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audioresample: implement support for non-interleaved buffers

Browse source 
https://bugzilla.gnome.org/show_bug.cgi?id=705986
This commit is contained in:
George Kiagiadakis 2018-02-07 18:17:07 +02:00
parent d57a572172
commit 114f0abbd7
1 changed files with 33 additions and 31 deletions
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64
gst/audioresample/gstaudioresample.c
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@ -548,8 +548,7 @@ gst_audio_resample_push_drain (GstAudioResample * resample, guint history_len)
GstFlowReturn res;
gint outsize;
gsize out_len;
GstMapInfo map;
gpointer out[1];
GstAudioBuffer abuf;
g_assert (resample->converter != NULL);
@ -565,13 +564,15 @@ gst_audio_resample_push_drain (GstAudioResample * resample, guint history_len)
outsize = out_len * resample->in.bpf;
outbuf = gst_buffer_new_and_alloc (outsize);
gst_buffer_map (outbuf, &map, GST_MAP_WRITE);
if (GST_AUDIO_INFO_LAYOUT (&resample->out) ==
GST_AUDIO_LAYOUT_NON_INTERLEAVED) {
gst_buffer_add_audio_meta (outbuf, &resample->out, out_len, NULL);
}
out[0] = map.data;
gst_audio_buffer_map (&abuf, &resample->out, outbuf, GST_MAP_WRITE);
gst_audio_converter_samples (resample->converter, 0, NULL, history_len,
out, out_len);
gst_buffer_unmap (outbuf, &map);
abuf.planes, out_len);
gst_audio_buffer_unmap (&abuf);
/* time */
if (GST_CLOCK_TIME_IS_VALID (resample->t0)) {
@ -703,10 +704,10 @@ static GstFlowReturn
gst_audio_resample_process (GstAudioResample * resample, GstBuffer * inbuf,
GstBuffer * outbuf)
{
GstMapInfo in_map, out_map;
GstAudioBuffer srcabuf, dstabuf;
gsize outsize;
guint32 in_len;
guint32 out_len;
gsize in_len;
gsize out_len;
guint filt_len =
gst_audio_converter_get_max_latency (resample->converter) * 2;
gboolean inbuf_writable;
@ -715,12 +716,21 @@ gst_audio_resample_process (GstAudioResample * resample, GstBuffer * inbuf,
&& gst_buffer_n_memory (inbuf) == 1
&& gst_memory_is_writable (gst_buffer_peek_memory (inbuf, 0));
gst_buffer_map (inbuf, &in_map,
gst_audio_buffer_map (&srcabuf, &resample->in, inbuf,
inbuf_writable ? GST_MAP_READWRITE : GST_MAP_READ);
gst_buffer_map (outbuf, &out_map, GST_MAP_WRITE);
in_len = in_map.size / resample->in.bpf;
out_len = out_map.size / resample->out.bpf;
in_len = srcabuf.n_samples;
out_len = gst_audio_converter_get_out_frames (resample->converter, in_len);
/* ensure that the output buffer is not bigger than what we need */
gst_buffer_set_size (outbuf, out_len * resample->in.bpf);
if (GST_AUDIO_INFO_LAYOUT (&resample->out) ==
GST_AUDIO_LAYOUT_NON_INTERLEAVED) {
gst_buffer_add_audio_meta (outbuf, &resample->out, out_len, NULL);
}
gst_audio_buffer_map (&dstabuf, &resample->out, outbuf, GST_MAP_WRITE);
if (GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_GAP)) {
resample->num_nongap_samples = 0;
@ -738,6 +748,7 @@ gst_audio_resample_process (GstAudioResample * resample, GstBuffer * inbuf,
{
guint num, den;
gint i;
num = resample->in.rate;
den = resample->out.rate;
@ -749,7 +760,9 @@ gst_audio_resample_process (GstAudioResample * resample, GstBuffer * inbuf,
else
out_len = 0;
memset (out_map.data, 0, out_map.size);
for (i = 0; i < dstabuf.n_planes; i++)
memset (dstabuf.planes[i], 0, GST_AUDIO_BUFFER_PLANE_SIZE (&dstabuf));
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_GAP);
resample->num_gap_samples += in_len;
}
@ -771,17 +784,14 @@ gst_audio_resample_process (GstAudioResample * resample, GstBuffer * inbuf,
}
{
/* process */
gpointer in[1], out[1];
GstAudioConverterFlags flags;
flags = 0;
if (inbuf_writable)
flags |= GST_AUDIO_CONVERTER_FLAG_IN_WRITABLE;
in[0] = in_map.data;
out[0] = out_map.data;
gst_audio_converter_samples (resample->converter, flags, in, in_len,
out, out_len);
gst_audio_converter_samples (resample->converter, flags, srcabuf.planes,
in_len, dstabuf.planes, out_len);
}
}
@ -809,13 +819,13 @@ gst_audio_resample_process (GstAudioResample * resample, GstBuffer * inbuf,
resample->samples_out += out_len;
resample->samples_in += in_len;
gst_buffer_unmap (inbuf, &in_map);
gst_buffer_unmap (outbuf, &out_map);
gst_audio_buffer_unmap (&srcabuf);
gst_audio_buffer_unmap (&dstabuf);
outsize = out_len * resample->in.bpf;
GST_LOG_OBJECT (resample,
"Converted to buffer of %" G_GUINT32_FORMAT
"Converted to buffer of %" G_GSIZE_FORMAT
" samples (%" G_GSIZE_FORMAT " bytes) with timestamp %" GST_TIME_FORMAT
", duration %" GST_TIME_FORMAT ", offset %" G_GUINT64_FORMAT
", offset_end %" G_GUINT64_FORMAT, out_len, outsize,
@ -847,16 +857,8 @@ gst_audio_resample_transform (GstBaseTransform * base, GstBuffer * inbuf,
* flag to resync timestamp and offset counters and send event
* downstream */
if (G_UNLIKELY (gst_audio_resample_check_discont (resample, inbuf))) {
gsize size;
gint bpf = GST_AUDIO_INFO_BPF (&resample->in);
gst_audio_resample_reset_state (resample);
resample->need_discont = TRUE;
/* need to recalculate the output size */
size = gst_buffer_get_size (inbuf) / bpf;
size = gst_audio_converter_get_out_frames (resample->converter, size);
gst_buffer_set_size (outbuf, size * bpf);
}
/* handle discontinuity */