels, src_rate, dst_rate); -#endif - - /* Start off with no conversion necessary */ - cvt->src_format = src_fmt; - cvt->dst_format = dst_fmt; - cvt->needed = 0; - cvt->filter_index = 0; - SDL_zero(cvt->filters); - cvt->len_mult = 1; - cvt->len_ratio = 1.0; - cvt->rate_incr = ((double) dst_rate) / ((double) src_rate); - - /* Make sure we've chosen audio conversion functions (MMX, scalar, etc.) */ - SDL_ChooseAudioConverters(); - - /* Type conversion goes like this now: - - byteswap to CPU native format first if necessary. - - convert to native Float32 if necessary. - - resample and change channel count if necessary. - - convert back to native format. - - byteswap back to foreign format if necessary. - - The expectation is we can process data faster in float32 - (possibly with SIMD), and making several passes over the same - buffer is likely to be CPU cache-friendly, avoiding the - biggest performance hit in modern times. Previously we had - (script-generated) custom converters for every data type and - it was a bloat on SDL compile times and final library size. */ - - /* see if we can skip float conversion entirely. */ - if (src_rate == dst_rate && src_channels == dst_channels) { - if (src_fmt == dst_fmt) { - return 0; - } - - /* just a byteswap needed? */ - if ((src_fmt & ~SDL_AUDIO_MASK_ENDIAN) == (dst_fmt & ~SDL_AUDIO_MASK_ENDIAN)) { - if (SDL_AddAudioCVTFilter(cvt, SDL_Convert_Byteswap) < 0) { - return -1; - } - cvt->needed = 1; - return 1; - } - } - - /* Convert data types, if necessary. Updates (cvt). */ - if (SDL_BuildAudioTypeCVTToFloat(cvt, src_fmt) < 0) { - return -1; /* shouldn't happen, but just in case... */ - } - - /* Channel conversion */ - if (src_channels < dst_channels) { - /* Upmixing */ - /* Mono -> Stereo [-> ...] */ - if ((src_channels == 1) && (dst_channels > 1)) { - if (SDL_AddAudioCVTFilter(cvt, SDL_ConvertMonoToStereo) < 0) { - return -1; - } - cvt->len_mult *= 2; - src_channels = 2; - cvt->len_ratio *= 2; - } - /* [Mono ->] Stereo -> 5.1 [-> 7.1] */ - if ((src_channels == 2) && (dst_channels >= 6)) { - if (SDL_AddAudioCVTFilter(cvt, SDL_ConvertStereoTo51) < 0) { - return -1; - } - src_channels = 6; - cvt->len_mult *= 3; - cvt->len_ratio *= 3; - } - /* Quad -> 5.1 [-> 7.1] */ - if ((src_channels == 4) && (dst_channels >= 6)) { - if (SDL_AddAudioCVTFilter(cvt, SDL_ConvertQuadTo51) < 0) { - return -1; - } - src_channels = 6; - cvt->len_mult = (cvt->len_mult * 3 + 1) / 2; - cvt->len_ratio *= 1.5; - } - /* [[Mono ->] Stereo ->] 5.1 -> 7.1 */ - if ((src_channels == 6) && (dst_channels == 8)) { - if (SDL_AddAudioCVTFilter(cvt, SDL_Convert51To71) < 0) { - return -1; - } - src_channels = 8; - cvt->len_mult = (cvt->len_mult * 4 + 2) / 3; - /* Should be numerically exact with every valid input to this - function */ - cvt->len_ratio = cvt->len_ratio * 4 / 3; - } - /* [Mono ->] Stereo -> Quad */ - if ((src_channels == 2) && (dst_channels == 4)) { - if (SDL_AddAudioCVTFilter(cvt, SDL_ConvertStereoToQuad) < 0) { - return -1; - } - src_channels = 4; - cvt->len_mult *= 2; - cvt->len_ratio *= 2; - } - } else if (src_channels > dst_channels) { - /* Downmixing */ - /* 7.1 -> 5.1 [-> Stereo [-> Mono]] */ - /* 7.1 -> 5.1 [-> Quad] */ - if ((src_channels == 8) && (dst_channels <= 6)) { - if (SDL_AddAudioCVTFilter(cvt, SDL_Convert71To51) < 0) { - return -1; - } - src_channels = 6; - cvt->len_ratio *= 0.75; - } - /* [7.1 ->] 5.1 -> Stereo [-> Mono] */ - if ((src_channels == 6) && (dst_channels <= 2)) { - if (SDL_AddAudioCVTFilter(cvt, SDL_Convert51ToStereo) < 0) { - return -1; - } - src_channels = 2; - cvt->len_ratio /= 3; - } - /* 5.1 -> Quad */ - if ((src_channels == 6) && (dst_channels == 4)) { - if (SDL_AddAudioCVTFilter(cvt, SDL_Convert51ToQuad) < 0) { - return -1; - } - src_channels = 4; - cvt->len_ratio = cvt->len_ratio * 2 / 3; - } - /* Quad -> Stereo [-> Mono] */ - if ((src_channels == 4) && (dst_channels <= 2)) { - if (SDL_AddAudioCVTFilter(cvt, SDL_ConvertQuadToStereo) < 0) { - return -1; - } - src_channels = 2; - cvt->len_ratio /= 2; - } - /* [... ->] Stereo -> Mono */ - if ((src_channels == 2) && (dst_channels == 1)) { - SDL_AudioFilter filter = NULL; - - #if HAVE_SSE3_INTRINSICS - if (SDL_HasSSE3()) { - filter = SDL_ConvertStereoToMono_SSE3; - } - #endif - - if (!filter) { - filter = SDL_ConvertStereoToMono; - } - - if (SDL_AddAudioCVTFilter(cvt, filter) < 0) { - return -1; - } - - src_channels = 1; - cvt->len_ratio /= 2; - } - } - - if (src_channels != dst_channels) { - /* All combinations of supported channel counts should have been - handled by now, but let's be defensive */ - return SDL_SetError("Invalid channel combination"); - } - - /* Do rate conversion, if necessary. Updates (cvt). */ - if (SDL_BuildAudioResampleCVT(cvt, dst_channels, src_rate, dst_rate) < 0) { - return -1; /* shouldn't happen, but just in case... */ - } - - /* Move to final data type. */ - if (SDL_BuildAudioTypeCVTFromFloat(cvt, dst_fmt) < 0) { - return -1; /* shouldn't happen, but just in case... */ - } - - cvt->needed = (cvt->filter_index != 0); - return (cvt->needed); -} - -typedef int (*SDL_ResampleAudioStreamFunc)(SDL_AudioStream *stream, const void *inbuf, const int inbuflen, void *outbuf, const int outbuflen); -typedef void (*SDL_ResetAudioStreamResamplerFunc)(SDL_AudioStream *stream); -typedef void (*SDL_CleanupAudioStreamResamplerFunc)(SDL_AudioStream *stream); - -struct _SDL_AudioStream -{ - SDL_AudioCVT cvt_before_resampling; - SDL_AudioCVT cvt_after_resampling; - SDL_DataQueue *queue; - SDL_bool first_run; - Uint8 *staging_buffer; - int staging_buffer_size; - int staging_buffer_filled; - Uint8 *work_buffer_base; /* maybe unaligned pointer from SDL_realloc(). */ - int work_buffer_len; - int src_sample_frame_size; - SDL_AudioFormat src_format; - Uint8 src_channels; - int src_rate; - int dst_sample_frame_size; - SDL_AudioFormat dst_format; - Uint8 dst_channels; - int dst_rate; - double rate_incr; - Uint8 pre_resample_channels; - int packetlen; - int resampler_padding_samples; - float *resampler_padding; - void *resampler_state; - SDL_ResampleAudioStreamFunc resampler_func; - SDL_ResetAudioStreamResamplerFunc reset_resampler_func; - SDL_CleanupAudioStreamResamplerFunc cleanup_resampler_func; -}; - -static Uint8 * -EnsureStreamBufferSize(SDL_AudioStream *stream, const int newlen) -{ - Uint8 *ptr; - size_t offset; - - if (stream->work_buffer_len >= newlen) { - ptr = stream->work_buffer_base; - } else { - ptr = (Uint8 *) SDL_realloc(stream->work_buffer_base, newlen + 32); - if (!ptr) { - SDL_OutOfMemory(); - return NULL; - } - /* Make sure we're aligned to 16 bytes for SIMD code. */ - stream->work_buffer_base = ptr; - stream->work_buffer_len = newlen; - } - - offset = ((size_t) ptr) & 15; - return offset ? ptr + (16 - offset) : ptr; -} - -#ifdef HAVE_LIBSAMPLERATE_H -static int -SDL_ResampleAudioStream_SRC(SDL_AudioStream *stream, const void *_inbuf, const int inbuflen, void *_outbuf, const int outbuflen) -{ - const float *inbuf = (const float *) _inbuf; - float *outbuf = (float *) _outbuf; - const int framelen = sizeof(float) * stream->pre_resample_channels; - SRC_STATE *state = (SRC_STATE *)stream->resampler_state; - SRC_DATA data; - int result; - - SDL_assert(inbuf != ((const float *) outbuf)); /* SDL_AudioStreamPut() shouldn't allow in-place resamples. */ - - data.data_in = (float *)inbuf; /* Older versions of libsamplerate had a non-const pointer, but didn't write to it */ - data.input_frames = inbuflen / framelen; - data.input_frames_used = 0; - - data.data_out = outbuf; - data.output_frames = outbuflen / framelen; - - data.end_of_input = 0; - data.src_ratio = stream->rate_incr; - - result = SRC_src_process(state, &data); - if (result != 0) { - SDL_SetError("src_process() failed: %s", SRC_src_strerror(result)); - return 0; - } - - /* If this fails, we need to store them off somewhere */ - SDL_assert(data.input_frames_used == data.input_frames); - - return data.output_frames_gen * (sizeof(float) * stream->pre_resample_channels); -} - -static void -SDL_ResetAudioStreamResampler_SRC(SDL_AudioStream *stream) -{ - SRC_src_reset((SRC_STATE *)stream->resampler_state); -} - -static void -SDL_CleanupAudioStreamResampler_SRC(SDL_AudioStream *stream) -{ - SRC_STATE *state = (SRC_STATE *)stream->resampler_state; - if (state) { - SRC_src_delete(state); - } - - stream->resampler_state = NULL; - stream->resampler_func = NULL; - stream->reset_resampler_func = NULL; - stream->cleanup_resampler_func = NULL; -} - -static SDL_bool -SetupLibSampleRateResampling(SDL_AudioStream *stream) -{ - int result = 0; - SRC_STATE *state = NULL; - - if (SRC_available) { - state = SRC_src_new(SRC_converter, stream->pre_resample_channels, &result); - if (!state) { - SDL_SetError("src_new() failed: %s", SRC_src_strerror(result)); - } - } - - if (!state) { - SDL_CleanupAudioStreamResampler_SRC(stream); - return SDL_FALSE; - } - - stream->resampler_state = state; - stream->resampler_func = SDL_ResampleAudioStream_SRC; - stream->reset_resampler_func = SDL_ResetAudioStreamResampler_SRC; - stream->cleanup_resampler_func = SDL_CleanupAudioStreamResampler_SRC; - - return SDL_TRUE; -} -#endif /* HAVE_LIBSAMPLERATE_H */ - - -static int -SDL_ResampleAudioStream(SDL_AudioStream *stream, const void *_inbuf, const int inbuflen, void *_outbuf, const int outbuflen) -{ - const Uint8 *inbufend = ((const Uint8 *) _inbuf) + inbuflen; - const float *inbuf = (const float *) _inbuf; - float *outbuf = (float *) _outbuf; - const int chans = (int) stream->pre_resample_channels; - const int inrate = stream->src_rate; - const int outrate = stream->dst_rate; - const int paddingsamples = stream->resampler_padding_samples; - const int paddingbytes = paddingsamples * sizeof (float); - float *lpadding = (float *) stream->resampler_state; - const float *rpadding = (const float *) inbufend; /* we set this up so there are valid padding samples at the end of the input buffer. */ - const int cpy = SDL_min(inbuflen, paddingbytes); - int retval; - - SDL_assert(inbuf != ((const float *) outbuf)); /* SDL_AudioStreamPut() shouldn't allow in-place resamples. */ - - retval = SDL_ResampleAudio(chans, inrate, outrate, lpadding, rpadding, inbuf, inbuflen, outbuf, outbuflen); - - /* update our left padding with end of current input, for next run. */ - SDL_memcpy((lpadding + paddingsamples) - (cpy / sizeof (float)), inbufend - cpy, cpy); - return retval; -} - -static void -SDL_ResetAudioStreamResampler(SDL_AudioStream *stream) -{ - /* set all the padding to silence. */ - const int len = stream->resampler_padding_samples; - SDL_memset(stream->resampler_state, '\0', len * sizeof (float)); -} - -static void -SDL_CleanupAudioStreamResampler(SDL_AudioStream *stream) -{ - SDL_free(stream->resampler_state); -} - -SDL_AudioStream * -SDL_NewAudioStream(const SDL_AudioFormat src_format, - const Uint8 src_channels, - const int src_rate, - const SDL_AudioFormat dst_format, - const Uint8 dst_channels, - const int dst_rate) -{ - const int packetlen = 4096; /* !!! FIXME: good enough for now. */ - Uint8 pre_resample_channels; - SDL_AudioStream *retval; - - retval = (SDL_AudioStream *) SDL_calloc(1, sizeof (SDL_AudioStream)); - if (!retval) { - return NULL; - } - - /* If increasing channels, do it after resampling, since we'd just - do more work to resample duplicate channels. If we're decreasing, do - it first so we resample the interpolated data instead of interpolating - the resampled data (!!! FIXME: decide if that works in practice, though!). */ - pre_resample_channels = SDL_min(src_channels, dst_channels); - - retval->first_run = SDL_TRUE; - retval->src_sample_frame_size = (SDL_AUDIO_BITSIZE(src_format) / 8) * src_channels; - retval->src_format = src_format; - retval->src_channels = src_channels; - retval->src_rate = src_rate; - retval->dst_sample_frame_size = (SDL_AUDIO_BITSIZE(dst_format) / 8) * dst_channels; - retval->dst_format = dst_format; - retval->dst_channels = dst_channels; - retval->dst_rate = dst_rate; - retval->pre_resample_channels = pre_resample_channels; - retval->packetlen = packetlen; - retval->rate_incr = ((double) dst_rate) / ((double) src_rate); - retval->resampler_padding_samples = ResamplerPadding(retval->src_rate, retval->dst_rate) * pre_resample_channels; - retval->resampler_padding = (float *) SDL_calloc(retval->resampler_padding_samples ? retval->resampler_padding_samples : 1, sizeof (float)); - - if (retval->resampler_padding == NULL) { - SDL_FreeAudioStream(retval); - SDL_OutOfMemory(); - return NULL; - } - - retval->staging_buffer_size = ((retval->resampler_padding_samples / retval->pre_resample_channels) * retval->src_sample_frame_size); - if (retval->staging_buffer_size > 0) { - retval->staging_buffer = (Uint8 *) SDL_malloc(retval->staging_buffer_size); - if (retval->staging_buffer == NULL) { - SDL_FreeAudioStream(retval); - SDL_OutOfMemory(); - return NULL; - } - } - - /* Not resampling? It's an easy conversion (and maybe not even that!) */ - if (src_rate == dst_rate) { - retval->cvt_before_resampling.needed = SDL_FALSE; - if (SDL_BuildAudioCVT(&retval->cvt_after_resampling, src_format, src_channels, dst_rate, dst_format, dst_channels, dst_rate) < 0) { - SDL_FreeAudioStream(retval); - return NULL; /* SDL_BuildAudioCVT should have called SDL_SetError. */ - } - } else { - /* Don't resample at first. Just get us to Float32 format. */ - /* !!! FIXME: convert to int32 on devices without hardware float. */ - if (SDL_BuildAudioCVT(&retval->cvt_before_resampling, src_format, src_channels, src_rate, AUDIO_F32SYS, pre_resample_channels, src_rate) < 0) { - SDL_FreeAudioStream(retval); - return NULL; /* SDL_BuildAudioCVT should have called SDL_SetError. */ - } - -#ifdef HAVE_LIBSAMPLERATE_H - SetupLibSampleRateResampling(retval); -#endif - - if (!retval->resampler_func) { - retval->resampler_state = SDL_calloc(retval->resampler_padding_samples, sizeof (float)); - if (!retval->resampler_state) { - SDL_FreeAudioStream(retval); - SDL_OutOfMemory(); - return NULL; - } - - if (SDL_PrepareResampleFilter() < 0) { - SDL_free(retval->resampler_state); - retval->resampler_state = NULL; - SDL_FreeAudioStream(retval); - return NULL; - } - - retval->resampler_func = SDL_ResampleAudioStream; - retval->reset_resampler_func = SDL_ResetAudioStreamResampler; - retval->cleanup_resampler_func = SDL_CleanupAudioStreamResampler; - } - - /* Convert us to the final format after resampling. */ - if (SDL_BuildAudioCVT(&retval->cvt_after_resampling, AUDIO_F32SYS, pre_resample_channels, dst_rate, dst_format, dst_channels, dst_rate) < 0) { - SDL_FreeAudioStream(retval); - return NULL; /* SDL_BuildAudioCVT should have called SDL_SetError. */ - } - } - - retval->queue = SDL_NewDataQueue(packetlen, packetlen * 2); - if (!retval->queue) { - SDL_FreeAudioStream(retval); - return NULL; /* SDL_NewDataQueue should have called SDL_SetError. */ - } - - return retval; -} - -static int -SDL_AudioStreamPutInternal(SDL_AudioStream *stream, const void *buf, int len, int *maxputbytes) -{ - int buflen = len; - int workbuflen; - Uint8 *workbuf; - Uint8 *resamplebuf = NULL; - int resamplebuflen = 0; - int neededpaddingbytes; - int paddingbytes; - - /* !!! FIXME: several converters can take advantage of SIMD, but only - !!! FIXME: if the data is aligned to 16 bytes. EnsureStreamBufferSize() - !!! FIXME: guarantees the buffer will align, but the - !!! FIXME: converters will iterate over the data backwards if - !!! FIXME: the output grows, and this means we won't align if buflen - !!! FIXME: isn't a multiple of 16. In these cases, we should chop off - !!! FIXME: a few samples at the end and convert them separately. */ - - /* no padding prepended on first run. */ - neededpaddingbytes = stream->resampler_padding_samples * sizeof (float); - paddingbytes = stream->first_run ? 0 : neededpaddingbytes; - stream->first_run = SDL_FALSE; - - /* Make sure the work buffer can hold all the data we need at once... */ - workbuflen = buflen; - if (stream->cvt_before_resampling.needed) { - workbuflen *= stream->cvt_before_resampling.len_mult; - } - - if (stream->dst_rate != stream->src_rate) { - /* resamples can't happen in place, so make space for second buf. */ - const int framesize = stream->pre_resample_channels * sizeof (float); - const int frames = workbuflen / framesize; - resamplebuflen = ((int) SDL_ceil(frames * stream->rate_incr)) * framesize; - #if DEBUG_AUDIOSTREAM - printf("AUDIOSTREAM: will resample %d bytes to %d (ratio=%.6f)\n", workbuflen, resamplebuflen, stream->rate_incr); - #endif - workbuflen += resamplebuflen; - } - - if (stream->cvt_after_resampling.needed) { - /* !!! FIXME: buffer might be big enough already? */ - workbuflen *= stream->cvt_after_resampling.len_mult; - } - - workbuflen += neededpaddingbytes; - - #if DEBUG_AUDIOSTREAM - printf("AUDIOSTREAM: Putting %d bytes of preconverted audio, need %d byte work buffer\n", buflen, workbuflen); - #endif - - workbuf = EnsureStreamBufferSize(stream, workbuflen); - if (!workbuf) { - return -1; /* probably out of memory. */ - } - - resamplebuf = workbuf; /* default if not resampling. */ - - SDL_memcpy(workbuf + paddingbytes, buf, buflen); - - if (stream->cvt_before_resampling.needed) { - stream->cvt_before_resampling.buf = workbuf + paddingbytes; - stream->cvt_before_resampling.len = buflen; - if (SDL_ConvertAudio(&stream->cvt_before_resampling) == -1) { - return -1; /* uhoh! */ - } - buflen = stream->cvt_before_resampling.len_cvt; - - #if DEBUG_AUDIOSTREAM - printf("AUDIOSTREAM: After initial conversion we have %d bytes\n", buflen); - #endif - } - - if (stream->dst_rate != stream->src_rate) { - /* save off some samples at the end; they are used for padding now so - the resampler is coherent and then used at the start of the next - put operation. Prepend last put operation's padding, too. */ - - /* prepend prior put's padding. :P */ - if (paddingbytes) { - SDL_memcpy(workbuf, stream->resampler_padding, paddingbytes); - buflen += paddingbytes; - } - - /* save off the data at the end for the next run. */ - SDL_memcpy(stream->resampler_padding, workbuf + (buflen - neededpaddingbytes), neededpaddingbytes); - - resamplebuf = workbuf + buflen; /* skip to second piece of workbuf. */ - SDL_assert(buflen >= neededpaddingbytes); - if (buflen > neededpaddingbytes) { - buflen = stream->resampler_func(stream, workbuf, buflen - neededpaddingbytes, resamplebuf, resamplebuflen); - } else { - buflen = 0; - } - - #if DEBUG_AUDIOSTREAM - printf("AUDIOSTREAM: After resampling we have %d bytes\n", buflen); - #endif - } - - if (stream->cvt_after_resampling.needed && (buflen > 0)) { - stream->cvt_after_resampling.buf = resamplebuf; - stream->cvt_after_resampling.len = buflen; - if (SDL_ConvertAudio(&stream->cvt_after_resampling) == -1) { - return -1; /* uhoh! */ - } - buflen = stream->cvt_after_resampling.len_cvt; - - #if DEBUG_AUDIOSTREAM - printf("AUDIOSTREAM: After final conversion we have %d bytes\n", buflen); - #endif - } - - #if DEBUG_AUDIOSTREAM - printf("AUDIOSTREAM: Final output is %d bytes\n", buflen); - #endif - - if (maxputbytes) { - const int maxbytes = *maxputbytes; - if (buflen > maxbytes) - buflen = maxbytes; - *maxputbytes -= buflen; - } - - /* resamplebuf holds the final output, even if we didn't resample. */ - return buflen ? SDL_WriteToDataQueue(stream->queue, resamplebuf, buflen) : 0; -} - -int -SDL_AudioStreamPut(SDL_AudioStream *stream, const void *buf, int len) -{ - /* !!! FIXME: several converters can take advantage of SIMD, but only - !!! FIXME: if the data is aligned to 16 bytes. EnsureStreamBufferSize() - !!! FIXME: guarantees the buffer will align, but the - !!! FIXME: converters will iterate over the data backwards if - !!! FIXME: the output grows, and this means we won't align if buflen - !!! FIXME: isn't a multiple of 16. In these cases, we should chop off - !!! FIXME: a few samples at the end and convert them separately. */ - - #if DEBUG_AUDIOSTREAM - printf("AUDIOSTREAM: wants to put %d preconverted bytes\n", buflen); - #endif - - if (!stream) { - return SDL_InvalidParamError("stream"); - } else if (!buf) { - return SDL_InvalidParamError("buf"); - } else if (len == 0) { - return 0; /* nothing to do. */ - } else if ((len % stream->src_sample_frame_size) != 0) { - return SDL_SetError("Can't add partial sample frames"); - } - - if (!stream->cvt_before_resampling.needed && - (stream->dst_rate == stream->src_rate) && - !stream->cvt_after_resampling.needed) { - #if DEBUG_AUDIOSTREAM - printf("AUDIOSTREAM: no conversion needed at all, queueing %d bytes.\n", len); - #endif - return SDL_WriteToDataQueue(stream->queue, buf, len); - } - - while (len > 0) { - int amount; - - /* If we don't have a staging buffer or we're given enough data that - we don't need to store it for later, skip the staging process. - */ - if (!stream->staging_buffer_filled && len >= stream->staging_buffer_size) { - return SDL_AudioStreamPutInternal(stream, buf, len, NULL); - } - - /* If there's not enough data to fill the staging buffer, just save it */ - if ((stream->staging_buffer_filled + len) < stream->staging_buffer_size) { - SDL_memcpy(stream->staging_buffer + stream->staging_buffer_filled, buf, len); - stream->staging_buffer_filled += len; - return 0; - } - - /* Fill the staging buffer, process it, and continue */ - amount = (stream->staging_buffer_size - stream->staging_buffer_filled); - SDL_assert(amount > 0); - SDL_memcpy(stream->staging_buffer + stream->staging_buffer_filled, buf, amount); - stream->staging_buffer_filled = 0; - if (SDL_AudioStreamPutInternal(stream, stream->staging_buffer, stream->staging_buffer_size, NULL) < 0) { - return -1; - } - buf = (void *)((Uint8 *)buf + amount); - len -= amount; - } - return 0; -} - -int SDL_AudioStreamFlush(SDL_AudioStream *stream) -{ - if (!stream) { - return SDL_InvalidParamError("stream"); - } - - #if DEBUG_AUDIOSTREAM - printf("AUDIOSTREAM: flushing! staging_buffer_filled=%d bytes\n", stream->staging_buffer_filled); - #endif - - /* shouldn't use a staging buffer if we're not resampling. */ - SDL_assert((stream->dst_rate != stream->src_rate) || (stream->staging_buffer_filled == 0)); - - if (stream->staging_buffer_filled > 0) { - /* push the staging buffer + silence. We need to flush out not just - the staging buffer, but the piece that the stream was saving off - for right-side resampler padding. */ - const SDL_bool first_run = stream->first_run; - const int filled = stream->staging_buffer_filled; - int actual_input_frames = filled / stream->src_sample_frame_size; - if (!first_run) - actual_input_frames += stream->resampler_padding_samples / stream->pre_resample_channels; - - if (actual_input_frames > 0) { /* don't bother if nothing to flush. */ - /* This is how many bytes we're expecting without silence appended. */ - int flush_remaining = ((int) SDL_ceil(actual_input_frames * stream->rate_incr)) * stream->dst_sample_frame_size; - - #if DEBUG_AUDIOSTREAM - printf("AUDIOSTREAM: flushing with padding to get max %d bytes!\n", flush_remaining); - #endif - - SDL_memset(stream->staging_buffer + filled, '\0', stream->staging_buffer_size - filled); - if (SDL_AudioStreamPutInternal(stream, stream->staging_buffer, stream->staging_buffer_size, &flush_remaining) < 0) { - return -1; - } - - /* we have flushed out (or initially filled) the pending right-side - resampler padding, but we need to push more silence to guarantee - the staging buffer is fully flushed out, too. */ - SDL_memset(stream->staging_buffer, '\0', filled); - if (SDL_AudioStreamPutInternal(stream, stream->staging_buffer, stream->staging_buffer_size, &flush_remaining) < 0) { - return -1; - } - } - } - - stream->staging_buffer_filled = 0; - stream->first_run = SDL_TRUE; - - return 0; -} - -/* get converted/resampled data from the stream */ -int -SDL_AudioStreamGet(SDL_AudioStream *stream, void *buf, int len) -{ - #if DEBUG_AUDIOSTREAM - printf("AUDIOSTREAM: want to get %d converted bytes\n", len); - #endif - - if (!stream) { - return SDL_InvalidParamError("stream"); - } else if (!buf) { - return SDL_InvalidParamError("buf"); - } else if (len <= 0) { - return 0; /* nothing to do. */ - } else if ((len % stream->dst_sample_frame_size) != 0) { - return SDL_SetError("Can't request partial sample frames"); - } - - return (int) SDL_ReadFromDataQueue(stream->queue, buf, len); -} - -/* number of converted/resampled bytes available */ -int -SDL_AudioStreamAvailable(SDL_AudioStream *stream) -{ - return stream ? (int) SDL_CountDataQueue(stream->queue) : 0; -} - -void -SDL_AudioStreamClear(SDL_AudioStream *stream) -{ - if (!stream) { - SDL_InvalidParamError("stream"); - } else { - SDL_ClearDataQueue(stream->queue, stream->packetlen * 2); - if (stream->reset_resampler_func) { - stream->reset_resampler_func(stream); - } - stream->first_run = SDL_TRUE; - stream->staging_buffer_filled = 0; - } -} - -/* dispose of a stream */ -void -SDL_FreeAudioStream(SDL_AudioStream *stream) -{ - if (stream) { - if (stream->cleanup_resampler_func) { - stream->cleanup_resampler_func(stream); - } - SDL_FreeDataQueue(stream->queue); - SDL_free(stream->staging_buffer); - SDL_free(stream->work_buffer_base); - SDL_free(stream->resampler_padding); - SDL_free(stream); - } -} - -/* vi: set ts=4 sw=4 expandtab: */ - |