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