/* GStreamer AAC encoder
* Copyright 2009 Collabora Multimedia,
* Copyright 2009 Nokia Corporation
* @author: Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>.
*
* 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.
*/
/* TODO non-GPL license */
/**
* SECTION:element-nokiaaacenc
* @seealso: nokiaaacdec
*
* nokiaaacenc encodes raw audio to AAC streams.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/gst.h>
#include <gst/audio/audio.h>
#include <string.h>
#include "gstaacenc.h"
GST_DEBUG_CATEGORY_STATIC (aac_enc);
#define GST_CAT_DEFAULT aac_enc
enum
{
AAC_PROFILE_AUTO = 0,
AAC_PROFILE_LC = 2,
AAC_PROFILE_HE = 5
};
#define GST_TYPE_AAC_ENC_PROFILE (gst_aac_enc_profile_get_type ())
static GType
gst_aac_enc_profile_get_type (void)
{
static GType gst_aac_enc_profile_type = 0;
if (!gst_aac_enc_profile_type) {
static GEnumValue gst_aac_enc_profile[] = {
{AAC_PROFILE_AUTO, "Codec selects LC or HE", "AUTO"},
{AAC_PROFILE_LC, "Low complexity profile", "LC"},
{AAC_PROFILE_HE, "High Efficiency", "HE"},
{0, NULL, NULL},
};
gst_aac_enc_profile_type = g_enum_register_static ("GstNokiaAacEncProfile",
gst_aac_enc_profile);
}
return gst_aac_enc_profile_type;
}
#define GST_TYPE_AAC_ENC_OUTPUTFORMAT (gst_aac_enc_outputformat_get_type ())
static GType
gst_aac_enc_outputformat_get_type (void)
{
static GType gst_aac_enc_outputformat_type = 0;
if (!gst_aac_enc_outputformat_type) {
static GEnumValue gst_aac_enc_outputformat[] = {
{RAW, "AAC Raw format", "RAW"},
{USE_ADTS, "Audio Data Transport Stream format", "ADTS"},
{USE_ADIF, "Audio Data Interchange Format", "ADIF"},
{0, NULL, NULL},
};
gst_aac_enc_outputformat_type =
g_enum_register_static ("GstNokiaAacEncOutputFormat",
gst_aac_enc_outputformat);
}
return gst_aac_enc_outputformat_type;
}
enum
{
PROP_0,
PROP_BITRATE,
PROP_PROFILE,
PROP_FORMAT
};
static GstStaticPadTemplate gst_aac_enc_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"endianness = (int) BYTE_ORDER, "
"signed = (bool) TRUE, "
"width = (int) 16, "
"depth = (int) 16, "
"rate = (int) [ 8000, 96000 ], channels = (int) [ 1, 2 ] ")
);
static GstStaticPadTemplate gst_aac_enc_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, 2 ] ")
);
static void gst_aac_enc_base_init (gpointer g_class);
static void gst_aac_enc_class_init (GstAACEncClass * klass);
static void gst_aac_enc_init (GstAACEnc * filter, GstAACEncClass * klass);
static void gst_aac_enc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_aac_enc_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static void gst_aac_enc_finalize (GObject * object);
static void gst_aac_enc_reset (GstAACEnc * enc);
static GstStateChangeReturn gst_aac_enc_change_state (GstElement * element,
GstStateChange transition);
static gboolean gst_aac_enc_sink_setcaps (GstPad * pad, GstCaps * caps);
static GstFlowReturn gst_aac_enc_chain (GstPad * pad, GstBuffer * buffer);
GST_BOILERPLATE (GstNokiaAACEnc, gst_aac_enc, GstElement, GST_TYPE_ELEMENT);
static void
gst_aac_enc_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_set_details_simple (element_class,
"Nokia AAC encoder", "Codec/Encoder/Audio",
"Nokia AAC encoder",
"MCC, Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>");
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_aac_enc_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_aac_enc_sink_template));
}
/* initialize the plugin's class */
static void
gst_aac_enc_class_init (GstAACEncClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
GST_DEBUG_CATEGORY_INIT (aac_enc, "nokiaaacenc", 0, "Nokia AAC encoder");
gobject_class->set_property = gst_aac_enc_set_property;
gobject_class->get_property = gst_aac_enc_get_property;
gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_aac_enc_finalize);
/* properties */
g_object_class_install_property (gobject_class, PROP_BITRATE,
g_param_spec_int ("bitrate", "Bitrate (bps)", "Bitrate in bits/sec",
8 * 1000, 320 * 1000, 128 * 1000,
(GParamFlags) (G_PARAM_READWRITE | G_PARAM_CONSTRUCT)));
g_object_class_install_property (gobject_class, PROP_PROFILE,
g_param_spec_enum ("profile", "Profile",
"MPEG/AAC encoding profile",
GST_TYPE_AAC_ENC_PROFILE, AAC_PROFILE_LC,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
g_object_class_install_property (gobject_class, PROP_FORMAT,
g_param_spec_enum ("output-format", "Output format",
"Format of output frames",
GST_TYPE_AAC_ENC_OUTPUTFORMAT, RAW,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_aac_enc_change_state);
}
static void
gst_aac_enc_init (GstAACEnc * enc, GstAACEncClass * klass)
{
enc->sinkpad =
gst_pad_new_from_static_template (&gst_aac_enc_sink_template, "sink");
gst_pad_set_setcaps_function (enc->sinkpad,
GST_DEBUG_FUNCPTR (gst_aac_enc_sink_setcaps));
gst_pad_set_chain_function (enc->sinkpad,
GST_DEBUG_FUNCPTR (gst_aac_enc_chain));
gst_element_add_pad (GST_ELEMENT (enc), enc->sinkpad);
enc->srcpad =
gst_pad_new_from_static_template (&gst_aac_enc_src_template, "src");
gst_pad_use_fixed_caps (enc->srcpad);
gst_element_add_pad (GST_ELEMENT (enc), enc->srcpad);
#ifndef GST_DISABLE_GST_DEBUG
gst_framed_audio_enc_init (&enc->enc, GST_ELEMENT (enc), aac_enc);
#else
gst_framed_audio_enc_init (&enc->enc, GST_ELEMENT (enc), NULL);
#endif
gst_aac_enc_reset (enc);
}
static void
gst_aac_enc_reset (GstAACEnc * enc)
{
gst_framed_audio_enc_reset (&enc->enc);
if (enc->encoder)
EnAACPlus_Enc_Delete (enc->encoder);
enc->encoder = NULL;
g_free (enc->buffer);
enc->buffer = NULL;
}
static void
gst_aac_enc_finalize (GObject * object)
{
GstAACEnc *enc = (GstAACEnc *) object;
gst_framed_audio_enc_finalize (&enc->enc);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gboolean
gst_aac_enc_setup_encoder (GstAACEnc * enc)
{
AACPLUS_ENC_CONFIG enc_params;
AACPLUS_ENC_MODE mode;
gint rate, channels;
guint maxbitrate;
rate = enc->rate;
channels = enc->channels;
/* only up to 2 channels supported */
enc_params.sampleRate = rate;
enc_params.bitRate = enc->bitrate;
enc_params.nChannels = channels;
enc_params.aac_tools = USE_ALL;
enc_params.pcm_mode = 16;
enc_params.format = enc->format;
/* check, warn and correct if the max bitrate for the given samplerate is
* exceeded. Maximum of 6144 bit for a channel */
maxbitrate =
(guint) (6144.0 * (gdouble) rate / (gdouble) 1024.0 + .5) * channels;
if (enc_params.bitRate > maxbitrate) {
GST_ELEMENT_INFO (enc, RESOURCE, SETTINGS, (NULL),
("bitrate %d exceeds maximum allowed bitrate of %d for samplerate %d "
"and %d channels. Setting bitrate to %d",
enc_params.bitRate, maxbitrate, rate, channels, maxbitrate));
enc_params.bitRate = maxbitrate;
}
/* set up encoder */
if (enc->encoder)
EnAACPlus_Enc_Delete (enc->encoder);
/* only these profiles are really known to and supported by codec */
switch (enc->profile) {
case AAC_PROFILE_LC:
mode = MODE_AACLC;
break;
case AAC_PROFILE_HE:
mode = MODE_EAACPLUS;
break;
case AAC_PROFILE_AUTO:
mode = MODE_AUTO;
break;
default:
mode = MODE_AACLC;
g_assert_not_reached ();
break;
}
enc->encoder = EnAACPlus_Enc_Create (&enc_params, mode);
if (!enc->encoder)
goto setup_failed;
/* query and setup params,
* also set up some buffers for fancy HE */
EnAACPlus_Enc_GetSetParam (enc->encoder, &enc->info);
#define DUMP_FIELD(f) \
GST_DEBUG_OBJECT (enc, "encoder info: " G_STRINGIFY (f) " = %d", enc->info.f);
DUMP_FIELD (InBufSize);
DUMP_FIELD (OutBufSize);
DUMP_FIELD (Frame_Size);
DUMP_FIELD (writeOffset);
DUMP_FIELD (InBufSize);
enc->raw_frame_size = enc->info.Frame_Size;
enc->codec_frame_size = enc->info.OutBufSize;
enc->frame_duration =
GST_FRAMES_TO_CLOCK_TIME (enc->raw_frame_size / enc->channels / 2,
enc->rate);
g_free (enc->buffer);
/* safety margin */
enc->buffer = g_malloc (enc->info.InBufSize * 2);
return TRUE;
/* ERRORS */
setup_failed:
{
GST_ELEMENT_ERROR (enc, LIBRARY, SETTINGS, (NULL), (NULL));
return FALSE;
}
}
static gint
gst_aac_enc_rate_idx (gint rate)
{
static int rates[] = {
96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050, 16000, 12000, 11025,
8000, 7350
};
guint i;
for (i = 0; i < G_N_ELEMENTS (rates); ++i)
if (rates[i] == rate)
return i;
return 0xF;
}
static gboolean
gst_aac_enc_sink_setcaps (GstPad * pad, GstCaps * caps)
{
GstAACEnc *enc;
gboolean ret = TRUE;
GstStructure *s;
GstBuffer *buf = NULL;
gint rate, channels;
enc = GST_AAC_ENC (GST_PAD_PARENT (pad));
/* extract stream properties */
s = gst_caps_get_structure (caps, 0);
if (!s)
goto refuse_caps;
ret = gst_structure_get_int (s, "rate", &rate);
ret &= gst_structure_get_int (s, "channels", &channels);
if (!ret)
goto refuse_caps;
enc->rate = rate;
enc->channels = channels;
/* NOTE:
* - codec only supports LC or HE (= LC + SBR etc)
* - HE has (more) restrictive samplerate/channels/bitrate combination
* - AUTO makes codec select between LC or HE (depending on settings)
*/
gst_aac_enc_setup_encoder (enc);
if (!enc->encoder)
return FALSE;
/* HE iff writeOffset <> 0 iff Frame_Size <> 1024 * 2 * channels */
if (enc->info.writeOffset)
rate /= 2;
/* create codec_data if raw output */
if (enc->format == RAW) {
gint rate_idx;
guint8 *data;
buf = gst_buffer_new_and_alloc (5);
data = GST_BUFFER_DATA (buf);
rate_idx = gst_aac_enc_rate_idx (rate);
GST_DEBUG_OBJECT (enc, "codec_data: profile=%d, sri=%d, channels=%d",
enc->profile, rate_idx, enc->channels);
/* always write LC profile, and use implicit signaling for HE SBR */
data[0] = ((2 & 0x1F) << 3) | ((rate_idx & 0xE) >> 1);
data[1] = ((rate_idx & 0x1) << 7);
if (rate_idx != 0x0F) {
data[1] |= ((channels & 0xF) << 3);
GST_BUFFER_SIZE (buf) = 2;
} else {
gint srate;
srate = rate << 7;
data[1] |= ((srate >> 24) & 0xFF);
data[2] = ((srate >> 16) & 0xFF);
data[3] = ((srate >> 8) & 0xFF);
data[4] = (srate & 0xFF);
data[4] |= ((channels & 0xF) << 3);
GST_BUFFER_SIZE (buf) = 5;
}
}
/* fix some in src template */
caps = gst_caps_copy (gst_pad_get_pad_template_caps (enc->srcpad));
gst_caps_set_simple (caps, "rate", G_TYPE_INT, rate,
"channels", G_TYPE_INT, channels, NULL);
if (buf) {
gst_caps_set_simple (caps, "codec_data", GST_TYPE_BUFFER, buf, NULL);
gst_buffer_unref (buf);
}
ret = gst_pad_set_caps (enc->srcpad, caps);
gst_caps_unref (caps);
return ret;
/* ERRORS */
refuse_caps:
{
GST_WARNING_OBJECT (enc, "refused caps %" GST_PTR_FORMAT, caps);
return FALSE;
}
}
static gint
gst_aac_enc_get_data (GstElement * element, const guint8 * in, guint8 * out,
GstDtxDecision * dtx)
{
GstAACEnc *enc;
gint res;
gint offset;
UWord32 used, encoded;
Word8 *inbuffer;
enc = GST_AAC_ENC_CAST (element);
offset = enc->info.writeOffset;
if (offset) {
memcpy (enc->buffer + offset, in, enc->raw_frame_size);
inbuffer = (Word8 *) enc->buffer;
} else {
inbuffer = (Word8 *) in;
}
res = EnAACPlus_Enc_Encode (enc->encoder, &enc->info, inbuffer, &used,
(UWord8 *) out, &encoded);
if (offset) {
memcpy (enc->buffer, enc->buffer + used, offset);
}
return res == 0 ? encoded : -1;
}
/* set parameters */
#define AUDIO_SAMPLE_RATE ((GST_AAC_ENC (enc->element))->rate)
#define RAW_FRAME_SIZE ((GST_AAC_ENC (enc->element))->raw_frame_size)
/* safe maximum frame size */
#define CODEC_FRAME_SIZE ((GST_AAC_ENC (enc->element))->codec_frame_size)
/* do not set variable frame;
* this will make every frame act as a silence frame and force output */
/* #define CODEC_FRAME_VARIABLE 1 */
#define FRAME_DURATION ((GST_AAC_ENC (enc->element))->frame_duration)
#define codec_get_data(enc, in, out, dtx) \
gst_aac_enc_get_data (enc, in, out, dtx)
/* and include code */
#include "gstframedaudioenc.c"
static GstFlowReturn
gst_aac_enc_chain (GstPad * pad, GstBuffer * buf)
{
GstAACEnc *enc;
enc = GST_AAC_ENC (GST_PAD_PARENT (pad));
if (G_UNLIKELY (enc->encoder == NULL))
goto not_negotiated;
return gst_framed_audio_enc_chain (&enc->enc, buf, enc->srcpad, &enc->cnpad);
/* ERRORS */
not_negotiated:
{
GST_ELEMENT_ERROR (enc, CORE, NEGOTIATION, (NULL),
("format wasn't negotiated before chain function"));
gst_buffer_unref (buf);
return GST_FLOW_NOT_NEGOTIATED;
}
}
static void
gst_aac_enc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAACEnc *enc;
enc = GST_AAC_ENC (object);
switch (prop_id) {
case PROP_BITRATE:
enc->bitrate = g_value_get_int (value);
break;
case PROP_PROFILE:
enc->profile = g_value_get_enum (value);
break;
case PROP_FORMAT:
enc->format = g_value_get_enum (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_aac_enc_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstAACEnc *enc;
enc = GST_AAC_ENC (object);
switch (prop_id) {
case PROP_BITRATE:
g_value_set_int (value, enc->bitrate);
break;
case PROP_PROFILE:
g_value_set_enum (value, enc->profile);
break;
case PROP_FORMAT:
g_value_set_enum (value, enc->format);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstStateChangeReturn
gst_aac_enc_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
GstAACEnc *enc = GST_AAC_ENC (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
if (ret == GST_STATE_CHANGE_FAILURE)
return ret;
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_aac_enc_reset (enc);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
break;
default:
break;
}
return ret;
}
static gboolean
plugin_init (GstPlugin * plugin)
{
if (!gst_element_register (plugin, "nokiaaacenc", GST_RANK_SECONDARY,
GST_TYPE_AAC_ENC))
return FALSE;
return TRUE;
}
/* this is the structure that gst-register looks for
* so keep the name plugin_desc, or you cannot get your plug-in registered */
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"nokiaaacenc",
"Nokia AAC MCC codec",
plugin_init, VERSION, "Proprietary", "gst-nokia-speech", "")
EXPORT_C GstPluginDesc* _GST_PLUGIN_DESC()
{
return &gst_plugin_desc;
}