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mssdemux: make codec private data from manifest attributes with AAC-LC

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When the codec is AAC-LC, some server implementation (e.g. Microsoft IIS) doesn't add the CodecPrivateData attribute. The element needs to re-create the codec data from the Quality Level attributes (channels and sampling rate).
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Louis-Francis Ratté-Boulianne 2013-01-30 00:22:50 -05:00 committed by Thiago Santos
parent 3ebb842388
commit ea5f2718bf
1 changed files with 54 additions and 0 deletions
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54
ext/smoothstreaming/gstmssmanifest.c
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@ -476,6 +476,56 @@ end:
return caps; return caps;
} }
static guint8
_frequency_index_from_sampling_rate (guint sampling_rate)
{
static const guint aac_sample_rates[] = { 96000, 88200, 64000, 48000, 44100,
32000, 24000, 22050, 16000, 12000, 11025, 8000, 7350
};
guint8 i;
for (i = 0; i < G_N_ELEMENTS (aac_sample_rates); i++) {
if (aac_sample_rates[i] == sampling_rate)
return i;
}
return 15;
}
static GstBuffer *
_make_accl_codec_data (guint sampling_rate, guint channels)
{
GstBuffer *buf;
guint8 *data;
guint8 frequency_index;
guint8 buf_size;
buf_size = 2;
frequency_index = _frequency_index_from_sampling_rate (sampling_rate);
if (frequency_index == 15)
buf_size += 3;
buf = gst_buffer_new_and_alloc (buf_size);
data = GST_BUFFER_DATA (buf);
data[0] = 2 << 3; /* AAC-LC object type is 2 */
data[0] += frequency_index >> 1;
data[1] = (frequency_index & 0x01) << 7;
/* Sampling rate is not in frequencies table, write manually */
if (frequency_index == 15) {
data[1] += sampling_rate >> 17;
data[2] = (sampling_rate >> 9) & 0xFF;
data[3] = (sampling_rate >> 1) & 0xFF;
data[4] = sampling_rate & 0x01;
data += 3;
}
data[1] += (channels & 0x0F) << 3;
return buf;
}
static GstCaps * static GstCaps *
_gst_mss_stream_audio_caps_from_qualitylevel_xml (xmlNodePtr node) _gst_mss_stream_audio_caps_from_qualitylevel_xml (xmlNodePtr node)
{ {
@ -509,6 +559,10 @@ _gst_mss_stream_audio_caps_from_qualitylevel_xml (xmlNodePtr node)
g_value_init (value, GST_TYPE_BUFFER); g_value_init (value, GST_TYPE_BUFFER);
gst_value_deserialize (value, (gchar *) codec_data); gst_value_deserialize (value, (gchar *) codec_data);
gst_structure_take_value (structure, "codec_data", value); gst_structure_take_value (structure, "codec_data", value);
} else if (strcmp (fourcc, "AACL") == 0) {
GstBuffer *buffer = _make_accl_codec_data (atoi (rate), atoi (channels));
gst_caps_set_simple (caps, "codec_data", GST_TYPE_BUFFER, buffer, NULL);
gst_buffer_unref (buffer);
} }
end: end: