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Refactored freerdp_dsp_resample, checks now done internally.

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This commit is contained in:
Armin Novak 2018-11-09 10:15:14 +01:00
parent 83f959ce6b
commit ab15e34032
1 changed files with 50 additions and 18 deletions
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68
libfreerdp/codec/dsp.c
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@ -29,6 +29,7 @@
#include <winpr/crt.h>
#include <freerdp/types.h>
#include <freerdp/log.h>
#include <freerdp/codec/dsp.h>
#if !defined(WITH_DSP_FFMPEG)
@ -109,29 +110,65 @@ struct _FREERDP_DSP_CONTEXT
static BOOL freerdp_dsp_resample(FREERDP_DSP_CONTEXT* context,
const BYTE* src, size_t size,
const AUDIO_FORMAT* srcFormat)
const AUDIO_FORMAT* srcFormat,
const BYTE** data, size_t* length)
{
BYTE* p;
size_t sframes, rframes;
size_t rsize;
size_t i, j;
size_t sbytes, rbytes;
size_t srcBytePerFrame;
size_t srcBytesPerFrame, dstBytesPerFrame;
size_t srcChannels, dstChannels;
AUDIO_FORMAT format;
if (!context || !src || !srcFormat)
if (!context || !src || !srcFormat || !data || !length)
return FALSE;
assert(srcFormat->wFormatTag == WAVE_FORMAT_PCM);
srcBytePerFrame = (srcFormat->wBitsPerSample > 8) ? 2 : 1;
sbytes = srcFormat->nChannels * srcBytePerFrame;
if (srcFormat->wFormatTag != WAVE_FORMAT_PCM)
{
WLog_ERR(TAG, "%s requires %s for sample input, got %s", __FUNCTION__,
audio_format_get_tag_string(WAVE_FORMAT_PCM),
audio_format_get_tag_string(srcFormat->wFormatTag));
return FALSE;
}
srcChannels = srcFormat->nChannels;
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;
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 = srcBytePerFrame * context->format.nChannels;
rbytes = dstBytesPerFrame * dstChannels;
/* Integer rounding correct division */
rframes = (sframes * context->format.nSamplesPerSec + (srcFormat->nSamplesPerSec + 1) / 2) /
srcFormat->nSamplesPerSec;
rsize = rbytes * rframes;
rsize = rframes * rbytes;
if (!Stream_EnsureCapacity(context->resample, rsize + 1024))
if (!Stream_EnsureCapacity(context->resample, rsize))
return FALSE;
p = Stream_Buffer(context->resample);
@ -159,6 +196,8 @@ static BOOL freerdp_dsp_resample(FREERDP_DSP_CONTEXT* context,
Stream_SetPointer(context->resample, p);
Stream_SealLength(context->resample);
*data = Stream_Buffer(context->resample);
*length = Stream_Length(context->resample);
return TRUE;
}
@ -1049,14 +1088,8 @@ BOOL freerdp_dsp_encode(FREERDP_DSP_CONTEXT* context, const AUDIO_FORMAT* srcFor
if (!context || !context->encoder || !srcFormat || !data || !out)
return FALSE;
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);
}
if (!freerdp_dsp_resample(context, data, length, srcFormat, &data, &length))
return FALSE;
switch (context->format.wFormatTag)
{
@ -1196,7 +1229,6 @@ BOOL freerdp_dsp_supports_format(const AUDIO_FORMAT* format, BOOL encode)
#endif
}
BOOL freerdp_dsp_context_reset(FREERDP_DSP_CONTEXT* context, const AUDIO_FORMAT* targetFormat)
{
#if defined(WITH_DSP_FFMPEG)