mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2025-06-04 22:48:54 +00:00
Merge remote-tracking branch 'origin/master' into 0.11
Conflicts: ext/faac/gstfaac.c ext/opus/gstopusdec.c ext/opus/gstopusenc.c gst/audiovisualizers/gstspacescope.c gst/colorspace/colorspace.c
This commit is contained in:
commit
a8024bb698
9 changed files with 603 additions and 138 deletions
|
@ -1,6 +1,6 @@
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|||
plugin_LTLIBRARIES = libgstopus.la
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||||
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libgstopus_la_SOURCES = gstopus.c gstopusdec.c gstopusenc.c gstopusparse.c gstopusheader.c
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libgstopus_la_SOURCES = gstopus.c gstopusdec.c gstopusenc.c gstopusparse.c gstopusheader.c gstopuscommon.c
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libgstopus_la_CFLAGS = \
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-DGST_USE_UNSTABLE_API \
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$(GST_PLUGINS_BASE_CFLAGS) \
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@ -15,4 +15,4 @@ libgstopus_la_LIBADD = \
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libgstopus_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) $(LIBM)
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libgstopus_la_LIBTOOLFLAGS = --tag=disable-static
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noinst_HEADERS = gstopusenc.h gstopusdec.h gstopusparse.h gstopusheader.h
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noinst_HEADERS = gstopusenc.h gstopusdec.h gstopusparse.h gstopusheader.h gstopuscommon.h
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|
|
72
ext/opus/gstopuscommon.c
Normal file
72
ext/opus/gstopuscommon.c
Normal file
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@ -0,0 +1,72 @@
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|||
/* GStreamer
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* Copyright (C) 2009 Sebastian Dröge <sebastian.droege@collabora.co.uk>
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*
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* 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,
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* Boston, MA 02111-1307, USA.
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*/
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#include "gstopuscommon.h"
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/* http://www.xiph.org/vorbis/doc/Vorbis_I_spec.html#x1-800004.3.9 */
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/* copy of the same structure in the vorbis plugin */
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const GstAudioChannelPosition gst_opus_channel_positions[][8] = {
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{ /* Mono */
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GST_AUDIO_CHANNEL_POSITION_FRONT_MONO},
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{ /* Stereo */
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT},
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{ /* Stereo + Centre */
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT},
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{ /* Quadraphonic */
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
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GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
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},
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{ /* Stereo + Centre + rear stereo */
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
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GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
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},
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{ /* Full 5.1 Surround */
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
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GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_LFE,
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},
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{ /* 6.1 Surround, in Vorbis spec since 2010-01-13 */
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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||||
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
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||||
GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT,
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||||
GST_AUDIO_CHANNEL_POSITION_REAR_CENTER,
|
||||
GST_AUDIO_CHANNEL_POSITION_LFE},
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||||
{ /* 7.1 Surround, in Vorbis spec since 2010-01-13 */
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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||||
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
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||||
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
|
||||
GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
|
||||
GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT,
|
||||
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
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||||
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
|
||||
GST_AUDIO_CHANNEL_POSITION_LFE},
|
||||
};
|
33
ext/opus/gstopuscommon.h
Normal file
33
ext/opus/gstopuscommon.h
Normal file
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@ -0,0 +1,33 @@
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|||
/* GStreamer Opus Encoder
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||||
* Copyright (C) 2009 Sebastian Dröge <sebastian.droege@collabora.co.uk>
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||||
*
|
||||
* 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.
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||||
*/
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#ifndef __GST_OPUS_COMMON_H__
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#define __GST_OPUS_COMMON_H__
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#include <gst/gst.h>
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#include <gst/audio/multichannel.h>
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G_BEGIN_DECLS
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extern const GstAudioChannelPosition gst_opus_channel_positions[][8];
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G_END_DECLS
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#endif /* __GST_OPUS_COMMON_H__ */
|
|
@ -41,9 +41,11 @@
|
|||
# include "config.h"
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#endif
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||||
#include <math.h>
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||||
#include <string.h>
|
||||
#include <gst/tag/tag.h>
|
||||
#include "gstopusheader.h"
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||||
#include "gstopuscommon.h"
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#include "gstopusdec.h"
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|
||||
GST_DEBUG_CATEGORY_STATIC (opusdec_debug);
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||||
|
@ -54,9 +56,9 @@ GST_STATIC_PAD_TEMPLATE ("src",
|
|||
GST_PAD_SRC,
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||||
GST_PAD_ALWAYS,
|
||||
GST_STATIC_CAPS ("audio/x-raw, "
|
||||
"format = (string) { S16LE }, "
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||||
"format = (string) { " GST_AUDIO_NE (S16) " }, "
|
||||
"rate = (int) { 8000, 12000, 16000, 24000, 48000 }, "
|
||||
"channels = (int) [ 1, 2 ] ")
|
||||
"channels = (int) [ 1, 8 ] ")
|
||||
);
|
||||
|
||||
static GstStaticPadTemplate opus_dec_sink_factory =
|
||||
|
@ -68,6 +70,19 @@ GST_STATIC_PAD_TEMPLATE ("sink",
|
|||
|
||||
G_DEFINE_TYPE (GstOpusDec, gst_opus_dec, GST_TYPE_AUDIO_DECODER);
|
||||
|
||||
#define DB_TO_LINEAR(x) pow (10., (x) / 20.)
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||||
|
||||
#define DEFAULT_USE_INBAND_FEC FALSE
|
||||
#define DEFAULT_APPLY_GAIN TRUE
|
||||
|
||||
enum
|
||||
{
|
||||
PROP_0,
|
||||
PROP_USE_INBAND_FEC,
|
||||
PROP_APPLY_GAIN
|
||||
};
|
||||
|
||||
|
||||
static GstFlowReturn gst_opus_dec_parse_header (GstOpusDec * dec,
|
||||
GstBuffer * buf);
|
||||
static gboolean gst_opus_dec_start (GstAudioDecoder * dec);
|
||||
|
@ -76,16 +91,26 @@ static GstFlowReturn gst_opus_dec_handle_frame (GstAudioDecoder * dec,
|
|||
GstBuffer * buffer);
|
||||
static gboolean gst_opus_dec_set_format (GstAudioDecoder * bdec,
|
||||
GstCaps * caps);
|
||||
static void gst_opus_dec_get_property (GObject * object, guint prop_id,
|
||||
GValue * value, GParamSpec * pspec);
|
||||
static void gst_opus_dec_set_property (GObject * object, guint prop_id,
|
||||
const GValue * value, GParamSpec * pspec);
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||||
|
||||
|
||||
static void
|
||||
gst_opus_dec_class_init (GstOpusDecClass * klass)
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||||
{
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||||
GObjectClass *gobject_class;
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GstAudioDecoderClass *adclass;
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||||
GstElementClass *element_class;
|
||||
|
||||
gobject_class = (GObjectClass *) klass;
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||||
adclass = (GstAudioDecoderClass *) klass;
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element_class = (GstElementClass *) klass;
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||||
|
||||
gobject_class->set_property = gst_opus_dec_set_property;
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||||
gobject_class->get_property = gst_opus_dec_get_property;
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|
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adclass->start = GST_DEBUG_FUNCPTR (gst_opus_dec_start);
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adclass->stop = GST_DEBUG_FUNCPTR (gst_opus_dec_stop);
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adclass->handle_frame = GST_DEBUG_FUNCPTR (gst_opus_dec_handle_frame);
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||||
|
@ -99,6 +124,15 @@ gst_opus_dec_class_init (GstOpusDecClass * klass)
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|||
"Codec/Decoder/Audio",
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||||
"decode opus streams to audio",
|
||||
"Sebastian Dröge <sebastian.droege@collabora.co.uk>");
|
||||
g_object_class_install_property (gobject_class, PROP_USE_INBAND_FEC,
|
||||
g_param_spec_boolean ("use-inband-fec", "Use in-band FEC",
|
||||
"Use forward error correction if available", DEFAULT_USE_INBAND_FEC,
|
||||
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
||||
|
||||
g_object_class_install_property (gobject_class, PROP_APPLY_GAIN,
|
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g_param_spec_boolean ("apply-gain", "Apply gain",
|
||||
"Apply gain if any is specified in the header", DEFAULT_APPLY_GAIN,
|
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
||||
|
||||
GST_DEBUG_CATEGORY_INIT (opusdec_debug, "opusdec", 0,
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||||
"opus decoding element");
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||||
|
@ -109,14 +143,17 @@ gst_opus_dec_reset (GstOpusDec * dec)
|
|||
{
|
||||
dec->packetno = 0;
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||||
if (dec->state) {
|
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opus_decoder_destroy (dec->state);
|
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opus_multistream_decoder_destroy (dec->state);
|
||||
dec->state = NULL;
|
||||
}
|
||||
|
||||
gst_buffer_replace (&dec->streamheader, NULL);
|
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gst_buffer_replace (&dec->vorbiscomment, NULL);
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gst_buffer_replace (&dec->last_buffer, NULL);
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||||
dec->primed = FALSE;
|
||||
|
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dec->pre_skip = 0;
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||||
dec->r128_gain = 0;
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||||
}
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||||
|
||||
static void
|
||||
|
@ -124,6 +161,8 @@ gst_opus_dec_init (GstOpusDec * dec)
|
|||
{
|
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dec->sample_rate = 0;
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dec->n_channels = 0;
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dec->use_inband_fec = FALSE;
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dec->apply_gain = DEFAULT_APPLY_GAIN;
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|
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gst_opus_dec_reset (dec);
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}
|
||||
|
@ -138,6 +177,11 @@ gst_opus_dec_start (GstAudioDecoder * dec)
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/* we know about concealment */
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gst_audio_decoder_set_plc_aware (dec, TRUE);
|
||||
|
||||
if (odec->use_inband_fec) {
|
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gst_audio_decoder_set_latency (dec, 2 * GST_MSECOND + GST_MSECOND / 2,
|
||||
120 * GST_MSECOND);
|
||||
}
|
||||
|
||||
return TRUE;
|
||||
}
|
||||
|
||||
|
@ -151,15 +195,111 @@ gst_opus_dec_stop (GstAudioDecoder * dec)
|
|||
return TRUE;
|
||||
}
|
||||
|
||||
static double
|
||||
gst_opus_dec_get_r128_gain (gint16 r128_gain)
|
||||
{
|
||||
return r128_gain / (double) (1 << 8);
|
||||
}
|
||||
|
||||
static double
|
||||
gst_opus_dec_get_r128_volume (gint16 r128_gain)
|
||||
{
|
||||
return DB_TO_LINEAR (gst_opus_dec_get_r128_gain (r128_gain));
|
||||
}
|
||||
|
||||
static GstFlowReturn
|
||||
gst_opus_dec_parse_header (GstOpusDec * dec, GstBuffer * buf)
|
||||
{
|
||||
g_return_val_if_fail (gst_opus_header_is_header (buf, "OpusHead", 8),
|
||||
GST_FLOW_ERROR);
|
||||
g_return_val_if_fail (GST_BUFFER_SIZE (buf) >= 19, GST_FLOW_ERROR);
|
||||
const guint8 *data;
|
||||
GstCaps *caps;
|
||||
GstStructure *s;
|
||||
const GstAudioChannelPosition *pos = NULL;
|
||||
|
||||
dec->pre_skip = GST_READ_UINT16_LE (GST_BUFFER_DATA (buf) + 10);
|
||||
GST_INFO_OBJECT (dec, "Found pre-skip of %u samples", dec->pre_skip);
|
||||
g_return_val_if_fail (gst_opus_header_is_id_header (buf), GST_FLOW_ERROR);
|
||||
|
||||
data = gst_buffer_map (buf, NULL, NULL, GST_MAP_READ);
|
||||
|
||||
g_return_val_if_fail (dec->n_channels != data[9], GST_FLOW_ERROR);
|
||||
|
||||
dec->n_channels = data[9];
|
||||
dec->pre_skip = GST_READ_UINT16_LE (data + 10);
|
||||
dec->r128_gain = GST_READ_UINT16_LE (data + 14);
|
||||
dec->r128_gain_volume = gst_opus_dec_get_r128_volume (dec->r128_gain);
|
||||
GST_INFO_OBJECT (dec,
|
||||
"Found pre-skip of %u samples, R128 gain %d (volume %f)",
|
||||
dec->pre_skip, dec->r128_gain, dec->r128_gain_volume);
|
||||
|
||||
dec->channel_mapping_family = data[18];
|
||||
if (dec->channel_mapping_family == 0) {
|
||||
/* implicit mapping */
|
||||
GST_INFO_OBJECT (dec, "Channel mapping family 0, implicit mapping");
|
||||
dec->n_streams = dec->n_stereo_streams = 1;
|
||||
dec->channel_mapping[0] = 0;
|
||||
dec->channel_mapping[1] = 1;
|
||||
} else {
|
||||
dec->n_streams = data[19];
|
||||
dec->n_stereo_streams = data[20];
|
||||
memcpy (dec->channel_mapping, data + 21, dec->n_channels);
|
||||
|
||||
if (dec->channel_mapping_family == 1) {
|
||||
GST_INFO_OBJECT (dec, "Channel mapping family 1, Vorbis mapping");
|
||||
switch (dec->n_channels) {
|
||||
case 1:
|
||||
case 2:
|
||||
/* nothing */
|
||||
break;
|
||||
case 3:
|
||||
case 4:
|
||||
case 5:
|
||||
case 6:
|
||||
case 7:
|
||||
case 8:
|
||||
pos = gst_opus_channel_positions[dec->n_channels - 1];
|
||||
break;
|
||||
default:{
|
||||
gint i;
|
||||
GstAudioChannelPosition *posn =
|
||||
g_new (GstAudioChannelPosition, dec->n_channels);
|
||||
|
||||
GST_ELEMENT_WARNING (GST_ELEMENT (dec), STREAM, DECODE,
|
||||
(NULL), ("Using NONE channel layout for more than 8 channels"));
|
||||
|
||||
for (i = 0; i < dec->n_channels; i++)
|
||||
posn[i] = GST_AUDIO_CHANNEL_POSITION_NONE;
|
||||
|
||||
pos = posn;
|
||||
}
|
||||
}
|
||||
} else {
|
||||
GST_INFO_OBJECT (dec, "Channel mapping family %d",
|
||||
dec->channel_mapping_family);
|
||||
}
|
||||
}
|
||||
|
||||
/* negotiate width with downstream */
|
||||
caps = gst_pad_get_allowed_caps (GST_AUDIO_DECODER_SRC_PAD (dec));
|
||||
s = gst_caps_get_structure (caps, 0);
|
||||
gst_structure_fixate_field_nearest_int (s, "rate", 48000);
|
||||
gst_structure_get_int (s, "rate", &dec->sample_rate);
|
||||
gst_structure_fixate_field_nearest_int (s, "channels", dec->n_channels);
|
||||
gst_structure_get_int (s, "channels", &dec->n_channels);
|
||||
|
||||
GST_INFO_OBJECT (dec, "Negotiated %d channels, %d Hz", dec->n_channels,
|
||||
dec->sample_rate);
|
||||
|
||||
if (pos) {
|
||||
gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos);
|
||||
}
|
||||
|
||||
if (dec->n_channels > 8) {
|
||||
g_free ((GstAudioChannelPosition *) pos);
|
||||
}
|
||||
|
||||
GST_INFO_OBJECT (dec, "Setting src caps to %" GST_PTR_FORMAT, caps);
|
||||
gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), caps);
|
||||
gst_caps_unref (caps);
|
||||
|
||||
gst_buffer_unmap (buf, (guint8 *) data, -1);
|
||||
|
||||
return GST_FLOW_OK;
|
||||
}
|
||||
|
@ -171,50 +311,8 @@ gst_opus_dec_parse_comments (GstOpusDec * dec, GstBuffer * buf)
|
|||
return GST_FLOW_OK;
|
||||
}
|
||||
|
||||
static void
|
||||
gst_opus_dec_setup_from_peer_caps (GstOpusDec * dec)
|
||||
{
|
||||
GstPad *srcpad, *peer;
|
||||
GstStructure *s;
|
||||
GstCaps *caps;
|
||||
const GstCaps *template_caps;
|
||||
const GstCaps *peer_caps;
|
||||
|
||||
srcpad = GST_AUDIO_DECODER_SRC_PAD (dec);
|
||||
peer = gst_pad_get_peer (srcpad);
|
||||
|
||||
if (peer) {
|
||||
template_caps = gst_pad_get_pad_template_caps (srcpad);
|
||||
peer_caps = gst_pad_get_caps (peer);
|
||||
GST_DEBUG_OBJECT (dec, "Peer caps: %" GST_PTR_FORMAT, peer_caps);
|
||||
caps = gst_caps_intersect (template_caps, peer_caps);
|
||||
gst_pad_fixate_caps (peer, caps);
|
||||
GST_DEBUG_OBJECT (dec, "Fixated caps: %" GST_PTR_FORMAT, caps);
|
||||
|
||||
s = gst_caps_get_structure (caps, 0);
|
||||
if (!gst_structure_get_int (s, "channels", &dec->n_channels)) {
|
||||
dec->n_channels = 2;
|
||||
GST_WARNING_OBJECT (dec, "Failed to get channels, using default %d",
|
||||
dec->n_channels);
|
||||
} else {
|
||||
GST_DEBUG_OBJECT (dec, "Got channels %d", dec->n_channels);
|
||||
}
|
||||
if (!gst_structure_get_int (s, "rate", &dec->sample_rate)) {
|
||||
dec->sample_rate = 48000;
|
||||
GST_WARNING_OBJECT (dec, "Failed to get rate, using default %d",
|
||||
dec->sample_rate);
|
||||
} else {
|
||||
GST_DEBUG_OBJECT (dec, "Got sample rate %d", dec->sample_rate);
|
||||
}
|
||||
|
||||
gst_audio_decoder_set_outcaps (GST_AUDIO_DECODER (dec), caps);
|
||||
} else {
|
||||
GST_WARNING_OBJECT (dec, "Failed to get src pad peer");
|
||||
}
|
||||
}
|
||||
|
||||
static GstFlowReturn
|
||||
opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buf)
|
||||
opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buffer)
|
||||
{
|
||||
GstFlowReturn res = GST_FLOW_OK;
|
||||
gsize size, out_size;
|
||||
|
@ -224,42 +322,51 @@ opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buf)
|
|||
int n, err;
|
||||
int samples;
|
||||
unsigned int packet_size;
|
||||
GstBuffer *buf;
|
||||
|
||||
if (dec->state == NULL) {
|
||||
gst_opus_dec_setup_from_peer_caps (dec);
|
||||
|
||||
GST_DEBUG_OBJECT (dec, "Creating decoder with %d channels, %d Hz",
|
||||
dec->n_channels, dec->sample_rate);
|
||||
dec->state = opus_decoder_create (dec->sample_rate, dec->n_channels, &err);
|
||||
dec->state = opus_multistream_decoder_create (dec->sample_rate,
|
||||
dec->n_channels, dec->n_streams, dec->n_stereo_streams,
|
||||
dec->channel_mapping, &err);
|
||||
if (!dec->state || err != OPUS_OK)
|
||||
goto creation_failed;
|
||||
}
|
||||
|
||||
if (buffer) {
|
||||
GST_DEBUG_OBJECT (dec, "Received buffer of size %u",
|
||||
gst_buffer_get_size (buffer));
|
||||
} else {
|
||||
GST_DEBUG_OBJECT (dec, "Received missing buffer");
|
||||
}
|
||||
|
||||
/* if using in-band FEC, we introdude one extra frame's delay as we need
|
||||
to potentially wait for next buffer to decode a missing buffer */
|
||||
if (dec->use_inband_fec && !dec->primed) {
|
||||
GST_DEBUG_OBJECT (dec, "First buffer received in FEC mode, early out");
|
||||
goto done;
|
||||
}
|
||||
|
||||
/* That's the buffer we'll be sending to the opus decoder. */
|
||||
buf = dec->use_inband_fec && dec->last_buffer ? dec->last_buffer : buffer;
|
||||
|
||||
if (buf) {
|
||||
data = gst_buffer_map (buf, &size, NULL, GST_MAP_READ);
|
||||
|
||||
GST_DEBUG_OBJECT (dec, "received buffer of size %u", size);
|
||||
GST_DEBUG_OBJECT (dec, "Using buffer of size %u", size);
|
||||
} else {
|
||||
/* concealment data, pass NULL as the bits parameters */
|
||||
GST_DEBUG_OBJECT (dec, "creating concealment data");
|
||||
GST_DEBUG_OBJECT (dec, "Using NULL buffer");
|
||||
data = NULL;
|
||||
size = 0;
|
||||
}
|
||||
|
||||
if (data) {
|
||||
samples =
|
||||
opus_packet_get_samples_per_frame (data,
|
||||
dec->sample_rate) * opus_packet_get_nb_frames (data, size);
|
||||
packet_size = samples * dec->n_channels * 2;
|
||||
GST_DEBUG_OBJECT (dec, "bandwidth %d", opus_packet_get_bandwidth (data));
|
||||
GST_DEBUG_OBJECT (dec, "samples %d", samples);
|
||||
} else {
|
||||
/* use maximum size (120 ms) as we do now know in advance how many samples
|
||||
will be returned */
|
||||
samples = 120 * dec->sample_rate / 1000;
|
||||
}
|
||||
|
||||
/* use maximum size (120 ms) as the number of returned samples is
|
||||
not constant over the stream. */
|
||||
samples = 120 * dec->sample_rate / 1000;
|
||||
packet_size = samples * dec->n_channels * 2;
|
||||
|
||||
outbuf = gst_buffer_new_and_alloc (packet_size);
|
||||
if (!outbuf) {
|
||||
goto buffer_failed;
|
||||
|
@ -267,39 +374,81 @@ opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buf)
|
|||
|
||||
out_data = (gint16 *) gst_buffer_map (outbuf, &out_size, NULL, GST_MAP_WRITE);
|
||||
|
||||
n = opus_decode (dec->state, data, size, out_data, samples, 0);
|
||||
if (dec->use_inband_fec) {
|
||||
if (dec->last_buffer) {
|
||||
/* normal delayed decode */
|
||||
n = opus_multistream_decode (dec->state, data, size, out_data, samples,
|
||||
0);
|
||||
} else {
|
||||
/* FEC reconstruction decode */
|
||||
n = opus_multistream_decode (dec->state, data, size, out_data, samples,
|
||||
1);
|
||||
}
|
||||
} else {
|
||||
/* normal decode */
|
||||
n = opus_multistream_decode (dec->state, data, size, out_data, samples, 0);
|
||||
}
|
||||
gst_buffer_unmap (buf, data, size);
|
||||
gst_buffer_unmap (outbuf, out_data, out_size);
|
||||
|
||||
if (n < 0) {
|
||||
GST_ELEMENT_ERROR (dec, STREAM, DECODE, ("Decoding error: %d", n), (NULL));
|
||||
return GST_FLOW_ERROR;
|
||||
}
|
||||
GST_DEBUG_OBJECT (dec, "decoded %d samples", n);
|
||||
GST_BUFFER_SIZE (outbuf) = n * 2 * dec->n_channels;
|
||||
gst_buffer_set_size (outbuf, n * 2 * dec->n_channels);
|
||||
|
||||
/* Skip any samples that need skipping */
|
||||
if (dec->pre_skip > 0) {
|
||||
guint scaled_pre_skip = dec->pre_skip * dec->sample_rate / 48000;
|
||||
guint skip = scaled_pre_skip > n ? n : scaled_pre_skip;
|
||||
guint scaled_skip = skip * 48000 / dec->sample_rate;
|
||||
GST_BUFFER_SIZE (outbuf) -= skip * 2 * dec->n_channels;
|
||||
GST_BUFFER_DATA (outbuf) += skip * 2 * dec->n_channels;
|
||||
|
||||
gst_buffer_resize (outbuf, skip * 2 * dec->n_channels, -1);
|
||||
dec->pre_skip -= scaled_skip;
|
||||
GST_INFO_OBJECT (dec,
|
||||
"Skipping %u samples (%u at 48000 Hz, %u left to skip)", skip,
|
||||
scaled_skip, dec->pre_skip);
|
||||
|
||||
if (GST_BUFFER_SIZE (outbuf) == 0) {
|
||||
if (gst_buffer_get_size (outbuf) == 0) {
|
||||
gst_buffer_unref (outbuf);
|
||||
outbuf = NULL;
|
||||
}
|
||||
}
|
||||
|
||||
/* Apply gain */
|
||||
/* Would be better off leaving this to a volume element, as this is
|
||||
a naive conversion that does too many int/float conversions.
|
||||
However, we don't have control over the pipeline...
|
||||
So make it optional if the user program wants to use a volume,
|
||||
but do it by default so the correct volume goes out by default */
|
||||
if (dec->apply_gain && outbuf && dec->r128_gain) {
|
||||
gsize rsize;
|
||||
unsigned int i, nsamples;
|
||||
double volume = dec->r128_gain_volume;
|
||||
gint16 *samples =
|
||||
(gint16 *) gst_buffer_map (outbuf, &rsize, NULL, GST_MAP_READWRITE);
|
||||
|
||||
GST_DEBUG_OBJECT (dec, "Applying gain: volume %f", volume);
|
||||
nsamples = rsize / 2;
|
||||
for (i = 0; i < nsamples; ++i) {
|
||||
int sample = (int) (samples[i] * volume + 0.5);
|
||||
samples[i] = sample < -32768 ? -32768 : sample > 32767 ? 32767 : sample;
|
||||
}
|
||||
gst_buffer_unmap (outbuf, samples, rsize);
|
||||
}
|
||||
|
||||
res = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (dec), outbuf, 1);
|
||||
|
||||
if (res != GST_FLOW_OK)
|
||||
GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (res));
|
||||
|
||||
done:
|
||||
if (dec->use_inband_fec) {
|
||||
gst_buffer_replace (&dec->last_buffer, buffer);
|
||||
dec->primed = TRUE;
|
||||
}
|
||||
|
||||
return res;
|
||||
|
||||
creation_failed:
|
||||
|
@ -436,3 +585,41 @@ gst_opus_dec_handle_frame (GstAudioDecoder * adec, GstBuffer * buf)
|
|||
|
||||
return res;
|
||||
}
|
||||
|
||||
static void
|
||||
gst_opus_dec_get_property (GObject * object, guint prop_id, GValue * value,
|
||||
GParamSpec * pspec)
|
||||
{
|
||||
GstOpusDec *dec = GST_OPUS_DEC (object);
|
||||
|
||||
switch (prop_id) {
|
||||
case PROP_USE_INBAND_FEC:
|
||||
g_value_set_boolean (value, dec->use_inband_fec);
|
||||
break;
|
||||
case PROP_APPLY_GAIN:
|
||||
g_value_set_boolean (value, dec->apply_gain);
|
||||
break;
|
||||
default:
|
||||
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
||||
break;
|
||||
}
|
||||
}
|
||||
|
||||
static void
|
||||
gst_opus_dec_set_property (GObject * object, guint prop_id,
|
||||
const GValue * value, GParamSpec * pspec)
|
||||
{
|
||||
GstOpusDec *dec = GST_OPUS_DEC (object);
|
||||
|
||||
switch (prop_id) {
|
||||
case PROP_USE_INBAND_FEC:
|
||||
dec->use_inband_fec = g_value_get_boolean (value);
|
||||
break;
|
||||
case PROP_APPLY_GAIN:
|
||||
dec->apply_gain = g_value_get_boolean (value);
|
||||
break;
|
||||
default:
|
||||
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
||||
break;
|
||||
}
|
||||
}
|
||||
|
|
|
@ -23,7 +23,7 @@
|
|||
|
||||
#include <gst/gst.h>
|
||||
#include <gst/audio/gstaudiodecoder.h>
|
||||
#include <opus/opus.h>
|
||||
#include <opus/opus_multistream.h>
|
||||
|
||||
G_BEGIN_DECLS
|
||||
|
||||
|
@ -44,7 +44,7 @@ typedef struct _GstOpusDecClass GstOpusDecClass;
|
|||
struct _GstOpusDec {
|
||||
GstAudioDecoder element;
|
||||
|
||||
OpusDecoder *state;
|
||||
OpusMSDecoder *state;
|
||||
|
||||
guint64 packetno;
|
||||
|
||||
|
@ -54,6 +54,19 @@ struct _GstOpusDec {
|
|||
int sample_rate;
|
||||
int n_channels;
|
||||
guint32 pre_skip;
|
||||
gint16 r128_gain;
|
||||
|
||||
guint8 n_streams;
|
||||
guint8 n_stereo_streams;
|
||||
guint8 channel_mapping_family;
|
||||
guint8 channel_mapping[256];
|
||||
|
||||
gboolean apply_gain;
|
||||
double r128_gain_volume;
|
||||
|
||||
gboolean use_inband_fec;
|
||||
GstBuffer *last_buffer;
|
||||
gboolean primed;
|
||||
};
|
||||
|
||||
struct _GstOpusDecClass {
|
||||
|
|
|
@ -49,6 +49,7 @@
|
|||
#include <gst/gsttagsetter.h>
|
||||
#include <gst/audio/audio.h>
|
||||
#include "gstopusheader.h"
|
||||
#include "gstopuscommon.h"
|
||||
#include "gstopusenc.h"
|
||||
|
||||
GST_DEBUG_CATEGORY_STATIC (opusenc_debug);
|
||||
|
@ -177,9 +178,6 @@ static gint64 gst_opus_enc_get_latency (GstOpusEnc * enc);
|
|||
|
||||
static GstFlowReturn gst_opus_enc_encode (GstOpusEnc * enc, GstBuffer * buffer);
|
||||
|
||||
static GstFlowReturn gst_opus_enc_encode (GstOpusEnc * enc, GstBuffer * buf);
|
||||
static gint64 gst_opus_enc_get_latency (GstOpusEnc * enc);
|
||||
|
||||
#define gst_opus_enc_parent_class parent_class
|
||||
G_DEFINE_TYPE_WITH_CODE (GstOpusEnc, gst_opus_enc, GST_TYPE_AUDIO_ENCODER,
|
||||
G_IMPLEMENT_INTERFACE (GST_TYPE_TAG_SETTER, NULL);
|
||||
|
@ -199,11 +197,11 @@ gst_opus_enc_class_init (GstOpusEncClass * klass)
|
|||
gobject_class->set_property = gst_opus_enc_set_property;
|
||||
gobject_class->get_property = gst_opus_enc_get_property;
|
||||
|
||||
gst_element_class_add_pad_template (element_class,
|
||||
gst_element_class_add_pad_template (gstelement_class,
|
||||
gst_static_pad_template_get (&src_factory));
|
||||
gst_element_class_add_pad_template (element_class,
|
||||
gst_element_class_add_pad_template (gstelement_class,
|
||||
gst_static_pad_template_get (&sink_factory));
|
||||
gst_element_class_set_details_simple (element_class, "Opus audio encoder",
|
||||
gst_element_class_set_details_simple (gstelement_class, "Opus audio encoder",
|
||||
"Codec/Encoder/Audio",
|
||||
"Encodes audio in Opus format",
|
||||
"Sebastian Dröge <sebastian.droege@collabora.co.uk>");
|
||||
|
@ -290,7 +288,7 @@ gst_opus_enc_finalize (GObject * object)
|
|||
}
|
||||
|
||||
static void
|
||||
gst_opus_enc_init (GstOpusEnc * enc, GstOpusEncClass * klass)
|
||||
gst_opus_enc_init (GstOpusEnc * enc)
|
||||
{
|
||||
GstAudioEncoder *benc = GST_AUDIO_ENCODER (enc);
|
||||
|
||||
|
@ -324,7 +322,7 @@ gst_opus_enc_start (GstAudioEncoder * benc)
|
|||
GstOpusEnc *enc = GST_OPUS_ENC (benc);
|
||||
|
||||
GST_DEBUG_OBJECT (enc, "start");
|
||||
enc->tags = gst_tag_list_new ();
|
||||
enc->tags = gst_tag_list_new_empty ();
|
||||
enc->header_sent = FALSE;
|
||||
|
||||
return TRUE;
|
||||
|
@ -338,7 +336,7 @@ gst_opus_enc_stop (GstAudioEncoder * benc)
|
|||
GST_DEBUG_OBJECT (enc, "stop");
|
||||
enc->header_sent = FALSE;
|
||||
if (enc->state) {
|
||||
opus_encoder_destroy (enc->state);
|
||||
opus_multistream_encoder_destroy (enc->state);
|
||||
enc->state = NULL;
|
||||
}
|
||||
gst_tag_list_free (enc->tags);
|
||||
|
@ -402,6 +400,82 @@ gst_opus_enc_get_frame_samples (GstOpusEnc * enc)
|
|||
return frame_samples;
|
||||
}
|
||||
|
||||
static void
|
||||
gst_opus_enc_setup_channel_mapping (GstOpusEnc * enc, const GstAudioInfo * info)
|
||||
{
|
||||
#define MAPS(idx,pos) (GST_AUDIO_INFO_POSITION (info, (idx)) == GST_AUDIO_CHANNEL_POSITION_##pos)
|
||||
|
||||
int n;
|
||||
|
||||
GST_DEBUG_OBJECT (enc, "Setting up channel mapping for %d channels",
|
||||
enc->n_channels);
|
||||
|
||||
/* Start by setting up a default trivial mapping */
|
||||
for (n = 0; n < 255; ++n)
|
||||
enc->channel_mapping[n] = n;
|
||||
|
||||
/* For one channel, use the basic RTP mapping */
|
||||
if (enc->n_channels == 1) {
|
||||
GST_INFO_OBJECT (enc, "Mono, trivial RTP mapping");
|
||||
enc->channel_mapping_family = 0;
|
||||
enc->channel_mapping[0] = 0;
|
||||
return;
|
||||
}
|
||||
|
||||
/* For two channels, use the basic RTP mapping if the channels are
|
||||
mapped as left/right. */
|
||||
if (enc->n_channels == 2) {
|
||||
if (MAPS (0, FRONT_LEFT) && MAPS (1, FRONT_RIGHT)) {
|
||||
GST_INFO_OBJECT (enc, "Stereo, canonical mapping");
|
||||
enc->channel_mapping_family = 0;
|
||||
/* The channel mapping is implicit for family 0, that's why we do not
|
||||
attempt to create one for right/left - this will be mapped to the
|
||||
Vorbis mapping below. */
|
||||
} else {
|
||||
GST_DEBUG_OBJECT (enc, "Stereo, but not canonical mapping, continuing");
|
||||
}
|
||||
}
|
||||
|
||||
/* For channels between 1 and 8, we use the Vorbis mapping if we can
|
||||
find a permutation that matches it. Mono will have been taken care
|
||||
of earlier, but this code also handles it. */
|
||||
if (enc->n_channels >= 1 && enc->n_channels <= 8) {
|
||||
GST_DEBUG_OBJECT (enc,
|
||||
"In range for the Vorbis mapping, checking channel positions");
|
||||
for (n = 0; n < enc->n_channels; ++n) {
|
||||
GstAudioChannelPosition pos = GST_AUDIO_INFO_POSITION (info, n);
|
||||
int c;
|
||||
|
||||
GST_DEBUG_OBJECT (enc, "Channel %d has position %d", n, pos);
|
||||
for (c = 0; c < enc->n_channels; ++c) {
|
||||
if (gst_opus_channel_positions[enc->n_channels - 1][c] == pos) {
|
||||
GST_DEBUG_OBJECT (enc, "Found in Vorbis mapping as channel %d", c);
|
||||
break;
|
||||
}
|
||||
}
|
||||
if (c == enc->n_channels) {
|
||||
/* We did not find that position, so use undefined */
|
||||
GST_WARNING_OBJECT (enc,
|
||||
"Position %d not found in Vorbis mapping, using unknown mapping",
|
||||
pos);
|
||||
enc->channel_mapping_family = 255;
|
||||
return;
|
||||
}
|
||||
GST_DEBUG_OBJECT (enc, "Mapping output channel %d to %d", c, n);
|
||||
enc->channel_mapping[c] = n;
|
||||
}
|
||||
GST_INFO_OBJECT (enc, "Permutation found, using Vorbis mapping");
|
||||
enc->channel_mapping_family = 1;
|
||||
return;
|
||||
}
|
||||
|
||||
/* For other cases, we use undefined, with the default trivial mapping */
|
||||
GST_WARNING_OBJECT (enc, "Unknown mapping");
|
||||
enc->channel_mapping_family = 255;
|
||||
|
||||
#undef MAPS
|
||||
}
|
||||
|
||||
static gboolean
|
||||
gst_opus_enc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
|
||||
{
|
||||
|
@ -413,12 +487,13 @@ gst_opus_enc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
|
|||
|
||||
enc->n_channels = GST_AUDIO_INFO_CHANNELS (info);
|
||||
enc->sample_rate = GST_AUDIO_INFO_RATE (info);
|
||||
gst_opus_enc_setup_channel_mapping (enc, info);
|
||||
GST_DEBUG_OBJECT (benc, "Setup with %d channels, %d Hz", enc->n_channels,
|
||||
enc->sample_rate);
|
||||
|
||||
/* handle reconfigure */
|
||||
if (enc->state) {
|
||||
opus_encoder_destroy (enc->state);
|
||||
opus_multistream_encoder_destroy (enc->state);
|
||||
enc->state = NULL;
|
||||
}
|
||||
if (!gst_opus_enc_setup (enc))
|
||||
|
@ -441,29 +516,30 @@ gst_opus_enc_setup (GstOpusEnc * enc)
|
|||
|
||||
GST_DEBUG_OBJECT (enc, "setup");
|
||||
|
||||
enc->setup = FALSE;
|
||||
|
||||
enc->state = opus_encoder_create (enc->sample_rate, enc->n_channels,
|
||||
enc->state =
|
||||
opus_multistream_encoder_create (enc->sample_rate, enc->n_channels,
|
||||
(enc->n_channels + 1) / 2, enc->n_channels / 2, enc->channel_mapping,
|
||||
enc->audio_or_voip ? OPUS_APPLICATION_AUDIO : OPUS_APPLICATION_VOIP,
|
||||
&error);
|
||||
if (!enc->state || error != OPUS_OK)
|
||||
goto encoder_creation_failed;
|
||||
|
||||
opus_encoder_ctl (enc->state, OPUS_SET_BITRATE (enc->bitrate), 0);
|
||||
opus_encoder_ctl (enc->state, OPUS_SET_BANDWIDTH (enc->bandwidth), 0);
|
||||
opus_encoder_ctl (enc->state, OPUS_SET_VBR (!enc->cbr), 0);
|
||||
opus_encoder_ctl (enc->state, OPUS_SET_VBR_CONSTRAINT (enc->constrained_vbr),
|
||||
opus_multistream_encoder_ctl (enc->state, OPUS_SET_BITRATE (enc->bitrate), 0);
|
||||
opus_multistream_encoder_ctl (enc->state, OPUS_SET_BANDWIDTH (enc->bandwidth),
|
||||
0);
|
||||
opus_encoder_ctl (enc->state, OPUS_SET_COMPLEXITY (enc->complexity), 0);
|
||||
opus_encoder_ctl (enc->state, OPUS_SET_INBAND_FEC (enc->inband_fec), 0);
|
||||
opus_encoder_ctl (enc->state, OPUS_SET_DTX (enc->dtx), 0);
|
||||
opus_encoder_ctl (enc->state,
|
||||
opus_multistream_encoder_ctl (enc->state, OPUS_SET_VBR (!enc->cbr), 0);
|
||||
opus_multistream_encoder_ctl (enc->state,
|
||||
OPUS_SET_VBR_CONSTRAINT (enc->constrained_vbr), 0);
|
||||
opus_multistream_encoder_ctl (enc->state,
|
||||
OPUS_SET_COMPLEXITY (enc->complexity), 0);
|
||||
opus_multistream_encoder_ctl (enc->state,
|
||||
OPUS_SET_INBAND_FEC (enc->inband_fec), 0);
|
||||
opus_multistream_encoder_ctl (enc->state, OPUS_SET_DTX (enc->dtx), 0);
|
||||
opus_multistream_encoder_ctl (enc->state,
|
||||
OPUS_SET_PACKET_LOSS_PERC (enc->packet_loss_percentage), 0);
|
||||
|
||||
GST_LOG_OBJECT (enc, "we have frame size %d", enc->frame_size);
|
||||
|
||||
enc->setup = TRUE;
|
||||
|
||||
return TRUE;
|
||||
|
||||
encoder_creation_failed:
|
||||
|
@ -501,7 +577,7 @@ gst_opus_enc_sink_event (GstAudioEncoder * benc, GstEvent * event)
|
|||
static GstFlowReturn
|
||||
gst_opus_enc_encode (GstOpusEnc * enc, GstBuffer * buf)
|
||||
{
|
||||
guint8 *bdata, *data, *mdata = NULL;
|
||||
guint8 *bdata = NULL, *data, *mdata = NULL;
|
||||
gsize bsize, size;
|
||||
gsize bytes = enc->frame_samples * enc->n_channels * 2;
|
||||
gint ret = GST_FLOW_OK;
|
||||
|
@ -535,17 +611,17 @@ gst_opus_enc_encode (GstOpusEnc * enc, GstBuffer * buf)
|
|||
gsize out_size;
|
||||
GstBuffer *outbuf;
|
||||
|
||||
outbuf = gst_buffer_new_and_alloc (enc->max_payload_size);
|
||||
outbuf = gst_buffer_new_and_alloc (enc->max_payload_size * enc->n_channels);
|
||||
if (!outbuf)
|
||||
goto done;
|
||||
|
||||
GST_DEBUG_OBJECT (enc, "encoding %d samples (%d bytes)",
|
||||
enc->frame_samples);
|
||||
enc->frame_samples, (int) bytes);
|
||||
|
||||
out_data = gst_buffer_map (outbuf, &out_size, NULL, GST_MAP_WRITE);
|
||||
encoded_size =
|
||||
opus_encode (enc->state, (const gint16 *) data, enc->frame_samples,
|
||||
out_data, enc->max_payload_size);
|
||||
opus_multistream_encode (enc->state, (const gint16 *) data,
|
||||
enc->frame_samples, out_data, enc->max_payload_size * enc->n_channels);
|
||||
gst_buffer_unmap (outbuf, out_data, out_size);
|
||||
|
||||
if (encoded_size < 0) {
|
||||
|
@ -555,12 +631,13 @@ gst_opus_enc_encode (GstOpusEnc * enc, GstBuffer * buf)
|
|||
} else if (encoded_size > enc->max_payload_size) {
|
||||
GST_WARNING_OBJECT (enc,
|
||||
"Encoded size %d is higher than max payload size (%d bytes)",
|
||||
outsize, enc->max_payload_size);
|
||||
out_size, enc->max_payload_size);
|
||||
ret = GST_FLOW_ERROR;
|
||||
goto done;
|
||||
}
|
||||
|
||||
GST_BUFFER_SIZE (outbuf) = outsize;
|
||||
GST_DEBUG_OBJECT (enc, "Output packet is %u bytes", encoded_size);
|
||||
gst_buffer_set_size (outbuf, encoded_size);
|
||||
|
||||
ret =
|
||||
gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (enc), outbuf,
|
||||
|
@ -601,7 +678,8 @@ gst_opus_enc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf)
|
|||
enc->headers = NULL;
|
||||
|
||||
gst_opus_header_create_caps (&caps, &enc->headers, enc->n_channels,
|
||||
enc->sample_rate, gst_tag_setter_get_tag_list (GST_TAG_SETTER (enc)));
|
||||
enc->sample_rate, enc->channel_mapping_family, enc->channel_mapping,
|
||||
gst_tag_setter_get_tag_list (GST_TAG_SETTER (enc)));
|
||||
|
||||
|
||||
/* negotiate with these caps */
|
||||
|
@ -613,7 +691,7 @@ gst_opus_enc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf)
|
|||
}
|
||||
|
||||
GST_DEBUG_OBJECT (enc, "received buffer %p of %u bytes", buf,
|
||||
buf ? GST_BUFFER_SIZE (buf) : 0);
|
||||
buf ? gst_buffer_get_size (buf) : 0);
|
||||
|
||||
ret = gst_opus_enc_encode (enc, buf);
|
||||
|
||||
|
@ -684,7 +762,7 @@ gst_opus_enc_set_property (GObject * object, guint prop_id,
|
|||
g_mutex_lock (enc->property_lock); \
|
||||
enc->prop = g_value_get_##type (value); \
|
||||
if (enc->state) { \
|
||||
opus_encoder_ctl (enc->state, OPUS_SET_##ctl (enc->prop)); \
|
||||
opus_multistream_encoder_ctl (enc->state, OPUS_SET_##ctl (enc->prop)); \
|
||||
} \
|
||||
g_mutex_unlock (enc->property_lock); \
|
||||
} while(0)
|
||||
|
@ -710,7 +788,7 @@ gst_opus_enc_set_property (GObject * object, guint prop_id,
|
|||
/* this one has an opposite meaning to the opus ctl... */
|
||||
g_mutex_lock (enc->property_lock);
|
||||
enc->cbr = g_value_get_boolean (value);
|
||||
opus_encoder_ctl (enc->state, OPUS_SET_VBR (!enc->cbr));
|
||||
opus_multistream_encoder_ctl (enc->state, OPUS_SET_VBR (!enc->cbr));
|
||||
g_mutex_unlock (enc->property_lock);
|
||||
break;
|
||||
case PROP_CONSTRAINED_VBR:
|
||||
|
|
|
@ -26,7 +26,7 @@
|
|||
#include <gst/gst.h>
|
||||
#include <gst/audio/gstaudioencoder.h>
|
||||
|
||||
#include <opus/opus.h>
|
||||
#include <opus/opus_multistream.h>
|
||||
|
||||
G_BEGIN_DECLS
|
||||
|
||||
|
@ -50,7 +50,7 @@ typedef struct _GstOpusEncClass GstOpusEncClass;
|
|||
struct _GstOpusEnc {
|
||||
GstAudioEncoder element;
|
||||
|
||||
OpusEncoder *state;
|
||||
OpusMSEncoder *state;
|
||||
|
||||
/* Locks those properties which may be changed at play time */
|
||||
GMutex *property_lock;
|
||||
|
@ -72,12 +72,14 @@ struct _GstOpusEnc {
|
|||
gint n_channels;
|
||||
gint sample_rate;
|
||||
|
||||
gboolean setup;
|
||||
gboolean header_sent;
|
||||
|
||||
GSList *headers;
|
||||
|
||||
GstTagList *tags;
|
||||
|
||||
guint8 channel_mapping_family;
|
||||
guint8 channel_mapping[256];
|
||||
};
|
||||
|
||||
struct _GstOpusEncClass {
|
||||
|
|
|
@ -27,7 +27,8 @@
|
|||
#include "gstopusheader.h"
|
||||
|
||||
static GstBuffer *
|
||||
gst_opus_enc_create_id_buffer (gint nchannels, gint sample_rate)
|
||||
gst_opus_enc_create_id_buffer (gint nchannels, gint sample_rate,
|
||||
guint8 channel_mapping_family, const guint8 * channel_mapping)
|
||||
{
|
||||
GstBuffer *buffer;
|
||||
GstByteWriter bw;
|
||||
|
@ -38,10 +39,15 @@ gst_opus_enc_create_id_buffer (gint nchannels, gint sample_rate)
|
|||
gst_byte_writer_put_data (&bw, (const guint8 *) "OpusHead", 8);
|
||||
gst_byte_writer_put_uint8 (&bw, 0); /* version number */
|
||||
gst_byte_writer_put_uint8 (&bw, nchannels);
|
||||
gst_byte_writer_put_uint16_le (&bw, 0); /* pre-skip *//* TODO: endianness ? */
|
||||
gst_byte_writer_put_uint16_le (&bw, 0); /* pre-skip */
|
||||
gst_byte_writer_put_uint32_le (&bw, sample_rate);
|
||||
gst_byte_writer_put_uint16_le (&bw, 0); /* output gain *//* TODO: endianness ? */
|
||||
gst_byte_writer_put_uint8 (&bw, 0); /* channel mapping *//* TODO: what is this ? */
|
||||
gst_byte_writer_put_uint16_le (&bw, 0); /* output gain */
|
||||
gst_byte_writer_put_uint8 (&bw, channel_mapping_family);
|
||||
if (channel_mapping_family > 0) {
|
||||
gst_byte_writer_put_uint8 (&bw, (nchannels + 1) / 2);
|
||||
gst_byte_writer_put_uint8 (&bw, nchannels / 2);
|
||||
gst_byte_writer_put_data (&bw, channel_mapping, nchannels);
|
||||
}
|
||||
|
||||
buffer = gst_byte_writer_reset_and_get_buffer (&bw);
|
||||
|
||||
|
@ -62,7 +68,7 @@ gst_opus_enc_create_metadata_buffer (const GstTagList * tags)
|
|||
if (tags == NULL) {
|
||||
/* FIXME: better fix chain of callers to not write metadata at all,
|
||||
* if there is none */
|
||||
empty_tags = gst_tag_list_new ();
|
||||
empty_tags = gst_tag_list_new_empty ();
|
||||
tags = empty_tags;
|
||||
}
|
||||
comments =
|
||||
|
@ -136,23 +142,11 @@ _gst_caps_set_buffer_array (GstCaps * caps, const gchar * field,
|
|||
}
|
||||
|
||||
void
|
||||
gst_opus_header_create_caps (GstCaps ** caps, GSList ** headers, gint nchannels,
|
||||
gint sample_rate, const GstTagList * tags)
|
||||
gst_opus_header_create_caps_from_headers (GstCaps ** caps, GSList ** headers,
|
||||
GstBuffer * buf1, GstBuffer * buf2)
|
||||
{
|
||||
GstBuffer *buf1, *buf2;
|
||||
|
||||
g_return_if_fail (caps);
|
||||
g_return_if_fail (headers && !*headers);
|
||||
g_return_if_fail (nchannels > 0);
|
||||
g_return_if_fail (sample_rate >= 0); /* 0 -> unset */
|
||||
|
||||
/* Opus streams in Ogg begin with two headers; the initial header (with
|
||||
most of the codec setup parameters) which is mandated by the Ogg
|
||||
bitstream spec. The second header holds any comment fields. */
|
||||
|
||||
/* create header buffers */
|
||||
buf1 = gst_opus_enc_create_id_buffer (nchannels, sample_rate);
|
||||
buf2 = gst_opus_enc_create_metadata_buffer (tags);
|
||||
|
||||
/* mark and put on caps */
|
||||
*caps = gst_caps_from_string ("audio/x-opus");
|
||||
|
@ -162,9 +156,87 @@ gst_opus_header_create_caps (GstCaps ** caps, GSList ** headers, gint nchannels,
|
|||
*headers = g_slist_prepend (*headers, buf1);
|
||||
}
|
||||
|
||||
void
|
||||
gst_opus_header_create_caps (GstCaps ** caps, GSList ** headers, gint nchannels,
|
||||
gint sample_rate, guint8 channel_mapping_family,
|
||||
const guint8 * channel_mapping, const GstTagList * tags)
|
||||
{
|
||||
GstBuffer *buf1, *buf2;
|
||||
|
||||
g_return_if_fail (caps);
|
||||
g_return_if_fail (headers && !*headers);
|
||||
g_return_if_fail (nchannels > 0);
|
||||
g_return_if_fail (sample_rate >= 0); /* 0 -> unset */
|
||||
g_return_if_fail (channel_mapping_family == 0 || channel_mapping);
|
||||
|
||||
/* Opus streams in Ogg begin with two headers; the initial header (with
|
||||
most of the codec setup parameters) which is mandated by the Ogg
|
||||
bitstream spec. The second header holds any comment fields. */
|
||||
|
||||
/* create header buffers */
|
||||
buf1 =
|
||||
gst_opus_enc_create_id_buffer (nchannels, sample_rate,
|
||||
channel_mapping_family, channel_mapping);
|
||||
buf2 = gst_opus_enc_create_metadata_buffer (tags);
|
||||
|
||||
gst_opus_header_create_caps_from_headers (caps, headers, buf1, buf2);
|
||||
}
|
||||
|
||||
gboolean
|
||||
gst_opus_header_is_header (GstBuffer * buf, const char *magic, guint magic_size)
|
||||
{
|
||||
return (GST_BUFFER_SIZE (buf) >= magic_size
|
||||
&& !memcmp (magic, GST_BUFFER_DATA (buf), magic_size));
|
||||
return (gst_buffer_get_size (buf) >= magic_size
|
||||
&& !gst_buffer_memcmp (buf, 0, magic, magic_size));
|
||||
}
|
||||
|
||||
gboolean
|
||||
gst_opus_header_is_id_header (GstBuffer * buf)
|
||||
{
|
||||
gsize size = gst_buffer_get_size (buf);
|
||||
guint8 *data = NULL;
|
||||
guint8 channels, channel_mapping_family, n_streams, n_stereo_streams;
|
||||
gboolean ret = FALSE;
|
||||
|
||||
if (size < 19)
|
||||
goto beach;
|
||||
if (!gst_opus_header_is_header (buf, "OpusHead", 8))
|
||||
goto beach;
|
||||
|
||||
data = gst_buffer_map (buf, &size, NULL, GST_MAP_READ);
|
||||
|
||||
channels = data[9];
|
||||
|
||||
if (channels == 0)
|
||||
goto beach;
|
||||
|
||||
channel_mapping_family = data[18];
|
||||
|
||||
if (channel_mapping_family == 0) {
|
||||
if (channels > 2)
|
||||
goto beach;
|
||||
} else {
|
||||
channels = data[9];
|
||||
if (size < 21 + channels)
|
||||
goto beach;
|
||||
n_streams = data[19];
|
||||
n_stereo_streams = data[20];
|
||||
if (n_streams == 0)
|
||||
goto beach;
|
||||
if (n_stereo_streams > n_streams)
|
||||
goto beach;
|
||||
if (n_streams + n_stereo_streams > 255)
|
||||
goto beach;
|
||||
}
|
||||
ret = TRUE;
|
||||
|
||||
beach:
|
||||
if (data)
|
||||
gst_buffer_unmap (buf, data, size);
|
||||
return ret;
|
||||
}
|
||||
|
||||
gboolean
|
||||
gst_opus_header_is_comment_header (GstBuffer * buf)
|
||||
{
|
||||
return gst_opus_header_is_header (buf, "OpusTags", 8);
|
||||
}
|
||||
|
|
|
@ -25,8 +25,16 @@
|
|||
|
||||
G_BEGIN_DECLS
|
||||
|
||||
extern void gst_opus_header_create_caps (GstCaps **caps, GSList **headers, gint nchannels, gint sample_rate, const GstTagList *tags);
|
||||
extern gboolean gst_opus_header_is_header (GstBuffer * buf, const char *magic, guint magic_size);
|
||||
extern void gst_opus_header_create_caps_from_headers (GstCaps **caps, GSList **headers,
|
||||
GstBuffer *id_header, GstBuffer *comment_header);
|
||||
extern void gst_opus_header_create_caps (GstCaps **caps, GSList **headers,
|
||||
gint nchannels, gint sample_rate,
|
||||
guint8 channel_mapping_family, const guint8 *channel_mapping,
|
||||
const GstTagList *tags);
|
||||
extern gboolean gst_opus_header_is_header (GstBuffer * buf,
|
||||
const char *magic, guint magic_size);
|
||||
extern gboolean gst_opus_header_is_id_header (GstBuffer * buf);
|
||||
extern gboolean gst_opus_header_is_comment_header (GstBuffer * buf);
|
||||
|
||||
|
||||
G_END_DECLS
|
||||
|
|
Loading…
Reference in a new issue