gstreamer/ext/opus/gstopusenc.c
Vincent Penquerc'h 15b5fecf30 opusenc: do not include variable fields in caps
Those can vary from one packet to the next, so have no reason
to be in the caps.
2011-11-16 18:51:16 +00:00

810 lines
24 KiB
C

/* GStreamer Opus Encoder
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
* Copyright (C) <2008> Sebastian Dröge <sebastian.droege@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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/*
* Based on the speexenc element
*/
/**
* SECTION:element-opusenc
* @see_also: opusdec, oggmux
*
* This element encodes raw audio to OPUS.
*
* <refsect2>
* <title>Example pipelines</title>
* |[
* gst-launch -v audiotestsrc wave=sine num-buffers=100 ! audioconvert ! opusenc ! oggmux ! filesink location=sine.ogg
* ]| Encode a test sine signal to Ogg/OPUS.
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <stdlib.h>
#include <string.h>
#include <time.h>
#include <math.h>
#include <opus/opus.h>
#include <gst/gsttagsetter.h>
#include <gst/tag/tag.h>
#include <gst/base/gstbytewriter.h>
#include <gst/audio/audio.h>
#include "gstopusenc.h"
GST_DEBUG_CATEGORY_STATIC (opusenc_debug);
#define GST_CAT_DEFAULT opusenc_debug
#define GST_OPUS_ENC_TYPE_BANDWIDTH (gst_opus_enc_bandwidth_get_type())
static GType
gst_opus_enc_bandwidth_get_type (void)
{
static const GEnumValue values[] = {
{OPUS_BANDWIDTH_NARROWBAND, "Narrow band", "narrowband"},
{OPUS_BANDWIDTH_MEDIUMBAND, "Medium band", "mediumband"},
{OPUS_BANDWIDTH_WIDEBAND, "Wide band", "wideband"},
{OPUS_BANDWIDTH_SUPERWIDEBAND, "Super wide band", "superwideband"},
{OPUS_BANDWIDTH_FULLBAND, "Full band", "fullband"},
{OPUS_AUTO, "Auto", "auto"},
{0, NULL, NULL}
};
static volatile GType id = 0;
if (g_once_init_enter ((gsize *) & id)) {
GType _id;
_id = g_enum_register_static ("GstOpusEncBandwidth", values);
g_once_init_leave ((gsize *) & id, _id);
}
return id;
}
static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"rate = (int) { 8000, 12000, 16000, 24000, 48000 }, "
"channels = (int) [ 1, 2 ], "
"endianness = (int) BYTE_ORDER, "
"signed = (boolean) TRUE, " "width = (int) 16, " "depth = (int) 16")
);
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-opus")
);
#define DEFAULT_AUDIO TRUE
#define DEFAULT_BITRATE 64000
#define DEFAULT_BANDWIDTH OPUS_BANDWIDTH_FULLBAND
#define DEFAULT_FRAMESIZE 20
#define DEFAULT_CBR TRUE
#define DEFAULT_CONSTRAINED_VBR TRUE
#define DEFAULT_COMPLEXITY 10
#define DEFAULT_INBAND_FEC FALSE
#define DEFAULT_DTX FALSE
#define DEFAULT_PACKET_LOSS_PERCENT 0
enum
{
PROP_0,
PROP_AUDIO,
PROP_BITRATE,
PROP_BANDWIDTH,
PROP_FRAME_SIZE,
PROP_CBR,
PROP_CONSTRAINED_VBR,
PROP_COMPLEXITY,
PROP_INBAND_FEC,
PROP_DTX,
PROP_PACKET_LOSS_PERCENT
};
static void gst_opus_enc_finalize (GObject * object);
static gboolean gst_opus_enc_sink_event (GstAudioEncoder * benc,
GstEvent * event);
static gboolean gst_opus_enc_setup (GstOpusEnc * enc);
static void gst_opus_enc_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static void gst_opus_enc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static gboolean gst_opus_enc_start (GstAudioEncoder * benc);
static gboolean gst_opus_enc_stop (GstAudioEncoder * benc);
static gboolean gst_opus_enc_set_format (GstAudioEncoder * benc,
GstAudioInfo * info);
static GstFlowReturn gst_opus_enc_handle_frame (GstAudioEncoder * benc,
GstBuffer * buf);
static GstFlowReturn gst_opus_enc_pre_push (GstAudioEncoder * benc,
GstBuffer ** buffer);
static gint64 gst_opus_enc_get_latency (GstOpusEnc * enc);
static GstFlowReturn gst_opus_enc_encode (GstOpusEnc * enc, GstBuffer * buffer);
static void
gst_opus_enc_setup_interfaces (GType opusenc_type)
{
static const GInterfaceInfo tag_setter_info = { NULL, NULL, NULL };
const GInterfaceInfo preset_interface_info = {
NULL, /* interface_init */
NULL, /* interface_finalize */
NULL /* interface_data */
};
g_type_add_interface_static (opusenc_type, GST_TYPE_TAG_SETTER,
&tag_setter_info);
g_type_add_interface_static (opusenc_type, GST_TYPE_PRESET,
&preset_interface_info);
GST_DEBUG_CATEGORY_INIT (opusenc_debug, "opusenc", 0, "Opus encoder");
}
GST_BOILERPLATE_FULL (GstOpusEnc, gst_opus_enc, GstAudioEncoder,
GST_TYPE_AUDIO_ENCODER, gst_opus_enc_setup_interfaces);
static void
gst_opus_enc_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&src_factory));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sink_factory));
gst_element_class_set_details_simple (element_class, "Opus audio encoder",
"Codec/Encoder/Audio",
"Encodes audio in Opus format",
"Sebastian Dröge <sebastian.droege@collabora.co.uk>");
}
static void
gst_opus_enc_class_init (GstOpusEncClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstAudioEncoderClass *base_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
base_class = (GstAudioEncoderClass *) klass;
gobject_class->set_property = gst_opus_enc_set_property;
gobject_class->get_property = gst_opus_enc_get_property;
base_class->start = GST_DEBUG_FUNCPTR (gst_opus_enc_start);
base_class->stop = GST_DEBUG_FUNCPTR (gst_opus_enc_stop);
base_class->set_format = GST_DEBUG_FUNCPTR (gst_opus_enc_set_format);
base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_opus_enc_handle_frame);
base_class->pre_push = GST_DEBUG_FUNCPTR (gst_opus_enc_pre_push);
base_class->event = GST_DEBUG_FUNCPTR (gst_opus_enc_sink_event);
g_object_class_install_property (gobject_class, PROP_AUDIO,
g_param_spec_boolean ("audio", "Audio or voice",
"Audio or voice", DEFAULT_AUDIO,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_BITRATE,
g_param_spec_int ("bitrate", "Encoding Bit-rate",
"Specify an encoding bit-rate (in bps).",
1, 320000, DEFAULT_BITRATE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_BANDWIDTH,
g_param_spec_enum ("bandwidth", "Band Width",
"Audio Band Width", GST_OPUS_ENC_TYPE_BANDWIDTH, DEFAULT_BANDWIDTH,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_FRAME_SIZE,
g_param_spec_int ("frame-size", "Frame Size",
"The duration of an audio frame, in ms", 2, 60, DEFAULT_FRAMESIZE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_CBR,
g_param_spec_boolean ("cbr", "Constant bit rate",
"Constant bit rate", DEFAULT_CBR,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_CONSTRAINED_VBR,
g_param_spec_boolean ("constrained-vbr", "Constrained VBR",
"Constrained VBR", DEFAULT_CONSTRAINED_VBR,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_COMPLEXITY,
g_param_spec_int ("complexity", "Complexity",
"Complexity", 0, 10, DEFAULT_COMPLEXITY,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_INBAND_FEC,
g_param_spec_boolean ("inband-fec", "In-band FEC",
"Enable forward error correction", DEFAULT_INBAND_FEC,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_DTX,
g_param_spec_boolean ("dtx", "DTX",
"DTX", DEFAULT_DTX, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass),
PROP_PACKET_LOSS_PERCENT, g_param_spec_int ("packet-loss-percentage",
"Loss percentage", "Packet loss percentage", 0, 100,
DEFAULT_PACKET_LOSS_PERCENT,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_opus_enc_finalize);
}
static void
gst_opus_enc_finalize (GObject * object)
{
GstOpusEnc *enc;
enc = GST_OPUS_ENC (object);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_opus_enc_init (GstOpusEnc * enc, GstOpusEncClass * klass)
{
GstAudioEncoder *benc = GST_AUDIO_ENCODER (enc);
GST_DEBUG_OBJECT (enc, "init");
enc->n_channels = -1;
enc->sample_rate = -1;
enc->frame_samples = 0;
enc->bitrate = DEFAULT_BITRATE;
enc->bandwidth = DEFAULT_BANDWIDTH;
enc->frame_size = DEFAULT_FRAMESIZE;
enc->cbr = DEFAULT_CBR;
enc->constrained_vbr = DEFAULT_CONSTRAINED_VBR;
enc->complexity = DEFAULT_COMPLEXITY;
enc->inband_fec = DEFAULT_INBAND_FEC;
enc->dtx = DEFAULT_DTX;
enc->packet_loss_percentage = DEFAULT_PACKET_LOSS_PERCENT;
/* arrange granulepos marking (and required perfect ts) */
gst_audio_encoder_set_mark_granule (benc, TRUE);
gst_audio_encoder_set_perfect_timestamp (benc, TRUE);
}
static gboolean
gst_opus_enc_start (GstAudioEncoder * benc)
{
GstOpusEnc *enc = GST_OPUS_ENC (benc);
GST_DEBUG_OBJECT (enc, "start");
enc->tags = gst_tag_list_new ();
enc->header_sent = FALSE;
return TRUE;
}
static gboolean
gst_opus_enc_stop (GstAudioEncoder * benc)
{
GstOpusEnc *enc = GST_OPUS_ENC (benc);
GST_DEBUG_OBJECT (enc, "stop");
enc->header_sent = FALSE;
if (enc->state) {
opus_encoder_destroy (enc->state);
enc->state = NULL;
}
gst_tag_list_free (enc->tags);
enc->tags = NULL;
g_slist_foreach (enc->headers, (GFunc) gst_buffer_unref, NULL);
enc->headers = NULL;
return TRUE;
}
static gint64
gst_opus_enc_get_latency (GstOpusEnc * enc)
{
gint64 latency = gst_util_uint64_scale (enc->frame_samples, GST_SECOND,
enc->sample_rate);
GST_DEBUG_OBJECT (enc, "Latency: %" GST_TIME_FORMAT, GST_TIME_ARGS (latency));
return latency;
}
static gint
gst_opus_enc_get_frame_samples (GstOpusEnc * enc)
{
gint frame_samples = 0;
switch (enc->frame_size) {
case 2:
frame_samples = enc->sample_rate / 400;
break;
case 5:
frame_samples = enc->sample_rate / 200;
break;
case 10:
frame_samples = enc->sample_rate / 100;
break;
case 20:
frame_samples = enc->sample_rate / 50;
break;
case 40:
frame_samples = enc->sample_rate / 25;
break;
case 60:
frame_samples = 3 * enc->sample_rate / 50;
break;
default:
GST_WARNING_OBJECT (enc, "Unsupported frame size: %d", enc->frame_size);
frame_samples = 0;
break;
}
return frame_samples;
}
static gboolean
gst_opus_enc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
{
GstOpusEnc *enc;
enc = GST_OPUS_ENC (benc);
enc->n_channels = GST_AUDIO_INFO_CHANNELS (info);
enc->sample_rate = GST_AUDIO_INFO_RATE (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);
enc->state = NULL;
}
if (!gst_opus_enc_setup (enc))
return FALSE;
enc->frame_samples = gst_opus_enc_get_frame_samples (enc);
/* feedback to base class */
gst_audio_encoder_set_latency (benc,
gst_opus_enc_get_latency (enc), gst_opus_enc_get_latency (enc));
gst_audio_encoder_set_frame_samples_min (benc,
enc->frame_samples * enc->n_channels * 2);
gst_audio_encoder_set_frame_samples_max (benc,
enc->frame_samples * enc->n_channels * 2);
gst_audio_encoder_set_frame_max (benc, 0);
return TRUE;
}
static GstBuffer *
gst_opus_enc_create_id_buffer (GstOpusEnc * enc)
{
GstBuffer *buffer;
GstByteWriter bw;
gst_byte_writer_init (&bw);
/* See http://wiki.xiph.org/OggOpus */
gst_byte_writer_put_string_utf8 (&bw, "OpusHead");
gst_byte_writer_put_uint8 (&bw, 0); /* version number */
gst_byte_writer_put_uint8 (&bw, enc->n_channels);
gst_byte_writer_put_uint16_le (&bw, 0); /* pre-skip *//* TODO: endianness ? */
gst_byte_writer_put_uint32_le (&bw, enc->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 ? */
buffer = gst_byte_writer_reset_and_get_buffer (&bw);
GST_BUFFER_OFFSET (buffer) = 0;
GST_BUFFER_OFFSET_END (buffer) = 0;
return buffer;
}
static GstBuffer *
gst_opus_enc_create_metadata_buffer (GstOpusEnc * enc)
{
const GstTagList *tags;
GstTagList *empty_tags = NULL;
GstBuffer *comments = NULL;
tags = gst_tag_setter_get_tag_list (GST_TAG_SETTER (enc));
GST_DEBUG_OBJECT (enc, "tags = %" GST_PTR_FORMAT, 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 ();
tags = empty_tags;
}
comments =
gst_tag_list_to_vorbiscomment_buffer (tags, (const guint8 *) "OpusTags",
8, "Encoded with GStreamer Opusenc");
GST_BUFFER_OFFSET (comments) = 0;
GST_BUFFER_OFFSET_END (comments) = 0;
if (empty_tags)
gst_tag_list_free (empty_tags);
return comments;
}
static gboolean
gst_opus_enc_setup (GstOpusEnc * enc)
{
int error = OPUS_OK;
GST_DEBUG_OBJECT (enc, "setup");
enc->setup = FALSE;
enc->state = opus_encoder_create (enc->sample_rate, enc->n_channels,
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),
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_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:
GST_ERROR_OBJECT (enc, "Encoder creation failed");
return FALSE;
}
static gboolean
gst_opus_enc_sink_event (GstAudioEncoder * benc, GstEvent * event)
{
GstOpusEnc *enc;
enc = GST_OPUS_ENC (benc);
GST_DEBUG_OBJECT (enc, "sink event: %s", GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_TAG:
{
GstTagList *list;
GstTagSetter *setter = GST_TAG_SETTER (enc);
const GstTagMergeMode mode = gst_tag_setter_get_tag_merge_mode (setter);
gst_event_parse_tag (event, &list);
gst_tag_setter_merge_tags (setter, list, mode);
break;
}
default:
break;
}
return FALSE;
}
static GstFlowReturn
gst_opus_enc_pre_push (GstAudioEncoder * benc, GstBuffer ** buffer)
{
GstFlowReturn ret = GST_FLOW_OK;
GstOpusEnc *enc;
enc = GST_OPUS_ENC (benc);
/* FIXME 0.11 ? get rid of this special ogg stuff and have it
* put and use 'codec data' in caps like anything else,
* with all the usual out-of-band advantage etc */
if (G_UNLIKELY (enc->headers)) {
GSList *header = enc->headers;
/* try to push all of these, if we lose one, might as well lose all */
while (header) {
if (ret == GST_FLOW_OK)
ret = gst_pad_push (GST_AUDIO_ENCODER_SRC_PAD (enc), header->data);
else
gst_pad_push (GST_AUDIO_ENCODER_SRC_PAD (enc), header->data);
header = g_slist_next (header);
}
g_slist_free (enc->headers);
enc->headers = NULL;
}
return ret;
}
static GstFlowReturn
gst_opus_enc_encode (GstOpusEnc * enc, GstBuffer * buf)
{
guint8 *bdata, *data, *mdata = NULL;
gsize bsize, size;
gsize bytes = enc->frame_samples * enc->n_channels * 2;
gsize bytes_per_packet =
(enc->bitrate * enc->frame_samples / enc->sample_rate + 4) / 8;
gint ret = GST_FLOW_OK;
if (G_LIKELY (buf)) {
bdata = GST_BUFFER_DATA (buf);
bsize = GST_BUFFER_SIZE (buf);
if (G_UNLIKELY (bsize % bytes)) {
GST_DEBUG_OBJECT (enc, "draining; adding silence samples");
size = ((bsize / bytes) + 1) * bytes;
mdata = g_malloc0 (size);
memcpy (mdata, bdata, bsize);
bdata = NULL;
data = mdata;
} else {
data = bdata;
size = bsize;
}
} else {
GST_DEBUG_OBJECT (enc, "nothing to drain");
goto done;
}
while (size) {
gint outsize;
GstBuffer *outbuf;
ret = gst_pad_alloc_buffer_and_set_caps (GST_AUDIO_ENCODER_SRC_PAD (enc),
GST_BUFFER_OFFSET_NONE, bytes_per_packet,
GST_PAD_CAPS (GST_AUDIO_ENCODER_SRC_PAD (enc)), &outbuf);
if (GST_FLOW_OK != ret)
goto done;
GST_DEBUG_OBJECT (enc, "encoding %d samples (%d bytes) to %d bytes",
enc->frame_samples, bytes, bytes_per_packet);
outsize =
opus_encode (enc->state, (const gint16 *) data, enc->frame_samples,
GST_BUFFER_DATA (outbuf), bytes_per_packet);
if (outsize < 0) {
GST_ERROR_OBJECT (enc, "Encoding failed: %d", outsize);
ret = GST_FLOW_ERROR;
goto done;
} else if (outsize != bytes_per_packet) {
GST_WARNING_OBJECT (enc,
"Encoded size %d is different from %d bytes per packet", outsize,
bytes_per_packet);
ret = GST_FLOW_ERROR;
goto done;
}
ret =
gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (enc), outbuf,
enc->frame_samples);
if ((GST_FLOW_OK != ret) && (GST_FLOW_NOT_LINKED != ret))
goto done;
data += bytes;
size -= bytes;
}
done:
if (mdata)
g_free (mdata);
return ret;
}
/*
* (really really) FIXME: move into core (dixit tpm)
*/
/**
* _gst_caps_set_buffer_array:
* @caps: a #GstCaps
* @field: field in caps to set
* @buf: header buffers
*
* Adds given buffers to an array of buffers set as the given @field
* on the given @caps. List of buffer arguments must be NULL-terminated.
*
* Returns: input caps with a streamheader field added, or NULL if some error
*/
static GstCaps *
_gst_caps_set_buffer_array (GstCaps * caps, const gchar * field,
GstBuffer * buf, ...)
{
GstStructure *structure = NULL;
va_list va;
GValue array = { 0 };
GValue value = { 0 };
g_return_val_if_fail (caps != NULL, NULL);
g_return_val_if_fail (gst_caps_is_fixed (caps), NULL);
g_return_val_if_fail (field != NULL, NULL);
caps = gst_caps_make_writable (caps);
structure = gst_caps_get_structure (caps, 0);
g_value_init (&array, GST_TYPE_ARRAY);
va_start (va, buf);
/* put buffers in a fixed list */
while (buf) {
g_assert (gst_buffer_is_writable (buf));
/* mark buffer */
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_IN_CAPS);
g_value_init (&value, GST_TYPE_BUFFER);
buf = gst_buffer_copy (buf);
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_IN_CAPS);
gst_value_set_buffer (&value, buf);
gst_buffer_unref (buf);
gst_value_array_append_value (&array, &value);
g_value_unset (&value);
buf = va_arg (va, GstBuffer *);
}
gst_structure_set_value (structure, field, &array);
g_value_unset (&array);
return caps;
}
static GstFlowReturn
gst_opus_enc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf)
{
GstOpusEnc *enc;
GstFlowReturn ret = GST_FLOW_OK;
enc = GST_OPUS_ENC (benc);
GST_DEBUG_OBJECT (enc, "handle_frame");
if (!enc->header_sent) {
/* 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. */
GstBuffer *buf1, *buf2;
GstCaps *caps;
/* create header buffers */
buf1 = gst_opus_enc_create_id_buffer (enc);
buf2 = gst_opus_enc_create_metadata_buffer (enc);
/* mark and put on caps */
caps = gst_caps_from_string ("audio/x-opus");
caps = _gst_caps_set_buffer_array (caps, "streamheader", buf1, buf2, NULL);
/* negotiate with these caps */
GST_DEBUG_OBJECT (enc, "here are the caps: %" GST_PTR_FORMAT, caps);
gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (enc), caps);
/* push out buffers */
/* store buffers for later pre_push sending */
g_slist_foreach (enc->headers, (GFunc) gst_buffer_unref, NULL);
enc->headers = NULL;
GST_DEBUG_OBJECT (enc, "storing header buffers");
enc->headers = g_slist_prepend (enc->headers, buf2);
enc->headers = g_slist_prepend (enc->headers, buf1);
enc->header_sent = TRUE;
}
GST_DEBUG_OBJECT (enc, "received buffer %p of %u bytes", buf,
buf ? GST_BUFFER_SIZE (buf) : 0);
ret = gst_opus_enc_encode (enc, buf);
return ret;
}
static void
gst_opus_enc_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
GstOpusEnc *enc;
enc = GST_OPUS_ENC (object);
switch (prop_id) {
case PROP_AUDIO:
g_value_set_boolean (value, enc->audio_or_voip);
break;
case PROP_BITRATE:
g_value_set_int (value, enc->bitrate);
break;
case PROP_BANDWIDTH:
g_value_set_enum (value, enc->bandwidth);
break;
case PROP_FRAME_SIZE:
g_value_set_int (value, enc->frame_size);
break;
case PROP_CBR:
g_value_set_boolean (value, enc->cbr);
break;
case PROP_CONSTRAINED_VBR:
g_value_set_boolean (value, enc->constrained_vbr);
break;
case PROP_COMPLEXITY:
g_value_set_int (value, enc->complexity);
break;
case PROP_INBAND_FEC:
g_value_set_boolean (value, enc->inband_fec);
break;
case PROP_DTX:
g_value_set_boolean (value, enc->dtx);
break;
case PROP_PACKET_LOSS_PERCENT:
g_value_set_int (value, enc->packet_loss_percentage);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_opus_enc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstOpusEnc *enc;
enc = GST_OPUS_ENC (object);
switch (prop_id) {
case PROP_AUDIO:
enc->audio_or_voip = g_value_get_boolean (value);
break;
case PROP_BITRATE:
enc->bitrate = g_value_get_int (value);
break;
case PROP_BANDWIDTH:
enc->bandwidth = g_value_get_enum (value);
break;
case PROP_FRAME_SIZE:
enc->frame_size = g_value_get_int (value);
break;
case PROP_CBR:
enc->cbr = g_value_get_boolean (value);
break;
case PROP_CONSTRAINED_VBR:
enc->constrained_vbr = g_value_get_boolean (value);
break;
case PROP_COMPLEXITY:
enc->complexity = g_value_get_int (value);
break;
case PROP_INBAND_FEC:
enc->inband_fec = g_value_get_boolean (value);
break;
case PROP_DTX:
enc->dtx = g_value_get_boolean (value);
break;
case PROP_PACKET_LOSS_PERCENT:
enc->packet_loss_percentage = g_value_get_int (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}