mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-28 19:20:35 +00:00
celt: port to 0.11
This commit is contained in:
parent
dbf0b994f2
commit
99a4e43c76
3 changed files with 97 additions and 111 deletions
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@ -310,7 +310,7 @@ GST_PLUGINS_NONPORTED=" aiff asfmux \
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sdi segmentclip siren speed subenc stereo tta videofilters \
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videomeasure videosignal vmnc \
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decklink fbdev linsys shm vcd \
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apexsink bz2 cdaudio celt cog curl dc1394 dirac directfb resindvd \
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apexsink bz2 cdaudio cog curl dc1394 dirac directfb resindvd \
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gsettings jp2k ladspa mimic \
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musepack musicbrainz nas neon ofa openal opencv rsvg sdl smooth sndfile soundtouch spandsp timidity \
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wildmidi xvid lv2 teletextdec opus dvb"
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@ -57,6 +57,7 @@ GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw, "
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"format = (string) " GST_AUDIO_NE (S16) ", "
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"layout = (string) interleaved, "
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"rate = (int) [ 32000, 64000 ], " "channels = (int) [ 1, 2 ]")
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);
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@ -80,8 +81,10 @@ static GstFlowReturn gst_celt_dec_handle_frame (GstAudioDecoder * dec,
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static void
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gst_celt_dec_class_init (GstCeltDecClass * klass)
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{
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GstElementClass *gstelement_class;
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GstAudioDecoderClass *gstbase_class;
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gstelement_class = (GstElementClass *) klass;
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gstbase_class = (GstAudioDecoderClass *) klass;
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gstbase_class->start = GST_DEBUG_FUNCPTR (gst_celt_dec_start);
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@ -161,13 +164,16 @@ gst_celt_dec_stop (GstAudioDecoder * dec)
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static GstFlowReturn
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gst_celt_dec_parse_header (GstCeltDec * dec, GstBuffer * buf)
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{
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GstCaps *caps;
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gint error = CELT_OK;
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GstMapInfo map;
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GstAudioInfo info;
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/* get the header */
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gst_buffer_map (buf, &map, GST_MAP_READ);
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error =
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celt_header_from_packet ((const unsigned char *) GST_BUFFER_DATA (buf),
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GST_BUFFER_SIZE (buf), &dec->header);
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celt_header_from_packet ((const unsigned char *) map.data,
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map.size, &dec->header);
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gst_buffer_unmap (buf, &map);
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if (error < 0)
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goto invalid_header;
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@ -206,21 +212,15 @@ gst_celt_dec_parse_header (GstCeltDec * dec, GstBuffer * buf)
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celt_mode_info (dec->mode, CELT_GET_FRAME_SIZE, &dec->frame_size);
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#endif
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/* set caps */
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caps = gst_caps_new_simple ("audio/x-raw-int",
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"rate", G_TYPE_INT, dec->header.sample_rate,
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"channels", G_TYPE_INT, dec->header.nb_channels,
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"signed", G_TYPE_BOOLEAN, TRUE,
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"endianness", G_TYPE_INT, G_BYTE_ORDER,
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"width", G_TYPE_INT, 16, "depth", G_TYPE_INT, 16, NULL);
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GST_DEBUG_OBJECT (dec, "rate=%d channels=%d frame-size=%d",
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dec->header.sample_rate, dec->header.nb_channels, dec->frame_size);
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if (!gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), caps))
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gst_audio_info_init (&info);
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gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_S16,
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dec->header.sample_rate, dec->header.nb_channels, NULL);
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if (!gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (dec), &info))
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goto nego_failed;
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gst_caps_unref (caps);
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return GST_FLOW_OK;
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/* ERRORS */
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@ -251,7 +251,6 @@ nego_failed:
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{
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GST_ELEMENT_ERROR (GST_ELEMENT (dec), STREAM, DECODE,
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(NULL), ("couldn't negotiate format"));
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gst_caps_unref (caps);
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return GST_FLOW_NOT_NEGOTIATED;
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}
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}
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@ -266,7 +265,7 @@ gst_celt_dec_parse_comments (GstCeltDec * dec, GstBuffer * buf)
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if (!list) {
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GST_WARNING_OBJECT (dec, "couldn't decode comments");
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list = gst_tag_list_new ();
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list = gst_tag_list_new_empty ();
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}
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if (encoder) {
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@ -292,8 +291,8 @@ gst_celt_dec_parse_comments (GstCeltDec * dec, GstBuffer * buf)
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GST_INFO_OBJECT (dec, "tags: %" GST_PTR_FORMAT, list);
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gst_element_found_tags_for_pad (GST_ELEMENT (dec),
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GST_AUDIO_DECODER_SRC_PAD (dec), list);
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gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (dec),
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gst_event_new_tag (list));
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g_free (encoder);
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g_free (ver);
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@ -311,13 +310,15 @@ gst_celt_dec_parse_data (GstCeltDec * dec, GstBuffer * buf)
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gint16 *out_data;
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gint error = CELT_OK;
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int skip = 0;
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GstMapInfo map, omap;
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if (!dec->frame_size)
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goto not_negotiated;
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if (G_LIKELY (GST_BUFFER_SIZE (buf))) {
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data = GST_BUFFER_DATA (buf);
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size = GST_BUFFER_SIZE (buf);
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if (G_LIKELY (buf && gst_buffer_get_size (buf))) {
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gst_buffer_map (buf, &map, GST_MAP_READ);
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data = map.data;
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size = map.size;
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} else {
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/* FIXME ? actually consider how much concealment is needed */
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/* concealment data, pass NULL as the bits parameters */
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@ -337,16 +338,10 @@ gst_celt_dec_parse_data (GstCeltDec * dec, GstBuffer * buf)
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#endif
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}
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res = gst_pad_alloc_buffer_and_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec),
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GST_BUFFER_OFFSET_NONE, dec->frame_size * dec->header.nb_channels * 2,
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GST_PAD_CAPS (GST_AUDIO_DECODER_SRC_PAD (dec)), &outbuf);
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if (res != GST_FLOW_OK) {
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GST_DEBUG_OBJECT (dec, "buf alloc flow: %s", gst_flow_get_name (res));
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return res;
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}
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out_data = (gint16 *) GST_BUFFER_DATA (outbuf);
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outbuf =
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gst_buffer_new_and_alloc (dec->frame_size * dec->header.nb_channels * 2);
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gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
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out_data = (gint16 *) omap.data;
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GST_LOG_OBJECT (dec, "decoding frame");
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@ -355,6 +350,11 @@ gst_celt_dec_parse_data (GstCeltDec * dec, GstBuffer * buf)
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#else
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error = celt_decode (dec->state, data, size, out_data);
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#endif
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gst_buffer_unmap (outbuf, &omap);
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if (buf)
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gst_buffer_unmap (buf, &map);
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#ifdef HAVE_CELT_0_11
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if (error < 0) {
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#else
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@ -366,10 +366,7 @@ gst_celt_dec_parse_data (GstCeltDec * dec, GstBuffer * buf)
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if (skip > 0) {
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GST_ERROR_OBJECT (dec, "skipping %d samples", skip);
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GST_BUFFER_DATA (outbuf) = GST_BUFFER_DATA (outbuf) +
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skip * dec->header.nb_channels * 2;
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GST_BUFFER_SIZE (outbuf) = GST_BUFFER_SIZE (outbuf) -
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skip * dec->header.nb_channels * 2;
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gst_buffer_resize (outbuf, skip * dec->header.nb_channels * 2, -1);
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}
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res = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (dec), outbuf, 1);
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@ -443,6 +440,22 @@ done:
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return ret;
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}
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static gint
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_gst_buffer_memcmp (GstBuffer * buf1, GstBuffer * buf2)
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{
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GstMapInfo map;
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gint ret;
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if (gst_buffer_get_size (buf1) == gst_buffer_get_size (buf2))
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return 1;
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gst_buffer_map (buf1, &map, GST_MAP_READ);
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ret = gst_buffer_memcmp (buf2, 0, map.data, map.size);
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gst_buffer_unmap (buf1, &map);
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return ret;
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}
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static GstFlowReturn
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gst_celt_dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buf)
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{
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@ -458,15 +471,11 @@ gst_celt_dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buf)
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/* If we have the streamheader and vorbiscomment from the caps already
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* ignore them here */
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if (dec->streamheader && dec->vorbiscomment) {
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if (GST_BUFFER_SIZE (dec->streamheader) == GST_BUFFER_SIZE (buf)
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&& memcmp (GST_BUFFER_DATA (dec->streamheader), GST_BUFFER_DATA (buf),
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GST_BUFFER_SIZE (buf)) == 0) {
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if (_gst_buffer_memcmp (dec->streamheader, buf) == 0) {
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GST_DEBUG_OBJECT (dec, "found streamheader");
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gst_audio_decoder_finish_frame (bdec, NULL, 1);
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res = GST_FLOW_OK;
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} else if (GST_BUFFER_SIZE (dec->vorbiscomment) == GST_BUFFER_SIZE (buf)
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&& memcmp (GST_BUFFER_DATA (dec->vorbiscomment), GST_BUFFER_DATA (buf),
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GST_BUFFER_SIZE (buf)) == 0) {
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} else if (_gst_buffer_memcmp (dec->vorbiscomment, buf) == 0) {
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GST_DEBUG_OBJECT (dec, "found vorbiscomments");
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gst_audio_decoder_finish_frame (bdec, NULL, 1);
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res = GST_FLOW_OK;
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@ -475,9 +484,7 @@ gst_celt_dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buf)
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for (l = dec->extra_headers; l; l = l->next) {
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GstBuffer *header = l->data;
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if (GST_BUFFER_SIZE (header) == GST_BUFFER_SIZE (buf) &&
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memcmp (GST_BUFFER_DATA (header), GST_BUFFER_DATA (buf),
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GST_BUFFER_SIZE (buf)) == 0) {
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if (_gst_buffer_memcmp (header, buf) == 0) {
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GST_DEBUG_OBJECT (dec, "found extra header buffer");
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gst_audio_decoder_finish_frame (bdec, NULL, 1);
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res = GST_FLOW_OK;
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@ -80,11 +80,10 @@ gst_celt_enc_prediction_get_type (void)
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static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw-int, "
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"rate = (int) [ 32000, 64000 ], "
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"channels = (int) [ 1, 2 ], "
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"endianness = (int) BYTE_ORDER, "
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"signed = (boolean) TRUE, " "width = (int) 16, " "depth = (int) 16")
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GST_STATIC_CAPS ("audio/x-raw, "
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"format = (string) " GST_AUDIO_NE (S16) ", "
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"layout = (string) interleaved, "
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"rate = (int) [ 32000, 64000 ], " "channels = (int) [ 1, 2 ]")
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);
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static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
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@ -131,42 +130,19 @@ static gboolean gst_celt_enc_sink_event (GstAudioEncoder * enc,
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static GstFlowReturn gst_celt_enc_pre_push (GstAudioEncoder * benc,
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GstBuffer ** buffer);
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static void
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gst_celt_enc_setup_interfaces (GType celtenc_type)
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{
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static const GInterfaceInfo tag_setter_info = { NULL, NULL, NULL };
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g_type_add_interface_static (celtenc_type, GST_TYPE_TAG_SETTER,
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&tag_setter_info);
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GST_DEBUG_CATEGORY_INIT (celtenc_debug, "celtenc", 0, "Celt encoder");
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}
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GST_BOILERPLATE_FULL (GstCeltEnc, gst_celt_enc, GstAudioEncoder,
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GST_TYPE_AUDIO_ENCODER, gst_celt_enc_setup_interfaces);
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static void
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gst_celt_enc_base_init (gpointer g_class)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&src_factory));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&sink_factory));
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gst_element_class_set_details_simple (element_class, "Celt audio encoder",
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"Codec/Encoder/Audio",
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"Encodes audio in Celt format",
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"Sebastian Dröge <sebastian.droege@collabora.co.uk>");
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}
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#define gst_celt_enc_parent_class parent_class
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G_DEFINE_TYPE_WITH_CODE (GstCeltEnc, gst_celt_enc, GST_TYPE_AUDIO_ENCODER,
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G_IMPLEMENT_INTERFACE (GST_TYPE_TAG_SETTER, NULL));
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static void
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gst_celt_enc_class_init (GstCeltEncClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstAudioEncoderClass *gstbase_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gstbase_class = (GstAudioEncoderClass *) klass;
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gobject_class->set_property = gst_celt_enc_set_property;
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@ -211,10 +187,21 @@ gst_celt_enc_class_init (GstCeltEncClass * klass)
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"Controls the start band that should be used",
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0, G_MAXINT, DEFAULT_START_BAND,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&src_factory));
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&sink_factory));
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gst_element_class_set_details_simple (gstelement_class, "Celt audio encoder",
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"Codec/Encoder/Audio",
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"Encodes audio in Celt format",
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"Sebastian Dröge <sebastian.droege@collabora.co.uk>");
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GST_DEBUG_CATEGORY_INIT (celtenc_debug, "celtenc", 0, "Celt encoder");
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}
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static void
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gst_celt_enc_init (GstCeltEnc * enc, GstCeltEncClass * klass)
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gst_celt_enc_init (GstCeltEnc * enc)
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{
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enc->bitrate = DEFAULT_BITRATE;
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enc->frame_size = DEFAULT_FRAMESIZE;
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@ -433,7 +420,7 @@ gst_celt_enc_create_metadata_buffer (GstCeltEnc * enc)
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if (tags == NULL) {
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/* FIXME: better fix chain of callers to not write metadata at all,
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* if there is none */
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empty_tags = gst_tag_list_new ();
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empty_tags = gst_tag_list_new_empty ();
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tags = empty_tags;
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}
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comments = gst_tag_list_to_vorbiscomment_buffer (tags, NULL,
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@ -458,16 +445,19 @@ gst_celt_enc_encode (GstCeltEnc * enc, GstBuffer * buf)
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gint16 *data, *data0 = NULL;
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gint outsize, size;
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GstBuffer *outbuf;
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GstMapInfo map, omap;
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if (G_LIKELY (buf)) {
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data = (gint16 *) GST_BUFFER_DATA (buf);
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size = GST_BUFFER_SIZE (buf);
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gst_buffer_map (buf, &map, GST_MAP_READ);
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data = (gint16 *) map.data;
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size = map.size;
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if (G_UNLIKELY (size % bytes)) {
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if (G_UNLIKELY (map.size % bytes)) {
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GST_DEBUG_OBJECT (enc, "draining; adding silence samples");
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size = ((size / bytes) + 1) * bytes;
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data0 = data = g_malloc0 (size);
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memcpy (data, GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
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data0 = g_malloc0 (size);
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memcpy (data0, data, size);
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data = data0;
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}
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} else {
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GST_DEBUG_OBJECT (enc, "nothing to drain");
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@ -481,25 +471,21 @@ gst_celt_enc_encode (GstCeltEnc * enc, GstBuffer * buf)
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bytes_per_packet = (enc->max_bitrate * frame_size / enc->rate + 4) / 8;
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}
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ret = gst_pad_alloc_buffer_and_set_caps (GST_AUDIO_ENCODER_SRC_PAD (enc),
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GST_BUFFER_OFFSET_NONE, bytes_per_packet,
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GST_PAD_CAPS (GST_AUDIO_ENCODER_SRC_PAD (enc)), &outbuf);
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if (GST_FLOW_OK != ret)
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goto done;
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outbuf = gst_buffer_new_and_alloc (bytes_per_packet);
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GST_DEBUG_OBJECT (enc, "encoding %d samples (%d bytes)", frame_size, bytes);
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gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
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#ifdef HAVE_CELT_0_8
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outsize =
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celt_encode (enc->state, data, frame_size,
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GST_BUFFER_DATA (outbuf), bytes_per_packet);
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celt_encode (enc->state, data, frame_size, omap.data, bytes_per_packet);
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#else
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outsize =
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celt_encode (enc->state, data, NULL,
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GST_BUFFER_DATA (outbuf), bytes_per_packet);
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outsize = celt_encode (enc->state, data, NULL, omap.data, bytes_per_packet);
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#endif
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gst_buffer_unmap (outbuf, &omap);
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gst_buffer_unmap (buf, &map);
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if (outsize < 0) {
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GST_ELEMENT_ERROR (enc, STREAM, ENCODE, (NULL),
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("encoding failed: %d", outsize));
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@ -552,14 +538,14 @@ _gst_caps_set_buffer_array (GstCaps * caps, const gchar * field,
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va_start (va, buf);
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/* put buffers in a fixed list */
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while (buf) {
|
||||
g_assert (gst_buffer_is_metadata_writable (buf));
|
||||
g_assert (gst_buffer_is_writable (buf));
|
||||
|
||||
/* mark buffer */
|
||||
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_IN_CAPS);
|
||||
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_HEADER);
|
||||
|
||||
g_value_init (&value, GST_TYPE_BUFFER);
|
||||
buf = gst_buffer_copy (buf);
|
||||
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_IN_CAPS);
|
||||
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_HEADER);
|
||||
gst_value_set_buffer (&value, buf);
|
||||
gst_buffer_unref (buf);
|
||||
gst_value_array_append_value (&array, &value);
|
||||
|
@ -601,9 +587,7 @@ gst_celt_enc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf)
|
|||
g_free (data);
|
||||
goto no_header;
|
||||
}
|
||||
buf1 = gst_buffer_new ();
|
||||
GST_BUFFER_DATA (buf1) = GST_BUFFER_MALLOCDATA (buf1) = data;
|
||||
GST_BUFFER_SIZE (buf1) = header_size;
|
||||
buf1 = gst_buffer_new_wrapped (data, header_size);
|
||||
GST_BUFFER_OFFSET_END (buf1) = 0;
|
||||
GST_BUFFER_OFFSET (buf1) = 0;
|
||||
|
||||
|
@ -611,8 +595,7 @@ gst_celt_enc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf)
|
|||
buf2 = gst_celt_enc_create_metadata_buffer (enc);
|
||||
|
||||
/* mark and put on caps */
|
||||
caps = gst_pad_get_caps (GST_AUDIO_ENCODER_SRC_PAD (enc));
|
||||
gst_caps_set_simple (caps,
|
||||
caps = gst_caps_new_simple ("audio/x-celt",
|
||||
"rate", G_TYPE_INT, enc->rate,
|
||||
"channels", G_TYPE_INT, enc->channels,
|
||||
"frame-size", G_TYPE_INT, enc->frame_size, NULL);
|
||||
|
@ -622,10 +605,7 @@ gst_celt_enc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf)
|
|||
GST_DEBUG_OBJECT (enc, "here are the caps: %" GST_PTR_FORMAT, caps);
|
||||
GST_LOG_OBJECT (enc, "rate=%d channels=%d frame-size=%d",
|
||||
enc->rate, enc->channels, enc->frame_size);
|
||||
gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (enc), caps);
|
||||
|
||||
gst_buffer_set_caps (buf1, caps);
|
||||
gst_buffer_set_caps (buf2, caps);
|
||||
gst_audio_encoder_set_output_format (GST_AUDIO_ENCODER (enc), caps);
|
||||
gst_caps_unref (caps);
|
||||
|
||||
/* push out buffers */
|
||||
|
@ -640,7 +620,7 @@ gst_celt_enc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf)
|
|||
}
|
||||
|
||||
GST_DEBUG_OBJECT (enc, "received buffer %p of %u bytes", buf,
|
||||
buf ? GST_BUFFER_SIZE (buf) : 0);
|
||||
buf ? gst_buffer_get_size (buf) : 0);
|
||||
|
||||
ret = gst_celt_enc_encode (enc, buf);
|
||||
|
||||
|
@ -663,10 +643,9 @@ gst_celt_enc_push_buffer (GstCeltEnc * enc, GstBuffer * buffer)
|
|||
{
|
||||
guint size;
|
||||
|
||||
size = GST_BUFFER_SIZE (buffer);
|
||||
size = gst_buffer_get_size (buffer);
|
||||
GST_DEBUG_OBJECT (enc, "pushing output buffer of size %u", size);
|
||||
|
||||
gst_buffer_set_caps (buffer, GST_PAD_CAPS (GST_AUDIO_ENCODER_SRC_PAD (enc)));
|
||||
return gst_pad_push (GST_AUDIO_ENCODER_SRC_PAD (enc), buffer);
|
||||
}
|
||||
|
||||
|
|
Loading…
Reference in a new issue