gstreamer/subprojects/gst-libav/ext/libav/gstavaudenc.c

877 lines
26 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 <libavutil/opt.h>
#include <gst/gst.h>
#include <gst/base/base.h>
#include "gstav.h"
#include "gstavcfg.h"
#include "gstavcodecmap.h"
#include "gstavutils.h"
#include "gstavaudenc.h"
enum
{
PROP_0,
PROP_CFG_BASE,
};
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 gboolean gst_ffmpegaudenc_set_format (GstAudioEncoder * encoder,
GstAudioInfo * info);
static GstFlowReturn gst_ffmpegaudenc_handle_frame (GstAudioEncoder * encoder,
GstBuffer * inbuf);
static gboolean gst_ffmpegaudenc_start (GstAudioEncoder * encoder);
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 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);
gst_caps_unref (sinkcaps);
gst_caps_unref (srccaps);
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;
gst_ffmpeg_cfg_install_properties (gobject_class, klass->in_plugin,
PROP_CFG_BASE, AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM);
gobject_class->finalize = gst_ffmpegaudenc_finalize;
gstaudioencoder_class->start = GST_DEBUG_FUNCPTR (gst_ffmpegaudenc_start);
gstaudioencoder_class->stop = GST_DEBUG_FUNCPTR (gst_ffmpegaudenc_stop);
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);
GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_ENCODER_SINK_PAD (ffmpegaudenc));
/* ffmpeg objects */
ffmpegaudenc->context = avcodec_alloc_context3 (klass->in_plugin);
ffmpegaudenc->refcontext = avcodec_alloc_context3 (klass->in_plugin);
ffmpegaudenc->opened = FALSE;
ffmpegaudenc->frame = av_frame_alloc ();
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_frame_free (&ffmpegaudenc->frame);
avcodec_free_context (&ffmpegaudenc->context);
avcodec_free_context (&ffmpegaudenc->refcontext);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gboolean
gst_ffmpegaudenc_start (GstAudioEncoder * encoder)
{
GstFFMpegAudEnc *ffmpegaudenc = (GstFFMpegAudEnc *) encoder;
GstFFMpegAudEncClass *oclass =
(GstFFMpegAudEncClass *) G_OBJECT_GET_CLASS (ffmpegaudenc);
ffmpegaudenc->opened = FALSE;
ffmpegaudenc->need_reopen = FALSE;
avcodec_free_context (&ffmpegaudenc->context);
ffmpegaudenc->context = avcodec_alloc_context3 (oclass->in_plugin);
if (ffmpegaudenc->context == NULL) {
GST_DEBUG_OBJECT (ffmpegaudenc, "Failed to set context defaults");
return FALSE;
}
return TRUE;
}
static gboolean
gst_ffmpegaudenc_stop (GstAudioEncoder * encoder)
{
GstFFMpegAudEnc *ffmpegaudenc = (GstFFMpegAudEnc *) encoder;
/* close old session */
gst_ffmpeg_avcodec_close (ffmpegaudenc->context);
ffmpegaudenc->opened = FALSE;
ffmpegaudenc->need_reopen = FALSE;
return TRUE;
}
static void
gst_ffmpegaudenc_flush (GstAudioEncoder * encoder)
{
GstFFMpegAudEnc *ffmpegaudenc = (GstFFMpegAudEnc *) encoder;
if (ffmpegaudenc->opened) {
avcodec_flush_buffers (ffmpegaudenc->context);
}
}
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);
ffmpegaudenc->need_reopen = FALSE;
/* close old session */
if (ffmpegaudenc->opened) {
avcodec_free_context (&ffmpegaudenc->context);
ffmpegaudenc->opened = FALSE;
ffmpegaudenc->context = avcodec_alloc_context3 (oclass->in_plugin);
if (ffmpegaudenc->context == NULL) {
GST_DEBUG_OBJECT (ffmpegaudenc, "Failed to set context defaults");
return FALSE;
}
}
gst_ffmpeg_cfg_fill_context (G_OBJECT (ffmpegaudenc), ffmpegaudenc->context);
/* 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);
}
/* 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);
/* open codec */
if (gst_ffmpeg_avcodec_open (ffmpegaudenc->context, oclass->in_plugin) < 0) {
gst_caps_unref (allowed_caps);
avcodec_free_context (&ffmpegaudenc->context);
GST_DEBUG_OBJECT (ffmpegaudenc, "avenc_%s: Failed to open FFMPEG codec",
oclass->in_plugin->name);
ffmpegaudenc->context = avcodec_alloc_context3 (oclass->in_plugin);
if (ffmpegaudenc->context == NULL)
GST_DEBUG_OBJECT (ffmpegaudenc, "Failed to set context defaults");
if ((oclass->in_plugin->capabilities & AV_CODEC_CAP_EXPERIMENTAL) &&
ffmpegaudenc->context->strict_std_compliance !=
FF_COMPLIANCE_EXPERIMENTAL) {
GST_ELEMENT_ERROR (ffmpegaudenc, LIBRARY, SETTINGS,
("Codec is experimental, but settings don't allow encoders to "
"produce output of experimental quality"),
("This codec may not create output that is conformant to the specs "
"or of good quality. If you must use it anyway, set the "
"compliance property to experimental"));
}
return FALSE;
}
/* 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);
avcodec_free_context (&ffmpegaudenc->context);
GST_DEBUG ("Unsupported codec - no caps found");
ffmpegaudenc->context = avcodec_alloc_context3 (oclass->in_plugin);
if (ffmpegaudenc->context == NULL)
GST_DEBUG_OBJECT (ffmpegaudenc, "Failed to set context defaults");
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_fixate (icaps);
if (!gst_audio_encoder_set_output_format (GST_AUDIO_ENCODER (ffmpegaudenc),
icaps)) {
avcodec_free_context (&ffmpegaudenc->context);
gst_caps_unref (icaps);
ffmpegaudenc->context = avcodec_alloc_context3 (oclass->in_plugin);
if (ffmpegaudenc->context == NULL)
GST_DEBUG_OBJECT (ffmpegaudenc, "Failed to set context defaults");
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);
}
/* Store some tags */
{
GstTagList *tags = gst_tag_list_new_empty ();
const gchar *codec;
gst_tag_list_add (tags, GST_TAG_MERGE_REPLACE, GST_TAG_NOMINAL_BITRATE,
(guint) ffmpegaudenc->context->bit_rate, NULL);
if ((codec =
gst_ffmpeg_get_codecid_longname (ffmpegaudenc->context->codec_id)))
gst_tag_list_add (tags, GST_TAG_MERGE_REPLACE, GST_TAG_AUDIO_CODEC, codec,
NULL);
gst_audio_encoder_merge_tags (encoder, tags, GST_TAG_MERGE_REPLACE);
gst_tag_list_unref (tags);
}
/* success! */
ffmpegaudenc->opened = TRUE;
ffmpegaudenc->need_reopen = FALSE;
return TRUE;
}
static void
gst_ffmpegaudenc_free_avpacket (gpointer pkt)
{
av_packet_unref ((AVPacket *) pkt);
g_free (pkt);
}
typedef struct
{
GstBuffer *buffer;
GstMapInfo map;
guint8 **ext_data_array, *ext_data;
} BufferInfo;
static void
buffer_info_free (void *opaque, guint8 * data)
{
BufferInfo *info = opaque;
if (info->buffer) {
gst_buffer_unmap (info->buffer, &info->map);
gst_buffer_unref (info->buffer);
} else {
av_freep (&info->ext_data);
av_freep (&info->ext_data_array);
}
g_free (info);
}
static GstFlowReturn
gst_ffmpegaudenc_send_frame (GstFFMpegAudEnc * ffmpegaudenc, GstBuffer * buffer)
{
GstAudioEncoder *enc;
AVCodecContext *ctx;
GstFlowReturn ret;
gint res;
GstAudioInfo *info;
AVFrame *frame = ffmpegaudenc->frame;
gboolean planar;
gint nsamples = -1;
enc = GST_AUDIO_ENCODER (ffmpegaudenc);
ctx = ffmpegaudenc->context;
if (buffer != NULL) {
BufferInfo *buffer_info = g_new0 (BufferInfo, 1);
guint8 *audio_in;
guint in_size;
buffer_info->buffer = buffer;
gst_buffer_map (buffer, &buffer_info->map, GST_MAP_READ);
audio_in = buffer_info->map.data;
in_size = buffer_info->map.size;
GST_LOG_OBJECT (ffmpegaudenc, "encoding buffer %p size:%u", audio_in,
in_size);
info = gst_audio_encoder_get_audio_info (enc);
planar = av_sample_fmt_is_planar (ffmpegaudenc->context->sample_fmt);
frame->format = ffmpegaudenc->context->sample_fmt;
frame->sample_rate = ffmpegaudenc->context->sample_rate;
frame->channels = ffmpegaudenc->context->channels;
frame->channel_layout = ffmpegaudenc->context->channel_layout;
if (planar && info->channels > 1) {
gint channels;
gint i, j;
nsamples = frame->nb_samples = in_size / info->bpf;
channels = info->channels;
frame->buf[0] =
av_buffer_create (NULL, 0, buffer_info_free, buffer_info, 0);
if (info->channels > AV_NUM_DATA_POINTERS) {
buffer_info->ext_data_array = frame->extended_data =
av_malloc_array (info->channels, sizeof (uint8_t *));
} else {
frame->extended_data = frame->data;
}
buffer_info->ext_data = frame->extended_data[0] = av_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;
}
gst_buffer_unmap (buffer, &buffer_info->map);
gst_buffer_unref (buffer);
buffer_info->buffer = NULL;
} else {
frame->data[0] = audio_in;
frame->extended_data = frame->data;
frame->linesize[0] = in_size;
frame->nb_samples = nsamples = in_size / info->bpf;
frame->buf[0] =
av_buffer_create (NULL, 0, buffer_info_free, buffer_info, 0);
}
/* we have a frame to feed the encoder */
res = avcodec_send_frame (ctx, frame);
av_frame_unref (frame);
} else {
GstFFMpegAudEncClass *oclass =
(GstFFMpegAudEncClass *) G_OBJECT_GET_CLASS (ffmpegaudenc);
GST_LOG_OBJECT (ffmpegaudenc, "draining");
/* flushing the encoder */
res = avcodec_send_frame (ctx, NULL);
/* If AV_CODEC_CAP_ENCODER_FLUSH wasn't set, we need to re-open
* encoder */
if (!(oclass->in_plugin->capabilities & AV_CODEC_CAP_ENCODER_FLUSH)) {
GST_DEBUG_OBJECT (ffmpegaudenc, "Encoder needs reopen later");
/* we will reopen later handle_frame() */
ffmpegaudenc->need_reopen = TRUE;
}
}
if (res == 0) {
ret = GST_FLOW_OK;
} else if (res == AVERROR_EOF) {
ret = GST_FLOW_EOS;
} else { /* Any other return value is an error in our context */
ret = GST_FLOW_OK;
GST_WARNING_OBJECT (ffmpegaudenc, "Failed to encode buffer");
}
return ret;
}
static GstFlowReturn
gst_ffmpegaudenc_receive_packet (GstFFMpegAudEnc * ffmpegaudenc,
gboolean * got_packet)
{
GstAudioEncoder *enc;
AVCodecContext *ctx;
gint res;
GstFlowReturn ret;
AVPacket *pkt;
enc = GST_AUDIO_ENCODER (ffmpegaudenc);
ctx = ffmpegaudenc->context;
pkt = g_new0 (AVPacket, 1);
res = avcodec_receive_packet (ctx, pkt);
if (res == 0) {
GstBuffer *outbuf;
const uint8_t *side_data;
#if LIBAVFORMAT_VERSION_INT >= AV_VERSION_INT(58,130,0)
size_t side_data_length = 0;
#else
int side_data_length = 0;
#endif
GST_LOG_OBJECT (ffmpegaudenc, "pushing size %d", pkt->size);
outbuf =
gst_buffer_new_wrapped_full (GST_MEMORY_FLAG_READONLY, pkt->data,
pkt->size, 0, pkt->size, pkt, gst_ffmpegaudenc_free_avpacket);
if ((side_data =
av_packet_get_side_data (pkt, AV_PKT_DATA_SKIP_SAMPLES,
&side_data_length)) && side_data_length == 10) {
GstByteReader reader = GST_BYTE_READER_INIT (pkt->data, pkt->size);
guint32 start, end;
start = gst_byte_reader_get_uint32_le_unchecked (&reader);
end = gst_byte_reader_get_uint32_le_unchecked (&reader);
GST_LOG_OBJECT (ffmpegaudenc,
"got skip samples side data with start %u and end %u", start, end);
gst_buffer_add_audio_clipping_meta (outbuf, GST_FORMAT_DEFAULT, start,
end);
}
ret =
gst_audio_encoder_finish_frame (enc, outbuf,
pkt->duration > 0 ? pkt->duration : -1);
*got_packet = TRUE;
} else {
GST_LOG_OBJECT (ffmpegaudenc, "no output produced");
g_free (pkt);
ret = GST_FLOW_OK;
*got_packet = FALSE;
}
return ret;
}
static GstFlowReturn
gst_ffmpegaudenc_drain (GstFFMpegAudEnc * ffmpegaudenc)
{
GstFlowReturn ret = GST_FLOW_OK;
gboolean got_packet;
ret = gst_ffmpegaudenc_send_frame (ffmpegaudenc, NULL);
if (ret == GST_FLOW_OK) {
do {
ret = gst_ffmpegaudenc_receive_packet (ffmpegaudenc, &got_packet);
if (ret != GST_FLOW_OK)
break;
} while (got_packet);
}
/* NOTE: this may or may not work depending on capability */
avcodec_flush_buffers (ffmpegaudenc->context);
/* FFMpeg will return AVERROR_EOF if it's internal was fully drained
* then we are translating it to GST_FLOW_EOS. However, because this behavior
* is fully internal stuff of this implementation and gstaudioencoder
* baseclass doesn't convert this GST_FLOW_EOS to GST_FLOW_OK,
* convert this flow returned here */
if (ret == GST_FLOW_EOS)
ret = GST_FLOW_OK;
return ret;
}
static GstFlowReturn
gst_ffmpegaudenc_handle_frame (GstAudioEncoder * encoder, GstBuffer * inbuf)
{
GstFFMpegAudEnc *ffmpegaudenc;
GstFlowReturn ret;
gboolean got_packet;
ffmpegaudenc = (GstFFMpegAudEnc *) encoder;
if (G_UNLIKELY (!ffmpegaudenc->opened))
goto not_negotiated;
if (!inbuf)
return gst_ffmpegaudenc_drain (ffmpegaudenc);
/* endoder was drained or flushed, and ffmpeg encoder doesn't support
* flushing. We need to re-open encoder then */
if (ffmpegaudenc->need_reopen) {
GST_DEBUG_OBJECT (ffmpegaudenc, "Open encoder again");
if (!gst_ffmpegaudenc_set_format (encoder,
gst_audio_encoder_get_audio_info (encoder))) {
GST_ERROR_OBJECT (ffmpegaudenc, "Couldn't re-open encoder");
return GST_FLOW_NOT_NEGOTIATED;
}
}
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);
}
ret = gst_ffmpegaudenc_send_frame (ffmpegaudenc, inbuf);
if (ret != GST_FLOW_OK)
goto send_frame_failed;
do {
ret = gst_ffmpegaudenc_receive_packet (ffmpegaudenc, &got_packet);
} while (got_packet);
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;
}
send_frame_failed:
{
GST_DEBUG_OBJECT (ffmpegaudenc, "Failed to send frame %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;
ffmpegaudenc = (GstFFMpegAudEnc *) (object);
if (ffmpegaudenc->opened) {
GST_WARNING_OBJECT (ffmpegaudenc,
"Can't change properties once encoder is setup !");
return;
}
switch (prop_id) {
default:
if (!gst_ffmpeg_cfg_set_property (ffmpegaudenc->refcontext, value, pspec))
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_ffmpegaudenc_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec)
{
GstFFMpegAudEnc *ffmpegaudenc;
ffmpegaudenc = (GstFFMpegAudEnc *) (object);
switch (prop_id) {
default:
if (!gst_ffmpeg_cfg_get_property (ffmpegaudenc->refcontext, value, pspec))
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;
void *i = 0;
GST_LOG ("Registering encoders");
while ((in_plugin = (AVCodec *) av_codec_iterate (&i))) {
gchar *type_name;
guint rank;
/* Skip non-AV codecs */
if (in_plugin->type != AVMEDIA_TYPE_AUDIO)
continue;
/* no quasi codecs, please */
if (in_plugin->id == AV_CODEC_ID_PCM_S16LE_PLANAR ||
(in_plugin->id >= AV_CODEC_ID_PCM_S16LE &&
in_plugin->id <= AV_CODEC_ID_PCM_BLURAY) ||
(in_plugin->id >= AV_CODEC_ID_PCM_S8_PLANAR &&
in_plugin->id <= AV_CODEC_ID_PCM_F24LE)) {
continue;
}
/* 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);
continue;
}
/* only encoders */
if (!av_codec_is_encoder (in_plugin)) {
continue;
}
/* 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);
continue;
}
/* 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 AV_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);
}
GST_LOG ("Finished registering encoders");
return TRUE;
}