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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2025-04-11 02:24:13 +00:00
sirendec: port to audiodecoder
This commit is contained in:
Mark Nauwelaerts
parent
6f8e60e24f
commit
22b9b13166
2 changed files with 92 additions and 165 deletions
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@ -69,14 +69,14 @@ enum
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ARG_0,
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};
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static void gst_siren_dec_finalize (GObject * object);
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static GstStateChangeReturn
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gst_siren_change_state (GstElement * element, GstStateChange transition);
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static gboolean gst_siren_dec_sink_setcaps (GstPad * pad, GstCaps * caps);
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static gboolean gst_siren_dec_sink_event (GstPad * pad, GstEvent * event);
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static GstFlowReturn gst_siren_dec_chain (GstPad * pad, GstBuffer * buf);
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static gboolean gst_siren_dec_start (GstAudioDecoder * dec);
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static gboolean gst_siren_dec_stop (GstAudioDecoder * dec);
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static gboolean gst_siren_dec_set_format (GstAudioDecoder * dec,
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GstCaps * caps);
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static gboolean gst_siren_dec_parse (GstAudioDecoder * dec,
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GstAdapter * adapter, gint * offset, gint * length);
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static GstFlowReturn gst_siren_dec_handle_frame (GstAudioDecoder * dec,
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GstBuffer * buffer);
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static void
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_do_init (GType type)
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@ -84,8 +84,8 @@ _do_init (GType type)
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GST_DEBUG_CATEGORY_INIT (sirendec_debug, "sirendec", 0, "sirendec");
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}
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GST_BOILERPLATE_FULL (GstSirenDec, gst_siren_dec, GstElement,
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GST_TYPE_ELEMENT, _do_init);
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GST_BOILERPLATE_FULL (GstSirenDec, gst_siren_dec, GstAudioDecoder,
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GST_TYPE_AUDIO_DECODER, _do_init);
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static void
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gst_siren_dec_base_init (gpointer klass)
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@ -104,17 +104,15 @@ gst_siren_dec_base_init (gpointer klass)
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static void
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gst_siren_dec_class_init (GstSirenDecClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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GstAudioDecoderClass *base_class = GST_AUDIO_DECODER_CLASS (klass);
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GST_DEBUG ("Initializing Class");
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gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_siren_dec_finalize);
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gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_siren_change_state);
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base_class->start = GST_DEBUG_FUNCPTR (gst_siren_dec_start);
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base_class->stop = GST_DEBUG_FUNCPTR (gst_siren_dec_stop);
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base_class->set_format = GST_DEBUG_FUNCPTR (gst_siren_dec_set_format);
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base_class->parse = GST_DEBUG_FUNCPTR (gst_siren_dec_parse);
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base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_siren_dec_handle_frame);
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GST_DEBUG ("Class Init done");
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}
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@ -122,119 +120,103 @@ gst_siren_dec_class_init (GstSirenDecClass * klass)
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static void
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gst_siren_dec_init (GstSirenDec * dec, GstSirenDecClass * klass)
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{
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GST_DEBUG_OBJECT (dec, "Initializing");
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dec->decoder = Siren7_NewDecoder (16000);;
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dec->sinkpad = gst_pad_new_from_static_template (&sinktemplate, "sink");
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dec->srcpad = gst_pad_new_from_static_template (&srctemplate, "src");
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gst_pad_set_setcaps_function (dec->sinkpad,
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GST_DEBUG_FUNCPTR (gst_siren_dec_sink_setcaps));
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gst_pad_set_event_function (dec->sinkpad,
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GST_DEBUG_FUNCPTR (gst_siren_dec_sink_event));
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gst_pad_set_chain_function (dec->sinkpad,
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GST_DEBUG_FUNCPTR (gst_siren_dec_chain));
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gst_element_add_pad (GST_ELEMENT (dec), dec->sinkpad);
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gst_element_add_pad (GST_ELEMENT (dec), dec->srcpad);
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dec->adapter = gst_adapter_new ();
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GST_DEBUG_OBJECT (dec, "Init done");
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}
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static void
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gst_siren_dec_finalize (GObject * object)
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{
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GstSirenDec *dec = GST_SIREN_DEC (object);
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GST_DEBUG_OBJECT (dec, "Finalize");
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Siren7_CloseDecoder (dec->decoder);
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g_object_unref (dec->adapter);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static gboolean
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gst_siren_dec_sink_setcaps (GstPad * pad, GstCaps * caps)
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gst_siren_dec_start (GstAudioDecoder * dec)
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{
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GstSirenDec *sdec = GST_SIREN_DEC (dec);
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GST_DEBUG_OBJECT (dec, "start");
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sdec->decoder = Siren7_NewDecoder (16000);;
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/* no flushing please */
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gst_audio_decoder_set_drainable (dec, FALSE);
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return TRUE;
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}
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static gboolean
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gst_siren_dec_stop (GstAudioDecoder * dec)
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{
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GstSirenDec *sdec = GST_SIREN_DEC (dec);
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GST_DEBUG_OBJECT (dec, "stop");
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Siren7_CloseDecoder (sdec->decoder);
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return TRUE;
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}
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static gboolean
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gst_siren_dec_negotiate (GstSirenDec * dec)
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{
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GstSirenDec *dec;
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gboolean res;
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GstCaps *outcaps;
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dec = GST_SIREN_DEC (GST_PAD_PARENT (pad));
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outcaps = gst_static_pad_template_get_caps (&srctemplate);
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res = gst_pad_set_caps (dec->srcpad, outcaps);
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res = gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), outcaps);
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gst_caps_unref (outcaps);
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return res;
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}
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static gboolean
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gst_siren_dec_sink_event (GstPad * pad, GstEvent * event)
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gst_siren_dec_set_format (GstAudioDecoder * bdec, GstCaps * caps)
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{
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GstSirenDec *dec;
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gboolean res;
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dec = GST_SIREN_DEC (GST_PAD_PARENT (pad));
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dec = GST_SIREN_DEC (bdec);
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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_EOS:
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gst_adapter_clear (dec->adapter);
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res = gst_pad_push_event (dec->srcpad, event);
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break;
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case GST_EVENT_FLUSH_STOP:
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gst_adapter_clear (dec->adapter);
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res = gst_pad_push_event (dec->srcpad, event);
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break;
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default:
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res = gst_pad_push_event (dec->srcpad, event);
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break;
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}
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return res;
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return gst_siren_dec_negotiate (dec);
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}
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static GstFlowReturn
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gst_siren_dec_chain (GstPad * pad, GstBuffer * buf)
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gst_siren_dec_parse (GstAudioDecoder * dec, GstAdapter * adapter,
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gint * offset, gint * length)
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{
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gint size;
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GstFlowReturn ret;
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size = gst_adapter_available (adapter);
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g_return_val_if_fail (size > 0, GST_FLOW_ERROR);
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/* accept any multiple of frames */
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if (size > 40) {
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ret = GST_FLOW_OK;
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*offset = 0;
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*length = size - (size % 40);
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} else {
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ret = GST_FLOW_UNEXPECTED;
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}
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return ret;
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}
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static GstFlowReturn
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gst_siren_dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buf)
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{
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GstSirenDec *dec;
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GstFlowReturn ret = GST_FLOW_OK;
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GstBuffer *out_buf;
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guint8 *in_data, *out_data;
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guint8 *to_free = NULL;
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guint i, size, num_frames;
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gint out_size, in_size;
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gint decode_ret;
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gboolean discont;
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GstClockTime timestamp;
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guint64 distance;
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GstCaps *outcaps;
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dec = GST_SIREN_DEC (GST_PAD_PARENT (pad));
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dec = GST_SIREN_DEC (bdec);
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discont = GST_BUFFER_IS_DISCONT (buf);
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if (discont) {
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GST_DEBUG_OBJECT (dec, "received DISCONT, flush adapter");
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gst_adapter_clear (dec->adapter);
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dec->discont = TRUE;
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}
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size = GST_BUFFER_SIZE (buf);
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gst_adapter_push (dec->adapter, buf);
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GST_LOG_OBJECT (dec, "Received buffer of size %u", size);
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size = gst_adapter_available (dec->adapter);
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GST_LOG_OBJECT (dec, "Received buffer of size %u with adapter of size : %u",
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GST_BUFFER_SIZE (buf), size);
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g_return_val_if_fail (size % 40 == 0, GST_FLOW_ERROR);
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g_return_val_if_fail (size > 0, GST_FLOW_ERROR);
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/* process 40 input bytes into 640 output bytes */
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num_frames = size / 40;
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if (num_frames == 0)
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goto done;
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/* this is the input/output size */
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in_size = num_frames * 40;
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out_size = num_frames * 640;
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@ -242,32 +224,19 @@ gst_siren_dec_chain (GstPad * pad, GstBuffer * buf)
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GST_LOG_OBJECT (dec, "we have %u frames, %u in, %u out", num_frames, in_size,
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out_size);
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/* set output caps when needed */
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if ((outcaps = GST_PAD_CAPS (dec->srcpad)) == NULL) {
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outcaps = gst_static_pad_template_get_caps (&srctemplate);
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gst_pad_set_caps (dec->srcpad, outcaps);
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gst_caps_unref (outcaps);
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/* allow and handle un-negotiated input */
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if (G_UNLIKELY (GST_PAD_CAPS (GST_AUDIO_DECODER_SRC_PAD (dec)) == NULL)) {
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gst_siren_dec_negotiate (dec);
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}
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/* get a buffer */
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ret = gst_pad_alloc_buffer_and_set_caps (dec->srcpad, -1,
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out_size, outcaps, &out_buf);
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ret = gst_pad_alloc_buffer_and_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), -1,
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out_size, GST_PAD_CAPS (GST_AUDIO_DECODER_SRC_PAD (dec)), &out_buf);
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if (ret != GST_FLOW_OK)
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goto alloc_failed;
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/* get the timestamp for the output buffer */
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timestamp = gst_adapter_prev_timestamp (dec->adapter, &distance);
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/* add the amount of time taken by the distance, each frame is 20ms */
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if (timestamp != -1)
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timestamp += (distance / 40) * FRAME_DURATION;
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GST_LOG_OBJECT (dec,
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"timestamp %" GST_TIME_FORMAT ", distance %" G_GUINT64_FORMAT,
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GST_TIME_ARGS (timestamp), distance);
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/* get the input data for all the frames */
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to_free = in_data = gst_adapter_take (dec->adapter, in_size);
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in_data = GST_BUFFER_DATA (buf);
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out_data = GST_BUFFER_DATA (out_buf);
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for (i = 0; i < num_frames; i++) {
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@ -285,21 +254,11 @@ gst_siren_dec_chain (GstPad * pad, GstBuffer * buf)
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GST_LOG_OBJECT (dec, "Finished decoding");
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/* mark discont */
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if (dec->discont) {
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GST_BUFFER_FLAG_SET (out_buf, GST_BUFFER_FLAG_DISCONT);
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dec->discont = FALSE;
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}
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GST_BUFFER_TIMESTAMP (out_buf) = timestamp;
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GST_BUFFER_DURATION (out_buf) = num_frames * FRAME_DURATION;
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ret = gst_pad_push (dec->srcpad, out_buf);
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/* might really be multiple frames,
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* but was treated as one for all purposes here */
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ret = gst_audio_decoder_finish_frame (bdec, out_buf, 1);
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done:
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if (to_free)
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g_free (to_free);
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return ret;
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/* ERRORS */
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@ -311,41 +270,15 @@ alloc_failed:
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}
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decode_error:
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{
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GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL),
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("Error decoding frame: %d", decode_ret));
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ret = GST_FLOW_ERROR;
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GST_AUDIO_DECODER_ERROR (bdec, 1, STREAM, DECODE, (NULL),
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("Error decoding frame: %d", decode_ret), ret);
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if (ret == GST_FLOW_OK)
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gst_audio_decoder_finish_frame (bdec, NULL, 1);
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gst_buffer_unref (out_buf);
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goto done;
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}
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}
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static GstStateChangeReturn
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gst_siren_change_state (GstElement * element, GstStateChange transition)
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{
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GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
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GstSirenDec *dec = GST_SIREN_DEC (element);
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switch (transition) {
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case GST_STATE_CHANGE_READY_TO_PAUSED:
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dec->discont = FALSE;
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break;
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default:
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break;
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}
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ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
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switch (transition) {
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case GST_STATE_CHANGE_PAUSED_TO_READY:
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gst_adapter_clear (dec->adapter);
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break;
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default:
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break;
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}
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return ret;
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}
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gboolean
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gst_siren_dec_plugin_init (GstPlugin * plugin)
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{
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@ -24,7 +24,7 @@
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#define __GST_SIREN_DEC_H__
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#include <gst/gst.h>
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#include <gst/base/gstadapter.h>
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#include <gst/audio/gstaudiodecoder.h>
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#include "siren7.h"
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@ -48,21 +48,15 @@ typedef struct _GstSirenDecPrivate GstSirenDecPrivate;
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struct _GstSirenDec
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{
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GstElement parent;
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GstAudioDecoder parent;
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/* Protected by stream lock */
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SirenDecoder decoder;
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GstAdapter *adapter;
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gboolean discont;
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GstPad *sinkpad;
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GstPad *srcpad;
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};
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struct _GstSirenDecClass
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{
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GstElementClass parent_class;
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GstAudioDecoderClass parent_class;
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};
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GType gst_siren_dec_get_type (void);
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