Commit 5e559685 authored by Wim Taymans's avatar Wim Taymans

audio-convert: improve converter API

Improve the converter API to allow for an max input and output number of
samples and return the number of consumed/produced samples.
parent 872f40d7
......@@ -374,36 +374,48 @@ gst_audio_converter_free (GstAudioConverter * convert)
* gst_audio_converter_samples:
* @convert: a #GstAudioConverter
* @flags: extra #GstAudioConverterFlags
* @src: source samples
* @dst: output samples
* @in: input samples
* @in_samples: number of input samples
* @out: output samples
* @out_samples: number of output samples
* @in_consumed: number of input samples consumed
* @out_produced: number of output samples produced
*
* Perform the conversion @src to @dst using @convert.
* Perform the conversion with @in_samples in @in to @out_samples in @out
* using @convert.
*
* The actual number of samples used from @in is returned in @in_consumed and
* can be less than @in_samples. The actual number of samples produced is
* returned in @out_produced and can be less than @out_samples.
*
* Returns: %TRUE is the conversion could be performed.
*/
gboolean
gst_audio_converter_samples (GstAudioConverter * convert,
GstAudioConverterFlags flags, gpointer src, gpointer dst,
gsize in_samples, gsize * out_samples)
GstAudioConverterFlags flags, gpointer in, gsize in_samples,
gpointer out, gsize out_samples, gsize * in_consumed, gsize * out_produced)
{
guint size;
gpointer outbuf, tmpbuf, tmpbuf2;
g_return_val_if_fail (convert != NULL, FALSE);
g_return_val_if_fail (src != NULL, FALSE);
g_return_val_if_fail (dst != NULL, FALSE);
g_return_val_if_fail (out_samples != NULL, FALSE);
g_return_val_if_fail (in != NULL, FALSE);
g_return_val_if_fail (out != NULL, FALSE);
g_return_val_if_fail (in_consumed != NULL, FALSE);
g_return_val_if_fail (out_produced != NULL, FALSE);
in_samples = MIN (in_samples, out_samples);
if (in_samples == 0) {
*out_samples = 0;
*in_consumed = 0;
*out_produced = 0;
return TRUE;
}
if (convert->passthrough) {
memcpy (dst, src, in_samples * convert->in.bpf);
*out_samples = in_samples;
memcpy (out, in, in_samples * convert->in.bpf);
*out_produced = in_samples;
*in_consumed = in_samples;
return TRUE;
}
......@@ -423,68 +435,69 @@ gst_audio_converter_samples (GstAudioConverter * convert,
if (!convert->in_default) {
if (!convert->convert_in && convert->mix_passthrough
&& !convert->convert_out && !convert->quant && convert->out_default)
outbuf = dst;
outbuf = out;
else
outbuf = tmpbuf;
convert->in.finfo->unpack_func (convert->in.finfo,
GST_AUDIO_PACK_FLAG_TRUNCATE_RANGE, outbuf, src,
GST_AUDIO_PACK_FLAG_TRUNCATE_RANGE, outbuf, in,
in_samples * convert->in.channels);
src = outbuf;
in = outbuf;
}
/* 2. optionally convert for mixing */
if (convert->convert_in) {
if (convert->mix_passthrough && !convert->convert_out && !convert->quant
&& convert->out_default)
outbuf = dst;
else if (src == tmpbuf)
outbuf = out;
else if (in == tmpbuf)
outbuf = tmpbuf2;
else
outbuf = tmpbuf;
convert->convert_in (outbuf, src, in_samples * convert->in.channels);
src = outbuf;
convert->convert_in (outbuf, in, in_samples * convert->in.channels);
in = outbuf;
}
/* step 3, channel mix if not passthrough */
if (!convert->mix_passthrough) {
if (!convert->convert_out && !convert->quant && convert->out_default)
outbuf = dst;
outbuf = out;
else
outbuf = tmpbuf;
gst_audio_channel_mix_samples (convert->mix, src, outbuf, in_samples);
src = outbuf;
gst_audio_channel_mix_samples (convert->mix, in, outbuf, in_samples);
in = outbuf;
}
/* step 4, optional convert F64 -> S32 for quantize */
if (convert->convert_out) {
if (!convert->quant && convert->out_default)
outbuf = dst;
outbuf = out;
else
outbuf = tmpbuf;
convert->convert_out (outbuf, src, in_samples * convert->out.channels);
src = outbuf;
convert->convert_out (outbuf, in, in_samples * convert->out.channels);
in = outbuf;
}
/* step 5, optional quantize */
if (convert->quant) {
if (convert->out_default)
outbuf = dst;
outbuf = out;
else
outbuf = tmpbuf;
gst_audio_quantize_samples (convert->quant, outbuf, src, in_samples);
src = outbuf;
gst_audio_quantize_samples (convert->quant, outbuf, in, in_samples);
in = outbuf;
}
/* step 6, pack */
if (!convert->out_default) {
convert->out.finfo->pack_func (convert->out.finfo, 0, src, dst,
convert->out.finfo->pack_func (convert->out.finfo, 0, in, out,
in_samples * convert->out.channels);
}
*out_samples = in_samples;
*out_produced = in_samples;
*in_consumed = in_samples;
return TRUE;
}
......@@ -81,7 +81,8 @@ const GstStructure * gst_audio_converter_get_config (GstAudioConverter * con
gboolean gst_audio_converter_samples (GstAudioConverter * convert,
GstAudioConverterFlags flags,
gpointer src, gpointer dst,
gsize in_samples, gsize *out_samples);
gpointer in, gsize in_samples,
gpointer out, gsize out_samples,
gsize *in_consumed, gsize *out_produced);
#endif /* __GST_AUDIO_CONVERTER_H__ */
......@@ -701,7 +701,7 @@ gst_audio_convert_transform (GstBaseTransform * base, GstBuffer * inbuf,
gint insize, outsize;
gboolean inbuf_writable;
GstAudioConverterFlags flags;
gsize samples, out;
gsize samples, consumed, produced;
/* get amount of samples to convert. */
samples = gst_buffer_get_size (inbuf) / this->in_info.bpf;
......@@ -736,7 +736,7 @@ gst_audio_convert_transform (GstBaseTransform * base, GstBuffer * inbuf,
if (!GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_GAP)) {
if (!gst_audio_converter_samples (this->convert, flags, srcmap.data,
dstmap.data, samples, &out))
samples, dstmap.data, samples, &consumed, &produced))
goto convert_error;
} else {
/* Create silence buffer */
......
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