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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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sirenenc: port to audioencoder
This commit is contained in:
Mark Nauwelaerts
parent
bc7442faa3
commit
6f8e60e24f
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@ -10,7 +10,7 @@ libgstsiren_la_SOURCES = gstsiren.c \
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libgstsiren_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) \
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$(GST_CFLAGS)
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libgstsiren_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) -lgstrtp-@GST_MAJORMINOR@ \
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libgstsiren_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) -lgstaudio-@GST_MAJORMINOR@ \
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$(GST_BASE_LIBS) $(GST_LIBS) $(LIBM)
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libgstsiren_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
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libgstsiren_la_LIBTOOLFLAGS = --tag=disable-static
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@ -69,17 +69,12 @@ enum
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ARG_0,
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};
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static void gst_siren_enc_finalize (GObject * object);
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static gboolean gst_siren_enc_sink_setcaps (GstPad * pad, GstCaps * caps);
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static gboolean gst_siren_enc_sink_event (GstPad * pad, GstEvent * event);
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static GstFlowReturn gst_siren_enc_chain (GstPad * pad, GstBuffer * buf);
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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_enc_start (GstAudioEncoder * enc);
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static gboolean gst_siren_enc_stop (GstAudioEncoder * enc);
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static gboolean gst_siren_enc_set_format (GstAudioEncoder * enc,
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GstAudioInfo * info);
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static GstFlowReturn gst_siren_enc_handle_frame (GstAudioEncoder * enc,
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GstBuffer * in_buf);
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static void
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_do_init (GType type)
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@ -87,8 +82,8 @@ _do_init (GType type)
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GST_DEBUG_CATEGORY_INIT (sirenenc_debug, "sirenenc", 0, "sirenenc");
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}
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GST_BOILERPLATE_FULL (GstSirenEnc, gst_siren_enc, GstElement,
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GST_TYPE_ELEMENT, _do_init);
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GST_BOILERPLATE_FULL (GstSirenEnc, gst_siren_enc, GstAudioEncoder,
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GST_TYPE_AUDIO_ENCODER, _do_init);
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static void
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gst_siren_enc_base_init (gpointer klass)
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@ -107,17 +102,14 @@ gst_siren_enc_base_init (gpointer klass)
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static void
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gst_siren_enc_class_init (GstSirenEncClass * 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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GstAudioEncoderClass *base_class = GST_AUDIO_ENCODER_CLASS (klass);
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GST_DEBUG ("Initializing Class");
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gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_siren_enc_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_enc_start);
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base_class->stop = GST_DEBUG_FUNCPTR (gst_siren_enc_stop);
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base_class->set_format = GST_DEBUG_FUNCPTR (gst_siren_enc_set_format);
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base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_siren_enc_handle_frame);
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GST_DEBUG ("Class Init done");
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}
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@ -125,120 +117,81 @@ gst_siren_enc_class_init (GstSirenEncClass * klass)
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static void
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gst_siren_enc_init (GstSirenEnc * enc, GstSirenEncClass * klass)
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{
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GST_DEBUG_OBJECT (enc, "Initializing");
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enc->encoder = Siren7_NewEncoder (16000);
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enc->adapter = gst_adapter_new ();
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enc->sinkpad = gst_pad_new_from_static_template (&sinktemplate, "sink");
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enc->srcpad = gst_pad_new_from_static_template (&srctemplate, "src");
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gst_pad_set_setcaps_function (enc->sinkpad,
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GST_DEBUG_FUNCPTR (gst_siren_enc_sink_setcaps));
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gst_pad_set_event_function (enc->sinkpad,
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GST_DEBUG_FUNCPTR (gst_siren_enc_sink_event));
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gst_pad_set_chain_function (enc->sinkpad,
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GST_DEBUG_FUNCPTR (gst_siren_enc_chain));
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gst_element_add_pad (GST_ELEMENT (enc), enc->sinkpad);
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gst_element_add_pad (GST_ELEMENT (enc), enc->srcpad);
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GST_DEBUG_OBJECT (enc, "Init done");
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}
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static void
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gst_siren_enc_finalize (GObject * object)
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{
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GstSirenEnc *enc = GST_SIREN_ENC (object);
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GST_DEBUG_OBJECT (object, "Disposing");
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Siren7_CloseEncoder (enc->encoder);
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g_object_unref (enc->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_enc_sink_setcaps (GstPad * pad, GstCaps * caps)
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gst_siren_enc_start (GstAudioEncoder * enc)
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{
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GstSirenEnc *senc = GST_SIREN_ENC (enc);
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GST_DEBUG_OBJECT (enc, "start");
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senc->encoder = Siren7_NewEncoder (16000);
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return TRUE;
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}
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static gboolean
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gst_siren_enc_stop (GstAudioEncoder * enc)
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{
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GstSirenEnc *senc = GST_SIREN_ENC (enc);
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GST_DEBUG_OBJECT (senc, "stop");
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Siren7_CloseEncoder (senc->encoder);
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return TRUE;
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}
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static gboolean
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gst_siren_enc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
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{
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GstSirenEnc *enc;
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gboolean res;
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GstCaps *outcaps;
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enc = GST_SIREN_ENC (GST_PAD_PARENT (pad));
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enc = GST_SIREN_ENC (benc);
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outcaps = gst_static_pad_template_get_caps (&srctemplate);
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res = gst_pad_set_caps (enc->srcpad, outcaps);
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res = gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (enc), outcaps);
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gst_caps_unref (outcaps);
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return res;
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}
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/* report needs to base class */
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gst_audio_encoder_set_frame_samples_min (benc, 320);
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gst_audio_encoder_set_frame_samples_max (benc, 320);
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/* no remainder or flushing please */
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gst_audio_encoder_set_hard_min (benc, TRUE);
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gst_audio_encoder_set_drainable (benc, FALSE);
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static gboolean
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gst_siren_enc_sink_event (GstPad * pad, GstEvent * event)
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{
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GstSirenEnc *enc;
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gboolean res;
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enc = GST_SIREN_ENC (GST_PAD_PARENT (pad));
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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_EOS:
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gst_adapter_clear (enc->adapter);
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res = gst_pad_push_event (enc->srcpad, event);
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break;
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case GST_EVENT_FLUSH_STOP:
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gst_adapter_clear (enc->adapter);
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res = gst_pad_push_event (enc->srcpad, event);
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break;
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default:
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res = gst_pad_push_event (enc->srcpad, event);
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break;
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}
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return res;
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}
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static GstFlowReturn
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gst_siren_enc_chain (GstPad * pad, GstBuffer * buf)
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gst_siren_enc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf)
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{
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GstSirenEnc *enc;
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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 encode_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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enc = GST_SIREN_ENC (GST_PAD_PARENT (pad));
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g_return_val_if_fail (buf != NULL, GST_FLOW_ERROR);
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discont = GST_BUFFER_IS_DISCONT (buf);
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if (discont) {
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GST_DEBUG_OBJECT (enc, "received DISCONT, flush adapter");
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gst_adapter_clear (enc->adapter);
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enc->discont = TRUE;
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}
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enc = GST_SIREN_ENC (benc);
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gst_adapter_push (enc->adapter, buf);
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size = GST_BUFFER_SIZE (buf);
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size = gst_adapter_available (enc->adapter);
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GST_LOG_OBJECT (enc, "Received buffer of size %d", GST_BUFFER_SIZE (buf));
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GST_LOG_OBJECT (enc, "Received buffer of size %d with adapter of size : %d",
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GST_BUFFER_SIZE (buf), size);
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g_return_val_if_fail (size > 0, GST_FLOW_ERROR);
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g_return_val_if_fail (size % 640 == 0, GST_FLOW_ERROR);
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/* we need to process 640 input bytes to produce 40 output bytes */
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/* calculate the amount of frames we will handle */
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num_frames = size / 640;
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/* no frames, wait some more */
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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 * 640;
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out_size = num_frames * 40;
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@ -246,32 +199,14 @@ gst_siren_enc_chain (GstPad * pad, GstBuffer * buf)
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GST_LOG_OBJECT (enc, "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 (enc->srcpad)) == NULL) {
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outcaps = gst_static_pad_template_get_caps (&srctemplate);
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gst_pad_set_caps (enc->srcpad, outcaps);
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gst_caps_unref (outcaps);
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}
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/* get a buffer */
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ret = gst_pad_alloc_buffer_and_set_caps (enc->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_ENCODER_SRC_PAD (benc),
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-1, out_size, GST_PAD_CAPS (GST_AUDIO_ENCODER_SRC_PAD (benc)), &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 (enc->adapter, &distance);
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/* add the amount of time taken by the distance */
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if (timestamp != -1)
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timestamp += gst_util_uint64_scale_int (distance / 2, GST_SECOND, 16000);
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GST_LOG_OBJECT (enc,
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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 (enc->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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@ -289,20 +224,10 @@ gst_siren_enc_chain (GstPad * pad, GstBuffer * buf)
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GST_LOG_OBJECT (enc, "Finished encoding");
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/* mark discont */
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if (enc->discont) {
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GST_BUFFER_FLAG_SET (out_buf, GST_BUFFER_FLAG_DISCONT);
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enc->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 (enc->srcpad, out_buf);
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/* we encode all we get, pass it along */
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ret = gst_audio_encoder_finish_frame (benc, 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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@ -322,33 +247,6 @@ encode_error:
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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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GstSirenEnc *enc = GST_SIREN_ENC (element);
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switch (transition) {
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case GST_STATE_CHANGE_READY_TO_PAUSED:
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enc->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 (enc->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_enc_plugin_init (GstPlugin * plugin)
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{
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@ -24,7 +24,7 @@
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#define __GST_SIREN_ENC_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/gstaudioencoder.h>
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#include "siren7.h"
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@ -48,21 +48,15 @@ typedef struct _GstSirenEncPrivate GstSirenEncPrivate;
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struct _GstSirenEnc
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{
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GstElement parent;
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GstAudioEncoder parent;
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/* protected by the stream lock */
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SirenEncoder encoder;
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GstAdapter *adapter;
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gboolean discont;
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GstPad *srcpad;
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GstPad *sinkpad;
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};
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struct _GstSirenEncClass
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{
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GstElementClass parent_class;
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GstAudioEncoderClass parent_class;
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};
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GType gst_siren_enc_get_type (void);
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