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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-23 16:50:47 +00:00
wavenc: port to 0.11
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parent
88f107b30b
commit
9cf287ebcc
1 changed files with 36 additions and 27 deletions
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@ -134,7 +134,8 @@ static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
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GST_STATIC_CAPS ("audio/x-wav")
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);
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GST_BOILERPLATE (GstWavEnc, gst_wavenc, GstElement, GST_TYPE_ELEMENT);
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#define gst_wavenc_parent_class parent_class
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G_DEFINE_TYPE (GstWavEnc, gst_wavenc, GST_TYPE_ELEMENT);
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static GstFlowReturn gst_wavenc_chain (GstPad * pad, GstBuffer * buf);
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static gboolean gst_wavenc_event (GstPad * pad, GstEvent * event);
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@ -143,9 +144,13 @@ static GstStateChangeReturn gst_wavenc_change_state (GstElement * element,
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static gboolean gst_wavenc_sink_setcaps (GstPad * pad, GstCaps * caps);
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static void
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gst_wavenc_base_init (gpointer g_class)
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gst_wavenc_class_init (GstWavEncClass * klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
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GstElementClass *element_class;
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element_class = (GstElementClass *) klass;
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element_class->change_state = GST_DEBUG_FUNCPTR (gst_wavenc_change_state);
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gst_element_class_set_details_simple (element_class, "WAV audio muxer",
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"Codec/Muxer/Audio",
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@ -160,25 +165,13 @@ gst_wavenc_base_init (gpointer g_class)
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}
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static void
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gst_wavenc_class_init (GstWavEncClass * klass)
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{
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GstElementClass *element_class;
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element_class = (GstElementClass *) klass;
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element_class->change_state = GST_DEBUG_FUNCPTR (gst_wavenc_change_state);
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}
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static void
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gst_wavenc_init (GstWavEnc * wavenc, GstWavEncClass * klass)
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gst_wavenc_init (GstWavEnc * wavenc)
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{
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wavenc->sinkpad = gst_pad_new_from_static_template (&sink_factory, "sink");
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gst_pad_set_chain_function (wavenc->sinkpad,
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GST_DEBUG_FUNCPTR (gst_wavenc_chain));
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gst_pad_set_event_function (wavenc->sinkpad,
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GST_DEBUG_FUNCPTR (gst_wavenc_event));
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gst_pad_set_setcaps_function (wavenc->sinkpad,
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GST_DEBUG_FUNCPTR (gst_wavenc_sink_setcaps));
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gst_pad_use_fixed_caps (wavenc->sinkpad);
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gst_element_add_pad (GST_ELEMENT (wavenc), wavenc->sinkpad);
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@ -199,7 +192,7 @@ gst_wavenc_create_header_buf (GstWavEnc * wavenc, guint audio_data_size)
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guint8 *header;
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buf = gst_buffer_new_and_alloc (WAV_HEADER_LEN);
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header = GST_BUFFER_DATA (buf);
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header = gst_buffer_map (buf, NULL, NULL, GST_MAP_WRITE);
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memset (header, 0, WAV_HEADER_LEN);
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wave.common.wChannels = wavenc->channels;
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@ -236,7 +229,7 @@ gst_wavenc_create_header_buf (GstWavEnc * wavenc, guint audio_data_size)
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memcpy (header + 36, (char *) wave.data.id, 4);
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GST_WRITE_UINT32_LE (header + 40, wave.data.len);
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gst_buffer_set_caps (buf, GST_PAD_CAPS (wavenc->srcpad));
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gst_buffer_unmap (buf, header, -1);
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return buf;
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}
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@ -246,10 +239,11 @@ gst_wavenc_push_header (GstWavEnc * wavenc, guint audio_data_size)
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{
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GstFlowReturn ret;
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GstBuffer *outbuf;
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GstSegment segment;
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/* seek to beginning of file */
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gst_pad_push_event (wavenc->srcpad,
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gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_BYTES, 0, -1, 0));
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gst_segment_init (&segment, GST_FORMAT_BYTES);
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gst_pad_push_event (wavenc->srcpad, gst_event_new_segment (&segment));
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GST_DEBUG_OBJECT (wavenc, "writing header with datasize=%u", audio_data_size);
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@ -273,13 +267,18 @@ gst_wavenc_sink_setcaps (GstPad * pad, GstCaps * caps)
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GstStructure *structure;
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const gchar *name;
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gint chans, rate, width;
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GstCaps *ccaps;
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wavenc = GST_WAVENC (gst_pad_get_parent (pad));
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if (wavenc->sent_header && !gst_caps_can_intersect (caps, GST_PAD_CAPS (pad))) {
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ccaps = gst_pad_get_current_caps (pad);
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if (wavenc->sent_header && ccaps && !gst_caps_can_intersect (caps, ccaps)) {
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gst_caps_unref (ccaps);
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GST_WARNING_OBJECT (wavenc, "cannot change format in middle of stream");
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goto fail;
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}
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if (ccaps)
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gst_caps_unref (ccaps);
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GST_DEBUG_OBJECT (wavenc, "got caps: %" GST_PTR_FORMAT, caps);
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@ -621,6 +620,17 @@ gst_wavenc_event (GstPad * pad, GstEvent * event)
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wavenc = GST_WAVENC (gst_pad_get_parent (pad));
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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_CAPS:
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{
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GstCaps *caps;
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gst_event_parse_caps (event, &caps);
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gst_wavenc_sink_setcaps (pad, caps);
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/* have our own src caps */
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gst_event_unref (event);
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break;
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}
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case GST_EVENT_EOS:{
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GST_DEBUG_OBJECT (wavenc, "got EOS");
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#if 0
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@ -641,7 +651,7 @@ gst_wavenc_event (GstPad * pad, GstEvent * event)
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res = gst_pad_event_default (pad, event);
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break;
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}
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case GST_EVENT_NEWSEGMENT:
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case GST_EVENT_SEGMENT:
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/* Just drop it, it's probably in TIME format
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* anyway. We'll send our own newsegment event */
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gst_event_unref (event);
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@ -679,15 +689,14 @@ gst_wavenc_chain (GstPad * pad, GstBuffer * buf)
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}
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GST_LOG_OBJECT (wavenc, "pushing %u bytes raw audio, ts=%" GST_TIME_FORMAT,
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GST_BUFFER_SIZE (buf), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)));
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gst_buffer_get_size (buf), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)));
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buf = gst_buffer_make_metadata_writable (buf);
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buf = gst_buffer_make_writable (buf);
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gst_buffer_set_caps (buf, GST_PAD_CAPS (wavenc->srcpad));
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GST_BUFFER_OFFSET (buf) = WAV_HEADER_LEN + wavenc->length;
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GST_BUFFER_OFFSET_END (buf) = GST_BUFFER_OFFSET_NONE;
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wavenc->length += GST_BUFFER_SIZE (buf);
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wavenc->length += gst_buffer_get_size (buf);
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flow = gst_pad_push (wavenc->srcpad, buf);
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@ -715,7 +724,7 @@ gst_wavenc_change_state (GstElement * element, GstStateChange transition)
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break;
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}
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ret = parent_class->change_state (element, transition);
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ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
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if (ret != GST_STATE_CHANGE_SUCCESS)
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return ret;
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