mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-12-28 03:00:35 +00:00
Code Issues 1 Projects Releases Wiki Activity

rtpmp4adepay: grab the sampling arte and put into caps

Browse source 
This is needed to be able to mux the received audio into mp4 (in the case of
aac). Fixes #625825.
This commit is contained in:
Stefan Kost 2010-08-12 12:33:06 +03:00
parent 22560c473d
commit 988f228da7
1 changed files with 13 additions and 0 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

13
gst/rtp/gstrtpmp4adepay.c
View file

@ -165,6 +165,10 @@ gst_rtp_mp4a_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
guint8 *data;
guint size;
gint i;
guint sr_idx;
static const guint aac_sample_rates[] = { 96000, 88200, 64000, 48000,
44100, 32000, 24000, 22050, 16000, 12000, 11025, 8000
};
buffer = gst_value_get_buffer (&v);
gst_buffer_ref (buffer);
@ -207,6 +211,15 @@ gst_rtp_mp4a_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
data[i] = ((data[i + 1] & 1) << 7) | ((data[i + 2] & 0xfe) >> 1);
}
/* grab and set sampling rate */
sr_idx = ((data[0] & 0x07) << 1) | ((data[1] & 0x80) >> 7);
if (sr_idx < G_N_ELEMENTS (aac_sample_rates)) {
gst_caps_set_simple (srccaps,
"rate", G_TYPE_INT, (gint) aac_sample_rates[sr_idx], NULL);
} else {
GST_WARNING ("Invalid sample rate index %u", sr_idx);
}
/* ignore remaining bit, we're only interested in full bytes */
GST_BUFFER_SIZE (buffer) = size;