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audio: Stream resampling now saves some samples from previous run for padding. 

Previously, the padding was silence, which was a problem when streaming since
you would sample a little bit of this silence between each buffer.

We still need a means to get padding data for the right hand side, but this
patch makes the resampler output more correct.
This commit is contained in:
Ryan C. Gordon 2017-09-22 07:42:24 -04:00
parent 466ba57d42
commit 6d206a7b28
1 changed files with 43 additions and 33 deletions
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74
src/audio/SDL_audiocvt.c
View File

@ -464,15 +464,21 @@ SDL_FreeResampleFilter(void)
ResamplerFilterDifference = NULL;
}
static int
ResamplerPadding(const int inrate, const int outrate)
{
return (inrate > outrate) ? (int) SDL_ceil(((float) (RESAMPLER_SAMPLES_PER_ZERO_CROSSING * inrate) / ((float) outrate))) : 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,
float *last_sample, const float *inbuf,
float *lpadding, float *rpadding, const float *inbuf,
const int inbuflen, float *outbuf, const int outbuflen)
{
const float outtimeincr = 1.0f / ((float) outrate);
const float ratio = ((float) outrate) / ((float) inrate);
/*const int padding_len = (ratio < 1.0f) ? (int) SDL_ceilf(((float) (RESAMPLER_SAMPLES_PER_ZERO_CROSSING * inrate) / ((float) outrate))) : RESAMPLER_SAMPLES_PER_ZERO_CROSSING;*/
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. */
@ -499,16 +505,16 @@ SDL_ResampleAudio(const int chans, const int inrate, const int outrate,
/* 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++) {
/* !!! FIXME: insample uses zero for padding samples, but it should use prior state from last_sample. */
const int srcframe = srcindex - j;
const float insample = (srcframe < 0) ? 0.0f : inbuf[(srcframe * chans) + chan]; /* !!! FIXME: we can bubble this conditional out of here by doing a pre loop. */
/* !!! 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 += (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: insample uses zero for padding samples, but it should use prior state from last_sample. */
const float insample = (srcframe >= inframes) ? 0.0f : inbuf[(srcframe * chans) + chan]; /* !!! FIXME: we can bubble this conditional out of here by doing a post loop. */
/* !!! 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 += (insample * (ResamplerFilter[filterindex2 + (j * RESAMPLER_SAMPLES_PER_ZERO_CROSSING)] + (interpolation2 * ResamplerFilterDifference[filterindex2 + (j * RESAMPLER_SAMPLES_PER_ZERO_CROSSING)])));
}
*(dst++) = outsample;
@ -693,8 +699,8 @@ 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 srcrate = (int) (size_t) cvt->filters[SDL_AUDIOCVT_MAX_FILTERS-1];
const int dstrate = (int) (size_t) cvt->filters[SDL_AUDIOCVT_MAX_FILTERS];
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;
@ -702,13 +708,15 @@ SDL_ResampleCVT(SDL_AudioCVT *cvt, const int chans, const SDL_AudioFormat format
/* !!! 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;
float state[8];
const int paddingsamples = (ResamplerPadding(inrate, outrate) * chans);
float *padding = SDL_stack_alloc(float, paddingsamples);
SDL_assert(format == AUDIO_F32SYS);
SDL_zero(state);
/* we keep no streaming state here, so pad with silence on both ends. */
SDL_memset(padding, '\0', paddingsamples * sizeof (float));
cvt->len_cvt = SDL_ResampleAudio(chans, srcrate, dstrate, state, src, srclen, dst, dstlen);
cvt->len_cvt = SDL_ResampleAudio(chans, inrate, outrate, padding, padding, src, srclen, dst, dstlen);
SDL_memcpy(cvt->buf, dst, cvt->len_cvt); /* !!! FIXME: remove this if we can get the resampler to work in-place again. */
@ -1195,25 +1203,19 @@ SetupLibSampleRateResampling(SDL_AudioStream *stream)
#endif /* HAVE_LIBSAMPLERATE_H */
typedef struct
{
SDL_bool resampler_seeded;
union
{
float f[8];
Sint16 si16[2];
} resampler_state;
} SDL_AudioStreamResamplerState;
static int
SDL_ResampleAudioStream(SDL_AudioStream *stream, const void *_inbuf, const int inbuflen, void *_outbuf, const int outbuflen)
{
const float *inbuf = (const float *) _inbuf;
float *outbuf = (float *) _outbuf;
SDL_AudioStreamResamplerState *state = (SDL_AudioStreamResamplerState*)stream->resampler_state;
const int chans = (int) stream->pre_resample_channels;
SDL_assert(chans <= SDL_arraysize(state->resampler_state.f));
const int inrate = stream->src_rate;
const int outrate = stream->dst_rate;
const int paddingsamples = ResamplerPadding(inrate, outrate) * chans;
const int paddingbytes = paddingsamples * sizeof (float);
float *lpadding = (float *) stream->resampler_state;
float *rpadding = SDL_stack_alloc(float, paddingsamples);
int retval;
if (inbuf == ((const float *) outbuf)) { /* !!! FIXME can't work in-place (for now!). */
Uint8 *ptr = EnsureStreamBufferSize(stream, inbuflen + outbuflen);
@ -1226,19 +1228,25 @@ SDL_ResampleAudioStream(SDL_AudioStream *stream, const void *_inbuf, const int i
outbuf = (float *) ptr;
}
if (!state->resampler_seeded) {
SDL_zero(state->resampler_state.f);
state->resampler_seeded = SDL_TRUE;
}
/* !!! FIXME: streaming current resamples on Put, because of probably good reasons I can't remember right now, but if we resample on Get, we'd be able to access legit right padding values. */
SDL_memset(rpadding, '\0', paddingbytes);
retval = SDL_ResampleAudio(chans, inrate, outrate, lpadding, rpadding, inbuf, inbuflen, outbuf, outbuflen);
return SDL_ResampleAudio(chans, stream->src_rate, stream->dst_rate, state->resampler_state.f, inbuf, inbuflen, outbuf, outbuflen);
/* update our left padding with end of current input, for next run. */
SDL_memcpy(lpadding, ((const Uint8 *) inbuf) + (inbuflen - paddingbytes), paddingbytes);
return retval;
}
static void
SDL_ResetAudioStreamResampler(SDL_AudioStream *stream)
{
SDL_AudioStreamResamplerState *state = (SDL_AudioStreamResamplerState*)stream->resampler_state;
state->resampler_seeded = SDL_FALSE;
/* set all the left padding to silence. */
const int inrate = stream->src_rate;
const int outrate = stream->dst_rate;
const int chans = (int) stream->pre_resample_channels;
const int len = ResamplerPadding(inrate, outrate) * chans;
SDL_memset(stream->resampler_state, '\0', len * sizeof (float));
}
static void
@ -1302,7 +1310,9 @@ SDL_NewAudioStream(const SDL_AudioFormat src_format,
#endif
if (!retval->resampler_func) {
retval->resampler_state = SDL_calloc(1, sizeof(SDL_AudioStreamResamplerState));
const int chans = (int) pre_resample_channels;
const int len = ResamplerPadding(src_rate, dst_rate) * chans;
retval->resampler_state = SDL_calloc(len, sizeof (float));
if (!retval->resampler_state) {
SDL_FreeAudioStream(retval);
SDL_OutOfMemory();