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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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omx: Add AAC audio decoder
This commit is contained in:
Sebastian Dröge
parent
f4961d0a72
commit
03cf0bc9a4
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@ -29,6 +29,7 @@ libgstomx_la_SOURCES = \
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gstomxmpeg4videoenc.c \
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gstomxh264enc.c \
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gstomxh263enc.c \
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gstomxaacdec.c \
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gstomxmp3dec.c \
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gstomxaacenc.c \
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gstomxaudiosink.c \
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@ -53,6 +54,7 @@ noinst_HEADERS = \
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gstomxmpeg4videoenc.h \
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gstomxh264enc.h \
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gstomxh263enc.h \
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gstomxaacdec.h \
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gstomxmp3dec.h \
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gstomxaacenc.h \
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gstomxaudiosink.h \
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@ -39,6 +39,7 @@
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#include "gstomxmpeg4videoenc.h"
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#include "gstomxh264enc.h"
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#include "gstomxh263enc.h"
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#include "gstomxaacdec.h"
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#include "gstomxmp3dec.h"
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#include "gstomxaacenc.h"
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#include "gstomxanalogaudiosink.h"
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@ -2246,7 +2247,7 @@ static const GGetTypeFunction types[] = {
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gst_omx_wmv_dec_get_type, gst_omx_mpeg4_video_enc_get_type,
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gst_omx_h264_enc_get_type, gst_omx_h263_enc_get_type,
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gst_omx_aac_enc_get_type, gst_omx_mjpeg_dec_get_type,
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gst_omx_mp3_dec_get_type
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gst_omx_aac_dec_get_type, gst_omx_mp3_dec_get_type
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#ifdef HAVE_VP8
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, gst_omx_vp8_dec_get_type
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#endif
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230
omx/gstomxaacdec.c
Normal file
230
omx/gstomxaacdec.c
Normal file
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@ -0,0 +1,230 @@
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/*
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* Copyright (C) 2014, Sebastian Dröge <sebastian@centricular.com>
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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 "gstomxaacdec.h"
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GST_DEBUG_CATEGORY_STATIC (gst_omx_aac_dec_debug_category);
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#define GST_CAT_DEFAULT gst_omx_aac_dec_debug_category
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/* prototypes */
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static gboolean gst_omx_aac_dec_set_format (GstOMXAudioDec * dec,
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GstOMXPort * port, GstCaps * caps);
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static gboolean gst_omx_aac_dec_is_format_change (GstOMXAudioDec * dec,
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GstOMXPort * port, GstCaps * caps);
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static gint gst_omx_aac_dec_get_samples_per_frame (GstOMXAudioDec * dec,
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GstOMXPort * port);
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/* class initialization */
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#define DEBUG_INIT \
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GST_DEBUG_CATEGORY_INIT (gst_omx_aac_dec_debug_category, "omxaacdec", 0, \
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"debug category for gst-omx aac audio decoder");
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G_DEFINE_TYPE_WITH_CODE (GstOMXAACDec, gst_omx_aac_dec,
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GST_TYPE_OMX_AUDIO_DEC, DEBUG_INIT);
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static void
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gst_omx_aac_dec_class_init (GstOMXAACDecClass * klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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GstOMXAudioDecClass *audiodec_class = GST_OMX_AUDIO_DEC_CLASS (klass);
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audiodec_class->set_format = GST_DEBUG_FUNCPTR (gst_omx_aac_dec_set_format);
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audiodec_class->is_format_change =
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GST_DEBUG_FUNCPTR (gst_omx_aac_dec_is_format_change);
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audiodec_class->get_samples_per_frame =
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GST_DEBUG_FUNCPTR (gst_omx_aac_dec_get_samples_per_frame);
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audiodec_class->cdata.default_sink_template_caps = "audio/mpeg, "
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"mpegversion=(int){2, 4}, "
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"stream-format=(string) { raw, adts, adif, loas }, "
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"rate=(int)[8000,48000], "
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"channels=(int)[1,9], " "framed=(boolean) true";
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gst_element_class_set_static_metadata (element_class,
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"OpenMAX AAC Audio Decoder",
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"Codec/Decoder/Audio",
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"Decode AAC audio streams",
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"Sebastian Dröge <sebastian@centricular.com>");
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gst_omx_set_default_role (&audiodec_class->cdata, "audio_decoder.aac");
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}
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static void
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gst_omx_aac_dec_init (GstOMXAACDec * self)
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{
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/* FIXME: Other values exist too! */
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self->spf = 1024;
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}
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static gboolean
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gst_omx_aac_dec_set_format (GstOMXAudioDec * dec, GstOMXPort * port,
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GstCaps * caps)
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{
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GstOMXAACDec *self = GST_OMX_AAC_DEC (dec);
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OMX_PARAM_PORTDEFINITIONTYPE port_def;
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OMX_AUDIO_PARAM_AACPROFILETYPE aac_param;
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OMX_ERRORTYPE err;
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GstStructure *s;
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gint rate, channels, mpegversion;
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const gchar *stream_format;
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gst_omx_port_get_port_definition (port, &port_def);
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port_def.format.audio.eEncoding = OMX_AUDIO_CodingAAC;
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err = gst_omx_port_update_port_definition (port, &port_def);
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if (err != OMX_ErrorNone) {
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GST_ERROR_OBJECT (self,
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"Failed to set AAC format on 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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GST_OMX_INIT_STRUCT (&aac_param);
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aac_param.nPortIndex = port->index;
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err =
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gst_omx_component_get_parameter (dec->dec, OMX_IndexParamAudioAac,
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&aac_param);
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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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s = gst_caps_get_structure (caps, 0);
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if (!gst_structure_get_int (s, "mpegversion", &mpegversion) ||
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!gst_structure_get_int (s, "rate", &rate) ||
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!gst_structure_get_int (s, "channels", &channels)) {
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GST_ERROR_OBJECT (self, "Incomplete caps");
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return FALSE;
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}
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stream_format = gst_structure_get_string (s, "stream-format");
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if (!stream_format) {
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GST_ERROR_OBJECT (self, "Incomplete caps");
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return FALSE;
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}
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aac_param.nChannels = channels;
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aac_param.nSampleRate = rate;
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aac_param.nBitRate = 0; /* unknown */
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aac_param.nAudioBandWidth = 0; /* decoder decision */
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aac_param.eChannelMode = 0; /* FIXME */
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if (mpegversion == 2)
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aac_param.eAACStreamFormat = OMX_AUDIO_AACStreamFormatMP2ADTS;
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else if (strcmp (stream_format, "adts") == 0)
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aac_param.eAACStreamFormat = OMX_AUDIO_AACStreamFormatMP4ADTS;
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else if (strcmp (stream_format, "loas") == 0)
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aac_param.eAACStreamFormat = OMX_AUDIO_AACStreamFormatMP4LOAS;
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else if (strcmp (stream_format, "adif") == 0)
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aac_param.eAACStreamFormat = OMX_AUDIO_AACStreamFormatADIF;
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else if (strcmp (stream_format, "raw") == 0)
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aac_param.eAACStreamFormat = OMX_AUDIO_AACStreamFormatRAW;
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else
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aac_param.eAACStreamFormat = OMX_AUDIO_AACStreamFormatRAW;
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err =
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gst_omx_component_set_parameter (dec->dec, OMX_IndexParamAudioAac,
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&aac_param);
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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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static gboolean
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gst_omx_aac_dec_is_format_change (GstOMXAudioDec * dec, GstOMXPort * port,
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GstCaps * caps)
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{
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GstOMXAACDec *self = GST_OMX_AAC_DEC (dec);
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OMX_AUDIO_PARAM_AACPROFILETYPE aac_param;
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OMX_ERRORTYPE err;
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GstStructure *s;
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gint rate, channels, mpegversion;
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const gchar *stream_format;
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GST_OMX_INIT_STRUCT (&aac_param);
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aac_param.nPortIndex = port->index;
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err =
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gst_omx_component_get_parameter (dec->dec, OMX_IndexParamAudioAac,
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&aac_param);
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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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s = gst_caps_get_structure (caps, 0);
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if (!gst_structure_get_int (s, "mpegversion", &mpegversion) ||
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!gst_structure_get_int (s, "rate", &rate) ||
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!gst_structure_get_int (s, "channels", &channels)) {
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GST_ERROR_OBJECT (self, "Incomplete caps");
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return FALSE;
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}
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stream_format = gst_structure_get_string (s, "stream-format");
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if (!stream_format) {
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GST_ERROR_OBJECT (self, "Incomplete caps");
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return FALSE;
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}
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if (aac_param.nChannels != channels)
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return TRUE;
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if (aac_param.nSampleRate != rate)
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return TRUE;
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if (mpegversion == 2
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&& aac_param.eAACStreamFormat != OMX_AUDIO_AACStreamFormatMP2ADTS)
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return TRUE;
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if (aac_param.eAACStreamFormat == OMX_AUDIO_AACStreamFormatMP4ADTS &&
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strcmp (stream_format, "adts") != 0)
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return TRUE;
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if (aac_param.eAACStreamFormat == OMX_AUDIO_AACStreamFormatMP4LOAS &&
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strcmp (stream_format, "loas") != 0)
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return TRUE;
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if (aac_param.eAACStreamFormat == OMX_AUDIO_AACStreamFormatADIF &&
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strcmp (stream_format, "adif") != 0)
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return TRUE;
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if (aac_param.eAACStreamFormat == OMX_AUDIO_AACStreamFormatRAW &&
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strcmp (stream_format, "raw") != 0)
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return TRUE;
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return FALSE;
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}
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static gint
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gst_omx_aac_dec_get_samples_per_frame (GstOMXAudioDec * dec, GstOMXPort * port)
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{
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return GST_OMX_AAC_DEC (dec)->spf;
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}
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60
omx/gstomxaacdec.h
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60
omx/gstomxaacdec.h
Normal file
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@ -0,0 +1,60 @@
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/*
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* Copyright (C) 2014, Sebastian Dröge <sebastian@centricular.com>
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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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#ifndef __GST_OMX_AAC_DEC_H__
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#define __GST_OMX_AAC_DEC_H__
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#include <gst/gst.h>
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#include "gstomxaudiodec.h"
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G_BEGIN_DECLS
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#define GST_TYPE_OMX_AAC_DEC \
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(gst_omx_aac_dec_get_type())
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#define GST_OMX_AAC_DEC(obj) \
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(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_OMX_AAC_DEC,GstOMXAACDec))
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#define GST_OMX_AAC_DEC_CLASS(klass) \
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(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_OMX_AAC_DEC,GstOMXAACDecClass))
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#define GST_OMX_AAC_DEC_GET_CLASS(obj) \
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(G_TYPE_INSTANCE_GET_CLASS((obj),GST_TYPE_OMX_AAC_DEC,GstOMXAACDecClass))
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#define GST_IS_OMX_AAC_DEC(obj) \
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(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_OMX_AAC_DEC))
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#define GST_IS_OMX_AAC_DEC_CLASS(obj) \
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(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_OMX_AAC_DEC))
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typedef struct _GstOMXAACDec GstOMXAACDec;
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typedef struct _GstOMXAACDecClass GstOMXAACDecClass;
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struct _GstOMXAACDec
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{
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GstOMXAudioDec parent;
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gint spf;
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};
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struct _GstOMXAACDecClass
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{
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GstOMXAudioDecClass parent_class;
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};
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GType gst_omx_aac_dec_get_type (void);
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G_END_DECLS
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#endif /* __GST_OMX_AAC_DEC_H__ */
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@ -40,7 +40,7 @@ static gint gst_omx_mp3_dec_get_samples_per_frame (GstOMXAudioDec * dec,
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#define DEBUG_INIT \
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GST_DEBUG_CATEGORY_INIT (gst_omx_mp3_dec_debug_category, "omxmp3dec", 0, \
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"debug category for gst-omx audio decoder base class");
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"debug category for gst-omx mp3 audio decoder");
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G_DEFINE_TYPE_WITH_CODE (GstOMXMP3Dec, gst_omx_mp3_dec,
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GST_TYPE_OMX_AUDIO_DEC, DEBUG_INIT);
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