From 95343e3ad9af7eecb2be61b1eb84c0fcb54a48db Mon Sep 17 00:00:00 2001
From: benjamin gaignard <benjamin.gaignard@linaro.org>
Date: Mon, 18 Apr 2011 17:19:00 +0200
Subject: [PATCH] [voaacenc] add new plugin for audio AAC encoder based on vo-aacenc lib
add plugin and unit
---
configure.ac | 8 +
ext/Makefile.am | 7 +
ext/voaacenc/Makefile.am | 18 +
ext/voaacenc/gstvoaac.c | 38 +++
ext/voaacenc/gstvoaacenc.c | 665 +++++++++++++++++++++++++++++++++++++++
ext/voaacenc/gstvoaacenc.h | 81 +++++
tests/check/Makefile.am | 11 +
tests/check/elements/voaacenc.c | 287 +++++++++++++++++
8 files changed, 1115 insertions(+), 0 deletions(-)
create mode 100644 ext/voaacenc/Makefile.am
create mode 100644 ext/voaacenc/gstvoaac.c
create mode 100644 ext/voaacenc/gstvoaacenc.c
create mode 100644 ext/voaacenc/gstvoaacenc.h
create mode 100644 tests/check/elements/voaacenc.c
@@ -569,6 +569,12 @@ AG_GST_CHECK_FEATURE(AMRWB, [amrwb library], amrwbenc, [
AC_SUBST(AMRWB_LIBS))
])
+dnl *** aac-enc ***
+translit(dnm, m, l) AM_CONDITIONAL(USE_VOAACENC, true)
+AG_GST_CHECK_FEATURE(VOAACENC, [vo-aacenc library], vo-aacenc, [
+ AG_GST_PKG_CHECK_MODULES(VOAACENC, vo-aacenc >= 0.1.0)
+])
+
dnl *** apexsink ***
translit(dnm, m, l) AM_CONDITIONAL(USE_APEXSINK, true)
AG_GST_CHECK_FEATURE(APEXSINK, [AirPort Express Wireless sink], apexsink, [
@@ -1585,6 +1591,7 @@ dnl but we still need to set the conditionals
AM_CONDITIONAL(USE_ASSRENDER, false)
AM_CONDITIONAL(USE_AMRWB, false)
+AM_CONDITIONAL(USE_VOAACENC, false)
AM_CONDITIONAL(USE_APEXSINK, false)
AM_CONDITIONAL(USE_BZ2, false)
AM_CONDITIONAL(USE_CDAUDIO, false)
@@ -1820,6 +1827,7 @@ tests/examples/mxf/Makefile
tests/examples/scaletempo/Makefile
tests/icles/Makefile
ext/amrwbenc/Makefile
+ext/voaacenc/Makefile
ext/assrender/Makefile
ext/apexsink/Makefile
ext/bz2/Makefile
@@ -112,6 +112,12 @@ else
FAAD_DIR=
endif
+if USE_VOAACENC
+ VOAACENC_DIR=voaacenc
+else
+ VOAACENC_DIR=
+endif
+
if USE_FLITE
FLITE_DIR=flite
else
@@ -368,6 +374,7 @@ endif
SUBDIRS=\
+ $(VOAACENC_DIR) \
$(ASSRENDER_DIR) \
$(AMRWB_DIR) \
$(APEXSINK_DIR) \
new file mode 100644
@@ -0,0 +1,18 @@
+plugin_LTLIBRARIES = libgstvoaacenc.la
+
+libgstvoaacenc_la_SOURCES = \
+ gstvoaac.c \
+ gstvoaacenc.c
+
+libgstvoaacenc_la_CFLAGS = $(GST_BASE_CFLAGS) $(GST_CFLAGS) $(VOAACENC_CFLAGS)
+libgstvoaacenc_la_LIBADD = -lgstaudio-$(GST_MAJORMINOR) \
+ $(GST_BASE_LIBS) $(GST_LIBS) $(VOAACENC_LIBS)
+libgstvoaacenc_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
+libgstvoaacenc_la_LIBTOOLFLAGS = --tag=disable-static
+
+noinst_HEADERS = \
+ gstvoaacenc.h
+
+presetdir = $(datadir)/gstreamer-$(GST_MAJORMINOR)/presets
+
+EXTRA_DIST = $(preset_DATA)
new file mode 100644
@@ -0,0 +1,38 @@
+/* GStreamer AAC encoder plugin
+ * Copyright (C) 2011 Kan Hu <kan.hu@linaro.org>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include "gstvoaacenc.h"
+
+static gboolean
+plugin_init (GstPlugin * plugin)
+{
+ return gst_element_register (plugin, "voaacenc",
+ GST_RANK_SECONDARY, GST_TYPE_VOAACENC);
+}
+
+
+GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
+ GST_VERSION_MINOR,
+ "voaacenc",
+ "AAC audio encoder",
+ plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN);
new file mode 100644
@@ -0,0 +1,665 @@
+/* GStreamer AAC encoder plugin
+ * Copyright (C) 2011 Kan Hu <kan.hu@linaro.org>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+/**
+ * SECTION:element-voaacenc
+ *
+ * AAC audio encoder based on vo-aacenc library
+ * <ulink url="http://sourceforge.net/projects/opencore-amr/files/vo-aacenc/">vo-aacenc library source file</ulink>.
+ *
+ * <refsect2>
+ * <title>Example launch line</title>
+ * |[
+ * gst-launch filesrc location=abc.wav ! wavparse ! audioresample ! audioconvert ! voaacenc ! filesink location=abc.aac
+ * ]|
+ * </refsect2>
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <string.h>
+
+#include <gst/audio/multichannel.h>
+
+#include "gstvoaacenc.h"
+
+#define VOAAC_ENC_DEFAULT_BITRATE (128000)
+#define VOAAC_ENC_DEFAULT_CHANNELS (2)
+#define VOAAC_ENC_DEFAULT_RATE (44100)
+#define VOAAC_ENC_DEFAULT_OUTPUTFORMAT (0) /* RAW */
+#define VOAAC_ENC_MPEGVERSION (4)
+#define VOAAC_ENC_CODECDATA_LEN (2)
+#define VOAAC_ENC_BITS_PER_SAMPLE (16)
+
+enum
+{
+ PROP_0,
+ PROP_BITRATE
+};
+
+static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/x-raw-int, "
+ "width = (int) 16, "
+ "depth = (int) 16, "
+ "signed = (boolean) TRUE, "
+ "endianness = (int) BYTE_ORDER, "
+ "rate = (int) [8000, 96000], " "channels = (int) [1, 6]")
+ );
+
+static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/mpeg, "
+ "mpegversion = (int) 4, "
+ "rate = (int) [8000, 96000], "
+ "channels = (int) [1, 6], " "stream-format = (string) { adts, raw } ")
+ );
+
+GST_DEBUG_CATEGORY_STATIC (gst_voaacenc_debug);
+#define GST_CAT_DEFAULT gst_voaacenc_debug
+
+static void gst_voaacenc_finalize (GObject * object);
+
+static GstFlowReturn gst_voaacenc_chain (GstPad * pad, GstBuffer * buffer);
+static gboolean gst_voaacenc_setcaps (GstPad * pad, GstCaps * caps);
+static GstStateChangeReturn gst_voaacenc_state_change (GstElement * element,
+ GstStateChange transition);
+static gboolean voaacenc_core_init (GstVoAacEnc * voaacenc);
+static gboolean voaacenc_core_set_parameter (GstVoAacEnc * voaacenc);
+static void voaacenc_core_uninit (GstVoAacEnc * voaacenc);
+static GstCaps *gst_voaacenc_getcaps (GstPad * pad);
+static GstCaps *gst_voaacenc_create_source_pad_caps (GstVoAacEnc * voaacenc);
+static gint voaacenc_get_rate_index (gint rate);
+
+#define VOAAC_ENC_MAX_CHANNELS 6
+
+/* describe the channels position */
+const GstAudioChannelPosition
+ gst_voaacenc_channel_position[][VOAAC_ENC_MAX_CHANNELS] = {
+ { /* 1 ch: Mono */
+ GST_AUDIO_CHANNEL_POSITION_FRONT_MONO},
+ { /* 2 ch: front left + front right (front stereo) */
+ GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
+ GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT},
+ { /* 3 ch: front center + front stereo */
+ GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
+ GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
+ GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT},
+ { /* 4 ch: front center + front stereo + back center */
+ GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
+ GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
+ GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
+ GST_AUDIO_CHANNEL_POSITION_REAR_CENTER},
+ { /* 5 ch: front center + front stereo + back stereo */
+ GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
+ GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
+ GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
+ GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
+ GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT},
+ { /* 6ch: front center + front stereo + back stereo + LFE */
+ GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
+ GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
+ GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
+ GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
+ GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
+ GST_AUDIO_CHANNEL_POSITION_LFE}
+};
+
+static void
+_do_init (GType object_type)
+{
+ const GInterfaceInfo preset_interface_info = {
+ NULL, /* interface init */
+ NULL, /* interface finalize */
+ NULL /* interface_data */
+ };
+
+ g_type_add_interface_static (object_type, GST_TYPE_PRESET,
+ &preset_interface_info);
+
+ GST_DEBUG_CATEGORY_INIT (gst_voaacenc_debug, "voaacenc", 0,
+ "AAC audio encoder");
+}
+
+GST_BOILERPLATE_FULL (GstVoAacEnc, gst_voaacenc, GstElement, GST_TYPE_ELEMENT,
+ _do_init);
+
+static void
+gst_voaacenc_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec)
+{
+ GstVoAacEnc *self = GST_VOAACENC (object);
+
+ switch (prop_id) {
+ case PROP_BITRATE:
+ self->bitrate = g_value_get_int (value);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+ return;
+}
+
+static void
+gst_voaacenc_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec)
+{
+ GstVoAacEnc *self = GST_VOAACENC (object);
+
+ switch (prop_id) {
+ case PROP_BITRATE:
+ g_value_set_int (value, self->bitrate);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+ return;
+}
+
+static void
+gst_voaacenc_base_init (gpointer klass)
+{
+ GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&sink_template));
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&src_template));
+
+ gst_element_class_set_details_simple (element_class, "AAC audio encoder",
+ "Codec/Encoder/Audio", "AAC audio encoder", "Kan Hu <kan.hu@linaro.org>");
+}
+
+static void
+gst_voaacenc_class_init (GstVoAacEncClass * klass)
+{
+ GObjectClass *object_class = G_OBJECT_CLASS (klass);
+ GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+
+ object_class->set_property = GST_DEBUG_FUNCPTR (gst_voaacenc_set_property);
+ object_class->get_property = GST_DEBUG_FUNCPTR (gst_voaacenc_get_property);
+ object_class->finalize = GST_DEBUG_FUNCPTR (gst_voaacenc_finalize);
+
+ g_object_class_install_property (object_class, PROP_BITRATE,
+ g_param_spec_int ("bitrate",
+ "Bitrate",
+ "Target Audio Bitrate",
+ 0, G_MAXINT, VOAAC_ENC_DEFAULT_BITRATE,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ element_class->change_state = GST_DEBUG_FUNCPTR (gst_voaacenc_state_change);
+}
+
+static void
+gst_voaacenc_init (GstVoAacEnc * voaacenc, GstVoAacEncClass * klass)
+{
+ /* create the sink pad */
+ voaacenc->sinkpad = gst_pad_new_from_static_template (&sink_template, "sink");
+ gst_pad_set_setcaps_function (voaacenc->sinkpad,
+ GST_DEBUG_FUNCPTR (gst_voaacenc_setcaps));
+ gst_pad_set_getcaps_function (voaacenc->sinkpad,
+ GST_DEBUG_FUNCPTR (gst_voaacenc_getcaps));
+ gst_pad_set_chain_function (voaacenc->sinkpad,
+ GST_DEBUG_FUNCPTR (gst_voaacenc_chain));
+ gst_element_add_pad (GST_ELEMENT (voaacenc), voaacenc->sinkpad);
+
+ /* create the src pad */
+ voaacenc->srcpad = gst_pad_new_from_static_template (&src_template, "src");
+ gst_pad_use_fixed_caps (voaacenc->srcpad);
+ gst_element_add_pad (GST_ELEMENT (voaacenc), voaacenc->srcpad);
+
+ voaacenc->adapter = gst_adapter_new ();
+
+ voaacenc->bitrate = VOAAC_ENC_DEFAULT_BITRATE;
+ voaacenc->rate = VOAAC_ENC_DEFAULT_RATE;
+ voaacenc->channels = VOAAC_ENC_DEFAULT_CHANNELS;
+ voaacenc->output_format = VOAAC_ENC_DEFAULT_OUTPUTFORMAT;
+
+ /* init rest */
+ voaacenc->handle = NULL;
+ voaacenc->sinkcaps = NULL;
+}
+
+static void
+gst_voaacenc_finalize (GObject * object)
+{
+ GstVoAacEnc *voaacenc;
+
+ voaacenc = GST_VOAACENC (object);
+
+ if (voaacenc->sinkcaps) {
+ gst_caps_unref (voaacenc->sinkcaps);
+ voaacenc->sinkcaps = NULL;
+ }
+
+ g_object_unref (G_OBJECT (voaacenc->adapter));
+ voaacenc->adapter = NULL;
+
+ G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+/* check downstream caps to configure format */
+static void
+gst_voaacenc_negotiate (GstVoAacEnc * voaacenc)
+{
+ GstCaps *caps;
+
+ caps = gst_pad_get_allowed_caps (voaacenc->srcpad);
+
+ GST_DEBUG_OBJECT (voaacenc, "allowed caps: %" GST_PTR_FORMAT, caps);
+
+ if (caps && gst_caps_get_size (caps) > 0) {
+ GstStructure *s = gst_caps_get_structure (caps, 0);
+ const gchar *str = NULL;
+
+ if ((str = gst_structure_get_string (s, "stream-format"))) {
+ if (strcmp (str, "adts") == 0) {
+ GST_DEBUG_OBJECT (voaacenc, "use ADTS format for output");
+ voaacenc->output_format = 1;
+ } else if (strcmp (str, "raw") == 0) {
+ GST_DEBUG_OBJECT (voaacenc, "use RAW format for output");
+ voaacenc->output_format = 0;
+ } else {
+ GST_DEBUG_OBJECT (voaacenc, "unknown stream-format: %s", str);
+ voaacenc->output_format = 0;
+ }
+ }
+ }
+
+ if (caps)
+ gst_caps_unref (caps);
+
+}
+
+
+static GstCaps *
+gst_voaacenc_generate_sink_caps (void)
+{
+ GstCaps *caps = gst_caps_new_empty ();
+ gint i, c;
+
+ for (i = 0; i < VOAAC_ENC_MAX_CHANNELS; i++) {
+ GValue chanpos = { 0 };
+ GValue pos = { 0 };
+ GstStructure *structure;
+
+ g_value_init (&chanpos, GST_TYPE_ARRAY);
+ g_value_init (&pos, GST_TYPE_AUDIO_CHANNEL_POSITION);
+
+ for (c = 0; c <= i; c++) {
+ g_value_set_enum (&pos, gst_voaacenc_channel_position[i][c]);
+ gst_value_array_append_value (&chanpos, &pos);
+ }
+
+ g_value_unset (&pos);
+
+ structure = gst_structure_new ("audio/x-raw-int",
+ "width", G_TYPE_INT, 16,
+ "depth", G_TYPE_INT, 16,
+ "signed", G_TYPE_BOOLEAN, TRUE,
+ "endianness", G_TYPE_INT, G_BYTE_ORDER,
+ "rate", GST_TYPE_INT_RANGE, 8000, 96000, "channels", G_TYPE_INT, i + 1);
+
+ gst_structure_set_value (structure, "channel-positions", &chanpos);
+ g_value_unset (&chanpos);
+
+ gst_caps_append_structure (caps, structure);
+ }
+
+ return caps;
+}
+
+
+static GstCaps *
+gst_voaacenc_getcaps (GstPad * pad)
+{
+ GstVoAacEnc *voaacenc = GST_VOAACENC (GST_PAD_PARENT (pad));
+
+ if (voaacenc->sinkcaps == NULL) {
+ voaacenc->sinkcaps = gst_voaacenc_generate_sink_caps ();
+ }
+
+ GST_DEBUG_OBJECT (voaacenc, "generated sink caps: %" GST_PTR_FORMAT,
+ voaacenc->sinkcaps);
+
+ return gst_caps_ref (voaacenc->sinkcaps);
+}
+
+
+static gboolean
+gst_voaacenc_setcaps (GstPad * pad, GstCaps * caps)
+{
+ gboolean ret = FALSE;
+ GstStructure *structure;
+ GstVoAacEnc *voaacenc;
+ GstCaps *src_caps;
+
+ voaacenc = GST_VOAACENC (GST_PAD_PARENT (pad));
+
+ structure = gst_caps_get_structure (caps, 0);
+
+ /* get channel count */
+ gst_structure_get_int (structure, "channels", &voaacenc->channels);
+ gst_structure_get_int (structure, "rate", &voaacenc->rate);
+
+ /* precalc duration as it's constant now */
+ voaacenc->duration =
+ gst_util_uint64_scale_int (1024, GST_SECOND, voaacenc->rate);
+ voaacenc->inbuf_size = voaacenc->channels * 2 * 1024;
+
+ gst_voaacenc_negotiate (voaacenc);
+
+ /* create reverse caps */
+ src_caps = gst_voaacenc_create_source_pad_caps (voaacenc);
+
+ if (src_caps) {
+ gst_pad_set_caps (voaacenc->srcpad, src_caps);
+ gst_caps_unref (src_caps);
+ ret = voaacenc_core_set_parameter (voaacenc);
+ }
+ return ret;
+}
+
+static GstFlowReturn
+gst_voaacenc_chain (GstPad * pad, GstBuffer * buffer)
+{
+ GstVoAacEnc *voaacenc;
+ GstFlowReturn ret;
+ guint64 timestamp, distance = 0;
+
+ voaacenc = GST_VOAACENC (GST_PAD_PARENT (pad));
+
+ g_return_val_if_fail (voaacenc->handle, GST_FLOW_WRONG_STATE);
+
+ if (voaacenc->rate == 0 || voaacenc->channels == 0)
+ goto not_negotiated;
+
+ /* discontinuity clears adapter, FIXME, maybe we can set some
+ * encoder flag to mask the discont. */
+ if (GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DISCONT)) {
+ gst_adapter_clear (voaacenc->adapter);
+ voaacenc->ts = 0;
+ voaacenc->discont = TRUE;
+ }
+
+ ret = GST_FLOW_OK;
+ gst_adapter_push (voaacenc->adapter, buffer);
+
+ /* Collect samples until we have enough for an output frame */
+ while (gst_adapter_available (voaacenc->adapter) >= voaacenc->inbuf_size) {
+ GstBuffer *out;
+ guint8 *data;
+ VO_CODECBUFFER input = { 0 }
+ , output = {
+ 0};
+ VO_AUDIO_OUTPUTINFO output_info = { {0}
+ };
+
+
+ /* max size */
+ if ((ret =
+ gst_pad_alloc_buffer_and_set_caps (voaacenc->srcpad, 0,
+ voaacenc->inbuf_size, GST_PAD_CAPS (voaacenc->srcpad),
+ &out)) != GST_FLOW_OK) {
+ return ret;
+ }
+
+ output.Buffer = GST_BUFFER_DATA (out);
+ output.Length = voaacenc->inbuf_size;
+
+ if (voaacenc->discont) {
+ GST_BUFFER_FLAG_SET (out, GST_BUFFER_FLAG_DISCONT);
+ voaacenc->discont = FALSE;
+ }
+
+ data =
+ (guint8 *) gst_adapter_peek (voaacenc->adapter, voaacenc->inbuf_size);
+ input.Buffer = data;
+ input.Length = voaacenc->inbuf_size;
+ voaacenc->codec_api.SetInputData (voaacenc->handle, &input);
+
+ /* encode */
+ if (voaacenc->codec_api.GetOutputData (voaacenc->handle, &output,
+ &output_info) != VO_ERR_NONE) {
+ gst_buffer_unref (out);
+ return GST_FLOW_ERROR;
+ }
+
+ /* get timestamp from adapter */
+ timestamp = gst_adapter_prev_timestamp (voaacenc->adapter, &distance);
+
+ if (G_LIKELY (GST_CLOCK_TIME_IS_VALID (timestamp))) {
+ GST_BUFFER_TIMESTAMP (out) =
+ timestamp +
+ GST_FRAMES_TO_CLOCK_TIME (distance / voaacenc->channels /
+ VOAAC_ENC_BITS_PER_SAMPLE, voaacenc->rate);
+ }
+
+ GST_BUFFER_DURATION (out) =
+ GST_FRAMES_TO_CLOCK_TIME (voaacenc->inbuf_size / voaacenc->channels /
+ VOAAC_ENC_BITS_PER_SAMPLE, voaacenc->rate);
+
+ voaacenc->ts = GST_BUFFER_TIMESTAMP (out) + GST_BUFFER_DURATION (out);
+
+ GST_LOG_OBJECT (voaacenc, "Pushing out buffer time: %" GST_TIME_FORMAT
+ " duration: %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (out)),
+ GST_TIME_ARGS (GST_BUFFER_DURATION (out)));
+
+ GST_BUFFER_SIZE (out) = output.Length;
+
+ /* flush the among of data we have peek */
+ gst_adapter_flush (voaacenc->adapter, voaacenc->inbuf_size);
+
+ /* play */
+ if ((ret = gst_pad_push (voaacenc->srcpad, out)) != GST_FLOW_OK)
+ break;
+ }
+ return ret;
+
+ /* ERRORS */
+not_negotiated:
+ {
+ GST_ELEMENT_ERROR (voaacenc, STREAM, TYPE_NOT_FOUND,
+ (NULL), ("unknown type"));
+ return GST_FLOW_NOT_NEGOTIATED;
+ }
+}
+
+static GstStateChangeReturn
+gst_voaacenc_state_change (GstElement * element, GstStateChange transition)
+{
+ GstVoAacEnc *voaacenc;
+ GstStateChangeReturn ret;
+
+ voaacenc = GST_VOAACENC (element);
+
+ switch (transition) {
+ case GST_STATE_CHANGE_NULL_TO_READY:
+ if (voaacenc_core_init (voaacenc) == FALSE)
+ return GST_STATE_CHANGE_FAILURE;
+ break;
+ case GST_STATE_CHANGE_READY_TO_PAUSED:
+ voaacenc->rate = 0;
+ voaacenc->channels = 0;
+ voaacenc->ts = 0;
+ voaacenc->discont = FALSE;
+ gst_adapter_clear (voaacenc->adapter);
+ break;
+ default:
+ break;
+ }
+
+ ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
+
+ switch (transition) {
+ case GST_STATE_CHANGE_READY_TO_NULL:
+ voaacenc_core_uninit (voaacenc);
+ gst_adapter_clear (voaacenc->adapter);
+ break;
+ default:
+ break;
+ }
+ return ret;
+}
+
+static GstCaps *
+gst_voaacenc_create_source_pad_caps (GstVoAacEnc * voaacenc)
+{
+ GstCaps *caps = NULL;
+ GstBuffer *codec_data;
+ gint index;
+
+ if ((index = voaacenc_get_rate_index (voaacenc->rate)) >= 0) {
+
+ caps = gst_caps_new_simple ("audio/mpeg",
+ "mpegversion", G_TYPE_INT, VOAAC_ENC_MPEGVERSION,
+ "channels", G_TYPE_INT, voaacenc->channels,
+ "rate", G_TYPE_INT, voaacenc->rate,
+ "stream-format", G_TYPE_STRING,
+ (voaacenc->output_format ? "adts" : "raw")
+ , NULL);
+
+ if (!voaacenc->output_format) {
+ codec_data = gst_buffer_new_and_alloc (VOAAC_ENC_CODECDATA_LEN);
+
+ GST_BUFFER_DATA (codec_data)[0] = ((0x02 << 3) | (index >> 1));
+ GST_BUFFER_DATA (codec_data)[1] =
+ ((index & 0x01) << 7) | (voaacenc->channels << 3);
+
+ gst_caps_set_simple (caps, "codec_data", GST_TYPE_BUFFER, codec_data,
+ NULL);
+
+ gst_buffer_unref (codec_data);
+ }
+ }
+
+ return caps;
+}
+
+
+
+static VO_U32
+voaacenc_core_mem_alloc (VO_S32 uID, VO_MEM_INFO * pMemInfo)
+{
+ if (!pMemInfo)
+ return VO_ERR_INVALID_ARG;
+
+ pMemInfo->VBuffer = g_malloc (pMemInfo->Size);
+ return 0;
+}
+
+static VO_U32
+voaacenc_core_mem_free (VO_S32 uID, VO_PTR pMem)
+{
+ g_free (pMem);
+ return 0;
+}
+
+static VO_U32
+voaacenc_core_mem_set (VO_S32 uID, VO_PTR pBuff, VO_U8 uValue, VO_U32 uSize)
+{
+ memset (pBuff, uValue, uSize);
+ return 0;
+}
+
+static VO_U32
+voaacenc_core_mem_copy (VO_S32 uID, VO_PTR pDest, VO_PTR pSource, VO_U32 uSize)
+{
+ memcpy (pDest, pSource, uSize);
+ return 0;
+}
+
+static VO_U32
+voaacenc_core_mem_check (VO_S32 uID, VO_PTR pBuffer, VO_U32 uSize)
+{
+ return 0;
+}
+
+static gboolean
+voaacenc_core_init (GstVoAacEnc * voaacenc)
+{
+ VO_CODEC_INIT_USERDATA user_data = { 0 };
+ voGetAACEncAPI (&voaacenc->codec_api);
+
+ voaacenc->mem_operator.Alloc = voaacenc_core_mem_alloc;
+ voaacenc->mem_operator.Copy = voaacenc_core_mem_copy;
+ voaacenc->mem_operator.Free = voaacenc_core_mem_free;
+ voaacenc->mem_operator.Set = voaacenc_core_mem_set;
+ voaacenc->mem_operator.Check = voaacenc_core_mem_check;
+ user_data.memflag = VO_IMF_USERMEMOPERATOR;
+ user_data.memData = &voaacenc->mem_operator;
+ voaacenc->codec_api.Init (&voaacenc->handle, VO_AUDIO_CodingAAC, &user_data);
+
+ if (voaacenc->handle == NULL) {
+ return FALSE;
+ }
+ return TRUE;
+
+}
+
+static gboolean
+voaacenc_core_set_parameter (GstVoAacEnc * voaacenc)
+{
+ AACENC_PARAM params = { 0 };
+ params.sampleRate = voaacenc->rate;
+ params.bitRate = voaacenc->bitrate;
+ params.nChannels = voaacenc->channels;
+ if (voaacenc->output_format) {
+ params.adtsUsed = 1;
+ } else {
+ params.adtsUsed = 0;
+ }
+ if (voaacenc->codec_api.SetParam (voaacenc->handle, VO_PID_AAC_ENCPARAM,
+ ¶ms) != VO_ERR_NONE) {
+ return FALSE;
+ }
+ return TRUE;
+}
+
+static void
+voaacenc_core_uninit (GstVoAacEnc * voaacenc)
+{
+ if (voaacenc->handle) {
+ voaacenc->codec_api.Uninit (voaacenc->handle);
+ voaacenc->handle = NULL;
+ }
+}
+
+static gint
+voaacenc_get_rate_index (gint rate)
+{
+ static const gint rate_table[] = {
+ 96000, 88200, 64000, 48000, 44100, 32000,
+ 24000, 22050, 16000, 12000, 11025, 8000
+ };
+ gint i;
+ for (i = 0; i < G_N_ELEMENTS (rate_table); ++i) {
+ if (rate == rate_table[i]) {
+ return i;
+ }
+ }
+ return -1;
+}
new file mode 100644
@@ -0,0 +1,81 @@
+/* GStreamer AAC encoder plugin
+ * Copyright (C) 2011 Kan Hu <kan.hu@linaro.org>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifndef __GST_VOAACENC_H__
+#define __GST_VOAACENC_H__
+
+#include <gst/gst.h>
+#include <vo-aacenc/voAAC.h>
+#include <vo-aacenc/cmnMemory.h>
+
+#include <gst/base/gstadapter.h>
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_VOAACENC \
+ (gst_voaacenc_get_type())
+#define GST_VOAACENC(obj) \
+ (G_TYPE_CHECK_INSTANCE_CAST((obj), GST_TYPE_VOAACENC, GstVoAacEnc))
+#define GST_VOAACENC_CLASS(klass) \
+ (G_TYPE_CHECK_CLASS_CAST((klass), GST_TYPE_VOAACENC, GstVoAacEncClass))
+#define GST_IS_VOAACENC(obj) \
+ (G_TYPE_CHECK_INSTANCE_TYPE((obj), GST_TYPE_VOAACENC))
+#define GST_IS_VOAACENC_CLASS(klass) \
+ (G_TYPE_CHECK_CLASS_TYPE((klass), GST_TYPE_VOAACENC))
+
+typedef struct _GstVoAacEnc GstVoAacEnc;
+typedef struct _GstVoAacEncClass GstVoAacEncClass;
+
+struct _GstVoAacEnc {
+ GstElement element;
+
+ /* pads */
+ GstPad *sinkpad, *srcpad;
+ GstCaps *sinkcaps;
+ guint64 ts;
+ gboolean discont;
+
+ GstAdapter *adapter;
+
+
+ /* desired bitrate */
+ gint bitrate;
+ gint channels;
+ gint rate;
+ gint output_format;
+ gint duration;
+
+ gint inbuf_size;
+
+ /* library handle */
+ VO_AUDIO_CODECAPI codec_api;
+ VO_HANDLE handle;
+ VO_MEM_OPERATOR mem_operator;
+
+};
+
+struct _GstVoAacEncClass {
+ GstElementClass parent_class;
+};
+
+GType gst_voaacenc_get_type (void);
+
+G_END_DECLS
+
+#endif /* __GST_VOAACENC_H__ */
@@ -46,6 +46,12 @@ else
check_faad =
endif
+if USE_VOAACENC
+check_voaacenc = elements/voaacenc
+else
+check_voaacenc =
+endif
+
if USE_EXIF
check_jifmux = elements/jifmux
else
@@ -143,6 +149,7 @@ check_PROGRAMS = \
$(check_assrender) \
$(check_faac) \
$(check_faad) \
+ $(check_voaacenc) \
$(check_mpeg2enc) \
$(check_mplex) \
$(check_ofa) \
@@ -184,6 +191,10 @@ AM_CFLAGS = $(GST_CHECK_CFLAGS) $(GST_OPTION_CFLAGS) \
-UG_DISABLE_ASSERT -UG_DISABLE_CAST_CHECKS
LDADD = $(GST_CHECK_LIBS)
+elements_voaacenc_LDADD = \
+ $(GST_BASE_LIBS) $(GST_LIBS) $(LDADD) \
+ -lgstaudio-@GST_MAJORMINOR@
+
elements_camerabin_CFLAGS = \
$(GST_PLUGINS_BAD_CFLAGS) $(GST_PLUGINS_BASE_CFLAGS) \
$(GST_BASE_CFLAGS) $(GST_CFLAGS) $(AM_CFLAGS) -DGST_USE_UNSTABLE_API
new file mode 100644
@@ -0,0 +1,287 @@
+/* GStreamer
+ *
+ * unit test for voaacenc
+ *
+ * Copyright (C) <2009> Mark Nauwelaerts <mnauw@users.sf.net>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#include <unistd.h>
+
+#include <gst/check/gstcheck.h>
+#include <gst/audio/multichannel.h>
+
+/* For ease of programming we use globals to keep refs for our floating
+ * src and sink pads we create; otherwise we always have to do get_pad,
+ * get_peer, and then remove references in every test function */
+static GstPad *mysrcpad, *mysinkpad;
+
+#define AUDIO_CAPS_STRING "audio/x-raw-int, " \
+ "rate = (int) 48000, " \
+ "channels = (int) 2, " \
+ "width = (int) 16, " \
+ "depth = (int) 16, " \
+ "signed = (boolean) true, " \
+ "endianness = (int) BYTE_ORDER "
+
+#define AAC_RAW_CAPS_STRING "audio/mpeg, " \
+ "mpegversion = (int) 4, " \
+ "rate = (int) 48000, " \
+ "channels = (int) 2, " \
+ "stream-format = \"raw\""
+
+#define AAC_ADTS_CAPS_STRING "audio/mpeg, " \
+ "mpegversion = (int) 4, " \
+ "rate = (int) 48000, " \
+ "channels = (int) 2, " \
+ "stream-format = \"adts\""
+
+
+static GstStaticPadTemplate sinktemplate_adts = GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS (AAC_ADTS_CAPS_STRING));
+
+static GstStaticPadTemplate sinktemplate_raw = GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS (AAC_RAW_CAPS_STRING));
+
+
+static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS (AUDIO_CAPS_STRING));
+
+
+static GstElement *
+setup_voaacenc (gboolean adts)
+{
+ GstElement *voaacenc;
+
+ GST_DEBUG ("setup_voaacenc");
+ voaacenc = gst_check_setup_element ("voaacenc");
+ mysrcpad = gst_check_setup_src_pad (voaacenc, &srctemplate, NULL);
+
+ if (adts)
+ mysinkpad = gst_check_setup_sink_pad (voaacenc, &sinktemplate_adts, NULL);
+ else
+ mysinkpad = gst_check_setup_sink_pad (voaacenc, &sinktemplate_raw, NULL);
+
+ gst_pad_set_active (mysrcpad, TRUE);
+ gst_pad_set_active (mysinkpad, TRUE);
+
+ return voaacenc;
+}
+
+static void
+cleanup_voaacenc (GstElement * voaacenc)
+{
+ GST_DEBUG ("cleanup_aacenc");
+ gst_element_set_state (voaacenc, GST_STATE_NULL);
+
+ gst_pad_set_active (mysrcpad, FALSE);
+ gst_pad_set_active (mysinkpad, FALSE);
+ gst_check_teardown_src_pad (voaacenc);
+ gst_check_teardown_sink_pad (voaacenc);
+ gst_check_teardown_element (voaacenc);
+}
+
+static void
+set_channel_positions (GstCaps * caps, int channels,
+ GstAudioChannelPosition * channelpositions)
+{
+ GValue chanpos = { 0 };
+ GValue pos = { 0 };
+ GstStructure *structure = gst_caps_get_structure (caps, 0);
+ int c;
+
+ g_value_init (&chanpos, GST_TYPE_ARRAY);
+ g_value_init (&pos, GST_TYPE_AUDIO_CHANNEL_POSITION);
+
+ for (c = 0; c < channels; c++) {
+ g_value_set_enum (&pos, channelpositions[c]);
+ gst_value_array_append_value (&chanpos, &pos);
+ }
+ g_value_unset (&pos);
+
+ gst_structure_set_value (structure, "channel-positions", &chanpos);
+ g_value_unset (&chanpos);
+}
+
+static void
+do_test (gboolean adts)
+{
+ GstElement *voaacenc;
+ GstBuffer *inbuffer, *outbuffer;
+ GstCaps *caps;
+ gint i, num_buffers;
+ const gint nbuffers = 10;
+ GstAudioChannelPosition channel_position_layout[2] =
+ { GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
+ GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT
+ };
+
+ voaacenc = setup_voaacenc (adts);
+ fail_unless (gst_element_set_state (voaacenc,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+ "could not set to playing");
+
+ /* corresponds to audio buffer mentioned in the caps */
+ inbuffer = gst_buffer_new_and_alloc (1024 * nbuffers * 2 * 2);
+ /* makes valgrind's memcheck happier */
+ memset (GST_BUFFER_DATA (inbuffer), 0, GST_BUFFER_SIZE (inbuffer));
+ caps = gst_caps_from_string (AUDIO_CAPS_STRING);
+
+ set_channel_positions (caps, 2, channel_position_layout);
+
+ gst_buffer_set_caps (inbuffer, caps);
+ gst_caps_unref (caps);
+ GST_BUFFER_TIMESTAMP (inbuffer) = 0;
+ ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
+ fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
+
+ /* send eos to have all flushed if needed */
+ fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()) == TRUE);
+
+ num_buffers = g_list_length (buffers);
+ fail_unless_equals_int (num_buffers, nbuffers);
+
+ /* clean up buffers */
+ for (i = 0; i < num_buffers; ++i) {
+ gint size, header = 0, id;
+ guint8 *data;
+
+ outbuffer = GST_BUFFER (buffers->data);
+ fail_if (outbuffer == NULL);
+
+ data = GST_BUFFER_DATA (outbuffer);
+ size = GST_BUFFER_SIZE (outbuffer);
+
+ if (adts) {
+ gboolean protection;
+ gint k;
+
+ fail_if (size < 7);
+ protection = !(data[1] & 0x1);
+ /* expect only 1 raw data block */
+ k = (data[6] & 0x3) + 1;
+ fail_if (k != 1);
+
+ header = 7;
+ if (protection)
+ header += (k - 1) * 2 + 2;
+
+ /* check header */
+ k = GST_READ_UINT16_BE (data) & 0xFFF6;
+ /* sync */
+ fail_unless (k == 0xFFF0);
+ k = data[2];
+ /* profile */
+ fail_unless ((k >> 6) == 0x1);
+ /* rate */
+ fail_unless (((k >> 2) & 0xF) == 0x3);
+ /* channels */
+ fail_unless ((k & 0x1) == 0);
+ k = data[3];
+ fail_unless ((k >> 6) == 0x2);
+
+ } else {
+ GstCaps *caps;
+ GstStructure *s;
+ const GValue *value;
+ GstBuffer *buf;
+ gint k;
+
+ caps = gst_buffer_get_caps (outbuffer);
+ fail_if (caps == NULL);
+ s = gst_caps_get_structure (caps, 0);
+ fail_if (s == NULL);
+ value = gst_structure_get_value (s, "codec_data");
+ fail_if (value == NULL);
+ buf = gst_value_get_buffer (value);
+ fail_if (buf == NULL);
+ data = GST_BUFFER_DATA (buf);
+ size = GST_BUFFER_SIZE (buf);
+ fail_if (size < 2);
+ k = GST_READ_UINT16_BE (data);
+ /* profile, rate, channels */
+ fail_unless ((k & 0xFFF8) == ((0x02 << 11) | (0x3 << 7) | (0x02 << 3)));
+ gst_caps_unref (caps);
+
+ }
+
+ fail_if (size <= header);
+ id = data[header] & (0x7 << 5);
+ /* allow all but ID_END or ID_LFE */
+ fail_if (id == 7 || id == 3);
+
+ buffers = g_list_remove (buffers, outbuffer);
+
+ ASSERT_BUFFER_REFCOUNT (outbuffer, "outbuffer", 1);
+ gst_buffer_unref (outbuffer);
+ outbuffer = NULL;
+ }
+
+ cleanup_voaacenc (voaacenc);
+ g_list_free (buffers);
+ buffers = NULL;
+}
+
+GST_START_TEST (test_adts)
+{
+ do_test (TRUE);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_raw)
+{
+ do_test (FALSE);
+}
+
+GST_END_TEST;
+
+static Suite *
+voaacenc_suite (void)
+{
+ Suite *s = suite_create ("voaacenc");
+ TCase *tc_chain = tcase_create ("general");
+
+ suite_add_tcase (s, tc_chain);
+ tcase_add_test (tc_chain, test_adts);
+ tcase_add_test (tc_chain, test_raw);
+
+ return s;
+}
+
+int
+main (int argc, char **argv)
+{
+ int nf;
+
+ Suite *s = voaacenc_suite ();
+ SRunner *sr = srunner_create (s);
+
+ gst_check_init (&argc, &argv);
+
+ srunner_run_all (sr, CK_NORMAL);
+ nf = srunner_ntests_failed (sr);
+ srunner_free (sr);
+
+ return nf;
+}
--
1.7.0.4