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gstreamer/ext/opus/gstopusenc.c

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/* 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> 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., 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/audio/audio.h>
#include "gstopusheader.h"
#include "gstopuscommon.h"
#include "gstopusenc.h"
GST_DEBUG_CATEGORY_STATIC (opusenc_debug);
#define GST_CAT_DEFAULT opusenc_debug
/* Some arbitrary bounds beyond which it really doesn't make sense.
The spec mentions 6 kb/s to 510 kb/s, so 4000 and 650000 ought to be
safe as property bounds. */
#define LOWEST_BITRATE 4000
#define HIGHEST_BITRATE 650000
#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;
}
#define GST_OPUS_ENC_TYPE_FRAME_SIZE (gst_opus_enc_frame_size_get_type())
static GType
gst_opus_enc_frame_size_get_type (void)
{
static const GEnumValue values[] = {
{2, "2.5", "2.5"},
{5, "5", "5"},
{10, "10", "10"},
{20, "20", "20"},
{40, "40", "40"},
{60, "60", "60"},
{0, NULL, NULL}
};
static volatile GType id = 0;
if (g_once_init_enter ((gsize *) & id)) {
GType _id;
_id = g_enum_register_static ("GstOpusEncFrameSize", 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) { 48000, 24000, 16000, 12000, 8000 }, "
"channels = (int) [ 1, 8 ], "
"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
#define DEFAULT_MAX_PAYLOAD_SIZE 1024
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,
PROP_MAX_PAYLOAD_SIZE
};
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 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_static_pad_template (element_class, &src_factory);
gst_element_class_add_static_pad_template (element_class, &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;
GstAudioEncoderClass *base_class;
gobject_class = (GObjectClass *) 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->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).",
LOWEST_BITRATE, HIGHEST_BITRATE, DEFAULT_BITRATE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
GST_PARAM_MUTABLE_PLAYING));
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 |
GST_PARAM_MUTABLE_PLAYING));
g_object_class_install_property (gobject_class, PROP_FRAME_SIZE,
g_param_spec_enum ("frame-size", "Frame Size",
"The duration of an audio frame, in ms", GST_OPUS_ENC_TYPE_FRAME_SIZE,
DEFAULT_FRAMESIZE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
GST_PARAM_MUTABLE_PLAYING));
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 |
GST_PARAM_MUTABLE_PLAYING));
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 |
GST_PARAM_MUTABLE_PLAYING));
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 |
GST_PARAM_MUTABLE_PLAYING));
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 |
GST_PARAM_MUTABLE_PLAYING));
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 |
GST_PARAM_MUTABLE_PLAYING));
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 |
GST_PARAM_MUTABLE_PLAYING));
g_object_class_install_property (G_OBJECT_CLASS (klass),
PROP_MAX_PAYLOAD_SIZE, g_param_spec_uint ("max-payload-size",
"Max payload size", "Maximum payload size in bytes", 2, 1275,
DEFAULT_MAX_PAYLOAD_SIZE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
GST_PARAM_MUTABLE_PLAYING));
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_mutex_free (enc->property_lock);
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->property_lock = g_mutex_new ();
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;
enc->max_payload_size = DEFAULT_MAX_PAYLOAD_SIZE;
/* 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_multistream_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;
gst_tag_setter_reset_tags (GST_TAG_SETTER (enc));
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 void
gst_opus_enc_setup_base_class (GstOpusEnc * enc, GstAudioEncoder * benc)
{
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);
}
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 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 (%s)", n, pos,
gst_opus_channel_names[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 (%s) not found in Vorbis mapping, using unknown mapping",
pos, gst_opus_channel_positions[pos]);
enc->channel_mapping_family = 255;
return;
}
GST_DEBUG_OBJECT (enc, "Mapping output channel %d to %d (%s)", c, n,
gst_opus_channel_names[pos]);
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)
{
GstOpusEnc *enc;
enc = GST_OPUS_ENC (benc);
g_mutex_lock (enc->property_lock);
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_multistream_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_opus_enc_setup_base_class (enc, benc);
g_mutex_unlock (enc->property_lock);
return TRUE;
}
static gboolean
gst_opus_enc_setup (GstOpusEnc * enc)
{
int error = OPUS_OK, n;
guint8 trivial_mapping[256];
GST_DEBUG_OBJECT (enc, "setup");
for (n = 0; n < 256; ++n)
trivial_mapping[n] = n;
enc->state =
opus_multistream_encoder_create (enc->sample_rate, enc->n_channels,
enc->n_channels, 0, trivial_mapping,
enc->audio_or_voip ? OPUS_APPLICATION_AUDIO : OPUS_APPLICATION_VOIP,
&error);
if (!enc->state || error != OPUS_OK)
goto encoder_creation_failed;
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_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);
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_encode (GstOpusEnc * enc, GstBuffer * buf)
{
guint8 *bdata, *data, *mdata = NULL;
gsize bsize, size;
gsize bytes;
gint ret = GST_FLOW_OK;
g_mutex_lock (enc->property_lock);
bytes = enc->frame_samples * enc->n_channels * 2;
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, enc->max_payload_size * enc->n_channels,
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)",
enc->frame_samples, (int) bytes);
outsize =
opus_multistream_encode (enc->state, (const gint16 *) data,
enc->frame_samples, GST_BUFFER_DATA (outbuf),
enc->max_payload_size * enc->n_channels);
if (outsize < 0) {
GST_ERROR_OBJECT (enc, "Encoding failed: %d", outsize);
ret = GST_FLOW_ERROR;
goto done;
} else if (outsize > enc->max_payload_size) {
GST_WARNING_OBJECT (enc,
"Encoded size %d is higher than max payload size (%d bytes)",
outsize, enc->max_payload_size);
ret = GST_FLOW_ERROR;
goto done;
}
GST_DEBUG_OBJECT (enc, "Output packet is %u bytes", outsize);
GST_BUFFER_SIZE (outbuf) = outsize;
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:
g_mutex_unlock (enc->property_lock);
if (mdata)
g_free (mdata);
return ret;
}
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) {
GstCaps *caps;
g_slist_foreach (enc->headers, (GFunc) gst_buffer_unref, NULL);
enc->headers = NULL;
gst_opus_header_create_caps (&caps, &enc->headers, enc->n_channels,
enc->sample_rate, enc->channel_mapping_family, enc->channel_mapping,
gst_tag_setter_get_tag_list (GST_TAG_SETTER (enc)));
/* 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);
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);
g_mutex_lock (enc->property_lock);
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_enum (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;
case PROP_MAX_PAYLOAD_SIZE:
g_value_set_uint (value, enc->max_payload_size);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
g_mutex_unlock (enc->property_lock);
}
static void
gst_opus_enc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstOpusEnc *enc;
enc = GST_OPUS_ENC (object);
#define GST_OPUS_UPDATE_PROPERTY(prop,type,ctl) do { \
g_mutex_lock (enc->property_lock); \
enc->prop = g_value_get_##type (value); \
if (enc->state) { \
opus_multistream_encoder_ctl (enc->state, OPUS_SET_##ctl (enc->prop)); \
} \
g_mutex_unlock (enc->property_lock); \
} while(0)
switch (prop_id) {
case PROP_AUDIO:
enc->audio_or_voip = g_value_get_boolean (value);
break;
case PROP_BITRATE:
GST_OPUS_UPDATE_PROPERTY (bitrate, int, BITRATE);
break;
case PROP_BANDWIDTH:
GST_OPUS_UPDATE_PROPERTY (bandwidth, enum, BANDWIDTH);
break;
case PROP_FRAME_SIZE:
g_mutex_lock (enc->property_lock);
enc->frame_size = g_value_get_enum (value);
enc->frame_samples = gst_opus_enc_get_frame_samples (enc);
gst_opus_enc_setup_base_class (enc, GST_AUDIO_ENCODER (enc));
g_mutex_unlock (enc->property_lock);
break;
case PROP_CBR:
/* this one has an opposite meaning to the opus ctl... */
g_mutex_lock (enc->property_lock);
enc->cbr = g_value_get_boolean (value);
opus_multistream_encoder_ctl (enc->state, OPUS_SET_VBR (!enc->cbr));
g_mutex_unlock (enc->property_lock);
break;
case PROP_CONSTRAINED_VBR:
GST_OPUS_UPDATE_PROPERTY (constrained_vbr, boolean, VBR_CONSTRAINT);
break;
case PROP_COMPLEXITY:
GST_OPUS_UPDATE_PROPERTY (complexity, int, COMPLEXITY);
break;
case PROP_INBAND_FEC:
GST_OPUS_UPDATE_PROPERTY (inband_fec, boolean, INBAND_FEC);
break;
case PROP_DTX:
GST_OPUS_UPDATE_PROPERTY (dtx, boolean, DTX);
break;
case PROP_PACKET_LOSS_PERCENT:
GST_OPUS_UPDATE_PROPERTY (packet_loss_percentage, int, PACKET_LOSS_PERC);
break;
case PROP_MAX_PAYLOAD_SIZE:
g_mutex_lock (enc->property_lock);
enc->max_payload_size = g_value_get_uint (value);
g_mutex_unlock (enc->property_lock);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
#undef GST_OPUS_UPDATE_PROPERTY
}
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