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a52dec: Reconcile code with dtsdec
Perform some cleanups based on the dtsdec code such as using the boilerplate macro and static pad template functions. Add some documentation. Don't register a change in flags until we synch on another frame successfully.
This commit is contained in:
Jan Schmidt
parent
c286272792
commit
dc7f71fb53
1 changed files with 42 additions and 56 deletions
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@ -25,7 +25,7 @@
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* <refsect2>
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* <title>Example launch line</title>
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* |[
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* gst-launch dvdreadsrc title=1 ! dvddemux ! a52dec ! audioresample ! audioconvert ! alsasink
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* gst-launch dvdreadsrc title=1 ! mpegpsdemux ! a52dec ! audioresample ! audioconvert ! alsasink
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* ]| Play audio track from a dvd.
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* |[
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* gst-launch filesrc location=abc.ac3 ! a52dec ! audioresample ! audioconvert ! alsasink
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@ -58,7 +58,7 @@ static GstElementDetails gst_a52dec_details = {
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"ATSC A/52 audio decoder",
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"Codec/Decoder/Audio",
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"Decodes ATSC A/52 encoded audio streams",
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"David I. Lehn <dlehn@users.sourceforge.net>",
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"David I. Lehn <dlehn@users.sourceforge.net>"
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};
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#ifdef LIBA52_DOUBLE
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@ -94,9 +94,7 @@ static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
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"rate = (int) [ 4000, 96000 ], " "channels = (int) [ 1, 6 ]")
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);
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static void gst_a52dec_base_init (GstA52DecClass * klass);
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static void gst_a52dec_class_init (GstA52DecClass * klass);
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static void gst_a52dec_init (GstA52Dec * a52dec);
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GST_BOILERPLATE (GstA52Dec, gst_a52dec, GstElement, GST_TYPE_ELEMENT);
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static GstFlowReturn gst_a52dec_chain (GstPad * pad, GstBuffer * buffer);
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static GstFlowReturn gst_a52dec_chain_raw (GstPad * pad, GstBuffer * buf);
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@ -110,8 +108,6 @@ static void gst_a52dec_set_property (GObject * object, guint prop_id,
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static void gst_a52dec_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static GstElementClass *parent_class = NULL;
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#define GST_TYPE_A52DEC_MODE (gst_a52dec_mode_get_type())
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static GType
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gst_a52dec_mode_get_type (void)
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@ -135,35 +131,10 @@ gst_a52dec_mode_get_type (void)
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return a52dec_mode_type;
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}
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GType
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gst_a52dec_get_type (void)
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{
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static GType a52dec_type = 0;
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if (!a52dec_type) {
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static const GTypeInfo a52dec_info = {
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sizeof (GstA52DecClass),
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(GBaseInitFunc) gst_a52dec_base_init,
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NULL,
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(GClassInitFunc) gst_a52dec_class_init,
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NULL,
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NULL,
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sizeof (GstA52Dec),
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0,
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(GInstanceInitFunc) gst_a52dec_init,
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};
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a52dec_type =
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g_type_register_static (GST_TYPE_ELEMENT, "GstA52Dec", &a52dec_info, 0);
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}
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return a52dec_type;
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}
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static void
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gst_a52dec_base_init (GstA52DecClass * klass)
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gst_a52dec_base_init (gpointer g_class)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&sink_factory));
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@ -185,19 +156,37 @@ gst_a52dec_class_init (GstA52DecClass * klass)
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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parent_class = g_type_class_peek_parent (klass);
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gobject_class->set_property = gst_a52dec_set_property;
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gobject_class->get_property = gst_a52dec_get_property;
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gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_a52dec_change_state);
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/**
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* GstA52Dec::drc
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*
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* Set to true to apply the recommended Dolby Digital dynamic range compression
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* to the audio stream. Dynamic range compression makes loud sounds
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* softer and soft sounds louder, so you can more easily listen
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* to the stream without disturbing other people.
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*/
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g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_DRC,
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g_param_spec_boolean ("drc", "Dynamic Range Compression",
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"Use Dynamic Range Compression", FALSE, G_PARAM_READWRITE));
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/**
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* GstA52Dec::mode
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*
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* Force a particular output channel configuration from the decoder. By default,
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* the channel downmix (if any) is chosen automatically based on the downstream
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* capabilities of the pipeline.
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*/
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g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_MODE,
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g_param_spec_enum ("mode", "Decoder Mode", "Decoding Mode (default 3f2r)",
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GST_TYPE_A52DEC_MODE, A52_3F2R, G_PARAM_READWRITE));
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/**
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* GstA52Dec::lfe
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*
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* Whether to output the LFE (Low Frequency Emitter) channel of the audio stream.
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*/
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g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_LFE,
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g_param_spec_boolean ("lfe", "LFE", "LFE", TRUE, G_PARAM_READWRITE));
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@ -216,14 +205,10 @@ gst_a52dec_class_init (GstA52DecClass * klass)
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}
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static void
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gst_a52dec_init (GstA52Dec * a52dec)
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gst_a52dec_init (GstA52Dec * a52dec, GstA52DecClass * g_class)
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{
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GstElementClass *klass = GST_ELEMENT_GET_CLASS (a52dec);
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/* create the sink and src pads */
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a52dec->sinkpad =
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gst_pad_new_from_template (gst_element_class_get_pad_template (klass,
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"sink"), "sink");
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a52dec->sinkpad = gst_pad_new_from_static_template (&sink_factory, "sink");
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gst_pad_set_setcaps_function (a52dec->sinkpad,
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GST_DEBUG_FUNCPTR (gst_a52dec_sink_setcaps));
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gst_pad_set_chain_function (a52dec->sinkpad,
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@ -232,21 +217,19 @@ gst_a52dec_init (GstA52Dec * a52dec)
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GST_DEBUG_FUNCPTR (gst_a52dec_sink_event));
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gst_element_add_pad (GST_ELEMENT (a52dec), a52dec->sinkpad);
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a52dec->srcpad =
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gst_pad_new_from_template (gst_element_class_get_pad_template (klass,
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"src"), "src");
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a52dec->srcpad = gst_pad_new_from_static_template (&src_factory, "src");
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gst_element_add_pad (GST_ELEMENT (a52dec), a52dec->srcpad);
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a52dec->request_channels = A52_CHANNEL;
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a52dec->dynamic_range_compression = FALSE;
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a52dec->cache = NULL;
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gst_segment_init (&a52dec->segment, GST_FORMAT_UNDEFINED);
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}
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static int
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static gint
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gst_a52dec_channels (int flags, GstAudioChannelPosition ** _pos)
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{
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int chans = 0;
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gint chans = 0;
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GstAudioChannelPosition *pos = NULL;
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/* allocated just for safety. Number makes no sense */
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@ -541,6 +524,7 @@ gst_a52dec_update_streaminfo (GstA52Dec * a52dec)
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taglist = gst_tag_list_new ();
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gst_tag_list_add (taglist, GST_TAG_MERGE_APPEND,
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GST_TAG_AUDIO_CODEC, "Dolby Digital (AC-3)",
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GST_TAG_BITRATE, (guint) a52dec->bit_rate, NULL);
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gst_element_found_tags_for_pad (GST_ELEMENT (a52dec),
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@ -619,6 +603,7 @@ gst_a52dec_handle_frame (GstA52Dec * a52dec, guint8 * data,
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flags = a52dec->using_channels;
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}
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/* process */
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flags |= A52_ADJUST_LEVEL;
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a52dec->level = 1;
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if (a52_frame (a52dec->state, data, &flags, &a52dec->level, a52dec->bias)) {
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GST_WARNING ("a52_frame error");
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@ -632,7 +617,7 @@ gst_a52dec_handle_frame (GstA52Dec * a52dec, guint8 * data,
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}
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/* negotiate if required */
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if (need_reneg == TRUE) {
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if (need_reneg) {
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GST_DEBUG ("a52dec reneg: sample_rate:%d stream_chans:%d using_chans:%d",
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a52dec->sample_rate, a52dec->stream_channels, a52dec->using_channels);
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if (!gst_a52dec_reneg (a52dec, a52dec->srcpad)) {
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@ -750,7 +735,6 @@ gst_a52dec_chain (GstPad * pad, GstBuffer * buf)
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}
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done:
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return ret;
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/* ERRORS */
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@ -771,12 +755,14 @@ bad_first_access_parameter:
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static GstFlowReturn
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gst_a52dec_chain_raw (GstPad * pad, GstBuffer * buf)
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{
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GstA52Dec *a52dec = GST_A52DEC (gst_pad_get_parent (pad));
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GstA52Dec *a52dec;
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guint8 *data;
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guint size;
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gint length = 0, flags, sample_rate, bit_rate;
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GstFlowReturn result = GST_FLOW_OK;
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a52dec = GST_A52DEC (GST_PAD_PARENT (pad));
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if (!a52dec->sent_segment) {
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GstSegment segment;
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@ -816,16 +802,17 @@ gst_a52dec_chain_raw (GstPad * pad, GstBuffer * buf)
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while (size >= 7) {
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length = a52_syncinfo (data, &flags, &sample_rate, &bit_rate);
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if (flags != a52dec->prev_flags)
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a52dec->flag_update = TRUE;
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a52dec->prev_flags = flags;
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if (length == 0) {
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/* no sync */
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data++;
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size--;
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} else if (length <= size) {
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GST_DEBUG ("Sync: %d", length);
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if (flags != a52dec->prev_flags)
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a52dec->flag_update = TRUE;
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a52dec->prev_flags = flags;
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result = gst_a52dec_handle_frame (a52dec, data,
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length, flags, sample_rate, bit_rate);
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if (result != GST_FLOW_OK) {
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@ -852,7 +839,6 @@ gst_a52dec_chain_raw (GstPad * pad, GstBuffer * buf)
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}
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gst_buffer_unref (buf);
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gst_object_unref (a52dec);
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return result;
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}
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