gstreamer/ext/libav/gstavaudenc.c
2013-01-22 12:57:41 +00:00

752 lines
22 KiB
C

/* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
* Copyright (C) <2012> Collabora Ltd.
* Author: 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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <assert.h>
#include <string.h>
/* for stats file handling */
#include <stdio.h>
#include <glib/gstdio.h>
#include <errno.h>
#include <libavcodec/avcodec.h>
#include <gst/gst.h>
#include "gstav.h"
#include "gstavcodecmap.h"
#include "gstavutils.h"
#include "gstavaudenc.h"
#define DEFAULT_AUDIO_BITRATE 128000
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
PROP_0,
PROP_BIT_RATE,
PROP_RTP_PAYLOAD_SIZE,
};
/* A number of function prototypes are given so we can refer to them later. */
static void gst_ffmpegaudenc_class_init (GstFFMpegAudEncClass * klass);
static void gst_ffmpegaudenc_base_init (GstFFMpegAudEncClass * klass);
static void gst_ffmpegaudenc_init (GstFFMpegAudEnc * ffmpegaudenc);
static void gst_ffmpegaudenc_finalize (GObject * object);
static GstCaps *gst_ffmpegaudenc_getcaps (GstAudioEncoder * encoder,
GstCaps * filter);
static gboolean gst_ffmpegaudenc_set_format (GstAudioEncoder * encoder,
GstAudioInfo * info);
static GstFlowReturn gst_ffmpegaudenc_handle_frame (GstAudioEncoder * encoder,
GstBuffer * inbuf);
static gboolean gst_ffmpegaudenc_stop (GstAudioEncoder * encoder);
static void gst_ffmpegaudenc_flush (GstAudioEncoder * encoder);
static void gst_ffmpegaudenc_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_ffmpegaudenc_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
#define GST_FFENC_PARAMS_QDATA g_quark_from_static_string("avenc-params")
static GstElementClass *parent_class = NULL;
/*static guint gst_ffmpegaudenc_signals[LAST_SIGNAL] = { 0 }; */
static void
gst_ffmpegaudenc_base_init (GstFFMpegAudEncClass * klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
AVCodec *in_plugin;
GstPadTemplate *srctempl = NULL, *sinktempl = NULL;
GstCaps *srccaps = NULL, *sinkcaps = NULL;
gchar *longname, *description;
in_plugin =
(AVCodec *) g_type_get_qdata (G_OBJECT_CLASS_TYPE (klass),
GST_FFENC_PARAMS_QDATA);
g_assert (in_plugin != NULL);
/* construct the element details struct */
longname = g_strdup_printf ("libav %s encoder", in_plugin->long_name);
description = g_strdup_printf ("libav %s encoder", in_plugin->name);
gst_element_class_set_metadata (element_class, longname,
"Codec/Encoder/Audio", description,
"Wim Taymans <wim.taymans@gmail.com>, "
"Ronald Bultje <rbultje@ronald.bitfreak.net>");
g_free (longname);
g_free (description);
if (!(srccaps = gst_ffmpeg_codecid_to_caps (in_plugin->id, NULL, TRUE))) {
GST_DEBUG ("Couldn't get source caps for encoder '%s'", in_plugin->name);
srccaps = gst_caps_new_empty_simple ("unknown/unknown");
}
sinkcaps = gst_ffmpeg_codectype_to_audio_caps (NULL,
in_plugin->id, TRUE, in_plugin);
if (!sinkcaps) {
GST_DEBUG ("Couldn't get sink caps for encoder '%s'", in_plugin->name);
sinkcaps = gst_caps_new_empty_simple ("unknown/unknown");
}
/* pad templates */
sinktempl = gst_pad_template_new ("sink", GST_PAD_SINK,
GST_PAD_ALWAYS, sinkcaps);
srctempl = gst_pad_template_new ("src", GST_PAD_SRC, GST_PAD_ALWAYS, srccaps);
gst_element_class_add_pad_template (element_class, srctempl);
gst_element_class_add_pad_template (element_class, sinktempl);
klass->in_plugin = in_plugin;
klass->srctempl = srctempl;
klass->sinktempl = sinktempl;
return;
}
static void
gst_ffmpegaudenc_class_init (GstFFMpegAudEncClass * klass)
{
GObjectClass *gobject_class;
GstAudioEncoderClass *gstaudioencoder_class;
gobject_class = (GObjectClass *) klass;
gstaudioencoder_class = (GstAudioEncoderClass *) klass;
parent_class = g_type_class_peek_parent (klass);
gobject_class->set_property = gst_ffmpegaudenc_set_property;
gobject_class->get_property = gst_ffmpegaudenc_get_property;
/* FIXME: could use -1 for a sensible per-codec defaults */
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_BIT_RATE,
g_param_spec_int ("bitrate", "Bit Rate",
"Target Audio Bitrate", 0, G_MAXINT, DEFAULT_AUDIO_BITRATE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gobject_class->finalize = gst_ffmpegaudenc_finalize;
gstaudioencoder_class->stop = GST_DEBUG_FUNCPTR (gst_ffmpegaudenc_stop);
gstaudioencoder_class->getcaps = GST_DEBUG_FUNCPTR (gst_ffmpegaudenc_getcaps);
gstaudioencoder_class->flush = GST_DEBUG_FUNCPTR (gst_ffmpegaudenc_flush);
gstaudioencoder_class->set_format =
GST_DEBUG_FUNCPTR (gst_ffmpegaudenc_set_format);
gstaudioencoder_class->handle_frame =
GST_DEBUG_FUNCPTR (gst_ffmpegaudenc_handle_frame);
}
static void
gst_ffmpegaudenc_init (GstFFMpegAudEnc * ffmpegaudenc)
{
GstFFMpegAudEncClass *klass =
(GstFFMpegAudEncClass *) G_OBJECT_GET_CLASS (ffmpegaudenc);
/* ffmpeg objects */
ffmpegaudenc->context = avcodec_alloc_context3 (klass->in_plugin);
ffmpegaudenc->opened = FALSE;
gst_audio_encoder_set_drainable (GST_AUDIO_ENCODER (ffmpegaudenc), TRUE);
}
static void
gst_ffmpegaudenc_finalize (GObject * object)
{
GstFFMpegAudEnc *ffmpegaudenc = (GstFFMpegAudEnc *) object;
/* clean up remaining allocated data */
av_free (ffmpegaudenc->context);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gboolean
gst_ffmpegaudenc_stop (GstAudioEncoder * encoder)
{
GstFFMpegAudEnc *ffmpegaudenc = (GstFFMpegAudEnc *) encoder;
/* close old session */
gst_ffmpeg_avcodec_close (ffmpegaudenc->context);
ffmpegaudenc->opened = FALSE;
return TRUE;
}
static void
gst_ffmpegaudenc_flush (GstAudioEncoder * encoder)
{
GstFFMpegAudEnc *ffmpegaudenc = (GstFFMpegAudEnc *) encoder;
if (ffmpegaudenc->opened) {
avcodec_flush_buffers (ffmpegaudenc->context);
}
}
static GstCaps *
gst_ffmpegaudenc_getcaps (GstAudioEncoder * encoder, GstCaps * filter)
{
GstFFMpegAudEnc *ffmpegaudenc = (GstFFMpegAudEnc *) encoder;
GstCaps *caps = NULL;
GST_DEBUG_OBJECT (ffmpegaudenc, "getting caps");
/* audio needs no special care */
caps = gst_audio_encoder_proxy_getcaps (encoder, NULL, filter);
GST_DEBUG_OBJECT (ffmpegaudenc, "audio caps, return %" GST_PTR_FORMAT, caps);
return caps;
}
static gboolean
gst_ffmpegaudenc_set_format (GstAudioEncoder * encoder, GstAudioInfo * info)
{
GstFFMpegAudEnc *ffmpegaudenc = (GstFFMpegAudEnc *) encoder;
GstCaps *other_caps;
GstCaps *allowed_caps;
GstCaps *icaps;
gsize frame_size;
GstFFMpegAudEncClass *oclass =
(GstFFMpegAudEncClass *) G_OBJECT_GET_CLASS (ffmpegaudenc);
/* close old session */
if (ffmpegaudenc->opened) {
gst_ffmpeg_avcodec_close (ffmpegaudenc->context);
ffmpegaudenc->opened = FALSE;
}
/* if we set it in _getcaps we should set it also in _link */
ffmpegaudenc->context->strict_std_compliance = -1;
/* user defined properties */
if (ffmpegaudenc->bitrate > 0) {
GST_INFO_OBJECT (ffmpegaudenc, "Setting avcontext to bitrate %d",
ffmpegaudenc->bitrate);
ffmpegaudenc->context->bit_rate = ffmpegaudenc->bitrate;
ffmpegaudenc->context->bit_rate_tolerance = ffmpegaudenc->bitrate;
} else {
GST_INFO_OBJECT (ffmpegaudenc, "Using avcontext default bitrate %d",
ffmpegaudenc->context->bit_rate);
}
/* RTP payload used for GOB production (for Asterisk) */
if (ffmpegaudenc->rtp_payload_size) {
ffmpegaudenc->context->rtp_payload_size = ffmpegaudenc->rtp_payload_size;
}
/* some other defaults */
ffmpegaudenc->context->rc_strategy = 2;
ffmpegaudenc->context->b_frame_strategy = 0;
ffmpegaudenc->context->coder_type = 0;
ffmpegaudenc->context->context_model = 0;
ffmpegaudenc->context->scenechange_threshold = 0;
ffmpegaudenc->context->inter_threshold = 0;
/* fetch pix_fmt and so on */
gst_ffmpeg_audioinfo_to_context (info, ffmpegaudenc->context);
if (!ffmpegaudenc->context->time_base.den) {
ffmpegaudenc->context->time_base.den = GST_AUDIO_INFO_RATE (info);
ffmpegaudenc->context->time_base.num = 1;
ffmpegaudenc->context->ticks_per_frame = 1;
}
if (ffmpegaudenc->context->channel_layout) {
gst_ffmpeg_channel_layout_to_gst (ffmpegaudenc->context->channel_layout,
ffmpegaudenc->context->channels, ffmpegaudenc->ffmpeg_layout);
ffmpegaudenc->needs_reorder =
(memcmp (ffmpegaudenc->ffmpeg_layout, info->position,
sizeof (GstAudioChannelPosition) *
ffmpegaudenc->context->channels) != 0);
}
/* open codec */
if (gst_ffmpeg_avcodec_open (ffmpegaudenc->context, oclass->in_plugin) < 0) {
if (ffmpegaudenc->context->priv_data)
gst_ffmpeg_avcodec_close (ffmpegaudenc->context);
GST_DEBUG_OBJECT (ffmpegaudenc, "avenc_%s: Failed to open FFMPEG codec",
oclass->in_plugin->name);
return FALSE;
}
/* some codecs support more than one format, first auto-choose one */
GST_DEBUG_OBJECT (ffmpegaudenc, "picking an output format ...");
allowed_caps = gst_pad_get_allowed_caps (GST_AUDIO_ENCODER_SRC_PAD (encoder));
if (!allowed_caps) {
GST_DEBUG_OBJECT (ffmpegaudenc, "... but no peer, using template caps");
/* we need to copy because get_allowed_caps returns a ref, and
* get_pad_template_caps doesn't */
allowed_caps =
gst_pad_get_pad_template_caps (GST_AUDIO_ENCODER_SRC_PAD (encoder));
}
GST_DEBUG_OBJECT (ffmpegaudenc, "chose caps %" GST_PTR_FORMAT, allowed_caps);
gst_ffmpeg_caps_with_codecid (oclass->in_plugin->id,
oclass->in_plugin->type, allowed_caps, ffmpegaudenc->context);
/* try to set this caps on the other side */
other_caps = gst_ffmpeg_codecid_to_caps (oclass->in_plugin->id,
ffmpegaudenc->context, TRUE);
if (!other_caps) {
gst_caps_unref (allowed_caps);
gst_ffmpeg_avcodec_close (ffmpegaudenc->context);
GST_DEBUG ("Unsupported codec - no caps found");
return FALSE;
}
icaps = gst_caps_intersect (allowed_caps, other_caps);
gst_caps_unref (allowed_caps);
gst_caps_unref (other_caps);
if (gst_caps_is_empty (icaps)) {
gst_caps_unref (icaps);
return FALSE;
}
icaps = gst_caps_truncate (icaps);
if (!gst_audio_encoder_set_output_format (GST_AUDIO_ENCODER (ffmpegaudenc),
icaps)) {
gst_ffmpeg_avcodec_close (ffmpegaudenc->context);
gst_caps_unref (icaps);
return FALSE;
}
gst_caps_unref (icaps);
frame_size = ffmpegaudenc->context->frame_size;
if (frame_size > 1) {
gst_audio_encoder_set_frame_samples_min (GST_AUDIO_ENCODER (ffmpegaudenc),
frame_size);
gst_audio_encoder_set_frame_samples_max (GST_AUDIO_ENCODER (ffmpegaudenc),
frame_size);
gst_audio_encoder_set_frame_max (GST_AUDIO_ENCODER (ffmpegaudenc), 1);
} else {
gst_audio_encoder_set_frame_samples_min (GST_AUDIO_ENCODER (ffmpegaudenc),
0);
gst_audio_encoder_set_frame_samples_max (GST_AUDIO_ENCODER (ffmpegaudenc),
0);
gst_audio_encoder_set_frame_max (GST_AUDIO_ENCODER (ffmpegaudenc), 0);
}
/* success! */
ffmpegaudenc->opened = TRUE;
return TRUE;
}
static GstFlowReturn
gst_ffmpegaudenc_encode_audio (GstFFMpegAudEnc * ffmpegaudenc,
guint8 * audio_in, guint in_size, gint * have_data)
{
GstAudioEncoder *enc;
AVCodecContext *ctx;
gint res;
GstFlowReturn ret;
GstAudioInfo *info;
AVPacket pkt;
AVFrame frame;
gboolean planar;
enc = GST_AUDIO_ENCODER (ffmpegaudenc);
ctx = ffmpegaudenc->context;
GST_LOG_OBJECT (ffmpegaudenc, "encoding buffer ");
memset (&pkt, 0, sizeof (pkt));
memset (&frame, 0, sizeof (frame));
avcodec_get_frame_defaults (&frame);
info = gst_audio_encoder_get_audio_info (enc);
planar = av_sample_fmt_is_planar (ffmpegaudenc->context->sample_fmt);
if (planar && info->channels > 1) {
gint channels, nsamples;
gint i, j;
nsamples = frame.nb_samples = in_size / info->bpf;
channels = info->channels;
if (info->channels > AV_NUM_DATA_POINTERS) {
frame.extended_data = g_new (uint8_t *, info->channels);
} else {
frame.extended_data = frame.data;
}
frame.extended_data[0] = g_malloc (in_size);
frame.linesize[0] = in_size / channels;
for (i = 1; i < channels; i++)
frame.extended_data[i] = frame.extended_data[i - 1] + frame.linesize[0];
switch (info->finfo->width) {
case 8:{
const guint8 *idata = (const guint8 *) audio_in;
for (i = 0; i < nsamples; i++) {
for (j = 0; j < channels; j++) {
((guint8 *) frame.extended_data[j])[i] = idata[j];
}
idata += channels;
}
break;
}
case 16:{
const guint16 *idata = (const guint16 *) audio_in;
for (i = 0; i < nsamples; i++) {
for (j = 0; j < channels; j++) {
((guint16 *) frame.extended_data[j])[i] = idata[j];
}
idata += channels;
}
break;
}
case 32:{
const guint32 *idata = (const guint32 *) audio_in;
for (i = 0; i < nsamples; i++) {
for (j = 0; j < channels; j++) {
((guint32 *) frame.extended_data[j])[i] = idata[j];
}
idata += channels;
}
break;
}
case 64:{
const guint64 *idata = (const guint64 *) audio_in;
for (i = 0; i < nsamples; i++) {
for (j = 0; j < channels; j++) {
((guint64 *) frame.extended_data[j])[i] = idata[j];
}
idata += channels;
}
break;
}
default:
g_assert_not_reached ();
break;
}
} else {
frame.data[0] = audio_in;
frame.extended_data = frame.data;
frame.linesize[0] = in_size;
frame.nb_samples = in_size / info->bpf;
}
res = avcodec_encode_audio2 (ctx, &pkt, &frame, have_data);
if (planar && info->channels > 1)
g_free (frame.data[0]);
if (frame.extended_data != frame.data)
g_free (frame.extended_data);
if (res < 0) {
char error_str[128] = { 0, };
av_strerror (res, error_str, sizeof (error_str));
GST_ERROR_OBJECT (enc, "Failed to encode buffer: %d - %s", res, error_str);
return GST_FLOW_OK;
}
GST_LOG_OBJECT (ffmpegaudenc, "got output size %d", res);
if (*have_data) {
GstBuffer *outbuf;
const AVCodec *codec;
GST_LOG_OBJECT (ffmpegaudenc, "pushing size %d", pkt.size);
outbuf =
gst_buffer_new_wrapped_full (0, pkt.data, pkt.size, 0, pkt.size,
pkt.data, av_free);
codec = ffmpegaudenc->context->codec;
if ((codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE)) {
ret = gst_audio_encoder_finish_frame (enc, outbuf, -1);
} else {
ret = gst_audio_encoder_finish_frame (enc, outbuf, frame.nb_samples);
}
} else {
GST_LOG_OBJECT (ffmpegaudenc, "no output produced");
ret = GST_FLOW_OK;
}
return ret;
}
static void
gst_ffmpegaudenc_drain (GstFFMpegAudEnc * ffmpegaudenc)
{
GstFFMpegAudEncClass *oclass;
oclass = (GstFFMpegAudEncClass *) (G_OBJECT_GET_CLASS (ffmpegaudenc));
if (oclass->in_plugin->capabilities & CODEC_CAP_DELAY) {
gint have_data, try = 0;
GST_LOG_OBJECT (ffmpegaudenc,
"codec has delay capabilities, calling until libav has drained everything");
do {
GstFlowReturn ret;
ret = gst_ffmpegaudenc_encode_audio (ffmpegaudenc, NULL, 0, &have_data);
if (ret != GST_FLOW_OK || have_data == 0)
break;
} while (try++ < 10);
}
}
static GstFlowReturn
gst_ffmpegaudenc_handle_frame (GstAudioEncoder * encoder, GstBuffer * inbuf)
{
GstFFMpegAudEnc *ffmpegaudenc;
gsize size;
GstFlowReturn ret;
guint8 *in_data;
GstMapInfo map;
gint have_data;
ffmpegaudenc = (GstFFMpegAudEnc *) encoder;
if (G_UNLIKELY (!ffmpegaudenc->opened))
goto not_negotiated;
if (!inbuf) {
gst_ffmpegaudenc_drain (ffmpegaudenc);
return GST_FLOW_OK;
}
inbuf = gst_buffer_ref (inbuf);
GST_DEBUG_OBJECT (ffmpegaudenc,
"Received time %" GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT
", size %" G_GSIZE_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (inbuf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (inbuf)), gst_buffer_get_size (inbuf));
/* Reorder channels to the GStreamer channel order */
if (ffmpegaudenc->needs_reorder) {
GstAudioInfo *info = gst_audio_encoder_get_audio_info (encoder);
inbuf = gst_buffer_make_writable (inbuf);
gst_audio_buffer_reorder_channels (inbuf, info->finfo->format,
info->channels, info->position, ffmpegaudenc->ffmpeg_layout);
}
gst_buffer_map (inbuf, &map, GST_MAP_READ);
in_data = map.data;
size = map.size;
ret = gst_ffmpegaudenc_encode_audio (ffmpegaudenc, in_data, size, &have_data);
gst_buffer_unmap (inbuf, &map);
gst_buffer_unref (inbuf);
if (ret != GST_FLOW_OK)
goto push_failed;
return GST_FLOW_OK;
/* ERRORS */
not_negotiated:
{
GST_ELEMENT_ERROR (ffmpegaudenc, CORE, NEGOTIATION, (NULL),
("not configured to input format before data start"));
gst_buffer_unref (inbuf);
return GST_FLOW_NOT_NEGOTIATED;
}
push_failed:
{
GST_DEBUG_OBJECT (ffmpegaudenc, "Failed to push buffer %d (%s)", ret,
gst_flow_get_name (ret));
return ret;
}
}
static void
gst_ffmpegaudenc_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec)
{
GstFFMpegAudEnc *ffmpegaudenc;
/* Get a pointer of the right type. */
ffmpegaudenc = (GstFFMpegAudEnc *) (object);
if (ffmpegaudenc->opened) {
GST_WARNING_OBJECT (ffmpegaudenc,
"Can't change properties once decoder is setup !");
return;
}
/* Check the argument id to see which argument we're setting. */
switch (prop_id) {
case PROP_BIT_RATE:
ffmpegaudenc->bitrate = g_value_get_int (value);
break;
case PROP_RTP_PAYLOAD_SIZE:
ffmpegaudenc->rtp_payload_size = g_value_get_int (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
/* The set function is simply the inverse of the get fuction. */
static void
gst_ffmpegaudenc_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec)
{
GstFFMpegAudEnc *ffmpegaudenc;
/* It's not null if we got it, but it might not be ours */
ffmpegaudenc = (GstFFMpegAudEnc *) (object);
switch (prop_id) {
case PROP_BIT_RATE:
g_value_set_int (value, ffmpegaudenc->bitrate);
break;
break;
case PROP_RTP_PAYLOAD_SIZE:
g_value_set_int (value, ffmpegaudenc->rtp_payload_size);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
gboolean
gst_ffmpegaudenc_register (GstPlugin * plugin)
{
GTypeInfo typeinfo = {
sizeof (GstFFMpegAudEncClass),
(GBaseInitFunc) gst_ffmpegaudenc_base_init,
NULL,
(GClassInitFunc) gst_ffmpegaudenc_class_init,
NULL,
NULL,
sizeof (GstFFMpegAudEnc),
0,
(GInstanceInitFunc) gst_ffmpegaudenc_init,
};
GType type;
AVCodec *in_plugin;
GST_LOG ("Registering encoders");
in_plugin = av_codec_next (NULL);
while (in_plugin) {
gchar *type_name;
guint rank;
/* Skip non-AV codecs */
if (in_plugin->type != AVMEDIA_TYPE_AUDIO)
goto next;
/* no quasi codecs, please */
if ((in_plugin->id >= CODEC_ID_PCM_S16LE &&
in_plugin->id <= CODEC_ID_PCM_BLURAY)) {
goto next;
}
/* No encoders depending on external libraries (we don't build them, but
* people who build against an external ffmpeg might have them.
* We have native gstreamer plugins for all of those libraries anyway. */
if (!strncmp (in_plugin->name, "lib", 3)) {
GST_DEBUG
("Not using external library encoder %s. Use the gstreamer-native ones instead.",
in_plugin->name);
goto next;
}
/* only encoders */
if (!av_codec_is_encoder (in_plugin)) {
goto next;
}
/* FIXME : We should have a method to know cheaply whether we have a mapping
* for the given plugin or not */
GST_DEBUG ("Trying plugin %s [%s]", in_plugin->name, in_plugin->long_name);
/* no codecs for which we're GUARANTEED to have better alternatives */
if (!strcmp (in_plugin->name, "vorbis")
|| !strcmp (in_plugin->name, "flac")) {
GST_LOG ("Ignoring encoder %s", in_plugin->name);
goto next;
}
/* construct the type */
type_name = g_strdup_printf ("avenc_%s", in_plugin->name);
type = g_type_from_name (type_name);
if (!type) {
/* create the glib type now */
type =
g_type_register_static (GST_TYPE_AUDIO_ENCODER, type_name, &typeinfo,
0);
g_type_set_qdata (type, GST_FFENC_PARAMS_QDATA, (gpointer) in_plugin);
{
static const GInterfaceInfo preset_info = {
NULL,
NULL,
NULL
};
g_type_add_interface_static (type, GST_TYPE_PRESET, &preset_info);
}
}
switch (in_plugin->id) {
/* avenc_aac: see https://bugzilla.gnome.org/show_bug.cgi?id=691617 */
case CODEC_ID_AAC:
rank = GST_RANK_NONE;
break;
default:
rank = GST_RANK_SECONDARY;
break;
}
if (!gst_element_register (plugin, type_name, rank, type)) {
g_free (type_name);
return FALSE;
}
g_free (type_name);
next:
in_plugin = av_codec_next (in_plugin);
}
GST_LOG ("Finished registering encoders");
return TRUE;
}