mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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a52dec: port to audiodecoder
This commit is contained in:
Mark Nauwelaerts
parent
260824b278
commit
d55d4054bd
3 changed files with 246 additions and 426 deletions
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@ -3,12 +3,15 @@ plugin_LTLIBRARIES = libgsta52dec.la
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libgsta52dec_la_SOURCES = gsta52dec.c
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libgsta52dec_la_CFLAGS = \
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$(GST_PLUGINS_BASE_CFLAGS) \
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$(GST_CFLAGS) \
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$(GST_BASE_CFLAGS) \
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$(GST_CFLAGS) \
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$(ORC_CFLAGS) \
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$(A52DEC_CFLAGS)
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libgsta52dec_la_LIBADD = \
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$(GST_PLUGINS_BASE_LIBS) \
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-lgstaudio-$(GST_MAJORMINOR) \
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$(GST_BASE_LIBS) \
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$(GST_LIBS) \
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-lgstaudio-$(GST_MAJORMINOR) \
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$(ORC_LIBS) \
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$(A52DEC_LIBS)
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libgsta52dec_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
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@ -86,14 +86,20 @@ 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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GST_BOILERPLATE (GstA52Dec, gst_a52dec, GstElement, GST_TYPE_ELEMENT);
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GST_BOILERPLATE (GstA52Dec, gst_a52dec, GstAudioDecoder,
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GST_TYPE_AUDIO_DECODER);
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static gboolean gst_a52dec_start (GstAudioDecoder * dec);
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static gboolean gst_a52dec_stop (GstAudioDecoder * dec);
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static gboolean gst_a52dec_set_format (GstAudioDecoder * bdec, GstCaps * caps);
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static gboolean gst_a52dec_parse (GstAudioDecoder * dec, GstAdapter * adapter,
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gint * offset, gint * length);
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static GstFlowReturn gst_a52dec_handle_frame (GstAudioDecoder * dec,
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GstBuffer * buffer);
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static GstFlowReturn gst_a52dec_pre_push (GstAudioDecoder * bdec,
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GstBuffer ** buffer);
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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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static gboolean gst_a52dec_sink_setcaps (GstPad * pad, GstCaps * caps);
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static gboolean gst_a52dec_sink_event (GstPad * pad, GstEvent * event);
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static GstStateChangeReturn gst_a52dec_change_state (GstElement * element,
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GstStateChange transition);
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static void gst_a52dec_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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@ -143,16 +149,21 @@ static void
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gst_a52dec_class_init (GstA52DecClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstAudioDecoderClass *gstbase_class;
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guint cpuflags;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gstbase_class = (GstAudioDecoderClass *) 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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gstbase_class->start = GST_DEBUG_FUNCPTR (gst_a52dec_start);
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gstbase_class->stop = GST_DEBUG_FUNCPTR (gst_a52dec_stop);
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gstbase_class->set_format = GST_DEBUG_FUNCPTR (gst_a52dec_set_format);
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gstbase_class->parse = GST_DEBUG_FUNCPTR (gst_a52dec_parse);
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gstbase_class->handle_frame = GST_DEBUG_FUNCPTR (gst_a52dec_handle_frame);
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gstbase_class->pre_push = GST_DEBUG_FUNCPTR (gst_a52dec_pre_push);
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/**
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* GstA52Dec::drc
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@ -213,27 +224,106 @@ gst_a52dec_class_init (GstA52DecClass * klass)
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static void
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gst_a52dec_init (GstA52Dec * a52dec, GstA52DecClass * g_class)
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{
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/* create the sink and src pads */
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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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GST_DEBUG_FUNCPTR (gst_a52dec_chain));
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gst_pad_set_event_function (a52dec->sinkpad,
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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 = gst_pad_new_from_static_template (&src_factory, "src");
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gst_pad_use_fixed_caps (a52dec->srcpad);
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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->state = NULL;
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a52dec->samples = NULL;
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gst_segment_init (&a52dec->segment, GST_FORMAT_UNDEFINED);
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/* retrieve and intercept base class chain.
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* Quite HACKish, but that's dvd specs/caps for you,
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* since one buffer needs to be split into 2 frames */
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a52dec->base_chain = GST_PAD_CHAINFUNC (GST_AUDIO_DECODER_SINK_PAD (a52dec));
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gst_pad_set_chain_function (GST_AUDIO_DECODER_SINK_PAD (a52dec),
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GST_DEBUG_FUNCPTR (gst_a52dec_chain));
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}
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static gboolean
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gst_a52dec_start (GstAudioDecoder * dec)
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{
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GstA52Dec *a52dec = GST_A52DEC (dec);
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GstA52DecClass *klass;
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GST_DEBUG_OBJECT (dec, "start");
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klass = GST_A52DEC_CLASS (G_OBJECT_GET_CLASS (a52dec));
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a52dec->state = a52_init (klass->a52_cpuflags);
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a52dec->samples = a52_samples (a52dec->state);
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a52dec->bit_rate = -1;
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a52dec->sample_rate = -1;
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a52dec->stream_channels = A52_CHANNEL;
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a52dec->using_channels = A52_CHANNEL;
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a52dec->level = 1;
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a52dec->bias = 0;
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a52dec->flag_update = TRUE;
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/* call upon legacy upstream byte support (e.g. seeking) */
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gst_audio_decoder_set_byte_time (dec, TRUE);
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return TRUE;
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}
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static gboolean
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gst_a52dec_stop (GstAudioDecoder * dec)
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{
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GstA52Dec *a52dec = GST_A52DEC (dec);
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GST_DEBUG_OBJECT (dec, "stop");
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a52dec->samples = NULL;
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if (a52dec->state) {
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a52_free (a52dec->state);
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a52dec->state = NULL;
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}
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if (a52dec->pending_tags) {
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gst_tag_list_free (a52dec->pending_tags);
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a52dec->pending_tags = NULL;
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}
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return TRUE;
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}
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static GstFlowReturn
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gst_a52dec_parse (GstAudioDecoder * bdec, GstAdapter * adapter,
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gint * _offset, gint * len)
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{
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GstA52Dec *a52dec;
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guint8 *data;
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gint av, size;
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gint length = 0, flags, sample_rate, bit_rate;
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GstFlowReturn result = GST_FLOW_UNEXPECTED;
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a52dec = GST_A52DEC (bdec);
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size = av = gst_adapter_available (adapter);
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data = (guint8 *) gst_adapter_peek (adapter, av);
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/* find and read header */
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bit_rate = a52dec->bit_rate;
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sample_rate = a52dec->sample_rate;
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flags = 0;
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while (av >= 7) {
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length = a52_syncinfo (data, &flags, &sample_rate, &bit_rate);
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if (length == 0) {
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/* shift window to re-find sync */
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data++;
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size--;
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} else if (length <= size) {
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GST_LOG_OBJECT (a52dec, "Sync: frame size %d", length);
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result = GST_FLOW_OK;
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break;
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} else {
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GST_LOG_OBJECT (a52dec, "Not enough data available (needed %d had %d)",
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length, size);
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break;
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}
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}
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*_offset = av - size;
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*len = length;
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return result;
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}
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static gint
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@ -324,106 +414,6 @@ gst_a52dec_channels (int flags, GstAudioChannelPosition ** _pos)
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return chans;
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}
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static void
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clear_queued (GstA52Dec * dec)
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{
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g_list_foreach (dec->queued, (GFunc) gst_mini_object_unref, NULL);
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g_list_free (dec->queued);
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dec->queued = NULL;
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}
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static GstFlowReturn
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flush_queued (GstA52Dec * dec)
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{
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GstFlowReturn ret = GST_FLOW_OK;
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while (dec->queued) {
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GstBuffer *buf = GST_BUFFER_CAST (dec->queued->data);
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GST_LOG_OBJECT (dec, "pushing buffer %p, timestamp %"
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GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT, buf,
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GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
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GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
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/* iterate ouput queue an push downstream */
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ret = gst_pad_push (dec->srcpad, buf);
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dec->queued = g_list_delete_link (dec->queued, dec->queued);
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}
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return ret;
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}
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static GstFlowReturn
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gst_a52dec_drain (GstA52Dec * dec)
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{
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GstFlowReturn ret = GST_FLOW_OK;
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if (dec->segment.rate < 0.0) {
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/* if we have some queued frames for reverse playback, flush
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* them now */
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ret = flush_queued (dec);
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}
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return ret;
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}
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static GstFlowReturn
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gst_a52dec_push (GstA52Dec * a52dec,
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GstPad * srcpad, int flags, sample_t * samples, GstClockTime timestamp)
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{
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GstBuffer *buf;
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int chans, n, c;
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GstFlowReturn result;
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flags &= (A52_CHANNEL_MASK | A52_LFE);
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chans = gst_a52dec_channels (flags, NULL);
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if (!chans) {
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GST_ELEMENT_ERROR (GST_ELEMENT (a52dec), STREAM, DECODE, (NULL),
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("invalid channel flags: %d", flags));
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return GST_FLOW_ERROR;
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}
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result =
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gst_pad_alloc_buffer_and_set_caps (srcpad, 0,
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256 * chans * (SAMPLE_WIDTH / 8), GST_PAD_CAPS (srcpad), &buf);
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if (result != GST_FLOW_OK)
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return result;
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for (n = 0; n < 256; n++) {
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for (c = 0; c < chans; c++) {
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((sample_t *) GST_BUFFER_DATA (buf))[n * chans + c] =
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samples[c * 256 + n];
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}
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}
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GST_BUFFER_TIMESTAMP (buf) = timestamp;
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GST_BUFFER_DURATION (buf) = 256 * GST_SECOND / a52dec->sample_rate;
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result = GST_FLOW_OK;
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if ((buf = gst_audio_buffer_clip (buf, &a52dec->segment,
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a52dec->sample_rate, (SAMPLE_WIDTH / 8) * chans))) {
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/* set discont when needed */
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if (a52dec->discont) {
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GST_LOG_OBJECT (a52dec, "marking DISCONT");
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GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
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a52dec->discont = FALSE;
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}
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if (a52dec->segment.rate > 0.0) {
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GST_DEBUG_OBJECT (a52dec,
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"Pushing buffer with ts %" GST_TIME_FORMAT " duration %"
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GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
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GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
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result = gst_pad_push (srcpad, buf);
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} else {
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/* reverse playback, queue frame till later when we get a discont. */
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GST_DEBUG_OBJECT (a52dec, "queued frame");
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a52dec->queued = g_list_prepend (a52dec->queued, buf);
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}
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}
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return result;
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}
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static gboolean
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gst_a52dec_reneg (GstA52Dec * a52dec, GstPad * pad)
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{
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@ -457,102 +447,69 @@ done:
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return result;
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}
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static gboolean
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gst_a52dec_sink_event (GstPad * pad, GstEvent * event)
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{
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GstA52Dec *a52dec = GST_A52DEC (gst_pad_get_parent (pad));
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gboolean ret = FALSE;
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GST_LOG ("Handling %s event", GST_EVENT_TYPE_NAME (event));
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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_NEWSEGMENT:
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{
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GstFormat fmt;
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gboolean update;
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gint64 start, end, pos;
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gdouble rate, arate;
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gst_event_parse_new_segment_full (event, &update, &rate, &arate, &fmt,
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&start, &end, &pos);
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/* drain queued buffers before activating the segment so that we can clip
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* against the old segment first */
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gst_a52dec_drain (a52dec);
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if (fmt != GST_FORMAT_TIME || !GST_CLOCK_TIME_IS_VALID (start)) {
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GST_WARNING ("No time in newsegment event %p (format is %s)",
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event, gst_format_get_name (fmt));
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gst_event_unref (event);
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a52dec->sent_segment = FALSE;
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/* set some dummy values, FIXME: do proper conversion */
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a52dec->time = start = pos = 0;
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fmt = GST_FORMAT_TIME;
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end = -1;
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} else {
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a52dec->time = start;
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a52dec->sent_segment = TRUE;
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GST_DEBUG_OBJECT (a52dec,
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"Pushing newseg rate %g, applied rate %g, format %d, start %"
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G_GINT64_FORMAT ", stop %" G_GINT64_FORMAT ", pos %"
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G_GINT64_FORMAT, rate, arate, fmt, start, end, pos);
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ret = gst_pad_push_event (a52dec->srcpad, event);
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}
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gst_segment_set_newsegment (&a52dec->segment, update, rate, fmt, start,
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end, pos);
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break;
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}
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case GST_EVENT_TAG:
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ret = gst_pad_push_event (a52dec->srcpad, event);
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break;
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case GST_EVENT_EOS:
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gst_a52dec_drain (a52dec);
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ret = gst_pad_push_event (a52dec->srcpad, event);
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break;
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case GST_EVENT_FLUSH_START:
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ret = gst_pad_push_event (a52dec->srcpad, event);
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break;
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case GST_EVENT_FLUSH_STOP:
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if (a52dec->cache) {
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gst_buffer_unref (a52dec->cache);
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a52dec->cache = NULL;
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}
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clear_queued (a52dec);
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gst_segment_init (&a52dec->segment, GST_FORMAT_UNDEFINED);
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ret = gst_pad_push_event (a52dec->srcpad, event);
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break;
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default:
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ret = gst_pad_push_event (a52dec->srcpad, event);
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break;
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}
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gst_object_unref (a52dec);
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return ret;
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}
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static void
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gst_a52dec_update_streaminfo (GstA52Dec * a52dec)
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{
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GstTagList *taglist;
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taglist = gst_tag_list_new ();
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gst_tag_list_add (taglist, GST_TAG_MERGE_APPEND, GST_TAG_BITRATE,
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(guint) a52dec->bit_rate, NULL);
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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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if (a52dec->pending_tags) {
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gst_tag_list_free (a52dec->pending_tags);
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a52dec->pending_tags = NULL;
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}
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gst_element_found_tags_for_pad (GST_ELEMENT (a52dec),
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GST_PAD (a52dec->srcpad), taglist);
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a52dec->pending_tags = taglist;
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}
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static GstFlowReturn
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gst_a52dec_handle_frame (GstA52Dec * a52dec, guint8 * data,
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guint length, gint flags, gint sample_rate, gint bit_rate)
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gst_a52dec_pre_push (GstAudioDecoder * bdec, GstBuffer ** buffer)
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{
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GstA52Dec *a52dec = GST_A52DEC (bdec);
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if (G_UNLIKELY (a52dec->pending_tags)) {
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gst_element_found_tags_for_pad (GST_ELEMENT (a52dec),
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GST_AUDIO_DECODER_SRC_PAD (a52dec), a52dec->pending_tags);
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a52dec->pending_tags = NULL;
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}
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return GST_FLOW_OK;
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}
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static GstFlowReturn
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gst_a52dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buffer)
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||||
{
|
||||
GstA52Dec *a52dec;
|
||||
gint channels, i;
|
||||
gboolean need_reneg = FALSE;
|
||||
gint size, chans;
|
||||
gint length = 0, flags, sample_rate, bit_rate;
|
||||
guint8 *data;
|
||||
GstFlowReturn result = GST_FLOW_OK;
|
||||
GstBuffer *outbuf;
|
||||
const gint num_blocks = 6;
|
||||
|
||||
a52dec = GST_A52DEC (bdec);
|
||||
|
||||
/* no fancy draining */
|
||||
if (G_UNLIKELY (!buffer))
|
||||
return GST_FLOW_OK;
|
||||
|
||||
/* parsed stuff already, so this should work out fine */
|
||||
data = GST_BUFFER_DATA (buffer);
|
||||
size = GST_BUFFER_SIZE (buffer);
|
||||
g_assert (size >= 7);
|
||||
|
||||
/* re-obtain some sync header info,
|
||||
* should be same as during _parse and could also be cached there,
|
||||
* but anyway ... */
|
||||
bit_rate = a52dec->bit_rate;
|
||||
sample_rate = a52dec->sample_rate;
|
||||
flags = 0;
|
||||
length = a52_syncinfo (data, &flags, &sample_rate, &bit_rate);
|
||||
g_assert (length == size);
|
||||
|
||||
/* update stream information, renegotiate or re-streaminfo if needed */
|
||||
need_reneg = FALSE;
|
||||
|
|
@ -582,7 +539,7 @@ gst_a52dec_handle_frame (GstA52Dec * a52dec, guint8 * data,
|
|||
|
||||
a52dec->flag_update = FALSE;
|
||||
|
||||
caps = gst_pad_get_allowed_caps (a52dec->srcpad);
|
||||
caps = gst_pad_get_allowed_caps (GST_AUDIO_DECODER_SRC_PAD (a52dec));
|
||||
if (caps && gst_caps_get_size (caps) > 0) {
|
||||
GstCaps *copy = gst_caps_copy_nth (caps, 0);
|
||||
GstStructure *structure = gst_caps_get_structure (copy, 0);
|
||||
|
|
@ -618,14 +575,16 @@ gst_a52dec_handle_frame (GstA52Dec * a52dec, guint8 * data,
|
|||
} else {
|
||||
flags = a52dec->using_channels;
|
||||
}
|
||||
|
||||
/* process */
|
||||
flags |= A52_ADJUST_LEVEL;
|
||||
a52dec->level = 1;
|
||||
if (a52_frame (a52dec->state, data, &flags, &a52dec->level, a52dec->bias)) {
|
||||
GST_WARNING ("a52_frame error");
|
||||
a52dec->discont = TRUE;
|
||||
return GST_FLOW_OK;
|
||||
GST_AUDIO_DECODER_ERROR (a52dec, 1, STREAM, DECODE, (NULL),
|
||||
("a52_frame error"), result);
|
||||
goto exit;
|
||||
}
|
||||
|
||||
channels = flags & (A52_CHANNEL_MASK | A52_LFE);
|
||||
if (a52dec->using_channels != channels) {
|
||||
need_reneg = TRUE;
|
||||
|
|
@ -634,43 +593,74 @@ gst_a52dec_handle_frame (GstA52Dec * a52dec, guint8 * data,
|
|||
|
||||
/* negotiate if required */
|
||||
if (need_reneg) {
|
||||
GST_DEBUG ("a52dec reneg: sample_rate:%d stream_chans:%d using_chans:%d",
|
||||
GST_DEBUG_OBJECT (a52dec,
|
||||
"a52dec reneg: sample_rate:%d stream_chans:%d using_chans:%d",
|
||||
a52dec->sample_rate, a52dec->stream_channels, a52dec->using_channels);
|
||||
if (!gst_a52dec_reneg (a52dec, a52dec->srcpad)) {
|
||||
GST_ELEMENT_ERROR (a52dec, CORE, NEGOTIATION, (NULL), (NULL));
|
||||
return GST_FLOW_ERROR;
|
||||
}
|
||||
if (!gst_a52dec_reneg (a52dec, GST_AUDIO_DECODER_SRC_PAD (a52dec)))
|
||||
goto failed_negotiation;
|
||||
}
|
||||
|
||||
if (a52dec->dynamic_range_compression == FALSE) {
|
||||
a52_dynrng (a52dec->state, NULL, NULL);
|
||||
}
|
||||
|
||||
/* each frame consists of 6 blocks */
|
||||
for (i = 0; i < 6; i++) {
|
||||
if (a52_block (a52dec->state)) {
|
||||
/* ignore errors but mark a discont */
|
||||
GST_WARNING ("a52_block error %d", i);
|
||||
a52dec->discont = TRUE;
|
||||
} else {
|
||||
GstFlowReturn ret;
|
||||
flags &= (A52_CHANNEL_MASK | A52_LFE);
|
||||
chans = gst_a52dec_channels (flags, NULL);
|
||||
if (!chans)
|
||||
goto invalid_flags;
|
||||
|
||||
/* push on */
|
||||
ret = gst_a52dec_push (a52dec, a52dec->srcpad, a52dec->using_channels,
|
||||
a52dec->samples, a52dec->time);
|
||||
if (ret != GST_FLOW_OK)
|
||||
return ret;
|
||||
/* handle decoded data;
|
||||
* each frame has 6 blocks, one block is 256 samples, ea */
|
||||
result =
|
||||
gst_pad_alloc_buffer_and_set_caps (GST_AUDIO_DECODER_SRC_PAD (a52dec), 0,
|
||||
256 * chans * (SAMPLE_WIDTH / 8) * num_blocks,
|
||||
GST_PAD_CAPS (GST_AUDIO_DECODER_SRC_PAD (a52dec)), &outbuf);
|
||||
if (result != GST_FLOW_OK)
|
||||
goto exit;
|
||||
|
||||
data = GST_BUFFER_DATA (outbuf);
|
||||
for (i = 0; i < num_blocks; i++) {
|
||||
if (a52_block (a52dec->state)) {
|
||||
/* also marks discont */
|
||||
GST_AUDIO_DECODER_ERROR (a52dec, 1, STREAM, DECODE, (NULL),
|
||||
("error decoding block %d", i), result);
|
||||
if (result != GST_FLOW_OK)
|
||||
goto exit;
|
||||
} else {
|
||||
gint n, c;
|
||||
|
||||
for (n = 0; n < 256; n++) {
|
||||
for (c = 0; c < chans; c++) {
|
||||
((sample_t *) data)[n * chans + c] = a52dec->samples[c * 256 + n];
|
||||
}
|
||||
}
|
||||
}
|
||||
a52dec->time += 256 * GST_SECOND / a52dec->sample_rate;
|
||||
data += 256 * chans * (SAMPLE_WIDTH / 8);
|
||||
}
|
||||
|
||||
return GST_FLOW_OK;
|
||||
result = gst_audio_decoder_finish_frame (bdec, outbuf, 1);
|
||||
|
||||
exit:
|
||||
return result;
|
||||
|
||||
/* ERRORS */
|
||||
failed_negotiation:
|
||||
{
|
||||
GST_ELEMENT_ERROR (a52dec, CORE, NEGOTIATION, (NULL), (NULL));
|
||||
return GST_FLOW_ERROR;
|
||||
}
|
||||
invalid_flags:
|
||||
{
|
||||
GST_ELEMENT_ERROR (GST_ELEMENT (a52dec), STREAM, DECODE, (NULL),
|
||||
("Invalid channel flags: %d", flags));
|
||||
return GST_FLOW_ERROR;
|
||||
}
|
||||
}
|
||||
|
||||
static gboolean
|
||||
gst_a52dec_sink_setcaps (GstPad * pad, GstCaps * caps)
|
||||
gst_a52dec_set_format (GstAudioDecoder * bdec, GstCaps * caps)
|
||||
{
|
||||
GstA52Dec *a52dec = GST_A52DEC (gst_pad_get_parent (pad));
|
||||
GstA52Dec *a52dec = GST_A52DEC (bdec);
|
||||
GstStructure *structure;
|
||||
|
||||
structure = gst_caps_get_structure (caps, 0);
|
||||
|
|
@ -680,8 +670,6 @@ gst_a52dec_sink_setcaps (GstPad * pad, GstCaps * caps)
|
|||
else
|
||||
a52dec->dvdmode = FALSE;
|
||||
|
||||
gst_object_unref (a52dec);
|
||||
|
||||
return TRUE;
|
||||
}
|
||||
|
||||
|
|
@ -689,20 +677,9 @@ static GstFlowReturn
|
|||
gst_a52dec_chain (GstPad * pad, GstBuffer * buf)
|
||||
{
|
||||
GstA52Dec *a52dec = GST_A52DEC (GST_PAD_PARENT (pad));
|
||||
GstFlowReturn ret;
|
||||
GstFlowReturn ret = GST_FLOW_OK;
|
||||
gint first_access;
|
||||
|
||||
if (GST_BUFFER_IS_DISCONT (buf)) {
|
||||
GST_LOG_OBJECT (a52dec, "received DISCONT");
|
||||
gst_a52dec_drain (a52dec);
|
||||
/* clear cache on discont and mark a discont in the element */
|
||||
if (a52dec->cache) {
|
||||
gst_buffer_unref (a52dec->cache);
|
||||
a52dec->cache = NULL;
|
||||
}
|
||||
a52dec->discont = TRUE;
|
||||
}
|
||||
|
||||
if (a52dec->dvdmode) {
|
||||
gint size = GST_BUFFER_SIZE (buf);
|
||||
guchar *data = GST_BUFFER_DATA (buf);
|
||||
|
|
@ -726,33 +703,36 @@ gst_a52dec_chain (GstPad * pad, GstBuffer * buf)
|
|||
goto bad_first_access_parameter;
|
||||
|
||||
subbuf = gst_buffer_create_sub (buf, offset, len);
|
||||
gst_buffer_copy_metadata (subbuf, buf, GST_BUFFER_COPY_ALL);
|
||||
GST_BUFFER_TIMESTAMP (subbuf) = GST_CLOCK_TIME_NONE;
|
||||
ret = gst_a52dec_chain_raw (pad, subbuf);
|
||||
if (ret != GST_FLOW_OK)
|
||||
ret = a52dec->base_chain (pad, subbuf);
|
||||
if (ret != GST_FLOW_OK) {
|
||||
gst_buffer_unref (buf);
|
||||
goto done;
|
||||
}
|
||||
|
||||
offset += len;
|
||||
len = size - offset;
|
||||
|
||||
if (len > 0) {
|
||||
subbuf = gst_buffer_create_sub (buf, offset, len);
|
||||
gst_buffer_copy_metadata (subbuf, buf, GST_BUFFER_COPY_ALL);
|
||||
GST_BUFFER_TIMESTAMP (subbuf) = GST_BUFFER_TIMESTAMP (buf);
|
||||
|
||||
ret = gst_a52dec_chain_raw (pad, subbuf);
|
||||
ret = a52dec->base_chain (pad, subbuf);
|
||||
}
|
||||
gst_buffer_unref (buf);
|
||||
} else {
|
||||
/* first_access = 0 or 1, so if there's a timestamp it applies to the first byte */
|
||||
subbuf = gst_buffer_create_sub (buf, offset, size - offset);
|
||||
GST_BUFFER_TIMESTAMP (subbuf) = GST_BUFFER_TIMESTAMP (buf);
|
||||
ret = gst_a52dec_chain_raw (pad, subbuf);
|
||||
gst_buffer_copy_metadata (subbuf, buf, GST_BUFFER_COPY_ALL);
|
||||
ret = a52dec->base_chain (pad, subbuf);
|
||||
}
|
||||
} else {
|
||||
gst_buffer_ref (buf);
|
||||
ret = gst_a52dec_chain_raw (pad, buf);
|
||||
ret = a52dec->base_chain (pad, buf);
|
||||
}
|
||||
|
||||
done:
|
||||
gst_buffer_unref (buf);
|
||||
return ret;
|
||||
|
||||
/* ERRORS */
|
||||
|
|
@ -772,162 +752,6 @@ bad_first_access_parameter:
|
|||
}
|
||||
}
|
||||
|
||||
static GstFlowReturn
|
||||
gst_a52dec_chain_raw (GstPad * pad, GstBuffer * buf)
|
||||
{
|
||||
GstA52Dec *a52dec;
|
||||
guint8 *data;
|
||||
guint size;
|
||||
gint length = 0, flags, sample_rate, bit_rate;
|
||||
GstFlowReturn result = GST_FLOW_OK;
|
||||
|
||||
a52dec = GST_A52DEC (GST_PAD_PARENT (pad));
|
||||
|
||||
if (!a52dec->sent_segment) {
|
||||
GstSegment segment;
|
||||
|
||||
/* Create a basic segment. Usually, we'll get a new-segment sent by
|
||||
* another element that will know more information (a demuxer). If we're
|
||||
* just looking at a raw AC3 stream, we won't - so we need to send one
|
||||
* here, but we don't know much info, so just send a minimal TIME
|
||||
* new-segment event
|
||||
*/
|
||||
gst_segment_init (&segment, GST_FORMAT_TIME);
|
||||
gst_pad_push_event (a52dec->srcpad, gst_event_new_new_segment (FALSE,
|
||||
segment.rate, segment.format, segment.start,
|
||||
segment.duration, segment.start));
|
||||
a52dec->sent_segment = TRUE;
|
||||
}
|
||||
|
||||
/* merge with cache, if any. Also make sure timestamps match */
|
||||
if (GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
|
||||
a52dec->time = GST_BUFFER_TIMESTAMP (buf);
|
||||
GST_DEBUG_OBJECT (a52dec,
|
||||
"Received buffer with ts %" GST_TIME_FORMAT " duration %"
|
||||
GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
|
||||
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
|
||||
}
|
||||
|
||||
if (a52dec->cache) {
|
||||
buf = gst_buffer_join (a52dec->cache, buf);
|
||||
a52dec->cache = NULL;
|
||||
}
|
||||
data = GST_BUFFER_DATA (buf);
|
||||
size = GST_BUFFER_SIZE (buf);
|
||||
|
||||
/* find and read header */
|
||||
bit_rate = a52dec->bit_rate;
|
||||
sample_rate = a52dec->sample_rate;
|
||||
flags = 0;
|
||||
while (size >= 7) {
|
||||
length = a52_syncinfo (data, &flags, &sample_rate, &bit_rate);
|
||||
|
||||
if (length == 0) {
|
||||
/* no sync */
|
||||
data++;
|
||||
size--;
|
||||
} else if (length <= size) {
|
||||
GST_DEBUG ("Sync: %d", length);
|
||||
|
||||
if (flags != a52dec->prev_flags)
|
||||
a52dec->flag_update = TRUE;
|
||||
a52dec->prev_flags = flags;
|
||||
|
||||
result = gst_a52dec_handle_frame (a52dec, data,
|
||||
length, flags, sample_rate, bit_rate);
|
||||
if (result != GST_FLOW_OK) {
|
||||
size = 0;
|
||||
break;
|
||||
}
|
||||
size -= length;
|
||||
data += length;
|
||||
} else {
|
||||
/* not enough data */
|
||||
GST_LOG ("Not enough data available");
|
||||
break;
|
||||
}
|
||||
}
|
||||
|
||||
/* keep cache */
|
||||
if (length == 0) {
|
||||
GST_LOG ("No sync found");
|
||||
}
|
||||
|
||||
if (size > 0) {
|
||||
a52dec->cache = gst_buffer_create_sub (buf,
|
||||
GST_BUFFER_SIZE (buf) - size, size);
|
||||
}
|
||||
|
||||
gst_buffer_unref (buf);
|
||||
|
||||
return result;
|
||||
}
|
||||
|
||||
static GstStateChangeReturn
|
||||
gst_a52dec_change_state (GstElement * element, GstStateChange transition)
|
||||
{
|
||||
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
|
||||
GstA52Dec *a52dec = GST_A52DEC (element);
|
||||
|
||||
switch (transition) {
|
||||
case GST_STATE_CHANGE_NULL_TO_READY:{
|
||||
GstA52DecClass *klass;
|
||||
|
||||
klass = GST_A52DEC_CLASS (G_OBJECT_GET_CLASS (a52dec));
|
||||
a52dec->state = a52_init (klass->a52_cpuflags);
|
||||
|
||||
if (!a52dec->state) {
|
||||
GST_ELEMENT_ERROR (GST_ELEMENT (a52dec), STREAM, DECODE, (NULL),
|
||||
("Failed to initialize a52 state"));
|
||||
ret = GST_STATE_CHANGE_FAILURE;
|
||||
}
|
||||
break;
|
||||
}
|
||||
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
||||
a52dec->samples = a52_samples (a52dec->state);
|
||||
a52dec->bit_rate = -1;
|
||||
a52dec->sample_rate = -1;
|
||||
a52dec->stream_channels = A52_CHANNEL;
|
||||
a52dec->using_channels = A52_CHANNEL;
|
||||
a52dec->level = 1;
|
||||
a52dec->bias = 0;
|
||||
a52dec->time = 0;
|
||||
a52dec->sent_segment = FALSE;
|
||||
a52dec->flag_update = TRUE;
|
||||
gst_segment_init (&a52dec->segment, GST_FORMAT_UNDEFINED);
|
||||
break;
|
||||
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
|
||||
break;
|
||||
default:
|
||||
break;
|
||||
}
|
||||
|
||||
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
||||
|
||||
switch (transition) {
|
||||
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
|
||||
break;
|
||||
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
||||
a52dec->samples = NULL;
|
||||
if (a52dec->cache) {
|
||||
gst_buffer_unref (a52dec->cache);
|
||||
a52dec->cache = NULL;
|
||||
}
|
||||
clear_queued (a52dec);
|
||||
break;
|
||||
case GST_STATE_CHANGE_READY_TO_NULL:
|
||||
if (a52dec->state) {
|
||||
a52_free (a52dec->state);
|
||||
a52dec->state = NULL;
|
||||
}
|
||||
break;
|
||||
default:
|
||||
break;
|
||||
}
|
||||
|
||||
return ret;
|
||||
}
|
||||
|
||||
static void
|
||||
gst_a52dec_set_property (GObject * object, guint prop_id, const GValue * value,
|
||||
GParamSpec * pspec)
|
||||
|
|
|
|||
|
|
@ -22,6 +22,7 @@
|
|||
#define __GST_A52DEC_H__
|
||||
|
||||
#include <gst/gst.h>
|
||||
#include <gst/audio/gstaudiodecoder.h>
|
||||
|
||||
G_BEGIN_DECLS
|
||||
|
||||
|
|
@ -40,41 +41,33 @@ typedef struct _GstA52Dec GstA52Dec;
|
|||
typedef struct _GstA52DecClass GstA52DecClass;
|
||||
|
||||
struct _GstA52Dec {
|
||||
GstElement element;
|
||||
GstAudioDecoder element;
|
||||
|
||||
/* pads */
|
||||
GstPad *sinkpad,
|
||||
*srcpad;
|
||||
GstSegment segment;
|
||||
GstPadChainFunction base_chain;
|
||||
|
||||
gboolean dvdmode;
|
||||
gboolean sent_segment;
|
||||
gboolean discont;
|
||||
|
||||
gboolean flag_update;
|
||||
int prev_flags;
|
||||
|
||||
/* stream properties */
|
||||
int bit_rate;
|
||||
int sample_rate;
|
||||
int stream_channels;
|
||||
int request_channels;
|
||||
int using_channels;
|
||||
|
||||
/* decoding properties */
|
||||
sample_t level;
|
||||
sample_t bias;
|
||||
gboolean dynamic_range_compression;
|
||||
sample_t *samples;
|
||||
a52_state_t *state;
|
||||
|
||||
GstBuffer *cache;
|
||||
GstClockTime time;
|
||||
|
||||
/* reverse */
|
||||
GList *queued;
|
||||
GstTagList *pending_tags;
|
||||
};
|
||||
|
||||
struct _GstA52DecClass {
|
||||
GstElementClass parent_class;
|
||||
GstAudioDecoderClass parent_class;
|
||||
|
||||
guint32 a52_cpuflags;
|
||||
};
|
||||
|
|
|
|||
Loading…
Reference in a new issue