This is a more o less stable element for aac audio decoding.

* It works for SN 23.i3.3 and 23.i3.8
* The input requires to be parsed (aacparse required if the stream is not in
  a container)
* The performance for an audio only pipeline is from 10% to 0% of arm usage
* A/V rendering looks quite good (IMO)

TODO:

* Fix caps renegotiation when parametric stereo stream is detected
* Tests for non-framed streams

Happy holidays.

vmjl

Víctor Manuel Jáquez Leal (2):
  base: add codec update_params() callback
  adec: aac decoder implementation

This patch add the callback update_params(), which will update the
SN parameters according to the received message.

1. http://en.wikipedia.org/wiki/Parametric_Stereo

Signed-off-by: Víctor Manuel Jáquez Leal <vjaq...@igalia.com>
---
 gstdspbase.c |    8 ++++++++
 gstdspbase.h |    1 +
 2 files changed, 9 insertions(+), 0 deletions(-)

diff --git a/gstdspbase.c b/gstdspbase.c
index 953328a..dd6b98f 100644
--- a/gstdspbase.c
+++ b/gstdspbase.c
@@ -227,6 +227,14 @@ got_message(GstDspBase *self,
                        pr_debug(self, "playback completed");
                        break;
                }
+
+               if (msg->arg_1 == 1 && (msg->arg_2 & 0x0600) == 0x0600) {
+                       struct td_codec *codec = self->codec;
+                       if (codec->update_params && self->node)
+                               codec->update_params(self, self->node, msg->arg_2);
+                       break;
+               }
+
                pr_warning(self, "DSP event: cmd=0x%04X, arg1=%u, arg2=0x%04X",
                           msg->cmd, msg->arg_1, msg->arg_2);
                if ((msg->arg_2 & 0x0F00) == 0x0F00)
diff --git a/gstdspbase.h b/gstdspbase.h
index 829b5b1..0014635 100644
--- a/gstdspbase.h
+++ b/gstdspbase.h
@@ -56,6 +56,7 @@ struct td_codec {
        bool (*handle_extra_data)(GstDspBase *base, GstBuffer *buf);
        void (*flush_buffer)(GstDspBase *base);
        void (*send_params)(GstDspBase *base, struct dsp_node *node);
+       void (*update_params) (GstDspBase *base, struct dsp_node *node, uint32_t msg);
 };

The performance of a simple pipeline goes among 10% and 0% of CPU
usage (montored with top and with gstreamer 0.10.31)

My testing pipeline is something like:

gst-launch filesrc location=test.aac ! aacparse ! dspadec ! alsasink

Issues: when a parametric stereo media is found, a downstream caps
renegotiation is done, but somehow (I don't understand why) the
renegotiation is not honored and the output is rendered wrong (because
of the reduced framerate).

Signed-off-by: Víctor Manuel Jáquez Leal <vjaq...@igalia.com>
---
 Makefile          |    4 +-
 gstdspadec.c      |  280 +++++++++++++++++++++++++++++++++++++++++++++++++++++
 gstdspadec.h      |   47 +++++++++
 gstdspbase.h      |    2 +
 plugin.c          |    4 +
 tidsp/td_aacdec.c |  175 +++++++++++++++++++++++++++++++++
 6 files changed, 510 insertions(+), 2 deletions(-)
 create mode 100644 gstdspadec.c
 create mode 100644 gstdspadec.h
 create mode 100644 tidsp/td_aacdec.c

diff --git a/Makefile b/Makefile
index c397101..4303eeb 100644
--- a/Makefile
+++ b/Makefile
@@ -23,7 +23,7 @@ prefix := /usr
 D = $(DESTDIR)

 tidsp.a: tidsp/td_mp4vdec.o tidsp/td_h264dec.o tidsp/td_wmvdec.o \
-       tidsp/td_jpegdec.o \
+       tidsp/td_jpegdec.o tidsp/td_aacdec.o \
        tidsp/td_mp4venc.o tidsp/td_jpegenc.o tidsp/td_h264enc.o
 tidsp.a: override CFLAGS += -fPIC -I.

@@ -34,7 +34,7 @@ gst_plugin := libgstdsp.so
 $(gst_plugin): plugin.o gstdspdummy.o gstdspbase.o gstdspvdec.o \
        gstdspvenc.o gstdsph263enc.o gstdspmp4venc.o gstdspjpegenc.o \
        dsp_bridge.o util.o log.o gstdspparse.o async_queue.o gstdsph264enc.o \
-       tidsp.a
+       gstdspadec.o tidsp.a
 $(gst_plugin): override CFLAGS += $(GST_CFLAGS) -fPIC \
        -D VERSION='"$(version)"' -D DSPDIR='"$(dspdir)"'
 $(gst_plugin): override LIBS += $(GST_LIBS)
diff --git a/gstdspadec.c b/gstdspadec.c
new file mode 100644
index 0000000..98ddcb6
--- /dev/null
+++ b/gstdspadec.c
@@ -0,0 +1,280 @@
+/*
+ * Copyright (C) 2010 Víctor M. Jáquez Leal
+ *
+ * Author: Víctor M. Jáquez Leal <vjaq...@igalia.com>
+ *
+ * This file may be used under the terms of the GNU Lesser General Public
+ * License version 2.1, a copy of which is found in LICENSE included in the
+ * packaging of this file.
+ */
+
+#include "gstdspadec.h"
+
+#include "log.h"
+
+#define GST_CAT_DEFAULT gstdsp_debug
+
+static GstDspBaseClass *parent_class;
+
+static inline GstCaps *
+generate_sink_template(void)
+{
+       GstCaps *caps;
+       GstStructure *struc;
+
+       caps = gst_caps_new_empty();
+
+       struc = gst_structure_new("audio/mpeg",
+                               "mpegversion", GST_TYPE_INT_RANGE, 1, 4,
+                               NULL);
+
+       gst_caps_append_structure(caps, struc);
+
+       return caps;
+}
+
+static inline GstCaps *
+generate_src_template(void)
+{
+       GstCaps *caps;
+       GstStructure *struc;
+
+       caps = gst_caps_new_empty();
+
+       struc = gst_structure_new("audio/x-raw-int",
+                               "endianness", G_TYPE_INT, G_BYTE_ORDER,
+                               "signed", G_TYPE_BOOLEAN, TRUE,
+                               "width", G_TYPE_INT, 16,
+                               "depth", G_TYPE_INT, 16,
+                               "rate", GST_TYPE_INT_RANGE, 8000, 96000,
+                               "channels", GST_TYPE_INT_RANGE, 1, 8,
+                               NULL);
+
+       gst_caps_append_structure(caps, struc);
+
+       return caps;
+}
+
+static void *
+create_node(GstDspBase *base)
+{
+       GstDspADec *self;
+       struct td_codec *codec;
+       int dsp_handle;
+       struct dsp_node *node;
+
+       const struct dsp_uuid usn_uuid = { 0x79A3C8B3, 0x95F2, 0x403F, 0x9A, 0x4B,
+               { 0xCF, 0x80, 0x57, 0x73, 0x05, 0x41 } };
+
+       self = GST_DSP_ADEC(base);
+       dsp_handle = base->dsp_handle;
+
+       if (!gstdsp_register(dsp_handle, &usn_uuid, DSP_DCD_LIBRARYTYPE, "usn.dll64P")) {
+               pr_err(self, "failed to register usn node library");
+               return NULL;
+       }
+
+       codec = base->codec;
+       if (!codec) {
+               pr_err(self, "unknown algorithm");
+               return NULL;
+       }
+
+       pr_info(base, "algo=%s", codec->filename);
+
+       if (!gstdsp_register(dsp_handle, codec->uuid, DSP_DCD_LIBRARYTYPE, codec->filename)) {
+               pr_err(self, "failed to register algo node library");
+               return NULL;
+       }
+
+       if (!gstdsp_register(dsp_handle, codec->uuid, DSP_DCD_NODETYPE, codec->filename)) {
+               pr_err(self, "failed to register algo node");
+               return NULL;
+       }
+
+       {
+               struct dsp_node_attr_in attrs = {
+                       .cb = sizeof(attrs),
+                       .priority = 10,
+                       .timeout = 10000,
+               };
+               void *arg_data;
+
+               codec->create_args(base, &attrs.profile_id, &arg_data);
+               if (!dsp_node_allocate(dsp_handle, base->proc, codec->uuid, arg_data, &attrs, &node)) {
+                       pr_err(self, "dsp node allocate failed");
+                       free(arg_data);
+                       return NULL;
+               }
+               free(arg_data);
+       }
+
+       if (!dsp_node_create(dsp_handle, node)) {
+               pr_err(self, "dsp node create failed");
+               dsp_node_free(dsp_handle, node);
+               return NULL;
+       }
+
+       pr_info(self, "dsp node created");
+
+       if (codec->send_params)
+               codec->send_params(base, node);
+
+       if (codec->setup_params)
+               codec->setup_params(base);
+
+       base->flush_buffer = codec->flush_buffer;
+
+       return node;
+}
+
+static inline void
+configure_caps(GstDspADec *self,
+              GstCaps *in,
+              GstCaps *out)
+{
+       GstDspBase *base;
+       GstStructure *out_struc, *in_struc;
+       uint channels;
+
+
+       base = GST_DSP_BASE(self);
+
+       in_struc = gst_caps_get_structure(in, 0);
+
+       out_struc = gst_structure_new("audio/x-raw-int",
+                               "endianness", G_TYPE_INT, G_BYTE_ORDER,
+                               "signed", G_TYPE_BOOLEAN, TRUE,
+                               "width", G_TYPE_INT, 16,
+                               "depth", G_TYPE_INT, 16,
+                               NULL);
+
+       if (gst_structure_get_int(in_struc, "channels", &channels))
+               gst_structure_set(out_struc, "channels", G_TYPE_INT, channels, NULL);
+
+       if (gst_structure_get_int(in_struc, "rate", &self->samplerate))
+               gst_structure_set(out_struc, "rate", G_TYPE_INT, self->samplerate, NULL);
+
+       if (base->alg == GSTDSP_AACDEC) {
+               const char *fmt;
+               gst_structure_get_boolean(in_struc, "framed", &self->packetised);
+               fmt = gst_structure_get_string(in_struc, "stream-format");
+               self->raw = strcmp(fmt, "raw") == 0;
+       }
+
+       base->output_buffer_size = 4 * 1024;
+
+       gst_caps_append_structure(out, out_struc);
+}
+
+static gboolean
+sink_setcaps(GstPad *pad,
+            GstCaps *caps)
+{
+       GstDspADec *self;
+       GstDspBase *base;
+       GstStructure *in_struc;
+       const char *name;
+       GstCaps *out_caps;
+
+       self = GST_DSP_ADEC(GST_PAD_PARENT(pad));
+       base = GST_DSP_BASE(self);
+
+       {
+               gchar *str = gst_caps_to_string(caps);
+               pr_info(self, "sink caps: %s", str);
+               g_free(str);
+       }
+
+       in_struc = gst_caps_get_structure(caps, 0);
+
+       name = gst_structure_get_name(in_struc);
+       if (strcmp(name, "audio/mpeg") == 0) {
+               int version = 1;
+
+               gst_structure_get_int(in_struc, "mpegversion", &version);
+               if (version == 2 || version == 4) {
+                       base->alg = GSTDSP_AACDEC;
+                       base->codec = &td_aacdec_codec;
+               }
+       }
+
+       du_port_alloc_buffers(base->ports[0], 4);
+       du_port_alloc_buffers(base->ports[1], 4);
+
+       out_caps = gst_caps_new_empty();
+       configure_caps(self, caps, out_caps);
+       base->tmp_caps = out_caps;
+
+       return TRUE;
+}
+
+static void
+instance_init(GTypeInstance *instance,
+             gpointer g_class)
+{
+       GstDspBase *base;
+
+       base = GST_DSP_BASE(instance);
+
+       base->use_pad_alloc = TRUE;
+       base->create_node = create_node;
+
+       gst_pad_set_setcaps_function(base->sinkpad, sink_setcaps);
+}
+
+static void
+base_init(gpointer g_class)
+{
+       GstElementClass *element_class;
+       GstPadTemplate *template;
+
+       element_class = GST_ELEMENT_CLASS(g_class);
+
+       gst_element_class_set_details_simple(element_class,
+                                            "DSP audio decoder",
+                                            "Codec/Decoder/Audio",
+                                            "Decodes audio with TI's DSP algorithms",
+                                            "Víctor M. Jáquez Leal");
+
+       template = gst_pad_template_new("src", GST_PAD_SRC,
+                                       GST_PAD_ALWAYS,
+                                       generate_src_template());
+
+       gst_element_class_add_pad_template(element_class, template);
+       gst_object_unref(template);
+
+       template = gst_pad_template_new("sink", GST_PAD_SINK,
+                                       GST_PAD_ALWAYS,
+                                       generate_sink_template());
+
+       gst_element_class_add_pad_template(element_class, template);
+       gst_object_unref(template);
+}
+
+static void
+class_init(gpointer g_class,
+          gpointer class_data)
+{
+       parent_class = g_type_class_peek_parent(g_class);
+}
+
+GType
+gst_dsp_adec_get_type(void)
+{
+       static GType type;
+
+       if (G_UNLIKELY(type == 0)) {
+               GTypeInfo type_info = {
+                       .class_size = sizeof(GstDspADecClass),
+                       .class_init = class_init,
+                       .base_init = base_init,
+                       .instance_size = sizeof(GstDspADec),
+                       .instance_init = instance_init,
+               };
+
+               type = g_type_register_static(GST_DSP_BASE_TYPE, "GstDspADec", &type_info, 0);
+       }
+
+       return type;
+}
diff --git a/gstdspadec.h b/gstdspadec.h
new file mode 100644
index 0000000..5f48463
--- /dev/null
+++ b/gstdspadec.h
@@ -0,0 +1,47 @@
+/*
+ * Copyright (C) 2010 Víctor M. Jáquez Leal
+ *
+ * Author: Víctor M. Jáquez Leal <vjaq...@igalia.com>
+ *
+ * This file may be used under the terms of the GNU Lesser General Public
+ * License version 2.1, a copy of which is found in LICENSE included in the
+ * packaging of this file.
+ */
+
+#ifndef GST_DSP_ADEC_H
+#define GST_DSP_ADEC_H
+
+#include <gst/gst.h>
+
+G_BEGIN_DECLS
+
+#define GST_DSP_ADEC(obj) (GstDspADec *)(obj)
+#define GST_DSP_ADEC_TYPE (gst_dsp_adec_get_type())
+#define GST_DSP_ADEC_CLASS(obj) (GstDspADecClass *)(obj)
+
+typedef struct GstDspADec GstDspADec;
+typedef struct GstDspADecClass GstDspADecClass;
+
+#include "gstdspbase.h"
+
+enum {
+       GSTDSP_AACDEC,
+};
+
+struct GstDspADec {
+       GstDspBase element;
+       guint samplerate;
+       gboolean parametric_stereo;
+       gboolean packetised;
+       gboolean raw;
+};
+
+struct GstDspADecClass {
+       GstDspBaseClass parent_class;
+};
+
+GType gst_dsp_adec_get_type(void);
+
+G_END_DECLS
+
+#endif /* GST_DSP_ADEC_H */
diff --git a/gstdspbase.h b/gstdspbase.h
index 0014635..90fece4 100644
--- a/gstdspbase.h
+++ b/gstdspbase.h
@@ -159,6 +159,8 @@ extern struct td_codec td_mp4venc_codec;
 extern struct td_codec td_jpegenc_codec;
 extern struct td_codec td_h264enc_codec;

+extern struct td_codec td_aacdec_codec;
+
 G_END_DECLS

vmjl

Do you plan to integrate the duration, vpp and this into your next branch?

vmjl