audio: allow stereo Sint16 resampling fast path in SDL_AudioStream.
authorRyan C. Gordon <icculus@icculus.org>
Tue, 24 Jan 2017 00:08:24 -0500
changeset 1084230b37eaf3b3c
parent 10841 b9d6a3d65394
child 10843 31c9dede7b9c
audio: allow stereo Sint16 resampling fast path in SDL_AudioStream.

This currently favors libsamplerate over the fast path (quality over speed),
but I'm not sure that's the correct approach, as there may be surprising
changes in performance metrics depending on what packages are available on
a user's system. That being said, currently, the only thing with access to
SDL_AudioStream is an SDL audio device's thread, and it might be mostly idle
otherwise, so maybe this is generally good.
src/audio/SDL_audiocvt.c
     1.1 --- a/src/audio/SDL_audiocvt.c	Tue Jan 24 00:03:36 2017 -0500
     1.2 +++ b/src/audio/SDL_audiocvt.c	Tue Jan 24 00:08:24 2017 -0500
     1.3 @@ -858,7 +858,7 @@
     1.4      return (cvt->needed);
     1.5  }
     1.6  
     1.7 -typedef int (*SDL_ResampleAudioStreamFunc)(SDL_AudioStream *stream, const float *inbuf, const int inbuflen, float *outbuf, const int outbuflen);
     1.8 +typedef int (*SDL_ResampleAudioStreamFunc)(SDL_AudioStream *stream, const void *inbuf, const int inbuflen, void *outbuf, const int outbuflen);
     1.9  typedef void (*SDL_ResetAudioStreamResamplerFunc)(SDL_AudioStream *stream);
    1.10  typedef void (*SDL_CleanupAudioStreamResamplerFunc)(SDL_AudioStream *stream);
    1.11  
    1.12 @@ -890,8 +890,10 @@
    1.13  
    1.14  #ifdef HAVE_LIBSAMPLERATE_H
    1.15  static int
    1.16 -SDL_ResampleAudioStream_SRC(SDL_AudioStream *stream, const float *inbuf, const int inbuflen, float *outbuf, const int outbuflen)
    1.17 +SDL_ResampleAudioStream_SRC(SDL_AudioStream *stream, const void *_inbuf, const int inbuflen, void *_outbuf, const int outbuflen)
    1.18  {
    1.19 +    const float *inbuf = (const float *) _inbuf;
    1.20 +    float *outbuf = (float *) _outbuf;
    1.21      const int framelen = sizeof(float) * stream->pre_resample_channels;
    1.22      SRC_STATE *state = (SRC_STATE *)stream->resampler_state;
    1.23      SRC_DATA data;
    1.24 @@ -970,26 +972,48 @@
    1.25  typedef struct
    1.26  {
    1.27      SDL_bool resampler_seeded;
    1.28 -    float resampler_state[8];
    1.29 +    union
    1.30 +    {
    1.31 +        float f[8];
    1.32 +        Sint16 si16[2];
    1.33 +    } resampler_state;
    1.34  } SDL_AudioStreamResamplerState;
    1.35  
    1.36  static int
    1.37 -SDL_ResampleAudioStream(SDL_AudioStream *stream, const float *inbuf, const int inbuflen, float *outbuf, const int outbuflen)
    1.38 +SDL_ResampleAudioStream(SDL_AudioStream *stream, const void *_inbuf, const int inbuflen, void *_outbuf, const int outbuflen)
    1.39  {
    1.40 +    const float *inbuf = (const float *) _inbuf;
    1.41 +    float *outbuf = (float *) _outbuf;
    1.42      SDL_AudioStreamResamplerState *state = (SDL_AudioStreamResamplerState*)stream->resampler_state;
    1.43      const int chans = (int)stream->pre_resample_channels;
    1.44  
    1.45 -    SDL_assert(chans <= SDL_arraysize(state->resampler_state));
    1.46 +    SDL_assert(chans <= SDL_arraysize(state->resampler_state.f));
    1.47  
    1.48      if (!state->resampler_seeded) {
    1.49 -        int i;
    1.50 -        for (i = 0; i < chans; i++) {
    1.51 -            state->resampler_state[i] = inbuf[i];
    1.52 -        }
    1.53 +        SDL_memcpy(state->resampler_state.f, inbuf, chans * sizeof (float));
    1.54          state->resampler_seeded = SDL_TRUE;
    1.55      }
    1.56  
    1.57 -    return SDL_ResampleAudioSimple(chans, stream->rate_incr, state->resampler_state, inbuf, inbuflen, outbuf, outbuflen);
    1.58 +    return SDL_ResampleAudioSimple(chans, stream->rate_incr, state->resampler_state.f, inbuf, inbuflen, outbuf, outbuflen);
    1.59 +}
    1.60 +
    1.61 +static int
    1.62 +SDL_ResampleAudioStream_si16_c2(SDL_AudioStream *stream, const void *_inbuf, const int inbuflen, void *_outbuf, const int outbuflen)
    1.63 +{
    1.64 +    const Sint16 *inbuf = (const Sint16 *) _inbuf;
    1.65 +    Sint16 *outbuf = (Sint16 *) _outbuf;
    1.66 +    SDL_AudioStreamResamplerState *state = (SDL_AudioStreamResamplerState*)stream->resampler_state;
    1.67 +    const int chans = (int)stream->pre_resample_channels;
    1.68 +
    1.69 +    SDL_assert(chans <= SDL_arraysize(state->resampler_state.si16));
    1.70 +
    1.71 +    if (!state->resampler_seeded) {
    1.72 +        state->resampler_state.si16[0] = inbuf[0];
    1.73 +        state->resampler_state.si16[1] = inbuf[1];
    1.74 +        state->resampler_seeded = SDL_TRUE;
    1.75 +    }
    1.76 +
    1.77 +    return SDL_ResampleAudioSimple_si16_c2(stream->rate_incr, state->resampler_state.si16, inbuf, inbuflen, outbuf, outbuflen);
    1.78  }
    1.79  
    1.80  static void
    1.81 @@ -1016,6 +1040,9 @@
    1.82      const int packetlen = 4096;  /* !!! FIXME: good enough for now. */
    1.83      Uint8 pre_resample_channels;
    1.84      SDL_AudioStream *retval;
    1.85 +#ifndef HAVE_LIBSAMPLERATE_H
    1.86 +    const SDL_bool SRC_available = SDL_FALSE;
    1.87 +#endif
    1.88  
    1.89      retval = (SDL_AudioStream *) SDL_calloc(1, sizeof (SDL_AudioStream));
    1.90      if (!retval) {
    1.91 @@ -1043,11 +1070,22 @@
    1.92      /* Not resampling? It's an easy conversion (and maybe not even that!). */
    1.93      if (src_rate == dst_rate) {
    1.94          retval->cvt_before_resampling.needed = SDL_FALSE;
    1.95 -        retval->cvt_before_resampling.len_mult = 1;
    1.96          if (SDL_BuildAudioCVT(&retval->cvt_after_resampling, src_format, src_channels, dst_rate, dst_format, dst_channels, dst_rate) < 0) {
    1.97              SDL_FreeAudioStream(retval);
    1.98              return NULL;  /* SDL_BuildAudioCVT should have called SDL_SetError. */
    1.99          }
   1.100 +    /* fast path special case for stereo Sint16 data that just needs resampling. */
   1.101 +    } else if ((!SRC_available) && (src_channels == 2) && (dst_channels == 2) && (src_format == AUDIO_S16SYS) && (dst_format == AUDIO_S16SYS)) {
   1.102 +        SDL_assert(src_rate != dst_rate);
   1.103 +        retval->resampler_state = SDL_calloc(1, sizeof(SDL_AudioStreamResamplerState));
   1.104 +        if (!retval->resampler_state) {
   1.105 +            SDL_FreeAudioStream(retval);
   1.106 +            SDL_OutOfMemory();
   1.107 +            return NULL;
   1.108 +        }
   1.109 +        retval->resampler_func = SDL_ResampleAudioStream_si16_c2;
   1.110 +        retval->reset_resampler_func = SDL_ResetAudioStreamResampler;
   1.111 +        retval->cleanup_resampler_func = SDL_CleanupAudioStreamResampler;
   1.112      } else {
   1.113          /* Don't resample at first. Just get us to Float32 format. */
   1.114          /* !!! FIXME: convert to int32 on devices without hardware float. */
   1.115 @@ -1136,16 +1174,16 @@
   1.116  
   1.117      if (stream->dst_rate != stream->src_rate) {
   1.118          const int workbuflen = buflen * ((int) SDL_ceil(stream->rate_incr));
   1.119 -        float *workbuf = (float *) EnsureBufferSize(&stream->resample_buffer, &stream->resample_buffer_len, workbuflen);
   1.120 +        void *workbuf = EnsureBufferSize(&stream->resample_buffer, &stream->resample_buffer_len, workbuflen);
   1.121          if (workbuf == NULL) {
   1.122              return -1;  /* probably out of memory. */
   1.123          }
   1.124 -        buflen = stream->resampler_func(stream, (float *) buf, buflen, workbuf, workbuflen);
   1.125 +        buflen = stream->resampler_func(stream, buf, buflen, workbuf, workbuflen);
   1.126          buf = workbuf;
   1.127      }
   1.128  
   1.129      if (stream->cvt_after_resampling.needed) {
   1.130 -        const int workbuflen = buflen * stream->cvt_before_resampling.len_mult;  /* will be "* 1" if not needed */
   1.131 +        const int workbuflen = buflen * stream->cvt_after_resampling.len_mult;  /* will be "* 1" if not needed */
   1.132          Uint8 *workbuf;
   1.133  
   1.134          if (buf == stream->resample_buffer) {