FaSTTUBe/GST-Tensordecoder-ov_ep
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audio-converter: add output size argument

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Make it possible to have a different number of output samples than input
samples when we, for example, want to add resampling later.
This commit is contained in:
Wim Taymans 2015-11-10 09:52:24 +01:00
parent 629b63d1f2
commit ff6d1a2a25
3 changed files with 24 additions and 18 deletions
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34
gst-libs/gst/audio/audio-converter.c
View File

@ -76,8 +76,7 @@ ensure_debug_category (void)
typedef void (*AudioConvertFunc) (gpointer dst, const gpointer src, gint count); typedef void (*AudioConvertFunc) (gpointer dst, const gpointer src, gint count);
/** /* int/int int/float float/int float/float
* int/int int/float float/int float/float
* *
* unpack S32 S32 F64 F64 * unpack S32 S32 F64 F64
* convert S32->F64 * convert S32->F64
@ -377,7 +376,8 @@ gst_audio_converter_free (GstAudioConverter * convert)
* @flags: extra #GstAudioConverterFlags * @flags: extra #GstAudioConverterFlags
* @src: source samples * @src: source samples
* @dst: output samples * @dst: output samples
* @samples: number of samples * @in_samples: number of input samples
* @out_samples: number of output samples
* *
* Perform the conversion @src to @dst using @convert. * Perform the conversion @src to @dst using @convert.
* *
@ -385,7 +385,8 @@ gst_audio_converter_free (GstAudioConverter * convert)
*/ */
gboolean gboolean
gst_audio_converter_samples (GstAudioConverter * convert, gst_audio_converter_samples (GstAudioConverter * convert,
GstAudioConverterFlags flags, gpointer src, gpointer dst, gint samples) GstAudioConverterFlags flags, gpointer src, gpointer dst,
gsize in_samples, gsize * out_samples)
{ {
guint size; guint size;
gpointer outbuf, tmpbuf, tmpbuf2; gpointer outbuf, tmpbuf, tmpbuf2;
@ -393,18 +394,22 @@ gst_audio_converter_samples (GstAudioConverter * convert,
g_return_val_if_fail (convert != NULL, FALSE); g_return_val_if_fail (convert != NULL, FALSE);
g_return_val_if_fail (src != NULL, FALSE); g_return_val_if_fail (src != NULL, FALSE);
g_return_val_if_fail (dst != NULL, FALSE); g_return_val_if_fail (dst != NULL, FALSE);
g_return_val_if_fail (samples >= 0, FALSE); g_return_val_if_fail (in_samples >= 0, FALSE);
g_return_val_if_fail (out_samples != NULL, FALSE);
if (samples == 0) if (in_samples == 0) {
*out_samples = 0;
return TRUE; return TRUE;
}
if (convert->passthrough) { if (convert->passthrough) {
memcpy (dst, src, samples * convert->in.bpf); memcpy (dst, src, in_samples * convert->in.bpf);
*out_samples = in_samples;
return TRUE; return TRUE;
} }
size = size =
sizeof (gdouble) * samples * MAX (convert->in.channels, sizeof (gdouble) * in_samples * MAX (convert->in.channels,
convert->out.channels); convert->out.channels);
if (size > convert->tmpbufsize) { if (size > convert->tmpbufsize) {
@ -425,7 +430,7 @@ gst_audio_converter_samples (GstAudioConverter * convert,
convert->in.finfo->unpack_func (convert->in.finfo, convert->in.finfo->unpack_func (convert->in.finfo,
GST_AUDIO_PACK_FLAG_TRUNCATE_RANGE, outbuf, src, GST_AUDIO_PACK_FLAG_TRUNCATE_RANGE, outbuf, src,
samples * convert->in.channels); in_samples * convert->in.channels);
src = outbuf; src = outbuf;
} }
@ -439,7 +444,7 @@ gst_audio_converter_samples (GstAudioConverter * convert,
else else
outbuf = tmpbuf; outbuf = tmpbuf;
convert->convert_in (outbuf, src, samples * convert->in.channels); convert->convert_in (outbuf, src, in_samples * convert->in.channels);
src = outbuf; src = outbuf;
} }
@ -450,7 +455,7 @@ gst_audio_converter_samples (GstAudioConverter * convert,
else else
outbuf = tmpbuf; outbuf = tmpbuf;
gst_audio_channel_mix_samples (convert->mix, src, outbuf, samples); gst_audio_channel_mix_samples (convert->mix, src, outbuf, in_samples);
src = outbuf; src = outbuf;
} }
/* step 4, optional convert F64 -> S32 for quantize */ /* step 4, optional convert F64 -> S32 for quantize */
@ -460,7 +465,7 @@ gst_audio_converter_samples (GstAudioConverter * convert,
else else
outbuf = tmpbuf; outbuf = tmpbuf;
convert->convert_out (outbuf, src, samples * convert->out.channels); convert->convert_out (outbuf, src, in_samples * convert->out.channels);
src = outbuf; src = outbuf;
} }
@ -471,15 +476,16 @@ gst_audio_converter_samples (GstAudioConverter * convert,
else else
outbuf = tmpbuf; outbuf = tmpbuf;
gst_audio_quantize_samples (convert->quant, outbuf, src, samples); gst_audio_quantize_samples (convert->quant, outbuf, src, in_samples);
src = outbuf; src = outbuf;
} }
/* step 6, pack */ /* step 6, pack */
if (!convert->out_default) { if (!convert->out_default) {
convert->out.finfo->pack_func (convert->out.finfo, 0, src, dst, convert->out.finfo->pack_func (convert->out.finfo, 0, src, dst,
samples * convert->out.channels); in_samples * convert->out.channels);
} }
*out_samples = in_samples;
return TRUE; return TRUE;
} }

3
gst-libs/gst/audio/audio-converter.h
View File

@ -57,6 +57,7 @@ typedef struct _GstAudioConverter GstAudioConverter;
/** /**
* GstAudioConverterFlags:
* @GST_AUDIO_CONVERTER_FLAG_NONE: no flag * @GST_AUDIO_CONVERTER_FLAG_NONE: no flag
* @GST_AUDIO_CONVERTER_FLAG_SOURCE_WRITABLE: the source is writable and can be * @GST_AUDIO_CONVERTER_FLAG_SOURCE_WRITABLE: the source is writable and can be
* used as temporary storage during conversion. * used as temporary storage during conversion.
@ -81,6 +82,6 @@ const GstStructure * gst_audio_converter_get_config (GstAudioConverter * con
gboolean gst_audio_converter_samples (GstAudioConverter * convert, gboolean gst_audio_converter_samples (GstAudioConverter * convert,
GstAudioConverterFlags flags, GstAudioConverterFlags flags,
gpointer src, gpointer dst, gpointer src, gpointer dst,
gint samples); gsize in_samples, gsize *out_samples);
#endif /* __GST_AUDIO_CONVERTER_H__ */ #endif /* __GST_AUDIO_CONVERTER_H__ */

5
gst/audioconvert/gstaudioconvert.c
View File

@ -701,8 +701,7 @@ gst_audio_convert_transform (GstBaseTransform * base, GstBuffer * inbuf,
gint insize, outsize; gint insize, outsize;
gboolean inbuf_writable; gboolean inbuf_writable;
GstAudioConverterFlags flags; GstAudioConverterFlags flags;
gsize samples, out;
gint samples;
/* get amount of samples to convert. */ /* get amount of samples to convert. */
samples = gst_buffer_get_size (inbuf) / this->in_info.bpf; samples = gst_buffer_get_size (inbuf) / this->in_info.bpf;
@ -737,7 +736,7 @@ gst_audio_convert_transform (GstBaseTransform * base, GstBuffer * inbuf,
if (!GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_GAP)) { if (!GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_GAP)) {
if (!gst_audio_converter_samples (this->convert, flags, srcmap.data, if (!gst_audio_converter_samples (this->convert, flags, srcmap.data,
dstmap.data, samples)) dstmap.data, samples, &out))
goto convert_error; goto convert_error;
} else { } else {
/* Create silence buffer */ /* Create silence buffer */