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Added libsoxr based resampling.

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This commit is contained in:
Armin Novak 2018-11-13 10:59:30 +01:00
parent e93aab2f7f
commit b362b61a1a
1 changed files with 38 additions and 45 deletions
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83
libfreerdp/codec/dsp.c
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@ -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);
error = soxr_process(context->sox, src, sframes, &idone,
Stream_Buffer(context->resample),
Stream_Capacity(context->resample)/rbytes, &odone);
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)]);
}
}
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))
@ -1071,6 +1047,9 @@ void freerdp_dsp_context_free(FREERDP_DSP_CONTEXT* context)
if (context->faac)
faacEncClose(context->faac);
#endif
#if defined(WITH_SOXR)
soxr_delete(context->sox);
#endif
free(context);
}
@ -1262,6 +1241,20 @@ BOOL freerdp_dsp_context_reset(FREERDP_DSP_CONTEXT* context, const AUDIO_FORMAT*
faacEncSetConfiguration(context->faac, cfg);
}
#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