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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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voamrwbenc: port to audioencoder
This commit is contained in:
Mark Nauwelaerts
parent
53723f81eb
commit
97279f1dfd
3 changed files with 88 additions and 162 deletions
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@ -4,8 +4,10 @@ libgstvoamrwbenc_la_SOURCES = \
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gstvoamrwb.c \
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gstvoamrwbenc.c
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libgstvoamrwbenc_la_CFLAGS = $(GST_CFLAGS) $(VOAMRWBENC_CFLAGS)
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libgstvoamrwbenc_la_LIBADD = $(GST_BASE_LIBS) $(VOAMRWBENC_LIBS)
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libgstvoamrwbenc_la_CFLAGS = -DGST_USE_UNSTABLE_API \
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$(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS) $(VOAMRWBENC_CFLAGS)
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libgstvoamrwbenc_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) \
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-lgstaudio-$(GST_MAJORMINOR) $(GST_BASE_LIBS) $(VOAMRWBENC_LIBS)
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libgstvoamrwbenc_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
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libgstvoamrwbenc_la_LIBTOOLFLAGS = --tag=disable-static
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@ -109,31 +109,15 @@ static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
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GST_DEBUG_CATEGORY_STATIC (gst_voamrwbenc_debug);
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#define GST_CAT_DEFAULT gst_voamrwbenc_debug
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static void gst_voamrwbenc_finalize (GObject * object);
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static gboolean gst_voamrwbenc_start (GstAudioEncoder * enc);
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static gboolean gst_voamrwbenc_stop (GstAudioEncoder * enc);
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static gboolean gst_voamrwbenc_set_format (GstAudioEncoder * enc,
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GstAudioInfo * info);
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static GstFlowReturn gst_voamrwbenc_handle_frame (GstAudioEncoder * enc,
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GstBuffer * in_buf);
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static GstFlowReturn gst_voamrwbenc_chain (GstPad * pad, GstBuffer * buffer);
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static gboolean gst_voamrwbenc_setcaps (GstPad * pad, GstCaps * caps);
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static GstStateChangeReturn gst_voamrwbenc_state_change (GstElement * element,
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GstStateChange transition);
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static void
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_do_init (GType object_type)
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{
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const GInterfaceInfo preset_interface_info = {
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NULL, /* interface init */
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NULL, /* interface finalize */
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NULL /* interface_data */
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};
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g_type_add_interface_static (object_type, GST_TYPE_PRESET,
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&preset_interface_info);
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GST_DEBUG_CATEGORY_INIT (gst_voamrwbenc_debug, "amrwbenc", 0,
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"AMR-WB audio encoder");
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}
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GST_BOILERPLATE_FULL (GstVoAmrWbEnc, gst_voamrwbenc, GstElement,
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GST_TYPE_ELEMENT, _do_init);
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GST_BOILERPLATE (GstVoAmrWbEnc, gst_voamrwbenc, GstAudioEncoder,
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GST_TYPE_AUDIO_ENCODER);
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static void
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gst_voamrwbenc_set_property (GObject * object, guint prop_id,
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@ -191,71 +175,74 @@ static void
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gst_voamrwbenc_class_init (GstVoAmrWbEncClass * klass)
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{
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GObjectClass *object_class = G_OBJECT_CLASS (klass);
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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GstAudioEncoderClass *base_class = GST_AUDIO_ENCODER_CLASS (klass);
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object_class->finalize = gst_voamrwbenc_finalize;
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object_class->set_property = gst_voamrwbenc_set_property;
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object_class->get_property = gst_voamrwbenc_get_property;
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base_class->start = GST_DEBUG_FUNCPTR (gst_voamrwbenc_start);
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base_class->stop = GST_DEBUG_FUNCPTR (gst_voamrwbenc_stop);
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base_class->set_format = GST_DEBUG_FUNCPTR (gst_voamrwbenc_set_format);
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base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_voamrwbenc_handle_frame);
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g_object_class_install_property (object_class, PROP_BANDMODE,
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g_param_spec_enum ("band-mode", "Band Mode",
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"Encoding Band Mode (Kbps)", GST_VOAMRWBENC_BANDMODE_TYPE,
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BANDMODE_DEFAULT,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
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element_class->change_state = GST_DEBUG_FUNCPTR (gst_voamrwbenc_state_change);
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}
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static void
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gst_voamrwbenc_init (GstVoAmrWbEnc * amrwbenc, GstVoAmrWbEncClass * klass)
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{
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/* create the sink pad */
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amrwbenc->sinkpad = gst_pad_new_from_static_template (&sink_template, "sink");
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gst_pad_set_setcaps_function (amrwbenc->sinkpad, gst_voamrwbenc_setcaps);
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gst_pad_set_chain_function (amrwbenc->sinkpad, gst_voamrwbenc_chain);
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gst_element_add_pad (GST_ELEMENT (amrwbenc), amrwbenc->sinkpad);
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/* create the src pad */
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amrwbenc->srcpad = gst_pad_new_from_static_template (&src_template, "src");
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gst_pad_use_fixed_caps (amrwbenc->srcpad);
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gst_element_add_pad (GST_ELEMENT (amrwbenc), amrwbenc->srcpad);
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amrwbenc->adapter = gst_adapter_new ();
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/* init rest */
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amrwbenc->handle = NULL;
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amrwbenc->channels = 0;
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amrwbenc->rate = 0;
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amrwbenc->ts = 0;
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}
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static void
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gst_voamrwbenc_finalize (GObject * object)
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{
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GstVoAmrWbEnc *amrwbenc;
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amrwbenc = GST_VOAMRWBENC (object);
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g_object_unref (G_OBJECT (amrwbenc->adapter));
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amrwbenc->adapter = NULL;
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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_voamrwbenc_setcaps (GstPad * pad, GstCaps * caps)
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gst_voamrwbenc_start (GstAudioEncoder * enc)
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{
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GstVoAmrWbEnc *voamrwbenc = GST_VOAMRWBENC (enc);
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GST_DEBUG_OBJECT (enc, "start");
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if (!(voamrwbenc->handle = E_IF_init ()))
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return FALSE;
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voamrwbenc->rate = 0;
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voamrwbenc->channels = 0;
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return TRUE;
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}
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static gboolean
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gst_voamrwbenc_stop (GstAudioEncoder * enc)
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{
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GstVoAmrWbEnc *voamrwbenc = GST_VOAMRWBENC (enc);
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GST_DEBUG_OBJECT (enc, "stop");
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if (voamrwbenc->handle) {
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E_IF_exit (voamrwbenc->handle);
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voamrwbenc->handle = NULL;
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}
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return TRUE;
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}
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static gboolean
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gst_voamrwbenc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
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{
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GstStructure *structure;
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GstVoAmrWbEnc *amrwbenc;
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GstCaps *copy;
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amrwbenc = GST_VOAMRWBENC (GST_PAD_PARENT (pad));
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structure = gst_caps_get_structure (caps, 0);
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amrwbenc = GST_VOAMRWBENC (benc);
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/* get channel count */
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gst_structure_get_int (structure, "channels", &amrwbenc->channels);
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gst_structure_get_int (structure, "rate", &amrwbenc->rate);
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amrwbenc->channels = GST_AUDIO_INFO_CHANNELS (info);
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amrwbenc->rate = GST_AUDIO_INFO_RATE (info);
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/* this is not wrong but will sound bad */
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if (amrwbenc->channels != 1) {
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@ -270,116 +257,59 @@ gst_voamrwbenc_setcaps (GstPad * pad, GstCaps * caps)
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"channels", G_TYPE_INT, amrwbenc->channels,
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"rate", G_TYPE_INT, amrwbenc->rate, NULL);
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gst_pad_set_caps (amrwbenc->srcpad, copy);
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gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (amrwbenc), copy);
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gst_caps_unref (copy);
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/* report needs to base class: one frame at a time */
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gst_audio_encoder_set_frame_samples_min (benc, L_FRAME16k);
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gst_audio_encoder_set_frame_samples_max (benc, L_FRAME16k);
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gst_audio_encoder_set_frame_max (benc, 1);
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return TRUE;
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}
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static GstFlowReturn
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gst_voamrwbenc_chain (GstPad * pad, GstBuffer * buffer)
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gst_voamrwbenc_handle_frame (GstAudioEncoder * benc, GstBuffer * buffer)
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{
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GstVoAmrWbEnc *amrwbenc;
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GstFlowReturn ret = GST_FLOW_OK;
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const int buffer_size = sizeof (short) * L_FRAME16k;
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GstBuffer *out;
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gint outsize;
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amrwbenc = GST_VOAMRWBENC (gst_pad_get_parent (pad));
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amrwbenc = GST_VOAMRWBENC (benc);
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g_return_val_if_fail (amrwbenc->handle, GST_FLOW_WRONG_STATE);
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g_return_val_if_fail (amrwbenc->handle, GST_FLOW_NOT_NEGOTIATED);
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if (amrwbenc->rate == 0 || amrwbenc->channels == 0) {
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ret = GST_FLOW_NOT_NEGOTIATED;
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goto done;
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}
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/* discontinuity clears adapter, FIXME, maybe we can set some
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* encoder flag to mask the discont. */
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if (GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DISCONT)) {
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gst_adapter_clear (amrwbenc->adapter);
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amrwbenc->ts = 0;
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amrwbenc->discont = TRUE;
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/* we don't deal with squeezing remnants, so simply discard those */
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if (G_UNLIKELY (buffer == NULL)) {
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GST_DEBUG_OBJECT (amrwbenc, "no data");
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goto done;
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}
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if (GST_BUFFER_TIMESTAMP_IS_VALID (buffer))
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amrwbenc->ts = GST_BUFFER_TIMESTAMP (buffer);
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ret = GST_FLOW_OK;
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gst_adapter_push (amrwbenc->adapter, buffer);
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/* Collect samples until we have enough for an output frame */
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while (gst_adapter_available (amrwbenc->adapter) >= buffer_size) {
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GstBuffer *out;
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guint8 *data;
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gint outsize;
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if (G_UNLIKELY (GST_BUFFER_SIZE (buffer) < buffer_size)) {
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GST_DEBUG_OBJECT (amrwbenc, "discarding trailing data %d",
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buffer ? GST_BUFFER_SIZE (buffer) : 0);
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ret = gst_audio_encoder_finish_frame (benc, NULL, -1);
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goto done;
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}
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out = gst_buffer_new_and_alloc (buffer_size);
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GST_BUFFER_DURATION (out) = GST_SECOND * L_FRAME16k /
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(amrwbenc->rate * amrwbenc->channels);
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GST_BUFFER_TIMESTAMP (out) = amrwbenc->ts;
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if (amrwbenc->ts != -1) {
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amrwbenc->ts += GST_BUFFER_DURATION (out);
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}
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if (amrwbenc->discont) {
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GST_BUFFER_FLAG_SET (out, GST_BUFFER_FLAG_DISCONT);
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amrwbenc->discont = FALSE;
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}
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gst_buffer_set_caps (out, gst_pad_get_caps (amrwbenc->srcpad));
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data = (guint8 *) gst_adapter_peek (amrwbenc->adapter, buffer_size);
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/* encode */
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outsize =
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E_IF_encode (amrwbenc->handle, amrwbenc->bandmode, (const short *) data,
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outsize = E_IF_encode (amrwbenc->handle, amrwbenc->bandmode,
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(const short *) GST_BUFFER_DATA (buffer),
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(unsigned char *) GST_BUFFER_DATA (out), 0);
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gst_adapter_flush (amrwbenc->adapter, buffer_size);
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GST_LOG_OBJECT (amrwbenc, "encoded to %d bytes", outsize);
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GST_BUFFER_SIZE (out) = outsize;
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/* play */
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if ((ret = gst_pad_push (amrwbenc->srcpad, out)) != GST_FLOW_OK)
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break;
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}
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ret = gst_audio_encoder_finish_frame (benc, out, L_FRAME16k);
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done:
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gst_object_unref (amrwbenc);
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return ret;
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}
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static GstStateChangeReturn
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gst_voamrwbenc_state_change (GstElement * element, GstStateChange transition)
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{
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GstVoAmrWbEnc *amrwbenc;
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GstStateChangeReturn ret;
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amrwbenc = GST_VOAMRWBENC (element);
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switch (transition) {
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case GST_STATE_CHANGE_NULL_TO_READY:
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if (!(amrwbenc->handle = E_IF_init ()))
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return GST_STATE_CHANGE_FAILURE;
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break;
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case GST_STATE_CHANGE_READY_TO_PAUSED:
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amrwbenc->rate = 0;
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amrwbenc->channels = 0;
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amrwbenc->ts = 0;
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amrwbenc->discont = FALSE;
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gst_adapter_clear (amrwbenc->adapter);
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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_READY_TO_NULL:
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E_IF_exit (amrwbenc->handle);
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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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@ -21,7 +21,8 @@
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#define __GST_VOAMRWBENC_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 <vo-amrwbenc/enc_if.h>
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G_BEGIN_DECLS
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@ -41,14 +42,7 @@ typedef struct _GstVoAmrWbEnc GstVoAmrWbEnc;
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typedef struct _GstVoAmrWbEncClass GstVoAmrWbEncClass;
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struct _GstVoAmrWbEnc {
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GstElement element;
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/* pads */
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GstPad *sinkpad, *srcpad;
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guint64 ts;
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gboolean discont;
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GstAdapter *adapter;
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GstAudioEncoder element;
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/* library handle */
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void *handle;
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@ -59,7 +53,7 @@ struct _GstVoAmrWbEnc {
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};
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struct _GstVoAmrWbEncClass {
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GstElementClass parent_class;
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GstAudioEncoderClass parent_class;
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};
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GType gst_voamrwbenc_get_type (void);
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