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opusdec: allow negotiation of rate/channels with downstream

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Since an opus stream may be decoded to any (sensible) rate,
and either stereo or mono, we try to accomodate downstream.
This commit is contained in:
Vincent Penquerc'h 2011-11-16 17:24:20 +00:00
parent d10cbd0268
commit a02e18917f
1 changed files with 47 additions and 27 deletions
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74
ext/opus/gstopusdec.c
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@ -129,8 +129,8 @@ gst_opus_dec_reset (GstOpusDec * dec)
static void static void
gst_opus_dec_init (GstOpusDec * dec, GstOpusDecClass * g_class) gst_opus_dec_init (GstOpusDec * dec, GstOpusDecClass * g_class)
{ {
dec->sample_rate = 48000; dec->sample_rate = 0;
dec->n_channels = 2; dec->n_channels = 0;
gst_opus_dec_reset (dec); gst_opus_dec_reset (dec);
} }
@ -170,6 +170,48 @@ gst_opus_dec_parse_comments (GstOpusDec * dec, GstBuffer * buf)
return GST_FLOW_OK; return GST_FLOW_OK;
} }
static void
gst_opus_dec_setup_from_peer_caps (GstOpusDec * dec)
{
GstPad *srcpad, *peer;
GstStructure *s;
GstCaps *caps;
const GstCaps *template_caps;
const GstCaps *peer_caps;
srcpad = GST_AUDIO_DECODER_SRC_PAD (dec);
peer = gst_pad_get_peer (srcpad);
if (peer) {
template_caps = gst_pad_get_pad_template_caps (srcpad);
peer_caps = gst_pad_get_caps (peer);
GST_DEBUG_OBJECT (dec, "Peer caps: %" GST_PTR_FORMAT, peer_caps);
caps = gst_caps_intersect (template_caps, peer_caps);
gst_pad_fixate_caps (peer, caps);
GST_DEBUG_OBJECT (dec, "Fixated caps: %" GST_PTR_FORMAT, caps);
s = gst_caps_get_structure (caps, 0);
if (!gst_structure_get_int (s, "channels", &dec->n_channels)) {
dec->n_channels = 2;
GST_WARNING_OBJECT (dec, "Failed to get channels, using default %d",
dec->n_channels);
} else {
GST_DEBUG_OBJECT (dec, "Got channels %d", dec->n_channels);
}
if (!gst_structure_get_int (s, "rate", &dec->sample_rate)) {
dec->sample_rate = 48000;
GST_WARNING_OBJECT (dec, "Failed to get rate, using default %d",
dec->sample_rate);
} else {
GST_DEBUG_OBJECT (dec, "Got sample rate %d", dec->sample_rate);
}
gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), caps);
} else {
GST_WARNING_OBJECT (dec, "Failed to get src pad peer");
}
}
static GstFlowReturn static GstFlowReturn
opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buf, opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buf,
GstClockTime timestamp, GstClockTime duration) GstClockTime timestamp, GstClockTime duration)
@ -184,27 +226,13 @@ opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buf,
unsigned int packet_size; unsigned int packet_size;
if (dec->state == NULL) { if (dec->state == NULL) {
GstCaps *caps; gst_opus_dec_setup_from_peer_caps (dec);
GST_DEBUG_OBJECT (dec, "Creating decoder with %d channels, %d Hz",
dec->n_channels, dec->sample_rate);
dec->state = opus_decoder_create (dec->sample_rate, dec->n_channels, &err); dec->state = opus_decoder_create (dec->sample_rate, dec->n_channels, &err);
if (!dec->state || err != OPUS_OK) if (!dec->state || err != OPUS_OK)
goto creation_failed; goto creation_failed;
/* set caps */
caps = gst_caps_new_simple ("audio/x-raw-int",
"rate", G_TYPE_INT, dec->sample_rate,
"channels", G_TYPE_INT, dec->n_channels,
"signed", G_TYPE_BOOLEAN, TRUE,
"endianness", G_TYPE_INT, G_BYTE_ORDER,
"width", G_TYPE_INT, 16, "depth", G_TYPE_INT, 16, NULL);
GST_DEBUG_OBJECT (dec, "rate=%d channels=%d",
dec->sample_rate, dec->n_channels);
if (!gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), caps))
GST_ERROR ("nego failure");
gst_caps_unref (caps);
} }
if (buf) { if (buf) {
@ -310,14 +338,6 @@ gst_opus_dec_set_format (GstAudioDecoder * bdec, GstCaps * caps)
gst_buffer_replace (&dec->vorbiscomment, buf); gst_buffer_replace (&dec->vorbiscomment, buf);
} }
} }
caps = gst_caps_new_simple ("audio/x-raw-int",
"rate", G_TYPE_INT, dec->sample_rate,
"channels", G_TYPE_INT, dec->n_channels,
"signed", G_TYPE_BOOLEAN, TRUE,
"endianness", G_TYPE_INT, G_BYTE_ORDER,
"width", G_TYPE_INT, 16, "depth", G_TYPE_INT, 16, NULL);
gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), caps);
gst_caps_unref (caps);
done: done:
return ret; return ret;