diff options
Diffstat (limited to 'source/3rd-party/SDL2/src/audio/SDL_audiocvt.c')
| -rw-r--r-- | source/3rd-party/SDL2/src/audio/SDL_audiocvt.c | 1673 |
1 files changed, 1673 insertions, 0 deletions
diff --git a/source/3rd-party/SDL2/src/audio/SDL_audiocvt.c b/source/3rd-party/SDL2/src/audio/SDL_audiocvt.c new file mode 100644 index 0000000..ee0ba32 --- /dev/null +++ b/source/3rd-party/SDL2/src/audio/SDL_audiocvt.c @@ -0,0 +1,1673 @@ +/* + 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: */ + |