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Added resampling to freerdp_dsp_encode.

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This commit is contained in:
Armin Novak 2018-11-07 14:14:15 +01:00
parent e980d20e8e
commit e8e4800c5e
1 changed files with 34 additions and 15 deletions
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49
libfreerdp/codec/dsp.c
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@ -21,6 +21,7 @@
#include "config.h"
#endif
#include <assert.h>
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
@ -104,19 +105,27 @@ struct _FREERDP_DSP_CONTEXT
*/
static BOOL freerdp_dsp_resample(FREERDP_DSP_CONTEXT* context,
const BYTE* src, size_t bytes_per_sample,
UINT32 schan, UINT32 srate, size_t sframes,
UINT32 rchan, UINT32 rrate)
const BYTE* src, size_t size,
const AUDIO_FORMAT* srcFormat)
{
BYTE* p;
int rframes;
int rsize;
int i, j;
int n1, n2;
int sbytes, rbytes;
sbytes = bytes_per_sample * schan;
rbytes = bytes_per_sample * rchan;
rframes = sframes * rrate / srate;
size_t sframes, rframes;
size_t rsize;
size_t i, j;
size_t sbytes, rbytes;
size_t srcBytePerFrame;
if (!context || !src || !srcFormat)
return FALSE;
assert(srcFormat->wFormatTag == WAVE_FORMAT_PCM);
srcBytePerFrame = (srcFormat->wBitsPerSample > 8) ? 2 : 1;
sbytes = srcFormat->nChannels * srcBytePerFrame;
sframes = size / sbytes;
rbytes = srcBytePerFrame * context->format.nChannels;
/* Integer rounding correct division */
rframes = (sframes * context->format.nSamplesPerSec + (srcFormat->nSamplesPerSec + 1) / 2) /
srcFormat->nSamplesPerSec;
rsize = rbytes * rframes;
if (!Stream_EnsureCapacity(context->resample, rsize + 1024))
@ -126,17 +135,20 @@ static BOOL freerdp_dsp_resample(FREERDP_DSP_CONTEXT* context,
for (i = 0; i < rframes; i++)
{
n1 = i * srate / rrate;
size_t n2;
size_t n1 = (i * srcFormat->nSamplesPerSec) / context->format.nSamplesPerSec;
if (n1 >= sframes)
n1 = sframes - 1;
n2 = (n1 * rrate == i * srate || n1 == sframes - 1 ? n1 : n1 + 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 * srate - n1 * rrate > n2 * rrate - i * srate ?
*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)]);
}
@ -1032,7 +1044,14 @@ BOOL freerdp_dsp_encode(FREERDP_DSP_CONTEXT* context, const AUDIO_FORMAT* srcFor
if (!context || !context->encoder || !srcFormat || !data || !out)
return FALSE;
// TODO: Resample
if (srcFormat->nSamplesPerSec != context->format.nSamplesPerSec)
{
if (!freerdp_dsp_resample(context, data, length, srcFormat))
return FALSE;
data = Stream_Buffer(context->resample);
length = Stream_GetPosition(context->resample);
}
switch (context->format.wFormatTag)
{