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faad: port to 0.11
This commit is contained in:
Wim Taymans
parent
7f4cf50496
commit
f03b320c8d
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@ -95,24 +95,12 @@ static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
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GST_STATIC_CAPS ("audio/mpeg, " "mpegversion = (int) { 2, 4 }")
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);
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#define STATIC_INT_CAPS(bpp) \
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"audio/x-raw-int, " \
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"endianness = (int) BYTE_ORDER, " \
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"signed = (bool) TRUE, " \
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"width = (int) " G_STRINGIFY (bpp) ", " \
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"depth = (int) " G_STRINGIFY (bpp) ", " \
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#define STATIC_RAW_CAPS(format) \
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"audio/x-raw, " \
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"format = (string) "GST_AUDIO_NE(format)", " \
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"rate = (int) [ 8000, 96000 ], " \
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"channels = (int) [ 1, 8 ]"
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#if 0
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#define STATIC_FLOAT_CAPS(bpp) \
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"audio/x-raw-float, " \
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"endianness = (int) BYTE_ORDER, " \
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"depth = (int) " G_STRINGIFY (bpp) ", " \
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"rate = (int) [ 8000, 96000 ], " \
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"channels = (int) [ 1, 8 ]"
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#endif
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/*
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* All except 16-bit integer are disabled until someone fixes FAAD.
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* FAAD allocates approximately 8*1024*2 bytes bytes, which is enough
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@ -122,16 +110,16 @@ static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
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*/
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#define STATIC_CAPS \
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STATIC_INT_CAPS (16)
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STATIC_RAW_CAPS (S16)
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#if 0
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#define NOTUSED "; " \
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STATIC_INT_CAPS (24) \
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STATIC_RAW_CAPS (S24) \
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"; " \
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STATIC_INT_CAPS (32) \
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STATIC_RAW_CAPS (S32) \
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"; " \
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STATIC_FLOAT_CAPS (32) \
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STATIC_RAW_CAPS (F32) \
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"; " \
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STATIC_FLOAT_CAPS (64)
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STATIC_RAW_CAPS (F64)
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#endif
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static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
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@ -154,12 +142,14 @@ static void gst_faad_flush (GstAudioDecoder * dec, gboolean hard);
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static gboolean gst_faad_open_decoder (GstFaad * faad);
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static void gst_faad_close_decoder (GstFaad * faad);
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GST_BOILERPLATE (GstFaad, gst_faad, GstAudioDecoder, GST_TYPE_AUDIO_DECODER);
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#define gst_faad_parent_class parent_class
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G_DEFINE_TYPE (GstFaad, gst_faad, GST_TYPE_AUDIO_DECODER);
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static void
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gst_faad_base_init (gpointer klass)
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gst_faad_class_init (GstFaadClass * klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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GstAudioDecoderClass *base_class = GST_AUDIO_DECODER_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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@ -171,26 +161,18 @@ gst_faad_base_init (gpointer klass)
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"Free MPEG-2/4 AAC decoder",
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"Ronald Bultje <rbultje@ronald.bitfreak.net>");
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GST_DEBUG_CATEGORY_INIT (faad_debug, "faad", 0, "AAC decoding");
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}
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static void
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gst_faad_class_init (GstFaadClass * klass)
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{
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GstAudioDecoderClass *base_class = GST_AUDIO_DECODER_CLASS (klass);
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parent_class = g_type_class_peek_parent (klass);
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base_class->start = GST_DEBUG_FUNCPTR (gst_faad_start);
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base_class->stop = GST_DEBUG_FUNCPTR (gst_faad_stop);
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base_class->set_format = GST_DEBUG_FUNCPTR (gst_faad_set_format);
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base_class->parse = GST_DEBUG_FUNCPTR (gst_faad_parse);
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base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_faad_handle_frame);
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base_class->flush = GST_DEBUG_FUNCPTR (gst_faad_flush);
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GST_DEBUG_CATEGORY_INIT (faad_debug, "faad", 0, "AAC decoding");
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}
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static void
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gst_faad_init (GstFaad * faad, GstFaadClass * klass)
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gst_faad_init (GstFaad * faad)
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{
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gst_faad_reset (faad);
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}
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@ -280,6 +262,8 @@ gst_faad_set_format (GstAudioDecoder * dec, GstCaps * caps)
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GstStructure *str = gst_caps_get_structure (caps, 0);
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GstBuffer *buf;
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const GValue *value;
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guint8 *cdata;
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gsize csize;
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/* clean up current decoder, rather than trying to reconfigure */
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gst_faad_close_decoder (faad);
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@ -294,8 +278,6 @@ gst_faad_set_format (GstAudioDecoder * dec, GstCaps * caps)
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guint32 samplerate;
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#endif
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guint8 channels;
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guint8 *cdata;
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guint csize;
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/* We have codec data, means packetised stream */
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faad->packetised = TRUE;
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@ -303,8 +285,7 @@ gst_faad_set_format (GstAudioDecoder * dec, GstCaps * caps)
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buf = gst_value_get_buffer (value);
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g_return_val_if_fail (buf != NULL, FALSE);
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cdata = GST_BUFFER_DATA (buf);
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csize = GST_BUFFER_SIZE (buf);
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cdata = gst_buffer_map (buf, &csize, NULL, GST_MAP_READ);
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if (csize < 2)
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goto wrong_length;
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@ -375,18 +356,21 @@ wrong_length:
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{
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GST_DEBUG_OBJECT (faad, "codec_data less than 2 bytes long");
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gst_object_unref (faad);
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gst_buffer_unmap (buf, cdata, csize);
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return FALSE;
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}
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open_failed:
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{
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GST_DEBUG_OBJECT (faad, "failed to create decoder");
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gst_object_unref (faad);
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gst_buffer_unmap (buf, cdata, csize);
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return FALSE;
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}
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init_failed:
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{
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GST_DEBUG_OBJECT (faad, "faacDecInit2() failed");
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gst_object_unref (faad);
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gst_buffer_unmap (buf, cdata, csize);
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return FALSE;
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}
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}
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@ -662,9 +646,13 @@ gst_faad_parse (GstAudioDecoder * dec, GstAdapter * adapter,
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*length = size;
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return GST_FLOW_OK;
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} else {
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data = gst_adapter_peek (adapter, size);
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return gst_faad_sync (faad, data, size, !eos, offset, length) ?
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GST_FLOW_OK : GST_FLOW_UNEXPECTED;
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gboolean ret;
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data = gst_adapter_map (adapter, size);
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ret = gst_faad_sync (faad, data, size, !eos, offset, length);
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gst_adapter_unmap (adapter, 0);
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return (ret ? GST_FLOW_OK : GST_FLOW_UNEXPECTED);
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}
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}
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@ -673,7 +661,7 @@ gst_faad_handle_frame (GstAudioDecoder * dec, GstBuffer * buffer)
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{
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GstFaad *faad;
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GstFlowReturn ret = GST_FLOW_OK;
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guint input_size;
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gsize input_size;
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guchar *input_data;
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GstBuffer *outbuf;
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faacDecFrameInfo info;
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@ -685,8 +673,7 @@ gst_faad_handle_frame (GstAudioDecoder * dec, GstBuffer * buffer)
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if (G_UNLIKELY (!buffer))
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return GST_FLOW_OK;
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input_data = GST_BUFFER_DATA (buffer);
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input_size = GST_BUFFER_SIZE (buffer);
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input_data = gst_buffer_map (buffer, &input_size, NULL, GST_MAP_READ);
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init:
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/* init if not already done during capsnego */
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@ -764,20 +751,17 @@ init:
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goto sample_overflow;
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/* note: info.samples is total samples, not per channel */
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ret =
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gst_pad_alloc_buffer_and_set_caps (GST_AUDIO_DECODER_SRC_PAD
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(faad), 0, info.samples * faad->bps,
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GST_PAD_CAPS (GST_AUDIO_DECODER_SRC_PAD (faad)), &outbuf);
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if (ret != GST_FLOW_OK)
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goto out;
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memcpy (GST_BUFFER_DATA (outbuf), out, GST_BUFFER_SIZE (outbuf));
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/* FIXME, add bufferpool and allocator support to the base class */
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outbuf = gst_buffer_new_allocate (NULL, info.samples * faad->bps, 0);
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gst_buffer_fill (outbuf, 0, out, info.samples * faad->bps);
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ret = gst_audio_decoder_finish_frame (dec, outbuf, 1);
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}
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} while (FALSE);
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out:
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gst_buffer_unmap (buffer, input_data, input_size);
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return ret;
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/* ERRORS */
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