mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-23 02:01:12 +00:00
Code Issues 1 Projects Releases Wiki Activity

qtdemux: Try to build AAC codec-data whenever it's possible

Browse source 
AAC codec_data is a just collection of AAC profile, samplerate, and
channels. We can know samplerate and channels from parsed
SampleEntry data. Although the AAC profile is unknown there,
let's assume it as AAC-LC like we've been doing for the version 1
atom.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1082>
This commit is contained in:
Seungha Yang 2021-09-14 17:26:27 +09:00
parent 44bfc00884
commit 4576abde67
1 changed files with 35 additions and 21 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

56
gst/isomp4/qtdemux.c
View file

@ -12730,32 +12730,46 @@ qtdemux_parse_trak (GstQTDemux * qtdemux, GNode * trak)
{
/* mp4a atom withtout ESDS; Attempt to build codec data from atom */
gint len = QT_UINT32 (stsd_entry_data);
guint16 sound_version = 0;
/* FIXME: Can this be determined somehow? There doesn't seem to be
* anything in mp4a atom that specifis compression */
gint profile = 2;
guint16 channels = entry->n_channels;
guint32 time_scale = (guint32) entry->rate;
gint sample_rate_index = -1;
if (len >= 34) {
guint16 sound_version = QT_UINT16 (stsd_entry_data + 16);
sound_version = QT_UINT16 (stsd_entry_data + 16);
if (sound_version == 1) {
guint16 channels = QT_UINT16 (stsd_entry_data + 24);
guint32 time_scale = QT_UINT32 (stsd_entry_data + 30);
guint8 codec_data[2];
GstBuffer *buf;
gint profile = 2; /* FIXME: Can this be determined somehow? There doesn't seem to be anything in mp4a atom that specifis compression */
gint sample_rate_index =
gst_codec_utils_aac_get_index_from_sample_rate (time_scale);
/* build AAC codec data */
codec_data[0] = profile << 3;
codec_data[0] |= ((sample_rate_index >> 1) & 0x7);
codec_data[1] = (sample_rate_index & 0x01) << 7;
codec_data[1] |= (channels & 0xF) << 3;
buf = gst_buffer_new_and_alloc (2);
gst_buffer_fill (buf, 0, codec_data, 2);
gst_caps_set_simple (entry->caps,
"codec_data", GST_TYPE_BUFFER, buf, NULL);
gst_buffer_unref (buf);
channels = QT_UINT16 (stsd_entry_data + 24);
time_scale = QT_UINT32 (stsd_entry_data + 30);
} else {
GST_FIXME_OBJECT (qtdemux, "Unhandled mp4a atom version %d",
sound_version);
}
} else {
GST_DEBUG_OBJECT (qtdemux, "Too small stsd entry data len %d",
len);
}
sample_rate_index =
gst_codec_utils_aac_get_index_from_sample_rate (time_scale);
if (sample_rate_index >= 0 && channels > 0) {
guint8 codec_data[2];
GstBuffer *buf;
/* build AAC codec data */
codec_data[0] = profile << 3;
codec_data[0] |= ((sample_rate_index >> 1) & 0x7);
codec_data[1] = (sample_rate_index & 0x01) << 7;
codec_data[1] |= (channels & 0xF) << 3;
buf = gst_buffer_new_and_alloc (2);
gst_buffer_fill (buf, 0, codec_data, 2);
gst_caps_set_simple (entry->caps,
"codec_data", GST_TYPE_BUFFER, buf, NULL);
gst_buffer_unref (buf);
}
break;
}