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aacparse: Avoid mismatch between src_caps and output_header_type

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If our downstream caps didn't intersect, we attempted to convert between
raw and ADTS stream formats, if possible. If the caps still did not
intersect, we then used the modified `src_caps` but left the
`output_header_type` unmodified.

This caused a mismatch between caps and actual stream format.

Avoid this by first copying the `src_caps` to `convcaps` for the
additional intersection tests, replacing `src_caps` if we succeed.

While we're here, clean up the code a bit and remove the `codec_data`
field from outgoing ADTS caps.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2550>
This commit is contained in:
Jan Alexander Steffens (heftig) 2022-06-03 18:35:54 +02:00 committed by GStreamer Marge Bot
parent e5f9bb973f
commit 637406cdb1
1 changed files with 27 additions and 26 deletions
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53
subprojects/gst-plugins-good/gst/audioparsers/gstaacparse.c
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@ -210,7 +210,9 @@ gst_aac_parse_set_src_caps (GstAacParse * aacparse, GstCaps * sink_caps)
stream_format = NULL;
}
/* Generate codec data to be able to set profile/level on the caps */
/* Generate codec data to be able to set profile/level on the caps.
* The codec_data data is according to AudioSpecificConfig,
* ISO/IEC 14496-3, 1.6.2.1 */
sample_rate_idx =
gst_codec_utils_aac_get_index_from_sample_rate (aacparse->sample_rate);
if (sample_rate_idx < 0)
@ -231,39 +233,38 @@ gst_aac_parse_set_src_caps (GstAacParse * aacparse, GstCaps * sink_caps)
peercaps = gst_pad_peer_query_caps (GST_BASE_PARSE_SRC_PAD (aacparse), NULL);
if (peercaps && !gst_caps_can_intersect (src_caps, peercaps)) {
GST_DEBUG_OBJECT (GST_BASE_PARSE (aacparse)->srcpad,
"Caps can not intersect");
GstCaps *convcaps = gst_caps_copy (src_caps);
GstStructure *cs = gst_caps_get_structure (convcaps, 0);
GST_DEBUG_OBJECT (aacparse, "Caps do not intersect: parsed %" GST_PTR_FORMAT
" and peer %" GST_PTR_FORMAT, src_caps, peercaps);
if (aacparse->header_type == DSPAAC_HEADER_ADTS) {
GST_DEBUG_OBJECT (GST_BASE_PARSE (aacparse)->srcpad,
"Input is ADTS, trying raw");
gst_caps_set_simple (src_caps, "stream-format", G_TYPE_STRING, "raw",
NULL);
if (gst_caps_can_intersect (src_caps, peercaps)) {
GstBuffer *codec_data_buffer;
GstBuffer *codec_data_buffer = gst_buffer_new_and_alloc (2);
GST_DEBUG_OBJECT (GST_BASE_PARSE (aacparse)->srcpad,
"Caps can intersect, we will drop the ADTS layer");
gst_buffer_fill (codec_data_buffer, 0, codec_data, 2);
gst_structure_set (cs, "stream-format", G_TYPE_STRING, "raw",
"codec_data", GST_TYPE_BUFFER, codec_data_buffer, NULL);
if (gst_caps_can_intersect (convcaps, peercaps)) {
GST_DEBUG_OBJECT (aacparse, "Converting from ADTS to raw");
aacparse->output_header_type = DSPAAC_HEADER_NONE;
/* The codec_data data is according to AudioSpecificConfig,
ISO/IEC 14496-3, 1.6.2.1 */
codec_data_buffer = gst_buffer_new_and_alloc (2);
gst_buffer_fill (codec_data_buffer, 0, codec_data, 2);
gst_caps_set_simple (src_caps, "codec_data", GST_TYPE_BUFFER,
codec_data_buffer, NULL);
gst_buffer_unref (codec_data_buffer);
gst_caps_replace (&src_caps, convcaps);
}
gst_buffer_unref (codec_data_buffer);
} else if (aacparse->header_type == DSPAAC_HEADER_NONE) {
GST_DEBUG_OBJECT (GST_BASE_PARSE (aacparse)->srcpad,
"Input is raw, trying ADTS");
gst_caps_set_simple (src_caps, "stream-format", G_TYPE_STRING, "adts",
NULL);
if (gst_caps_can_intersect (src_caps, peercaps)) {
GST_DEBUG_OBJECT (GST_BASE_PARSE (aacparse)->srcpad,
"Caps can intersect, we will prepend ADTS headers");
gst_structure_set (cs, "stream-format", G_TYPE_STRING, "adts", NULL);
gst_structure_remove_field (cs, "codec_data");
if (gst_caps_can_intersect (convcaps, peercaps)) {
GST_DEBUG_OBJECT (aacparse, "Converting from raw to ADTS");
aacparse->output_header_type = DSPAAC_HEADER_ADTS;
gst_caps_replace (&src_caps, convcaps);
}
}
gst_caps_unref (convcaps);
}
if (peercaps)
gst_caps_unref (peercaps);