mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-27 04:01:08 +00:00
twolame: port to audioencoder
This commit is contained in:
parent
442d993877
commit
8f58c2cbab
3 changed files with 97 additions and 257 deletions
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@ -1,8 +1,11 @@
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plugin_LTLIBRARIES = libgsttwolame.la
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libgsttwolame_la_SOURCES = gsttwolame.c
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libgsttwolame_la_CFLAGS = $(GST_CFLAGS) $(TWOLAME_CFLAGS)
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libgsttwolame_la_LIBADD = $(TWOLAME_LIBS) $(GST_LIBS)
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libgsttwolame_la_CFLAGS = -DGST_USE_UNSTABLE_API \
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$(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS) $(TWOLAME_CFLAGS)
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libgsttwolame_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) \
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-lgstaudio-@GST_MAJORMINOR@ -lgstpbutils-@GST_MAJORMINOR@ \
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$(GST_LIBS) $(TWOLAME_LIBS)
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libgsttwolame_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
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libgsttwolame_la_LIBTOOLFLAGS = --tag=disable-static
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@ -191,31 +191,22 @@ enum
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ARG_QUICK_MODE_COUNT
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};
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static gboolean gst_two_lame_start (GstAudioEncoder * enc);
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static gboolean gst_two_lame_stop (GstAudioEncoder * enc);
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static gboolean gst_two_lame_set_format (GstAudioEncoder * enc,
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GstAudioInfo * info);
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static GstFlowReturn gst_two_lame_handle_frame (GstAudioEncoder * enc,
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GstBuffer * in_buf);
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static void gst_two_lame_flush (GstAudioEncoder * enc);
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static void gst_two_lame_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_two_lame_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static gboolean gst_two_lame_sink_event (GstPad * pad, GstEvent * event);
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static GstFlowReturn gst_two_lame_chain (GstPad * pad, GstBuffer * buf);
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static gboolean gst_two_lame_setup (GstTwoLame * twolame);
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static GstStateChangeReturn gst_two_lame_change_state (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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}
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GST_BOILERPLATE_FULL (GstTwoLame, gst_two_lame, GstElement, GST_TYPE_ELEMENT,
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_do_init);
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GST_BOILERPLATE (GstTwoLame, gst_two_lame, GstAudioEncoder,
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GST_TYPE_AUDIO_ENCODER);
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static void
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gst_two_lame_release_memory (GstTwoLame * twolame)
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@ -254,9 +245,11 @@ gst_two_lame_class_init (GstTwoLameClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstAudioEncoderClass *gstbase_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gstbase_class = (GstAudioEncoderClass *) klass;
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parent_class = g_type_class_peek_parent (klass);
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@ -264,6 +257,12 @@ gst_two_lame_class_init (GstTwoLameClass * klass)
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gobject_class->get_property = gst_two_lame_get_property;
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gobject_class->finalize = gst_two_lame_finalize;
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gstbase_class->start = GST_DEBUG_FUNCPTR (gst_two_lame_start);
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gstbase_class->stop = GST_DEBUG_FUNCPTR (gst_two_lame_stop);
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gstbase_class->set_format = GST_DEBUG_FUNCPTR (gst_two_lame_set_format);
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gstbase_class->handle_frame = GST_DEBUG_FUNCPTR (gst_two_lame_handle_frame);
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gstbase_class->flush = GST_DEBUG_FUNCPTR (gst_two_lame_flush);
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g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_MODE,
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g_param_spec_enum ("mode", "Mode", "Encoding mode",
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GST_TYPE_TWO_LAME_MODE, gst_two_lame_default_settings.mode,
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@ -349,39 +348,25 @@ gst_two_lame_class_init (GstTwoLameClass * klass)
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"Calculate Psymodel every n frames",
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0, G_MAXINT, gst_two_lame_default_settings.quick_mode_count,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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gstelement_class->change_state =
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GST_DEBUG_FUNCPTR (gst_two_lame_change_state);
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}
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static gboolean
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gst_two_lame_src_setcaps (GstPad * pad, GstCaps * caps)
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{
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GST_DEBUG_OBJECT (pad, "caps: %" GST_PTR_FORMAT, caps);
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return TRUE;
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}
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static gboolean
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gst_two_lame_sink_setcaps (GstPad * pad, GstCaps * caps)
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gst_two_lame_set_format (GstAudioEncoder * enc, GstAudioInfo * info)
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{
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GstTwoLame *twolame;
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gint out_samplerate;
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gint version;
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GstStructure *structure;
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GstCaps *othercaps;
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twolame = GST_TWO_LAME (GST_PAD_PARENT (pad));
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structure = gst_caps_get_structure (caps, 0);
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twolame = GST_TWO_LAME (enc);
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if (strcmp (gst_structure_get_name (structure), "audio/x-raw-int") == 0)
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twolame->float_input = FALSE;
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else
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twolame->float_input = TRUE;
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/* parameters already parsed for us */
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twolame->samplerate = GST_AUDIO_INFO_RATE (info);
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twolame->num_channels = GST_AUDIO_INFO_CHANNELS (info);
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twolame->float_input = !GST_AUDIO_INFO_IS_INTEGER (info);
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if (!gst_structure_get_int (structure, "rate", &twolame->samplerate))
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goto no_rate;
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if (!gst_structure_get_int (structure, "channels", &twolame->num_channels))
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goto no_channels;
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/* but we might be asked to reconfigure, so reset */
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gst_two_lame_release_memory (twolame);
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GST_DEBUG_OBJECT (twolame, "setting up twolame");
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if (!gst_two_lame_setup (twolame))
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@ -413,21 +398,14 @@ gst_two_lame_sink_setcaps (GstPad * pad, GstCaps * caps)
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G_TYPE_INT, out_samplerate, NULL);
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/* and use these caps */
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gst_pad_set_caps (twolame->srcpad, othercaps);
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gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (twolame), othercaps);
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gst_caps_unref (othercaps);
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/* not much base class feedback:
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* - we will handle buffers, just hand us all available */
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return TRUE;
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no_rate:
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{
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GST_ERROR_OBJECT (twolame, "input caps have no sample rate field");
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return FALSE;
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}
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no_channels:
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{
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GST_ERROR_OBJECT (twolame, "input caps have no channels field");
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return FALSE;
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}
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zero_output_rate:
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{
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GST_ELEMENT_ERROR (twolame, LIBRARY, SETTINGS, (NULL),
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@ -447,26 +425,6 @@ gst_two_lame_init (GstTwoLame * twolame, GstTwoLameClass * klass)
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{
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GST_DEBUG_OBJECT (twolame, "starting initialization");
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twolame->sinkpad =
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gst_pad_new_from_static_template (&gst_two_lame_sink_template, "sink");
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gst_pad_set_event_function (twolame->sinkpad,
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GST_DEBUG_FUNCPTR (gst_two_lame_sink_event));
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gst_pad_set_chain_function (twolame->sinkpad,
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GST_DEBUG_FUNCPTR (gst_two_lame_chain));
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gst_pad_set_setcaps_function (twolame->sinkpad,
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GST_DEBUG_FUNCPTR (gst_two_lame_sink_setcaps));
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gst_element_add_pad (GST_ELEMENT (twolame), twolame->sinkpad);
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twolame->srcpad =
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gst_pad_new_from_static_template (&gst_two_lame_src_template, "src");
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gst_pad_set_setcaps_function (twolame->srcpad,
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GST_DEBUG_FUNCPTR (gst_two_lame_src_setcaps));
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gst_element_add_pad (GST_ELEMENT (twolame), twolame->srcpad);
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twolame->samplerate = 44100;
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twolame->num_channels = 2;
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twolame->setup = FALSE;
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twolame->mode = gst_two_lame_default_settings.mode;
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twolame->psymodel = gst_two_lame_default_settings.psymodel;
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twolame->bitrate = gst_two_lame_default_settings.bitrate;
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@ -487,6 +445,26 @@ gst_two_lame_init (GstTwoLame * twolame, GstTwoLameClass * klass)
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GST_DEBUG_OBJECT (twolame, "done initializing");
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}
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static gboolean
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gst_two_lame_start (GstAudioEncoder * enc)
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{
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GstTwoLame *twolame = GST_TWO_LAME (enc);
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GST_DEBUG_OBJECT (twolame, "start");
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return TRUE;
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}
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static gboolean
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gst_two_lame_stop (GstAudioEncoder * enc)
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{
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GstTwoLame *twolame = GST_TWO_LAME (enc);
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GST_DEBUG_OBJECT (twolame, "stop");
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gst_two_lame_release_memory (twolame);
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return TRUE;
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}
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/* <php-emulation-mode>three underscores for ___rate is really really really
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* private as opposed to one underscore<php-emulation-mode> */
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/* call this MACRO outside of the NULL state so that we have a higher chance
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@ -638,106 +616,54 @@ gst_two_lame_get_property (GObject * object, guint prop_id, GValue * value,
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}
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}
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static gboolean
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gst_two_lame_sink_event (GstPad * pad, GstEvent * event)
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static GstFlowReturn
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gst_two_lame_flush_full (GstTwoLame * lame, gboolean push)
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{
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gboolean ret;
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GstTwoLame *twolame;
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twolame = GST_TWO_LAME (gst_pad_get_parent (pad));
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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_EOS:{
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GST_DEBUG_OBJECT (twolame, "handling EOS event");
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if (twolame->glopts != NULL) {
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GstBuffer *buf;
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gint size;
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GstFlowReturn result = GST_FLOW_OK;
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if (!lame->glopts)
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return GST_FLOW_OK;
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buf = gst_buffer_new_and_alloc (16384);
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size =
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twolame_encode_flush (twolame->glopts, GST_BUFFER_DATA (buf),
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16394);
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size = twolame_encode_flush (lame->glopts, GST_BUFFER_DATA (buf), 16384);
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if (size > 0 && twolame->last_flow == GST_FLOW_OK) {
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gint64 duration;
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duration = gst_util_uint64_scale (size, 8 * GST_SECOND,
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1000 * twolame->bitrate);
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if (twolame->last_ts == GST_CLOCK_TIME_NONE) {
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twolame->last_ts = twolame->eos_ts;
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twolame->last_duration = duration;
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} else {
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twolame->last_duration += duration;
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}
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GST_BUFFER_TIMESTAMP (buf) = twolame->last_ts;
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GST_BUFFER_DURATION (buf) = twolame->last_duration;
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twolame->last_ts = GST_CLOCK_TIME_NONE;
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if (size > 0 && push) {
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GST_BUFFER_SIZE (buf) = size;
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GST_DEBUG_OBJECT (twolame, "pushing final packet of %u bytes", size);
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gst_buffer_set_caps (buf, GST_PAD_CAPS (twolame->srcpad));
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gst_pad_push (twolame->srcpad, buf);
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GST_DEBUG_OBJECT (lame, "pushing final packet of %u bytes", size);
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result = gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (lame), buf, -1);
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} else {
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GST_DEBUG_OBJECT (twolame, "no final packet (size=%d, last_flow=%s)",
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size, gst_flow_get_name (twolame->last_flow));
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GST_DEBUG_OBJECT (lame, "no final packet (size=%d, push=%d)", size, push);
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gst_buffer_unref (buf);
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result = GST_FLOW_OK;
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}
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return result;
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}
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ret = gst_pad_event_default (pad, event);
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break;
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}
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case GST_EVENT_FLUSH_START:
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GST_DEBUG_OBJECT (twolame, "handling FLUSH start event");
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/* forward event */
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ret = gst_pad_push_event (twolame->srcpad, event);
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break;
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case GST_EVENT_FLUSH_STOP:
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static void
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gst_two_lame_flush (GstAudioEncoder * enc)
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{
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guchar *mp3_data = NULL;
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gint mp3_buffer_size;
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GST_DEBUG_OBJECT (twolame, "handling FLUSH stop event");
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/* clear buffers */
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mp3_buffer_size = 16384;
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mp3_data = g_malloc (mp3_buffer_size);
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twolame_encode_flush (twolame->glopts, mp3_data, mp3_buffer_size);
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ret = gst_pad_push_event (twolame->srcpad, event);
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g_free (mp3_data);
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break;
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}
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default:
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ret = gst_pad_event_default (pad, event);
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break;
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}
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gst_object_unref (twolame);
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return ret;
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gst_two_lame_flush_full (GST_TWO_LAME (enc), FALSE);
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}
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static GstFlowReturn
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gst_two_lame_chain (GstPad * pad, GstBuffer * buf)
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gst_two_lame_handle_frame (GstAudioEncoder * enc, GstBuffer * buf)
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{
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GstTwoLame *twolame;
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guchar *mp3_data;
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gint mp3_buffer_size, mp3_size;
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gint64 duration;
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GstBuffer *mp3_buf;
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GstFlowReturn result;
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gint num_samples;
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guint8 *data;
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guint size;
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twolame = GST_TWO_LAME (GST_PAD_PARENT (pad));
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twolame = GST_TWO_LAME (enc);
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GST_LOG_OBJECT (twolame, "entered chain");
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if (!twolame->setup)
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goto not_setup;
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/* squeeze remaining and push */
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if (G_UNLIKELY (buf == NULL))
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return gst_two_lame_flush_full (twolame, TRUE);
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data = GST_BUFFER_DATA (buf);
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size = GST_BUFFER_SIZE (buf);
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@ -749,7 +675,8 @@ gst_two_lame_chain (GstPad * pad, GstBuffer * buf)
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/* allocate space for output */
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mp3_buffer_size = 1.25 * num_samples + 16384;
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mp3_data = g_malloc (mp3_buffer_size);
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mp3_buf = gst_buffer_new_and_alloc (mp3_buffer_size);
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mp3_data = GST_BUFFER_DATA (mp3_buf);
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if (twolame->num_channels == 1) {
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if (twolame->float_input)
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@ -774,73 +701,22 @@ gst_two_lame_chain (GstPad * pad, GstBuffer * buf)
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GST_LOG_OBJECT (twolame, "encoded %d bytes of audio to %d bytes of mp3",
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size, mp3_size);
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if (twolame->float_input)
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duration = gst_util_uint64_scale_int (size, GST_SECOND,
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4 * twolame->samplerate * twolame->num_channels);
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else
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duration = gst_util_uint64_scale_int (size, GST_SECOND,
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2 * twolame->samplerate * twolame->num_channels);
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if (GST_BUFFER_DURATION (buf) != GST_CLOCK_TIME_NONE &&
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GST_BUFFER_DURATION (buf) != duration) {
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GST_DEBUG_OBJECT (twolame, "incoming buffer had incorrect duration %"
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GST_TIME_FORMAT ", outgoing buffer will have correct duration %"
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GST_TIME_FORMAT,
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GST_TIME_ARGS (GST_BUFFER_DURATION (buf)), GST_TIME_ARGS (duration));
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}
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if (twolame->last_ts == GST_CLOCK_TIME_NONE) {
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twolame->last_ts = GST_BUFFER_TIMESTAMP (buf);
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twolame->last_offs = GST_BUFFER_OFFSET (buf);
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twolame->last_duration = duration;
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} else {
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twolame->last_duration += duration;
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}
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gst_buffer_unref (buf);
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if (mp3_size < 0) {
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g_warning ("error %d", mp3_size);
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}
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if (mp3_size > 0) {
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GstBuffer *outbuf;
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outbuf = gst_buffer_new ();
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GST_BUFFER_DATA (outbuf) = mp3_data;
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GST_BUFFER_MALLOCDATA (outbuf) = mp3_data;
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GST_BUFFER_SIZE (outbuf) = mp3_size;
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GST_BUFFER_TIMESTAMP (outbuf) = twolame->last_ts;
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GST_BUFFER_OFFSET (outbuf) = twolame->last_offs;
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GST_BUFFER_DURATION (outbuf) = twolame->last_duration;
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gst_buffer_set_caps (outbuf, GST_PAD_CAPS (twolame->srcpad));
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result = gst_pad_push (twolame->srcpad, outbuf);
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twolame->last_flow = result;
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if (result != GST_FLOW_OK) {
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GST_DEBUG_OBJECT (twolame, "flow return: %s", gst_flow_get_name (result));
|
||||
}
|
||||
|
||||
if (GST_CLOCK_TIME_IS_VALID (twolame->last_ts))
|
||||
twolame->eos_ts = twolame->last_ts + twolame->last_duration;
|
||||
else
|
||||
twolame->eos_ts = GST_CLOCK_TIME_NONE;
|
||||
twolame->last_ts = GST_CLOCK_TIME_NONE;
|
||||
GST_BUFFER_SIZE (mp3_buf) = mp3_size;
|
||||
result = gst_audio_encoder_finish_frame (enc, mp3_buf, -1);
|
||||
} else {
|
||||
g_free (mp3_data);
|
||||
if (mp3_size < 0) {
|
||||
/* eat error ? */
|
||||
g_warning ("error %d", mp3_size);
|
||||
}
|
||||
gst_buffer_unref (mp3_buf);
|
||||
result = GST_FLOW_OK;
|
||||
}
|
||||
|
||||
return result;
|
||||
|
||||
/* ERRORS */
|
||||
not_setup:
|
||||
{
|
||||
gst_buffer_unref (buf);
|
||||
GST_ELEMENT_ERROR (twolame, CORE, NEGOTIATION, (NULL),
|
||||
("encoder not initialized (input is not audio?)"));
|
||||
return GST_FLOW_ERROR;
|
||||
}
|
||||
}
|
||||
|
||||
/* set up the encoder state */
|
||||
|
@ -877,7 +753,7 @@ gst_two_lame_setup (GstTwoLame * twolame)
|
|||
twolame_set_in_samplerate (twolame->glopts, twolame->samplerate);
|
||||
|
||||
/* let twolame choose default samplerate unless outgoing sample rate is fixed */
|
||||
allowed_caps = gst_pad_get_allowed_caps (twolame->srcpad);
|
||||
allowed_caps = gst_pad_get_allowed_caps (GST_AUDIO_ENCODER_SRC_PAD (twolame));
|
||||
|
||||
if (allowed_caps != NULL) {
|
||||
GstStructure *structure;
|
||||
|
@ -949,37 +825,6 @@ gst_two_lame_setup (GstTwoLame * twolame)
|
|||
#undef CHECK_ERROR
|
||||
}
|
||||
|
||||
static GstStateChangeReturn
|
||||
gst_two_lame_change_state (GstElement * element, GstStateChange transition)
|
||||
{
|
||||
GstTwoLame *twolame;
|
||||
GstStateChangeReturn result;
|
||||
|
||||
twolame = GST_TWO_LAME (element);
|
||||
|
||||
switch (transition) {
|
||||
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
||||
twolame->last_flow = GST_FLOW_OK;
|
||||
twolame->last_ts = GST_CLOCK_TIME_NONE;
|
||||
twolame->eos_ts = GST_CLOCK_TIME_NONE;
|
||||
break;
|
||||
default:
|
||||
break;
|
||||
}
|
||||
|
||||
result = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
||||
|
||||
switch (transition) {
|
||||
case GST_STATE_CHANGE_READY_TO_NULL:
|
||||
gst_two_lame_release_memory (twolame);
|
||||
break;
|
||||
default:
|
||||
break;
|
||||
}
|
||||
|
||||
return result;
|
||||
}
|
||||
|
||||
static gboolean
|
||||
gst_two_lame_get_default_settings (void)
|
||||
{
|
||||
|
|
|
@ -24,6 +24,7 @@
|
|||
|
||||
|
||||
#include <gst/gst.h>
|
||||
#include <gst/audio/gstaudioencoder.h>
|
||||
|
||||
G_BEGIN_DECLS
|
||||
|
||||
|
@ -49,10 +50,7 @@ typedef struct _GstTwoLameClass GstTwoLameClass;
|
|||
* Opaque data structure.
|
||||
*/
|
||||
struct _GstTwoLame {
|
||||
GstElement element;
|
||||
|
||||
/*< private >*/
|
||||
GstPad *srcpad, *sinkpad;
|
||||
GstAudioEncoder element;
|
||||
|
||||
gint samplerate;
|
||||
gint num_channels;
|
||||
|
@ -75,17 +73,11 @@ struct _GstTwoLame {
|
|||
gboolean quick_mode;
|
||||
gint quick_mode_count;
|
||||
|
||||
/* track this so we don't send a last buffer in eos handler after error */
|
||||
GstFlowReturn last_flow;
|
||||
|
||||
twolame_options *glopts;
|
||||
|
||||
/* time tracker */
|
||||
guint64 last_ts, last_offs, last_duration, eos_ts;
|
||||
};
|
||||
|
||||
struct _GstTwoLameClass {
|
||||
GstElementClass parent_class;
|
||||
GstAudioEncoderClass parent_class;
|
||||
};
|
||||
|
||||
GType gst_two_lame_get_type(void);
|
||||
|
|
Loading…
Reference in a new issue