gstreamer/omx/gstomxaacdec.c

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2014-07-13 16:22:39 +00:00
/*
* Copyright (C) 2014, Sebastian Dröge <sebastian@centricular.com>
*
* 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 "gstomxaacdec.h"
GST_DEBUG_CATEGORY_STATIC (gst_omx_aac_dec_debug_category);
#define GST_CAT_DEFAULT gst_omx_aac_dec_debug_category
/* prototypes */
static gboolean gst_omx_aac_dec_set_format (GstOMXAudioDec * dec,
GstOMXPort * port, GstCaps * caps);
static gboolean gst_omx_aac_dec_is_format_change (GstOMXAudioDec * dec,
GstOMXPort * port, GstCaps * caps);
static gint gst_omx_aac_dec_get_samples_per_frame (GstOMXAudioDec * dec,
GstOMXPort * port);
static gboolean gst_omx_aac_dec_get_channel_positions (GstOMXAudioDec * dec,
GstOMXPort * port, GstAudioChannelPosition position[OMX_AUDIO_MAXCHANNELS]);
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/* class initialization */
#define DEBUG_INIT \
GST_DEBUG_CATEGORY_INIT (gst_omx_aac_dec_debug_category, "omxaacdec", 0, \
"debug category for gst-omx aac audio decoder");
G_DEFINE_TYPE_WITH_CODE (GstOMXAACDec, gst_omx_aac_dec,
GST_TYPE_OMX_AUDIO_DEC, DEBUG_INIT);
static void
gst_omx_aac_dec_class_init (GstOMXAACDecClass * klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstOMXAudioDecClass *audiodec_class = GST_OMX_AUDIO_DEC_CLASS (klass);
audiodec_class->set_format = GST_DEBUG_FUNCPTR (gst_omx_aac_dec_set_format);
audiodec_class->is_format_change =
GST_DEBUG_FUNCPTR (gst_omx_aac_dec_is_format_change);
audiodec_class->get_samples_per_frame =
GST_DEBUG_FUNCPTR (gst_omx_aac_dec_get_samples_per_frame);
audiodec_class->get_channel_positions =
GST_DEBUG_FUNCPTR (gst_omx_aac_dec_get_channel_positions);
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audiodec_class->cdata.default_sink_template_caps = "audio/mpeg, "
"mpegversion=(int){2, 4}, "
"stream-format=(string) { raw, adts, adif, loas }, "
"rate=(int)[8000,48000], "
"channels=(int)[1,9], " "framed=(boolean) true";
gst_element_class_set_static_metadata (element_class,
"OpenMAX AAC Audio Decoder",
"Codec/Decoder/Audio",
"Decode AAC audio streams",
"Sebastian Dröge <sebastian@centricular.com>");
gst_omx_set_default_role (&audiodec_class->cdata, "audio_decoder.aac");
}
static void
gst_omx_aac_dec_init (GstOMXAACDec * self)
{
/* FIXME: Other values exist too! */
self->spf = 1024;
}
static gboolean
gst_omx_aac_dec_set_format (GstOMXAudioDec * dec, GstOMXPort * port,
GstCaps * caps)
{
GstOMXAACDec *self = GST_OMX_AAC_DEC (dec);
OMX_PARAM_PORTDEFINITIONTYPE port_def;
OMX_AUDIO_PARAM_AACPROFILETYPE aac_param;
OMX_ERRORTYPE err;
GstStructure *s;
gint rate, channels, mpegversion;
const gchar *stream_format;
gst_omx_port_get_port_definition (port, &port_def);
port_def.format.audio.eEncoding = OMX_AUDIO_CodingAAC;
err = gst_omx_port_update_port_definition (port, &port_def);
if (err != OMX_ErrorNone) {
GST_ERROR_OBJECT (self,
"Failed to set AAC format on component: %s (0x%08x)",
gst_omx_error_to_string (err), err);
return FALSE;
}
GST_OMX_INIT_STRUCT (&aac_param);
aac_param.nPortIndex = port->index;
err =
gst_omx_component_get_parameter (dec->dec, OMX_IndexParamAudioAac,
&aac_param);
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;
}
s = gst_caps_get_structure (caps, 0);
if (!gst_structure_get_int (s, "mpegversion", &mpegversion) ||
!gst_structure_get_int (s, "rate", &rate) ||
!gst_structure_get_int (s, "channels", &channels)) {
GST_ERROR_OBJECT (self, "Incomplete caps");
return FALSE;
}
stream_format = gst_structure_get_string (s, "stream-format");
if (!stream_format) {
GST_ERROR_OBJECT (self, "Incomplete caps");
return FALSE;
}
aac_param.nChannels = channels;
aac_param.nSampleRate = rate;
aac_param.nBitRate = 0; /* unknown */
aac_param.nAudioBandWidth = 0; /* decoder decision */
aac_param.eChannelMode = 0; /* FIXME */
if (mpegversion == 2)
aac_param.eAACStreamFormat = OMX_AUDIO_AACStreamFormatMP2ADTS;
else if (strcmp (stream_format, "adts") == 0)
aac_param.eAACStreamFormat = OMX_AUDIO_AACStreamFormatMP4ADTS;
else if (strcmp (stream_format, "loas") == 0)
aac_param.eAACStreamFormat = OMX_AUDIO_AACStreamFormatMP4LOAS;
else if (strcmp (stream_format, "adif") == 0)
aac_param.eAACStreamFormat = OMX_AUDIO_AACStreamFormatADIF;
else if (strcmp (stream_format, "raw") == 0)
aac_param.eAACStreamFormat = OMX_AUDIO_AACStreamFormatRAW;
else {
GST_ERROR_OBJECT (self, "Unexpected format: %s", stream_format);
return FALSE;
}
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err =
gst_omx_component_set_parameter (dec->dec, OMX_IndexParamAudioAac,
&aac_param);
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;
}
static gboolean
gst_omx_aac_dec_is_format_change (GstOMXAudioDec * dec, GstOMXPort * port,
GstCaps * caps)
{
GstOMXAACDec *self = GST_OMX_AAC_DEC (dec);
OMX_AUDIO_PARAM_AACPROFILETYPE aac_param;
OMX_ERRORTYPE err;
GstStructure *s;
gint rate, channels, mpegversion;
const gchar *stream_format;
GST_OMX_INIT_STRUCT (&aac_param);
aac_param.nPortIndex = port->index;
err =
gst_omx_component_get_parameter (dec->dec, OMX_IndexParamAudioAac,
&aac_param);
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;
}
s = gst_caps_get_structure (caps, 0);
if (!gst_structure_get_int (s, "mpegversion", &mpegversion) ||
!gst_structure_get_int (s, "rate", &rate) ||
!gst_structure_get_int (s, "channels", &channels)) {
GST_ERROR_OBJECT (self, "Incomplete caps");
return FALSE;
}
stream_format = gst_structure_get_string (s, "stream-format");
if (!stream_format) {
GST_ERROR_OBJECT (self, "Incomplete caps");
return FALSE;
}
if (aac_param.nChannels != channels)
return TRUE;
if (aac_param.nSampleRate != rate)
return TRUE;
if (mpegversion == 2
&& aac_param.eAACStreamFormat != OMX_AUDIO_AACStreamFormatMP2ADTS)
return TRUE;
if (aac_param.eAACStreamFormat == OMX_AUDIO_AACStreamFormatMP4ADTS &&
strcmp (stream_format, "adts") != 0)
return TRUE;
if (aac_param.eAACStreamFormat == OMX_AUDIO_AACStreamFormatMP4LOAS &&
strcmp (stream_format, "loas") != 0)
return TRUE;
if (aac_param.eAACStreamFormat == OMX_AUDIO_AACStreamFormatADIF &&
strcmp (stream_format, "adif") != 0)
return TRUE;
if (aac_param.eAACStreamFormat == OMX_AUDIO_AACStreamFormatRAW &&
strcmp (stream_format, "raw") != 0)
return TRUE;
return FALSE;
}
static gint
gst_omx_aac_dec_get_samples_per_frame (GstOMXAudioDec * dec, GstOMXPort * port)
{
return GST_OMX_AAC_DEC (dec)->spf;
}
static gboolean
gst_omx_aac_dec_get_channel_positions (GstOMXAudioDec * dec,
GstOMXPort * port, GstAudioChannelPosition position[OMX_AUDIO_MAXCHANNELS])
{
OMX_AUDIO_PARAM_PCMMODETYPE pcm_param;
OMX_ERRORTYPE err;
GST_OMX_INIT_STRUCT (&pcm_param);
pcm_param.nPortIndex = port->index;
err =
gst_omx_component_get_parameter (dec->dec, OMX_IndexParamAudioPcm,
&pcm_param);
if (err != OMX_ErrorNone) {
GST_ERROR_OBJECT (dec, "Failed to get PCM parameters: %s (0x%08x)",
gst_omx_error_to_string (err), err);
return FALSE;
}
/* FIXME: Rather arbitrary values here, based on what we do in gstfaac.c */
switch (pcm_param.nChannels) {
case 1:
position[0] = GST_AUDIO_CHANNEL_POSITION_MONO;
break;
case 2:
position[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
position[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
break;
case 3:
position[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
position[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
position[2] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
break;
case 4:
position[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
position[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
position[2] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
position[3] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER;
break;
case 5:
position[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
position[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
position[2] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
position[3] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
position[4] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
break;
case 6:
position[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
position[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
position[2] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
position[3] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
position[4] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
position[5] = GST_AUDIO_CHANNEL_POSITION_LFE1;
break;
default:
return FALSE;
}
return TRUE;
}