mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-22 16:26:39 +00:00
4df2ffaad6
The first frame has lookahead less samples, the last frame might have some padding or we might have to encode another frame of silence to get all our input into the encoded data. This is because of a) the lookahead at the beginning of the encoding, which shifts all data by that amount of samples and b) the padding needed to fill the very last frame completely. Ideally we would use LPC to calculate something better than silence for the padding to make the encoding as smooth as possible. With this we get exactly the same amount of samples again in an opusenc ! opusdec pipeline. https://bugzilla.gnome.org/show_bug.cgi?id=757153
102 lines
3 KiB
C
102 lines
3 KiB
C
/* GStreamer Opus Encoder
|
|
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
|
|
* Copyright (C) <2008> Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
* Copyright (C) <2011-2012> Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
|
|
*
|
|
* 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., 51 Franklin St, Fifth Floor,
|
|
* Boston, MA 02110-1301, USA.
|
|
*/
|
|
|
|
|
|
#ifndef __GST_OPUS_ENC_H__
|
|
#define __GST_OPUS_ENC_H__
|
|
|
|
|
|
#include <gst/gst.h>
|
|
#include <gst/audio/gstaudioencoder.h>
|
|
|
|
#include <opus_multistream.h>
|
|
|
|
G_BEGIN_DECLS
|
|
|
|
#define GST_TYPE_OPUS_ENC \
|
|
(gst_opus_enc_get_type())
|
|
#define GST_OPUS_ENC(obj) \
|
|
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_OPUS_ENC,GstOpusEnc))
|
|
#define GST_OPUS_ENC_CLASS(klass) \
|
|
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_OPUS_ENC,GstOpusEncClass))
|
|
#define GST_IS_OPUS_ENC(obj) \
|
|
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_OPUS_ENC))
|
|
#define GST_IS_OPUS_ENC_CLASS(klass) \
|
|
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_OPUS_ENC))
|
|
|
|
#define MAX_FRAME_SIZE 2000*2
|
|
#define MAX_FRAME_BYTES 2000
|
|
|
|
typedef enum
|
|
{
|
|
BITRATE_TYPE_CBR,
|
|
BITRATE_TYPE_VBR,
|
|
BITRATE_TYPE_CONSTRAINED_VBR,
|
|
} GstOpusEncBitrateType;
|
|
|
|
typedef struct _GstOpusEnc GstOpusEnc;
|
|
typedef struct _GstOpusEncClass GstOpusEncClass;
|
|
|
|
struct _GstOpusEnc {
|
|
GstAudioEncoder element;
|
|
|
|
OpusMSEncoder *state;
|
|
|
|
/* Locks those properties which may be changed at play time */
|
|
GMutex property_lock;
|
|
|
|
/* properties */
|
|
gint audio_type;
|
|
gint bitrate;
|
|
gint bandwidth;
|
|
gint frame_size;
|
|
GstOpusEncBitrateType bitrate_type;
|
|
gint complexity;
|
|
gboolean inband_fec;
|
|
gboolean dtx;
|
|
gint packet_loss_percentage;
|
|
guint max_payload_size;
|
|
|
|
gint frame_samples;
|
|
gint n_channels;
|
|
gint sample_rate;
|
|
|
|
guint64 encoded_samples, consumed_samples;
|
|
guint16 lookahead, pending_lookahead;
|
|
|
|
guint8 channel_mapping_family;
|
|
guint8 encoding_channel_mapping[256];
|
|
guint8 decoding_channel_mapping[256];
|
|
guint8 n_stereo_streams;
|
|
};
|
|
|
|
struct _GstOpusEncClass {
|
|
GstAudioEncoderClass parent_class;
|
|
|
|
/* signals */
|
|
void (*frame_encoded) (GstElement *element);
|
|
};
|
|
|
|
GType gst_opus_enc_get_type (void);
|
|
|
|
G_END_DECLS
|
|
|
|
#endif /* __GST_OPUS_ENC_H__ */
|