Added libsoxr based resampling.

diff --git a/libfreerdp/codec/dsp.c b/libfreerdp/codec/dsp.c
index d5cbb6c..3a58009 100644 (file)
--- a/libfreerdp/codec/dsp.c
+++ b/libfreerdp/codec/dsp.c
@@ -49,6 +49,10 @@
 #include <faac.h>
 #endif
 
+#if defined(WITH_SOXR)
+#include <soxr.h>
+#endif
+
 #else
 #include "dsp_ffmpeg.h"
 #endif
@@ -101,6 +105,10 @@ struct _FREERDP_DSP_CONTEXT
        unsigned long faacInputSamples;
        unsigned long faacMaxOutputBytes;
 #endif
+
+#if defined(WITH_SOXR)
+       soxr_t sox;
+#endif
 };
 
 /**
@@ -113,10 +121,11 @@ static BOOL freerdp_dsp_resample(FREERDP_DSP_CONTEXT* context,
                                  const AUDIO_FORMAT* srcFormat,
                                  const BYTE** data, size_t* length)
 {
-       BYTE* p;
+       soxr_error_t error;
+       size_t idone, odone;
        size_t sframes, rframes;
        size_t rsize;
-       size_t i, j;
+       size_t j;
        size_t sbytes, rbytes;
        size_t srcBytesPerFrame, dstBytesPerFrame;
        size_t srcChannels, dstChannels;
@@ -137,6 +146,7 @@ static BOOL freerdp_dsp_resample(FREERDP_DSP_CONTEXT* context,
        dstChannels = context->format.nChannels;
        srcBytesPerFrame = (srcFormat->wBitsPerSample > 8) ? 2 : 1;
        dstBytesPerFrame = (context->format.wBitsPerSample > 8) ? 2 : 1;
+
        /* We want to ignore differences of source and destination format. */
        format = *srcFormat;
        format.wFormatTag = WAVE_FORMAT_UNKNOWN;
@@ -144,22 +154,6 @@ static BOOL freerdp_dsp_resample(FREERDP_DSP_CONTEXT* context,
        if (audio_format_compatible(&format, &context->format))
                return TRUE;
 
-       /* TODO: Currently sample size conversion is not implemented. */
-       if (srcFormat->wBitsPerSample != context->format.wBitsPerSample)
-       {
-               WLog_ERR(TAG, "%s does not support changing sample bit width [in=%"PRId16", out=%"PRId16"]",
-                        __FUNCTION__, srcFormat->wBitsPerSample, context->format.wBitsPerSample);
-               return FALSE;
-       }
-
-       /* TODO: Currently sample size conversion is not implemented. */
-       if (srcChannels != dstChannels)
-       {
-               WLog_ERR(TAG, "%s does not support changing number of channels [in=%"PRIdz", out=%"PRIdz"]",
-                        __FUNCTION__, srcChannels, dstChannels);
-               return FALSE;
-       }
-
        sbytes = srcChannels * srcBytesPerFrame;
        sframes = size / sbytes;
        rbytes = dstBytesPerFrame * dstChannels;
@@ -171,34 +165,14 @@ static BOOL freerdp_dsp_resample(FREERDP_DSP_CONTEXT* context,
        if (!Stream_EnsureCapacity(context->resample, rsize))
                return FALSE;
 
-       p = Stream_Buffer(context->resample);
-
-       for (i = 0; i < rframes; i++)
-       {
-               size_t n2;
-               size_t n1 = (i * srcFormat->nSamplesPerSec) / context->format.nSamplesPerSec;
-
-               if (n1 >= sframes)
-                       n1 = sframes - 1;
-
-               n2 = ((n1 * context->format.nSamplesPerSec == i * srcFormat->nSamplesPerSec) ||
-                     ((n1 == sframes - 1) ? n1 : n1 + 1));
-
-               for (j = 0; j < rbytes; j++)
-               {
-                       /* Nearest Interpolation, probably the easiest, but works */
-                       *p++ = (i * srcFormat->nSamplesPerSec - n1 * context->format.nSamplesPerSec > n2 *
-                               context->format.nSamplesPerSec - i * srcFormat->nSamplesPerSec ?
-                               src[n2 * sbytes + (j % sbytes)] :
-                               src[n1 * sbytes + (j % sbytes)]);
-               }
-       }
+       error = soxr_process(context->sox, src, sframes, &idone,
+                                       Stream_Buffer(context->resample),
+                                       Stream_Capacity(context->resample)/rbytes, &odone);
 
-       Stream_SetPointer(context->resample, p);
-       Stream_SealLength(context->resample);
+       Stream_SetLength(context->resample, odone * rbytes);
        *data = Stream_Buffer(context->resample);
        *length = Stream_Length(context->resample);
-       return TRUE;
+       return (error == 0) ? TRUE : FALSE;
 }
 
 /**
@@ -465,16 +439,18 @@ static BOOL freerdp_dsp_encode_faac(FREERDP_DSP_CONTEXT* context,
 {
        int16_t* inSamples = (int16_t*)src;
        int32_t* outSamples;
+       unsigned int bpp;
        unsigned int nrSamples, x;
        int rc;
 
        if (!context || !src || !out)
                return FALSE;
 
-       nrSamples = size / context->format.nChannels / context->format.wBitsPerSample / 8;
+       bpp = context->format.wBitsPerSample / 8 * context->format.nChannels;
+       nrSamples = size / bpp;
 
        if (!Stream_EnsureCapacity(context->buffer,
-                                  context->faacInputSamples * sizeof(int32_t) * context->format.nChannels))
+                                  nrSamples * sizeof(int32_t) * context->format.nChannels))
                return FALSE;
 
        if (!Stream_EnsureRemainingCapacity(out, context->faacMaxOutputBytes))
@@ -1072,6 +1048,9 @@ void freerdp_dsp_context_free(FREERDP_DSP_CONTEXT* context)
                        faacEncClose(context->faac);
 
 #endif
+#if defined(WITH_SOXR)
+               soxr_delete(context->sox);
+#endif
                free(context);
        }
 
@@ -1263,6 +1242,20 @@ BOOL freerdp_dsp_context_reset(FREERDP_DSP_CONTEXT* context, const AUDIO_FORMAT*
        }
 
 #endif
+#if defined(WITH_SOXR)
+       {
+               soxr_io_spec_t iospec = soxr_io_spec(SOXR_INT16, SOXR_INT16);
+               soxr_error_t error;
+               soxr_delete(context->sox);
+
+               context->sox = soxr_create(context->format.nSamplesPerSec, targetFormat->nSamplesPerSec,
+                                          targetFormat->nChannels, &error, &iospec, NULL, NULL);
+
+               if (!context->sox || (error != 0))
+                       return FALSE;
+
+       }
+#endif
        return TRUE;
 #endif
 }