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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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faac: port to 0.11
This commit is contained in:
Wim Taymans
parent
0806e5cbb9
commit
e856f6f475
2 changed files with 41 additions and 42 deletions
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@ -305,7 +305,7 @@ GST_PLUGINS_NONPORTED=" adpcmdec adpcmenc aiff asfmux \
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videomaxrate videomeasure videoparsers videosignal vmnc y4m \
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decklink fbdev linsys shm vcd \
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voaacenc apexsink bz2 camerabin2 cdaudio celt cog curl dc1394 dirac directfb dts resindvd \
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faac gsettings gsm jp2k ladspa modplug mpeg2enc mplex mimic \
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gsettings gsm jp2k ladspa modplug mpeg2enc mplex mimic \
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musepack musicbrainz nas neon ofa openal rsvg schro sdl sndfile soundtouch timidity \
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vp8 wildmidi xvid "
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AC_SUBST(GST_PLUGINS_NONPORTED)
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@ -56,11 +56,8 @@
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"88200, " \
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"96000"
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#define SINK_CAPS \
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"audio/x-raw-int, " \
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"endianness = (int) BYTE_ORDER, " \
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"signed = (boolean) true, " \
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"width = (int) 16, " \
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"depth = (int) 16, " \
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"audio/x-raw, " \
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"format = (string) "GST_AUDIO_NE (S16) ", " \
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"rate = (int) {" SAMPLE_RATES "}, " \
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"channels = (int) [ 1, 6 ] "
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@ -125,7 +122,7 @@ static void gst_faac_get_property (GObject * object,
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guint prop_id, GValue * value, GParamSpec * pspec);
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static gboolean gst_faac_configure_source_pad (GstFaac * faac);
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static GstCaps *gst_faac_getcaps (GstAudioEncoder * enc);
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static GstCaps *gst_faac_getcaps (GstAudioEncoder * enc, GstCaps * filter);
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static gboolean gst_faac_start (GstAudioEncoder * enc);
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static gboolean gst_faac_stop (GstAudioEncoder * enc);
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@ -144,25 +141,8 @@ GST_DEBUG_CATEGORY_STATIC (faac_debug);
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#define FAAC_DEFAULT_MIDSIDE TRUE
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#define FAAC_DEFAULT_SHORTCTL SHORTCTL_NORMAL
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GST_BOILERPLATE (GstFaac, gst_faac, GstAudioEncoder, GST_TYPE_AUDIO_ENCODER);
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static void
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gst_faac_base_init (gpointer klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&src_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&sink_template));
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gst_element_class_set_details_simple (element_class, "AAC audio encoder",
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"Codec/Encoder/Audio",
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"Free MPEG-2/4 AAC encoder",
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"Ronald Bultje <rbultje@ronald.bitfreak.net>");
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GST_DEBUG_CATEGORY_INIT (faac_debug, "faac", 0, "AAC encoding");
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}
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#define gst_faac_parent_class parent_class
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G_DEFINE_TYPE (GstFaac, gst_faac, GST_TYPE_AUDIO_ENCODER);
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#define GST_TYPE_FAAC_RATE_CONTROL (gst_faac_brtype_get_type ())
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static GType
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@ -209,13 +189,22 @@ static void
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gst_faac_class_init (GstFaacClass * klass)
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{
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GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
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GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
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GstAudioEncoderClass *base_class = GST_AUDIO_ENCODER_CLASS (klass);
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parent_class = g_type_class_peek_parent (klass);
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gobject_class->set_property = gst_faac_set_property;
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gobject_class->get_property = gst_faac_get_property;
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&src_template));
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&sink_template));
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gst_element_class_set_details_simple (gstelement_class, "AAC audio encoder",
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"Codec/Encoder/Audio",
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"Free MPEG-2/4 AAC encoder",
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"Ronald Bultje <rbultje@ronald.bitfreak.net>");
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base_class->start = GST_DEBUG_FUNCPTR (gst_faac_start);
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base_class->stop = GST_DEBUG_FUNCPTR (gst_faac_stop);
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base_class->set_format = GST_DEBUG_FUNCPTR (gst_faac_set_format);
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@ -251,10 +240,12 @@ gst_faac_class_init (GstFaacClass * klass)
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"Block type encorcing",
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GST_TYPE_FAAC_SHORTCTL, FAAC_DEFAULT_SHORTCTL,
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G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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GST_DEBUG_CATEGORY_INIT (faac_debug, "faac", 0, "AAC encoding");
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}
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static void
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gst_faac_init (GstFaac * faac, GstFaacClass * klass)
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gst_faac_init (GstFaac * faac)
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{
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}
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@ -315,7 +306,7 @@ static const GstAudioChannelPosition aac_channel_positions[][8] = {
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};
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static GstCaps *
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gst_faac_getcaps (GstAudioEncoder * enc)
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gst_faac_getcaps (GstAudioEncoder * enc, GstCaps * filter)
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{
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static volatile gsize sinkcaps = 0;
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@ -338,10 +329,8 @@ gst_faac_getcaps (GstAudioEncoder * enc)
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}
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g_value_unset (&tmp_v);
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s = gst_structure_new ("audio/x-raw-int",
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"endianness", G_TYPE_INT, G_BYTE_ORDER,
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"signed", G_TYPE_BOOLEAN, TRUE,
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"width", G_TYPE_INT, 16, "depth", G_TYPE_INT, 16, NULL);
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s = gst_structure_new ("audio/x-raw",
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"format", G_TYPE_STRING, GST_AUDIO_NE (S16), NULL);
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gst_structure_set_value (s, "rate", &rates_arr);
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for (i = 1; i <= 6; i++) {
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@ -588,7 +577,7 @@ gst_faac_configure_source_pad (GstFaac * faac)
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/* copy it into a buffer */
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codec_data = gst_buffer_new_and_alloc (config_len);
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memcpy (GST_BUFFER_DATA (codec_data), config, config_len);
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gst_buffer_fill (codec_data, 0, config, config_len);
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/* add to caps */
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gst_caps_set_simple (srccaps,
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@ -647,28 +636,35 @@ gst_faac_handle_frame (GstAudioEncoder * enc, GstBuffer * in_buf)
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GstFaac *faac = GST_FAAC (enc);
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GstFlowReturn ret = GST_FLOW_OK;
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GstBuffer *out_buf;
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gint size, ret_size;
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const guint8 *data;
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gsize size, ret_size;
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guint8 *data;
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guint8 *out_data;
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gsize out_size;
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out_buf = gst_buffer_new_and_alloc (faac->bytes);
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out_data = gst_buffer_map (out_buf, &out_size, NULL, GST_MAP_WRITE);
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if (G_LIKELY (in_buf)) {
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data = GST_BUFFER_DATA (in_buf);
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size = GST_BUFFER_SIZE (in_buf);
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data = gst_buffer_map (in_buf, &size, NULL, GST_MAP_READ);
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} else {
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data = NULL;
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size = 0;
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}
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if (G_UNLIKELY ((ret_size = faacEncEncode (faac->handle, (gint32 *) data,
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size / faac->bps, GST_BUFFER_DATA (out_buf),
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GST_BUFFER_SIZE (out_buf))) < 0))
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size / faac->bps, out_data, out_size)) < 0))
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goto encode_failed;
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gst_buffer_unmap (in_buf, data, size);
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GST_LOG_OBJECT (faac, "encoder return: %d", ret_size);
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if (ret_size > 0) {
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GST_BUFFER_SIZE (out_buf) = ret_size;
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gst_buffer_unmap (out_buf, out_data, ret_size);
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ret = gst_audio_encoder_finish_frame (enc, out_buf, faac->samples);
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} else {
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gst_buffer_unmap (out_buf, out_data, 0);
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gst_buffer_unref (out_buf);
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}
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return ret;
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@ -677,6 +673,9 @@ gst_faac_handle_frame (GstAudioEncoder * enc, GstBuffer * in_buf)
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encode_failed:
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{
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GST_ELEMENT_ERROR (faac, LIBRARY, ENCODE, (NULL), (NULL));
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gst_buffer_unmap (in_buf, data, size);
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gst_buffer_unmap (out_buf, out_data, 0);
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gst_buffer_unref (out_buf);
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return GST_FLOW_ERROR;
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}
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}
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