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gstreamer/omx/gstomxaacenc.c

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/*
* Copyright (C) 2011, Hewlett-Packard Development Company, L.P.
* Author: Sebastian Dröge <sebastian.droege@collabora.co.uk>, Collabora Ltd.
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation
* version 2.1 of the License.
*
* 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/gst.h>
#include "gstomxaacenc.h"
GST_DEBUG_CATEGORY_STATIC (gst_omx_aac_enc_debug_category);
#define GST_CAT_DEFAULT gst_omx_aac_enc_debug_category
/* prototypes */
static void gst_omx_aac_enc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_omx_aac_enc_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static gboolean gst_omx_aac_enc_set_format (GstOMXAudioEnc * enc,
GstOMXPort * port, GstAudioInfo * info);
static GstCaps *gst_omx_aac_enc_get_caps (GstOMXAudioEnc * enc,
GstOMXPort * port, GstAudioInfo * info);
static guint gst_omx_aac_enc_get_num_samples (GstOMXAudioEnc * enc,
GstOMXPort * port, GstAudioInfo * info, GstOMXBuffer * buf);
enum
{
PROP_0,
PROP_BITRATE,
PROP_AAC_TOOLS,
PROP_AAC_ERROR_RESILIENCE_TOOLS
};
#define DEFAULT_BITRATE (128000)
#define DEFAULT_AAC_TOOLS (OMX_AUDIO_AACToolMS | OMX_AUDIO_AACToolIS | OMX_AUDIO_AACToolTNS | OMX_AUDIO_AACToolPNS | OMX_AUDIO_AACToolLTP)
#define DEFAULT_AAC_ER_TOOLS (OMX_AUDIO_AACERNone)
#define GST_TYPE_OMX_AAC_TOOLS (gst_omx_aac_tools_get_type ())
static GType
gst_omx_aac_tools_get_type (void)
{
static gsize id = 0;
static const GFlagsValue values[] = {
{OMX_AUDIO_AACToolMS, "Mid/side joint coding", "ms"},
{OMX_AUDIO_AACToolIS, "Intensity stereo", "is"},
{OMX_AUDIO_AACToolTNS, "Temporal noise shaping", "tns"},
{OMX_AUDIO_AACToolPNS, "Perceptual noise substitution", "pns"},
{OMX_AUDIO_AACToolLTP, "Long term prediction", "ltp"},
{0, NULL, NULL}
};
if (g_once_init_enter (&id)) {
GType tmp = g_flags_register_static ("GstOMXAACTools", values);
g_once_init_leave (&id, tmp);
}
return (GType) id;
}
#define GST_TYPE_OMX_AAC_ER_TOOLS (gst_omx_aac_er_tools_get_type ())
static GType
gst_omx_aac_er_tools_get_type (void)
{
static gsize id = 0;
static const GFlagsValue values[] = {
{OMX_AUDIO_AACERVCB11, "Virtual code books", "vcb11"},
{OMX_AUDIO_AACERRVLC, "Reversible variable length coding", "rvlc"},
{OMX_AUDIO_AACERHCR, "Huffman codeword reordering", "hcr"},
{0, NULL, NULL}
};
if (g_once_init_enter (&id)) {
GType tmp = g_flags_register_static ("GstOMXAACERTools", values);
g_once_init_leave (&id, tmp);
}
return (GType) id;
}
/* class initialization */
#define DEBUG_INIT \
GST_DEBUG_CATEGORY_INIT (gst_omx_aac_enc_debug_category, "omxaacenc", 0, \
"debug category for gst-omx audio encoder base class");
G_DEFINE_TYPE_WITH_CODE (GstOMXAACEnc, gst_omx_aac_enc,
GST_TYPE_OMX_AUDIO_ENC, DEBUG_INIT);
static void
gst_omx_aac_enc_class_init (GstOMXAACEncClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstOMXAudioEncClass *audioenc_class = GST_OMX_AUDIO_ENC_CLASS (klass);
gobject_class->set_property = gst_omx_aac_enc_set_property;
gobject_class->get_property = gst_omx_aac_enc_get_property;
g_object_class_install_property (gobject_class, PROP_BITRATE,
g_param_spec_uint ("bitrate", "Bitrate",
"Bitrate",
0, G_MAXUINT, DEFAULT_BITRATE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
GST_PARAM_MUTABLE_READY));
g_object_class_install_property (gobject_class, PROP_AAC_TOOLS,
g_param_spec_flags ("aac-tools", "AAC Tools",
"Allowed AAC tools",
GST_TYPE_OMX_AAC_TOOLS,
DEFAULT_AAC_TOOLS,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
GST_PARAM_MUTABLE_READY));
g_object_class_install_property (gobject_class,
PROP_AAC_ERROR_RESILIENCE_TOOLS,
g_param_spec_flags ("aac-error-resilience-tools",
"AAC Error Resilience Tools", "Allowed AAC error resilience tools",
GST_TYPE_OMX_AAC_ER_TOOLS, DEFAULT_AAC_ER_TOOLS,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
GST_PARAM_MUTABLE_READY));
audioenc_class->set_format = GST_DEBUG_FUNCPTR (gst_omx_aac_enc_set_format);
audioenc_class->get_caps = GST_DEBUG_FUNCPTR (gst_omx_aac_enc_get_caps);
audioenc_class->get_num_samples =
GST_DEBUG_FUNCPTR (gst_omx_aac_enc_get_num_samples);
audioenc_class->cdata.default_src_template_caps = "audio/mpeg, "
"mpegversion=(int){2, 4}, "
"stream-format=(string){raw, adts, adif, loas, latm}";
gst_element_class_set_static_metadata (element_class,
"OpenMAX AAC Audio Encoder",
"Codec/Encoder/Audio",
"Encode AAC audio streams",
"Sebastian Dröge <sebastian.droege@collabora.co.uk>");
gst_omx_set_default_role (&audioenc_class->cdata, "audio_encoder.aac");
}
static void
gst_omx_aac_enc_init (GstOMXAACEnc * self)
{
self->bitrate = DEFAULT_BITRATE;
self->aac_tools = DEFAULT_AAC_TOOLS;
self->aac_er_tools = DEFAULT_AAC_ER_TOOLS;
}
static void
gst_omx_aac_enc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstOMXAACEnc *self = GST_OMX_AAC_ENC (object);
switch (prop_id) {
case PROP_BITRATE:
self->bitrate = g_value_get_uint (value);
break;
case PROP_AAC_TOOLS:
self->aac_tools = g_value_get_flags (value);
break;
case PROP_AAC_ERROR_RESILIENCE_TOOLS:
self->aac_er_tools = g_value_get_flags (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_omx_aac_enc_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
GstOMXAACEnc *self = GST_OMX_AAC_ENC (object);
switch (prop_id) {
case PROP_BITRATE:
g_value_set_uint (value, self->bitrate);
break;
case PROP_AAC_TOOLS:
g_value_set_flags (value, self->aac_tools);
break;
case PROP_AAC_ERROR_RESILIENCE_TOOLS:
g_value_set_flags (value, self->aac_er_tools);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static gboolean
gst_omx_aac_enc_set_format (GstOMXAudioEnc * enc, GstOMXPort * port,
GstAudioInfo * info)
{
GstOMXAACEnc *self = GST_OMX_AAC_ENC (enc);
OMX_AUDIO_PARAM_AACPROFILETYPE aac_profile;
GstCaps *peercaps;
OMX_ERRORTYPE err;
GST_OMX_INIT_STRUCT (&aac_profile);
aac_profile.nPortIndex = enc->enc_out_port->index;
err =
gst_omx_component_get_parameter (enc->enc, OMX_IndexParamAudioAac,
&aac_profile);
if (err != OMX_ErrorNone) {
GST_ERROR_OBJECT (self,
"Failed to get AAC parameters from component: %s (0x%08x)",
gst_omx_error_to_string (err), err);
return FALSE;
}
peercaps = gst_pad_peer_query_caps (GST_AUDIO_ENCODER_SRC_PAD (self),
gst_pad_get_pad_template_caps (GST_AUDIO_ENCODER_SRC_PAD (self)));
if (peercaps) {
GstStructure *s;
gint mpegversion = 0;
const gchar *profile_string, *stream_format_string;
if (gst_caps_is_empty (peercaps)) {
gst_caps_unref (peercaps);
GST_ERROR_OBJECT (self, "Empty caps");
return FALSE;
}
s = gst_caps_get_structure (peercaps, 0);
if (gst_structure_get_int (s, "mpegversion", &mpegversion)) {
profile_string =
gst_structure_get_string (s,
((mpegversion == 2) ? "profile" : "base-profile"));
if (profile_string) {
if (g_str_equal (profile_string, "main")) {
aac_profile.eAACProfile = OMX_AUDIO_AACObjectMain;
} else if (g_str_equal (profile_string, "lc")) {
aac_profile.eAACProfile = OMX_AUDIO_AACObjectLC;
} else if (g_str_equal (profile_string, "ssr")) {
aac_profile.eAACProfile = OMX_AUDIO_AACObjectSSR;
} else if (g_str_equal (profile_string, "ltp")) {
aac_profile.eAACProfile = OMX_AUDIO_AACObjectLTP;
} else {
GST_ERROR_OBJECT (self, "Unsupported profile '%s'", profile_string);
gst_caps_unref (peercaps);
return FALSE;
}
}
}
stream_format_string = gst_structure_get_string (s, "stream-format");
if (stream_format_string) {
if (g_str_equal (stream_format_string, "raw")) {
aac_profile.eAACStreamFormat = OMX_AUDIO_AACStreamFormatRAW;
} else if (g_str_equal (stream_format_string, "adts")) {
if (mpegversion == 2) {
aac_profile.eAACStreamFormat = OMX_AUDIO_AACStreamFormatMP2ADTS;
} else {
aac_profile.eAACStreamFormat = OMX_AUDIO_AACStreamFormatMP4ADTS;
}
} else if (g_str_equal (stream_format_string, "loas")) {
aac_profile.eAACStreamFormat = OMX_AUDIO_AACStreamFormatMP4LOAS;
} else if (g_str_equal (stream_format_string, "latm")) {
aac_profile.eAACStreamFormat = OMX_AUDIO_AACStreamFormatMP4LATM;
} else if (g_str_equal (stream_format_string, "adif")) {
aac_profile.eAACStreamFormat = OMX_AUDIO_AACStreamFormatADIF;
} else {
GST_ERROR_OBJECT (self, "Unsupported stream-format '%s'",
stream_format_string);
gst_caps_unref (peercaps);
return FALSE;
}
}
gst_caps_unref (peercaps);
}
aac_profile.nAACtools = self->aac_tools;
aac_profile.nAACERtools = self->aac_er_tools;
aac_profile.nBitRate = self->bitrate;
err =
gst_omx_component_set_parameter (enc->enc, OMX_IndexParamAudioAac,
&aac_profile);
if (err != OMX_ErrorNone) {
GST_ERROR_OBJECT (self, "Error setting AAC parameters: %s (0x%08x)",
gst_omx_error_to_string (err), err);
return FALSE;
}
return TRUE;
}
typedef enum adts_sample_index__
{
ADTS_SAMPLE_INDEX_96000 = 0x0,
ADTS_SAMPLE_INDEX_88200,
ADTS_SAMPLE_INDEX_64000,
ADTS_SAMPLE_INDEX_48000,
ADTS_SAMPLE_INDEX_44100,
ADTS_SAMPLE_INDEX_32000,
ADTS_SAMPLE_INDEX_24000,
ADTS_SAMPLE_INDEX_22050,
ADTS_SAMPLE_INDEX_16000,
ADTS_SAMPLE_INDEX_12000,
ADTS_SAMPLE_INDEX_11025,
ADTS_SAMPLE_INDEX_8000,
ADTS_SAMPLE_INDEX_7350,
ADTS_SAMPLE_INDEX_MAX
} adts_sample_index;
static adts_sample_index
map_adts_sample_index (guint32 srate)
{
adts_sample_index ret;
switch (srate) {
case 96000:
ret = ADTS_SAMPLE_INDEX_96000;
break;
case 88200:
ret = ADTS_SAMPLE_INDEX_88200;
break;
case 64000:
ret = ADTS_SAMPLE_INDEX_64000;
break;
case 48000:
ret = ADTS_SAMPLE_INDEX_48000;
break;
case 44100:
ret = ADTS_SAMPLE_INDEX_44100;
break;
case 32000:
ret = ADTS_SAMPLE_INDEX_32000;
break;
case 24000:
ret = ADTS_SAMPLE_INDEX_24000;
break;
case 22050:
ret = ADTS_SAMPLE_INDEX_22050;
break;
case 16000:
ret = ADTS_SAMPLE_INDEX_16000;
break;
case 12000:
ret = ADTS_SAMPLE_INDEX_12000;
break;
case 11025:
ret = ADTS_SAMPLE_INDEX_11025;
break;
case 8000:
ret = ADTS_SAMPLE_INDEX_8000;
break;
case 7350:
ret = ADTS_SAMPLE_INDEX_7350;
break;
default:
ret = ADTS_SAMPLE_INDEX_44100;
break;
}
return ret;
}
static GstCaps *
gst_omx_aac_enc_get_caps (GstOMXAudioEnc * enc, GstOMXPort * port,
GstAudioInfo * info)
{
GstCaps *caps;
OMX_ERRORTYPE err;
OMX_AUDIO_PARAM_AACPROFILETYPE aac_profile;
gint mpegversion = 4;
const gchar *stream_format = NULL, *profile = NULL;
GST_OMX_INIT_STRUCT (&aac_profile);
aac_profile.nPortIndex = enc->enc_out_port->index;
err =
gst_omx_component_get_parameter (enc->enc, OMX_IndexParamAudioAac,
&aac_profile);
if (err != OMX_ErrorNone) {
GST_ERROR_OBJECT (enc,
"Failed to get AAC parameters from component: %s (0x%08x)",
gst_omx_error_to_string (err), err);
return NULL;
}
switch (aac_profile.eAACProfile) {
case OMX_AUDIO_AACObjectMain:
profile = "main";
break;
case OMX_AUDIO_AACObjectLC:
profile = "lc";
break;
case OMX_AUDIO_AACObjectSSR:
profile = "ssr";
break;
case OMX_AUDIO_AACObjectLTP:
profile = "ltp";
break;
case OMX_AUDIO_AACObjectHE:
case OMX_AUDIO_AACObjectScalable:
case OMX_AUDIO_AACObjectERLC:
case OMX_AUDIO_AACObjectLD:
case OMX_AUDIO_AACObjectHE_PS:
default:
GST_ERROR_OBJECT (enc, "Unsupported profile %d", aac_profile.eAACProfile);
break;
}
switch (aac_profile.eAACStreamFormat) {
case OMX_AUDIO_AACStreamFormatMP2ADTS:
mpegversion = 2;
stream_format = "adts";
break;
case OMX_AUDIO_AACStreamFormatMP4ADTS:
mpegversion = 4;
stream_format = "adts";
break;
case OMX_AUDIO_AACStreamFormatMP4LOAS:
mpegversion = 4;
stream_format = "loas";
break;
case OMX_AUDIO_AACStreamFormatMP4LATM:
mpegversion = 4;
stream_format = "latm";
break;
case OMX_AUDIO_AACStreamFormatADIF:
mpegversion = 4;
stream_format = "adif";
break;
case OMX_AUDIO_AACStreamFormatRAW:
mpegversion = 4;
stream_format = "raw";
break;
case OMX_AUDIO_AACStreamFormatMP4FF:
default:
GST_ERROR_OBJECT (enc, "Unsupported stream-format %u",
aac_profile.eAACStreamFormat);
break;
}
caps = gst_caps_new_empty_simple ("audio/mpeg");
if (mpegversion != 0)
gst_caps_set_simple (caps, "mpegversion", G_TYPE_INT, mpegversion,
"stream-format", G_TYPE_STRING, stream_format, NULL);
if (profile != NULL && mpegversion == 2)
gst_caps_set_simple (caps, "profile", G_TYPE_STRING, profile, NULL);
if (profile != NULL && mpegversion == 4)
gst_caps_set_simple (caps, "base-profile", G_TYPE_STRING, profile, NULL);
if (aac_profile.nChannels != 0)
gst_caps_set_simple (caps, "channels", G_TYPE_INT, aac_profile.nChannels,
NULL);
if (aac_profile.nSampleRate != 0)
gst_caps_set_simple (caps, "rate", G_TYPE_INT, aac_profile.nSampleRate,
NULL);
if (aac_profile.eAACStreamFormat == OMX_AUDIO_AACStreamFormatRAW) {
GstBuffer *codec_data;
adts_sample_index sr_idx;
GstMapInfo map = GST_MAP_INFO_INIT;
codec_data = gst_buffer_new_and_alloc (2);
gst_buffer_map (codec_data, &map, GST_MAP_WRITE);
sr_idx = map_adts_sample_index (aac_profile.nSampleRate);
map.data[0] = ((aac_profile.eAACProfile & 0x1F) << 3) |
((sr_idx & 0xE) >> 1);
map.data[1] = ((sr_idx & 0x1) << 7) | ((aac_profile.nChannels & 0xF) << 3);
gst_buffer_unmap (codec_data, &map);
GST_DEBUG_OBJECT (enc, "setting new codec_data");
gst_caps_set_simple (caps, "codec_data", GST_TYPE_BUFFER, codec_data, NULL);
gst_buffer_unref (codec_data);
}
return caps;
}
static guint
gst_omx_aac_enc_get_num_samples (GstOMXAudioEnc * enc, GstOMXPort * port,
GstAudioInfo * info, GstOMXBuffer * buf)
{
/* FIXME: Depends on the profile at least */
return 1024;
}
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