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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2025-04-01 13:49:58 +00:00
audio-converter: add output size argument
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:
parent
629b63d1f2
commit
ff6d1a2a25
3 changed files with 24 additions and 18 deletions
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@ -76,8 +76,7 @@ ensure_debug_category (void)
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typedef void (*AudioConvertFunc) (gpointer dst, const gpointer src, gint count);
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/**
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* int/int int/float float/int float/float
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/* int/int int/float float/int float/float
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*
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* unpack S32 S32 F64 F64
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* convert S32->F64
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@ -377,7 +376,8 @@ gst_audio_converter_free (GstAudioConverter * convert)
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* @flags: extra #GstAudioConverterFlags
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* @src: source samples
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* @dst: output samples
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* @samples: number of samples
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* @in_samples: number of input samples
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* @out_samples: number of output samples
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*
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* Perform the conversion @src to @dst using @convert.
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*
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@ -385,7 +385,8 @@ gst_audio_converter_free (GstAudioConverter * convert)
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*/
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gboolean
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gst_audio_converter_samples (GstAudioConverter * convert,
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GstAudioConverterFlags flags, gpointer src, gpointer dst, gint samples)
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GstAudioConverterFlags flags, gpointer src, gpointer dst,
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gsize in_samples, gsize * out_samples)
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{
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guint size;
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gpointer outbuf, tmpbuf, tmpbuf2;
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@ -393,18 +394,22 @@ gst_audio_converter_samples (GstAudioConverter * convert,
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g_return_val_if_fail (convert != NULL, FALSE);
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g_return_val_if_fail (src != NULL, FALSE);
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g_return_val_if_fail (dst != NULL, FALSE);
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g_return_val_if_fail (samples >= 0, FALSE);
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g_return_val_if_fail (in_samples >= 0, FALSE);
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g_return_val_if_fail (out_samples != NULL, FALSE);
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if (samples == 0)
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if (in_samples == 0) {
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*out_samples = 0;
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return TRUE;
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}
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if (convert->passthrough) {
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memcpy (dst, src, samples * convert->in.bpf);
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memcpy (dst, src, in_samples * convert->in.bpf);
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*out_samples = in_samples;
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return TRUE;
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}
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size =
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sizeof (gdouble) * samples * MAX (convert->in.channels,
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sizeof (gdouble) * in_samples * MAX (convert->in.channels,
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convert->out.channels);
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if (size > convert->tmpbufsize) {
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@ -425,7 +430,7 @@ gst_audio_converter_samples (GstAudioConverter * convert,
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convert->in.finfo->unpack_func (convert->in.finfo,
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GST_AUDIO_PACK_FLAG_TRUNCATE_RANGE, outbuf, src,
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samples * convert->in.channels);
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in_samples * convert->in.channels);
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src = outbuf;
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}
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@ -439,7 +444,7 @@ gst_audio_converter_samples (GstAudioConverter * convert,
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else
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outbuf = tmpbuf;
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convert->convert_in (outbuf, src, samples * convert->in.channels);
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convert->convert_in (outbuf, src, in_samples * convert->in.channels);
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src = outbuf;
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}
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@ -450,7 +455,7 @@ gst_audio_converter_samples (GstAudioConverter * convert,
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else
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outbuf = tmpbuf;
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gst_audio_channel_mix_samples (convert->mix, src, outbuf, samples);
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gst_audio_channel_mix_samples (convert->mix, src, outbuf, in_samples);
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src = outbuf;
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}
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/* step 4, optional convert F64 -> S32 for quantize */
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@ -460,7 +465,7 @@ gst_audio_converter_samples (GstAudioConverter * convert,
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else
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outbuf = tmpbuf;
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convert->convert_out (outbuf, src, samples * convert->out.channels);
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convert->convert_out (outbuf, src, in_samples * convert->out.channels);
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src = outbuf;
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}
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@ -471,15 +476,16 @@ gst_audio_converter_samples (GstAudioConverter * convert,
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else
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outbuf = tmpbuf;
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gst_audio_quantize_samples (convert->quant, outbuf, src, samples);
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gst_audio_quantize_samples (convert->quant, outbuf, src, in_samples);
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src = outbuf;
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}
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/* step 6, pack */
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if (!convert->out_default) {
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convert->out.finfo->pack_func (convert->out.finfo, 0, src, dst,
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samples * convert->out.channels);
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in_samples * convert->out.channels);
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}
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*out_samples = in_samples;
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return TRUE;
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}
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@ -57,6 +57,7 @@ typedef struct _GstAudioConverter GstAudioConverter;
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/**
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* GstAudioConverterFlags:
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* @GST_AUDIO_CONVERTER_FLAG_NONE: no flag
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* @GST_AUDIO_CONVERTER_FLAG_SOURCE_WRITABLE: the source is writable and can be
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* used as temporary storage during conversion.
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@ -81,6 +82,6 @@ const GstStructure * gst_audio_converter_get_config (GstAudioConverter * con
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gboolean gst_audio_converter_samples (GstAudioConverter * convert,
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GstAudioConverterFlags flags,
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gpointer src, gpointer dst,
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gint samples);
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gsize in_samples, gsize *out_samples);
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#endif /* __GST_AUDIO_CONVERTER_H__ */
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@ -701,8 +701,7 @@ gst_audio_convert_transform (GstBaseTransform * base, GstBuffer * inbuf,
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gint insize, outsize;
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gboolean inbuf_writable;
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GstAudioConverterFlags flags;
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gint samples;
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gsize samples, out;
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/* get amount of samples to convert. */
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samples = gst_buffer_get_size (inbuf) / this->in_info.bpf;
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@ -737,7 +736,7 @@ gst_audio_convert_transform (GstBaseTransform * base, GstBuffer * inbuf,
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if (!GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_GAP)) {
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if (!gst_audio_converter_samples (this->convert, flags, srcmap.data,
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dstmap.data, samples))
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dstmap.data, samples, &out))
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goto convert_error;
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} else {
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/* Create silence buffer */
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