mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-10-02 16:52:42 +00:00
gsmdec: port to audiodecoder
This commit is contained in:
Mark Nauwelaerts
parent
0acfa3cc1e
commit
bc1c77395e
2 changed files with 92 additions and 195 deletions
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@ -43,43 +43,16 @@ enum
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ARG_0
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};
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static void gst_gsmdec_base_init (gpointer g_class);
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static void gst_gsmdec_class_init (GstGSMDec * klass);
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static void gst_gsmdec_init (GstGSMDec * gsmdec);
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static void gst_gsmdec_finalize (GObject * object);
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static gboolean gst_gsmdec_sink_setcaps (GstPad * pad, GstCaps * caps);
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static gboolean gst_gsmdec_sink_event (GstPad * pad, GstEvent * event);
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static GstFlowReturn gst_gsmdec_chain (GstPad * pad, GstBuffer * buf);
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static GstElementClass *parent_class = NULL;
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static gboolean gst_gsmdec_start (GstAudioDecoder * dec);
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static gboolean gst_gsmdec_stop (GstAudioDecoder * dec);
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static gboolean gst_gsmdec_set_format (GstAudioDecoder * dec, GstCaps * caps);
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static GstFlowReturn gst_gsmdec_parse (GstAudioDecoder * dec,
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GstAdapter * adapter, gint * offset, gint * length);
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static GstFlowReturn gst_gsmdec_handle_frame (GstAudioDecoder * dec,
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GstBuffer * in_buf);
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/*static guint gst_gsmdec_signals[LAST_SIGNAL] = { 0 }; */
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GType
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gst_gsmdec_get_type (void)
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{
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static GType gsmdec_type = 0;
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if (!gsmdec_type) {
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static const GTypeInfo gsmdec_info = {
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sizeof (GstGSMDecClass),
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gst_gsmdec_base_init,
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NULL,
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(GClassInitFunc) gst_gsmdec_class_init,
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NULL,
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NULL,
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sizeof (GstGSMDec),
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0,
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(GInstanceInitFunc) gst_gsmdec_init,
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};
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gsmdec_type =
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g_type_register_static (GST_TYPE_ELEMENT, "GstGSMDec", &gsmdec_info, 0);
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}
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return gsmdec_type;
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}
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#define ENCODED_SAMPLES 160
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static GstStaticPadTemplate gsmdec_sink_template =
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@ -101,6 +74,9 @@ GST_STATIC_PAD_TEMPLATE ("src",
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"depth = (int) 16, " "rate = (int) [1, MAX], " "channels = (int) 1")
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);
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GST_BOILERPLATE (GstGSMDec, gst_gsmdec, GstAudioDecoder,
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GST_TYPE_AUDIO_DECODER);
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static void
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gst_gsmdec_base_init (gpointer g_class)
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{
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@ -116,63 +92,60 @@ gst_gsmdec_base_init (gpointer g_class)
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}
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static void
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gst_gsmdec_class_init (GstGSMDec * klass)
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gst_gsmdec_class_init (GstGSMDecClass * klass)
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{
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GObjectClass *gobject_class;
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GstAudioDecoderClass *base_class;
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gobject_class = (GObjectClass *) klass;
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base_class = (GstAudioDecoderClass *) klass;
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parent_class = g_type_class_peek_parent (klass);
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gobject_class->finalize = gst_gsmdec_finalize;
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base_class->start = GST_DEBUG_FUNCPTR (gst_gsmdec_start);
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base_class->stop = GST_DEBUG_FUNCPTR (gst_gsmdec_stop);
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base_class->set_format = GST_DEBUG_FUNCPTR (gst_gsmdec_set_format);
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base_class->parse = GST_DEBUG_FUNCPTR (gst_gsmdec_parse);
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base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_gsmdec_handle_frame);
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GST_DEBUG_CATEGORY_INIT (gsmdec_debug, "gsmdec", 0, "GSM Decoder");
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}
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static void
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gst_gsmdec_init (GstGSMDec * gsmdec)
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gst_gsmdec_init (GstGSMDec * gsmdec, GstGSMDecClass * klass)
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{
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/* create the sink and src pads */
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gsmdec->sinkpad =
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gst_pad_new_from_static_template (&gsmdec_sink_template, "sink");
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gst_pad_set_setcaps_function (gsmdec->sinkpad, gst_gsmdec_sink_setcaps);
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gst_pad_set_event_function (gsmdec->sinkpad, gst_gsmdec_sink_event);
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gst_pad_set_chain_function (gsmdec->sinkpad, gst_gsmdec_chain);
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gst_element_add_pad (GST_ELEMENT (gsmdec), gsmdec->sinkpad);
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gsmdec->srcpad =
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gst_pad_new_from_static_template (&gsmdec_src_template, "src");
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gst_element_add_pad (GST_ELEMENT (gsmdec), gsmdec->srcpad);
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gsmdec->state = gsm_create ();
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gsmdec->adapter = gst_adapter_new ();
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gsmdec->next_of = 0;
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gsmdec->next_ts = 0;
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}
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static void
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gst_gsmdec_finalize (GObject * object)
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{
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GstGSMDec *gsmdec;
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gsmdec = GST_GSMDEC (object);
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g_object_unref (gsmdec->adapter);
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gsm_destroy (gsmdec->state);
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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_gsmdec_sink_setcaps (GstPad * pad, GstCaps * caps)
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gst_gsmdec_start (GstAudioDecoder * dec)
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{
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GstGSMDec *gsmdec = GST_GSMDEC (dec);
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GST_DEBUG_OBJECT (dec, "start");
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gsmdec->state = gsm_create ();
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return TRUE;
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}
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static gboolean
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gst_gsmdec_stop (GstAudioDecoder * dec)
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{
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GstGSMDec *gsmdec = GST_GSMDEC (dec);
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GST_DEBUG_OBJECT (dec, "stop");
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gsm_destroy (gsmdec->state);
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return TRUE;
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}
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static gboolean
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gst_gsmdec_set_format (GstAudioDecoder * dec, GstCaps * caps)
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{
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GstGSMDec *gsmdec;
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GstCaps *srccaps;
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GstStructure *s;
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gboolean ret = FALSE;
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gint rate;
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gsmdec = GST_GSMDEC (gst_pad_get_parent (pad));
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gsmdec = GST_GSMDEC (dec);
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s = gst_caps_get_structure (caps, 0);
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if (s == NULL)
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@ -186,7 +159,9 @@ gst_gsmdec_sink_setcaps (GstPad * pad, GstCaps * caps)
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else
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goto wrong_caps;
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if (!gst_structure_get_int (s, "rate", &gsmdec->rate)) {
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gsmdec->needed = 33;
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if (!gst_structure_get_int (s, "rate", &rate)) {
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GST_WARNING_OBJECT (gsmdec, "missing sample rate parameter from sink caps");
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goto beach;
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}
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@ -194,21 +169,16 @@ gst_gsmdec_sink_setcaps (GstPad * pad, GstCaps * caps)
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/* MSGSM needs different framing */
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gsm_option (gsmdec->state, GSM_OPT_WAV49, &gsmdec->use_wav49);
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gsmdec->duration = gst_util_uint64_scale (ENCODED_SAMPLES,
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GST_SECOND, gsmdec->rate);
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/* Setting up src caps based on the input sample rate. */
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srccaps = gst_caps_new_simple ("audio/x-raw-int",
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"endianness", G_TYPE_INT, G_BYTE_ORDER,
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"signed", G_TYPE_BOOLEAN, TRUE,
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"width", G_TYPE_INT, 16,
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"depth", G_TYPE_INT, 16,
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"rate", G_TYPE_INT, gsmdec->rate, "channels", G_TYPE_INT, 1, NULL);
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ret = gst_pad_set_caps (gsmdec->srcpad, srccaps);
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"rate", G_TYPE_INT, rate, "channels", G_TYPE_INT, 1, NULL);
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ret = gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), srccaps);
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gst_caps_unref (srccaps);
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gst_object_unref (gsmdec);
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return ret;
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@ -218,127 +188,66 @@ wrong_caps:
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GST_ERROR_OBJECT (gsmdec, "invalid caps received");
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beach:
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gst_object_unref (gsmdec);
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return ret;
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}
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static gboolean
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gst_gsmdec_sink_event (GstPad * pad, GstEvent * event)
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static GstFlowReturn
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gst_gsmdec_parse (GstAudioDecoder * dec, GstAdapter * adapter,
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gint * offset, gint * length)
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{
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gboolean res;
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GstGSMDec *gsmdec;
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GstGSMDec *gsmdec = GST_GSMDEC (dec);
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guint size;
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gsmdec = GST_GSMDEC (gst_pad_get_parent (pad));
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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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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_FLUSH_START:
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res = gst_pad_push_event (gsmdec->srcpad, event);
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break;
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case GST_EVENT_FLUSH_STOP:
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gst_segment_init (&gsmdec->segment, GST_FORMAT_UNDEFINED);
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res = gst_pad_push_event (gsmdec->srcpad, event);
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break;
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case GST_EVENT_NEWSEGMENT:
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{
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gboolean update;
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GstFormat format;
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gdouble rate, arate;
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gint64 start, stop, time;
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gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
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&start, &stop, &time);
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/* now configure the values */
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gst_segment_set_newsegment_full (&gsmdec->segment, update,
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rate, arate, format, start, stop, time);
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/* and forward */
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res = gst_pad_push_event (gsmdec->srcpad, event);
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break;
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}
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case GST_EVENT_EOS:
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default:
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res = gst_pad_push_event (gsmdec->srcpad, event);
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break;
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/* WAV49 requires alternating 33 and 32 bytes of input */
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if (gsmdec->use_wav49) {
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gsmdec->needed = (gsmdec->needed == 33 ? 32 : 33);
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}
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gst_object_unref (gsmdec);
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if (size < gsmdec->needed)
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return GST_FLOW_UNEXPECTED;
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return res;
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*offset = 0;
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*length = gsmdec->needed;
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return GST_FLOW_OK;
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}
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static GstFlowReturn
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gst_gsmdec_chain (GstPad * pad, GstBuffer * buf)
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gst_gsmdec_handle_frame (GstAudioDecoder * dec, GstBuffer * buffer)
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{
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GstGSMDec *gsmdec;
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gsm_byte *data;
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GstFlowReturn ret = GST_FLOW_OK;
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GstClockTime timestamp;
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gint needed;
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GstBuffer *outbuf;
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gsmdec = GST_GSMDEC (gst_pad_get_parent (pad));
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/* no fancy draining */
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if (G_UNLIKELY (!buffer))
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return GST_FLOW_OK;
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timestamp = GST_BUFFER_TIMESTAMP (buf);
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gsmdec = GST_GSMDEC (dec);
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if (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT)) {
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gst_adapter_clear (gsmdec->adapter);
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gsmdec->next_ts = GST_CLOCK_TIME_NONE;
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/* FIXME, do some good offset */
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gsmdec->next_of = 0;
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}
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gst_adapter_push (gsmdec->adapter, buf);
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/* always the same amount of output samples */
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outbuf = gst_buffer_new_and_alloc (ENCODED_SAMPLES * sizeof (gsm_signal));
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needed = 33;
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/* do we have enough bytes to read a frame */
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while (gst_adapter_available (gsmdec->adapter) >= needed) {
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GstBuffer *outbuf;
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/* always the same amount of output samples */
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outbuf = gst_buffer_new_and_alloc (ENCODED_SAMPLES * sizeof (gsm_signal));
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/* If we are not given any timestamp, interpolate from last seen
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* timestamp (if any). */
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if (timestamp == GST_CLOCK_TIME_NONE)
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timestamp = gsmdec->next_ts;
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GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
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/* interpolate in the next run */
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if (timestamp != GST_CLOCK_TIME_NONE)
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gsmdec->next_ts = timestamp + gsmdec->duration;
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timestamp = GST_CLOCK_TIME_NONE;
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GST_BUFFER_DURATION (outbuf) = gsmdec->duration;
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GST_BUFFER_OFFSET (outbuf) = gsmdec->next_of;
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if (gsmdec->next_of != -1)
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gsmdec->next_of += ENCODED_SAMPLES;
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GST_BUFFER_OFFSET_END (outbuf) = gsmdec->next_of;
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gst_buffer_set_caps (outbuf, GST_PAD_CAPS (gsmdec->srcpad));
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/* now encode frame into the output buffer */
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data = (gsm_byte *) gst_adapter_peek (gsmdec->adapter, needed);
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if (gsm_decode (gsmdec->state, data,
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(gsm_signal *) GST_BUFFER_DATA (outbuf)) < 0) {
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/* invalid frame */
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GST_WARNING_OBJECT (gsmdec, "tried to decode an invalid frame, skipping");
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}
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gst_adapter_flush (gsmdec->adapter, needed);
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/* WAV49 requires alternating 33 and 32 bytes of input */
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if (gsmdec->use_wav49)
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needed = (needed == 33 ? 32 : 33);
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GST_DEBUG_OBJECT (gsmdec, "Pushing buffer of size %d ts %" GST_TIME_FORMAT,
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GST_BUFFER_SIZE (outbuf),
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GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)));
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/* push */
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ret = gst_pad_push (gsmdec->srcpad, outbuf);
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/* now encode frame into the output buffer */
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data = (gsm_byte *) GST_BUFFER_DATA (buffer);
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if (gsm_decode (gsmdec->state, data,
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(gsm_signal *) GST_BUFFER_DATA (outbuf)) < 0) {
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/* invalid frame */
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GST_AUDIO_DECODER_ERROR (gsmdec, 1, STREAM, DECODE, (NULL),
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("tried to decode an invalid frame"), ret);
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if (ret != GST_FLOW_OK)
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goto exit;
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gst_buffer_unref (outbuf);
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outbuf = NULL;
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}
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gst_object_unref (gsmdec);
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gst_audio_decoder_finish_frame (dec, outbuf, 1);
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exit:
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return ret;
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}
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@ -21,7 +21,7 @@
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#define __GST_GSMDEC_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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#ifdef GSM_HEADER_IN_SUBDIR
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#include <gsm/gsm.h>
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@ -47,28 +47,16 @@ typedef struct _GstGSMDecClass GstGSMDecClass;
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struct _GstGSMDec
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{
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GstElement element;
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/* pads */
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GstPad *sinkpad, *srcpad;
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GstAudioDecoder element;
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gsm state;
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gint use_wav49;
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gint64 next_of;
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GstClockTime next_ts;
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GstAdapter *adapter;
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GstSegment segment;
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gint rate;
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GstClockTime duration;
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gint needed;
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};
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struct _GstGSMDecClass
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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_gsmdec_get_type (void);
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