static gboolean
gst_opus_dec_set_format (GstAudioDecoder * bdec, GstCaps * caps)
{
GstOpusDec *dec = GST_OPUS_DEC (bdec);
gboolean ret = TRUE;
GstStructure *s;
const GValue *streamheader;
GstCaps *old_caps;
GST_DEBUG_OBJECT (dec, "set_format: %" GST_PTR_FORMAT, caps);
if ((old_caps = gst_pad_get_current_caps (GST_AUDIO_DECODER_SINK_PAD (bdec)))) {
if (gst_caps_is_equal (caps, old_caps)) {
gst_caps_unref (old_caps);
GST_DEBUG_OBJECT (dec, "caps didn't change");
goto done;
}
GST_DEBUG_OBJECT (dec, "caps have changed, resetting decoder");
gst_opus_dec_reset (dec);
gst_caps_unref (old_caps);
}
s = gst_caps_get_structure (caps, 0);
if ((streamheader = gst_structure_get_value (s, "streamheader")) &&
G_VALUE_HOLDS (streamheader, GST_TYPE_ARRAY) &&
gst_value_array_get_size (streamheader) >= 2) {
const GValue *header, *vorbiscomment;
GstBuffer *buf;
GstFlowReturn res = GST_FLOW_OK;
header = gst_value_array_get_value (streamheader, 0);
if (header && G_VALUE_HOLDS (header, GST_TYPE_BUFFER)) {
buf = gst_value_get_buffer (header);
res = gst_opus_dec_parse_header (dec, buf);
if (res != GST_FLOW_OK) {
ret = FALSE;
goto done;
}
gst_buffer_replace (&dec->streamheader, buf);
}
vorbiscomment = gst_value_array_get_value (streamheader, 1);
if (vorbiscomment && G_VALUE_HOLDS (vorbiscomment, GST_TYPE_BUFFER)) {
buf = gst_value_get_buffer (vorbiscomment);
res = gst_opus_dec_parse_comments (dec, buf);
if (res != GST_FLOW_OK) {
ret = FALSE;
goto done;
}
gst_buffer_replace (&dec->vorbiscomment, buf);
}
} else {
const GstAudioChannelPosition *posn = NULL;
if (!gst_codec_utils_opus_parse_caps (caps, &dec->sample_rate,
&dec->n_channels, &dec->channel_mapping_family, &dec->n_streams,
&dec->n_stereo_streams, dec->channel_mapping)) {
ret = FALSE;
goto done;
}
if (dec->channel_mapping_family == 1 && dec->n_channels <= 8)
posn = gst_opus_channel_positions[dec->n_channels - 1];
if (!gst_opus_dec_negotiate (dec, posn))
return FALSE;
}
done:
return ret;
}
static GstFlowReturn
gst_opus_parse_parse_frame (GstBaseParse * base, GstBaseParseFrame * frame)
{
guint64 duration;
GstOpusParse *parse;
gboolean is_idheader, is_commentheader;
GstMapInfo map;
GstAudioClippingMeta *cmeta =
gst_buffer_get_audio_clipping_meta (frame->buffer);
parse = GST_OPUS_PARSE (base);
g_assert (!cmeta || cmeta->format == GST_FORMAT_DEFAULT);
is_idheader = gst_opus_header_is_id_header (frame->buffer);
is_commentheader = gst_opus_header_is_comment_header (frame->buffer);
if (!parse->got_headers || !parse->header_sent) {
GstCaps *caps;
/* Opus streams can decode to 1 or 2 channels, so use the header
value if we have one, or 2 otherwise */
if (is_idheader) {
gst_buffer_replace (&parse->id_header, frame->buffer);
GST_DEBUG_OBJECT (parse, "Found ID header, keeping");
return GST_BASE_PARSE_FLOW_DROPPED;
} else if (is_commentheader) {
gst_buffer_replace (&parse->comment_header, frame->buffer);
GST_DEBUG_OBJECT (parse, "Found comment header, keeping");
return GST_BASE_PARSE_FLOW_DROPPED;
}
parse->got_headers = TRUE;
if (cmeta && cmeta->start) {
parse->pre_skip += cmeta->start;
gst_buffer_map (frame->buffer, &map, GST_MAP_READ);
duration = packet_duration_opus (map.data, map.size);
gst_buffer_unmap (frame->buffer, &map);
/* Queue frame for later once we know all initial padding */
if (duration == cmeta->start) {
frame->flags |= GST_BASE_PARSE_FRAME_FLAG_QUEUE;
}
}
if (!(frame->flags & GST_BASE_PARSE_FRAME_FLAG_QUEUE)) {
if (FALSE && parse->id_header && parse->comment_header) {
guint16 pre_skip;
gst_buffer_map (parse->id_header, &map, GST_MAP_READWRITE);
pre_skip = GST_READ_UINT16_LE (map.data + 10);
if (pre_skip != parse->pre_skip) {
GST_DEBUG_OBJECT (parse,
"Fixing up pre-skip %u -> %" G_GUINT64_FORMAT, pre_skip,
parse->pre_skip);
GST_WRITE_UINT16_LE (map.data + 10, parse->pre_skip);
}
gst_buffer_unmap (parse->id_header, &map);
caps =
gst_codec_utils_opus_create_caps_from_header (parse->id_header,
parse->comment_header);
} else {
GstCaps *sink_caps;
guint32 sample_rate = 48000;
guint8 n_channels, n_streams, n_stereo_streams, channel_mapping_family;
guint8 channel_mapping[256];
GstBuffer *id_header;
sink_caps = gst_pad_get_current_caps (GST_BASE_PARSE_SINK_PAD (parse));
if (!sink_caps
|| !gst_codec_utils_opus_parse_caps (sink_caps, &sample_rate,
&n_channels, &channel_mapping_family, &n_streams,
&n_stereo_streams, channel_mapping)) {
GST_INFO_OBJECT (parse,
"No headers and no caps, blindly setting up canonical stereo");
n_channels = 2;
n_streams = 1;
n_stereo_streams = 1;
channel_mapping_family = 0;
channel_mapping[0] = 0;
channel_mapping[1] = 1;
}
if (sink_caps)
gst_caps_unref (sink_caps);
id_header =
gst_codec_utils_opus_create_header (sample_rate, n_channels,
channel_mapping_family, n_streams, n_stereo_streams,
channel_mapping, parse->pre_skip, 0);
caps = gst_codec_utils_opus_create_caps_from_header (id_header, NULL);
gst_buffer_unref (id_header);
}
gst_buffer_replace (&parse->id_header, NULL);
gst_buffer_replace (&parse->comment_header, NULL);
gst_pad_set_caps (GST_BASE_PARSE_SRC_PAD (parse), caps);
gst_caps_unref (caps);
parse->header_sent = TRUE;
}
}
GST_BUFFER_TIMESTAMP (frame->buffer) = parse->next_ts;
gst_buffer_map (frame->buffer, &map, GST_MAP_READ);
duration = packet_duration_opus (map.data, map.size);
gst_buffer_unmap (frame->buffer, &map);
parse->next_ts += duration;
GST_BUFFER_DURATION (frame->buffer) = duration;
GST_BUFFER_OFFSET_END (frame->buffer) =
gst_util_uint64_scale (parse->next_ts, 48000, GST_SECOND);
GST_BUFFER_OFFSET (frame->buffer) = parse->next_ts;
return GST_FLOW_OK;
}
