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* @param[in] frame AVFrame containing the raw audio data to be encoded. * May be NULL when flushing an encoder that has the * CODEC_CAP_DELAY capability set. The AVFrame itself should be null, not the frame.data pointer https://bugzilla.gnome.org/show_bug.cgi?id=724536
818 lines
25 KiB
C
818 lines
25 KiB
C
/* GStreamer
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* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
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* Copyright (C) <2012> Collabora Ltd.
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* Author: Sebastian Dröge <sebastian.droege@collabora.co.uk>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <assert.h>
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#include <string.h>
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/* for stats file handling */
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#include <stdio.h>
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#include <glib/gstdio.h>
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#include <errno.h>
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#include <libavcodec/avcodec.h>
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#include <gst/gst.h>
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#include "gstav.h"
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#include "gstavcodecmap.h"
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#include "gstavutils.h"
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#include "gstavaudenc.h"
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#define DEFAULT_AUDIO_BITRATE 128000
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enum
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{
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/* FILL ME */
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LAST_SIGNAL
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};
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enum
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{
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PROP_0,
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PROP_BIT_RATE,
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PROP_RTP_PAYLOAD_SIZE,
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PROP_COMPLIANCE,
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};
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/* A number of function prototypes are given so we can refer to them later. */
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static void gst_ffmpegaudenc_class_init (GstFFMpegAudEncClass * klass);
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static void gst_ffmpegaudenc_base_init (GstFFMpegAudEncClass * klass);
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static void gst_ffmpegaudenc_init (GstFFMpegAudEnc * ffmpegaudenc);
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static void gst_ffmpegaudenc_finalize (GObject * object);
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static GstCaps *gst_ffmpegaudenc_getcaps (GstAudioEncoder * encoder,
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GstCaps * filter);
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static gboolean gst_ffmpegaudenc_set_format (GstAudioEncoder * encoder,
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GstAudioInfo * info);
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static GstFlowReturn gst_ffmpegaudenc_handle_frame (GstAudioEncoder * encoder,
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GstBuffer * inbuf);
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static gboolean gst_ffmpegaudenc_start (GstAudioEncoder * encoder);
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static gboolean gst_ffmpegaudenc_stop (GstAudioEncoder * encoder);
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static void gst_ffmpegaudenc_flush (GstAudioEncoder * encoder);
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static void gst_ffmpegaudenc_set_property (GObject * object,
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guint prop_id, const GValue * value, GParamSpec * pspec);
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static void gst_ffmpegaudenc_get_property (GObject * object,
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guint prop_id, GValue * value, GParamSpec * pspec);
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#define GST_FFENC_PARAMS_QDATA g_quark_from_static_string("avenc-params")
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static GstElementClass *parent_class = NULL;
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/*static guint gst_ffmpegaudenc_signals[LAST_SIGNAL] = { 0 }; */
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static void
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gst_ffmpegaudenc_base_init (GstFFMpegAudEncClass * klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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AVCodec *in_plugin;
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GstPadTemplate *srctempl = NULL, *sinktempl = NULL;
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GstCaps *srccaps = NULL, *sinkcaps = NULL;
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gchar *longname, *description;
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in_plugin =
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(AVCodec *) g_type_get_qdata (G_OBJECT_CLASS_TYPE (klass),
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GST_FFENC_PARAMS_QDATA);
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g_assert (in_plugin != NULL);
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/* construct the element details struct */
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longname = g_strdup_printf ("libav %s encoder", in_plugin->long_name);
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description = g_strdup_printf ("libav %s encoder", in_plugin->name);
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gst_element_class_set_metadata (element_class, longname,
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"Codec/Encoder/Audio", description,
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"Wim Taymans <wim.taymans@gmail.com>, "
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"Ronald Bultje <rbultje@ronald.bitfreak.net>");
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g_free (longname);
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g_free (description);
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if (!(srccaps = gst_ffmpeg_codecid_to_caps (in_plugin->id, NULL, TRUE))) {
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GST_DEBUG ("Couldn't get source caps for encoder '%s'", in_plugin->name);
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srccaps = gst_caps_new_empty_simple ("unknown/unknown");
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}
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sinkcaps = gst_ffmpeg_codectype_to_audio_caps (NULL,
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in_plugin->id, TRUE, in_plugin);
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if (!sinkcaps) {
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GST_DEBUG ("Couldn't get sink caps for encoder '%s'", in_plugin->name);
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sinkcaps = gst_caps_new_empty_simple ("unknown/unknown");
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}
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/* pad templates */
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sinktempl = gst_pad_template_new ("sink", GST_PAD_SINK,
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GST_PAD_ALWAYS, sinkcaps);
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srctempl = gst_pad_template_new ("src", GST_PAD_SRC, GST_PAD_ALWAYS, srccaps);
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gst_element_class_add_pad_template (element_class, srctempl);
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gst_element_class_add_pad_template (element_class, sinktempl);
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klass->in_plugin = in_plugin;
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klass->srctempl = srctempl;
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klass->sinktempl = sinktempl;
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return;
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}
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static void
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gst_ffmpegaudenc_class_init (GstFFMpegAudEncClass * klass)
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{
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GObjectClass *gobject_class;
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GstAudioEncoderClass *gstaudioencoder_class;
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gobject_class = (GObjectClass *) klass;
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gstaudioencoder_class = (GstAudioEncoderClass *) klass;
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parent_class = g_type_class_peek_parent (klass);
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gobject_class->set_property = gst_ffmpegaudenc_set_property;
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gobject_class->get_property = gst_ffmpegaudenc_get_property;
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/* FIXME: could use -1 for a sensible per-codec defaults */
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_BIT_RATE,
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g_param_spec_int ("bitrate", "Bit Rate",
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"Target Audio Bitrate", 0, G_MAXINT, DEFAULT_AUDIO_BITRATE,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_COMPLIANCE,
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g_param_spec_enum ("compliance", "Compliance",
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"Adherence of the encoder to the specifications",
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GST_TYPE_FFMPEG_COMPLIANCE, FFMPEG_DEFAULT_COMPLIANCE,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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gobject_class->finalize = gst_ffmpegaudenc_finalize;
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gstaudioencoder_class->start = GST_DEBUG_FUNCPTR (gst_ffmpegaudenc_start);
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gstaudioencoder_class->stop = GST_DEBUG_FUNCPTR (gst_ffmpegaudenc_stop);
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gstaudioencoder_class->getcaps = GST_DEBUG_FUNCPTR (gst_ffmpegaudenc_getcaps);
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gstaudioencoder_class->flush = GST_DEBUG_FUNCPTR (gst_ffmpegaudenc_flush);
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gstaudioencoder_class->set_format =
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GST_DEBUG_FUNCPTR (gst_ffmpegaudenc_set_format);
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gstaudioencoder_class->handle_frame =
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GST_DEBUG_FUNCPTR (gst_ffmpegaudenc_handle_frame);
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}
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static void
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gst_ffmpegaudenc_init (GstFFMpegAudEnc * ffmpegaudenc)
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{
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GstFFMpegAudEncClass *klass =
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(GstFFMpegAudEncClass *) G_OBJECT_GET_CLASS (ffmpegaudenc);
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/* ffmpeg objects */
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ffmpegaudenc->context = avcodec_alloc_context3 (klass->in_plugin);
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ffmpegaudenc->opened = FALSE;
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ffmpegaudenc->compliance = FFMPEG_DEFAULT_COMPLIANCE;
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gst_audio_encoder_set_drainable (GST_AUDIO_ENCODER (ffmpegaudenc), TRUE);
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}
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static void
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gst_ffmpegaudenc_finalize (GObject * object)
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{
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GstFFMpegAudEnc *ffmpegaudenc = (GstFFMpegAudEnc *) object;
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/* clean up remaining allocated data */
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gst_ffmpeg_avcodec_close (ffmpegaudenc->context);
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av_free (ffmpegaudenc->context);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static gboolean
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gst_ffmpegaudenc_start (GstAudioEncoder * encoder)
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{
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GstFFMpegAudEnc *ffmpegaudenc = (GstFFMpegAudEnc *) encoder;
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GstFFMpegAudEncClass *oclass =
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(GstFFMpegAudEncClass *) G_OBJECT_GET_CLASS (ffmpegaudenc);
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gst_ffmpeg_avcodec_close (ffmpegaudenc->context);
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if (avcodec_get_context_defaults3 (ffmpegaudenc->context,
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oclass->in_plugin) < 0) {
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GST_DEBUG_OBJECT (ffmpegaudenc, "Failed to set context defaults");
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return FALSE;
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}
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return TRUE;
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}
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static gboolean
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gst_ffmpegaudenc_stop (GstAudioEncoder * encoder)
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{
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GstFFMpegAudEnc *ffmpegaudenc = (GstFFMpegAudEnc *) encoder;
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/* close old session */
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gst_ffmpeg_avcodec_close (ffmpegaudenc->context);
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ffmpegaudenc->opened = FALSE;
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return TRUE;
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}
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static void
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gst_ffmpegaudenc_flush (GstAudioEncoder * encoder)
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{
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GstFFMpegAudEnc *ffmpegaudenc = (GstFFMpegAudEnc *) encoder;
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if (ffmpegaudenc->opened) {
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avcodec_flush_buffers (ffmpegaudenc->context);
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}
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}
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static GstCaps *
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gst_ffmpegaudenc_getcaps (GstAudioEncoder * encoder, GstCaps * filter)
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{
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GstFFMpegAudEnc *ffmpegaudenc = (GstFFMpegAudEnc *) encoder;
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GstCaps *caps = NULL;
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GST_DEBUG_OBJECT (ffmpegaudenc, "getting caps");
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/* audio needs no special care */
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caps = gst_audio_encoder_proxy_getcaps (encoder, NULL, filter);
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GST_DEBUG_OBJECT (ffmpegaudenc, "audio caps, return %" GST_PTR_FORMAT, caps);
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return caps;
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}
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static gboolean
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gst_ffmpegaudenc_set_format (GstAudioEncoder * encoder, GstAudioInfo * info)
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{
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GstFFMpegAudEnc *ffmpegaudenc = (GstFFMpegAudEnc *) encoder;
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GstCaps *other_caps;
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GstCaps *allowed_caps;
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GstCaps *icaps;
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gsize frame_size;
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GstFFMpegAudEncClass *oclass =
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(GstFFMpegAudEncClass *) G_OBJECT_GET_CLASS (ffmpegaudenc);
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/* close old session */
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if (ffmpegaudenc->opened) {
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gst_ffmpeg_avcodec_close (ffmpegaudenc->context);
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ffmpegaudenc->opened = FALSE;
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if (avcodec_get_context_defaults3 (ffmpegaudenc->context,
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oclass->in_plugin) < 0) {
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GST_DEBUG_OBJECT (ffmpegaudenc, "Failed to set context defaults");
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return FALSE;
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}
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}
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/* if we set it in _getcaps we should set it also in _link */
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ffmpegaudenc->context->strict_std_compliance = ffmpegaudenc->compliance;
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/* user defined properties */
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if (ffmpegaudenc->bitrate > 0) {
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GST_INFO_OBJECT (ffmpegaudenc, "Setting avcontext to bitrate %d",
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ffmpegaudenc->bitrate);
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ffmpegaudenc->context->bit_rate = ffmpegaudenc->bitrate;
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ffmpegaudenc->context->bit_rate_tolerance = ffmpegaudenc->bitrate;
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} else {
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GST_INFO_OBJECT (ffmpegaudenc, "Using avcontext default bitrate %d",
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ffmpegaudenc->context->bit_rate);
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}
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/* RTP payload used for GOB production (for Asterisk) */
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if (ffmpegaudenc->rtp_payload_size) {
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ffmpegaudenc->context->rtp_payload_size = ffmpegaudenc->rtp_payload_size;
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}
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/* some other defaults */
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ffmpegaudenc->context->rc_strategy = 2;
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ffmpegaudenc->context->b_frame_strategy = 0;
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ffmpegaudenc->context->coder_type = 0;
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ffmpegaudenc->context->context_model = 0;
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ffmpegaudenc->context->scenechange_threshold = 0;
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/* fetch pix_fmt and so on */
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gst_ffmpeg_audioinfo_to_context (info, ffmpegaudenc->context);
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if (!ffmpegaudenc->context->time_base.den) {
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ffmpegaudenc->context->time_base.den = GST_AUDIO_INFO_RATE (info);
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ffmpegaudenc->context->time_base.num = 1;
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ffmpegaudenc->context->ticks_per_frame = 1;
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}
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if (ffmpegaudenc->context->channel_layout) {
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gst_ffmpeg_channel_layout_to_gst (ffmpegaudenc->context->channel_layout,
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ffmpegaudenc->context->channels, ffmpegaudenc->ffmpeg_layout);
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ffmpegaudenc->needs_reorder =
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(memcmp (ffmpegaudenc->ffmpeg_layout, info->position,
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sizeof (GstAudioChannelPosition) *
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ffmpegaudenc->context->channels) != 0);
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}
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/* open codec */
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if (gst_ffmpeg_avcodec_open (ffmpegaudenc->context, oclass->in_plugin) < 0) {
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gst_ffmpeg_avcodec_close (ffmpegaudenc->context);
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GST_DEBUG_OBJECT (ffmpegaudenc, "avenc_%s: Failed to open FFMPEG codec",
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oclass->in_plugin->name);
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if (avcodec_get_context_defaults3 (ffmpegaudenc->context,
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oclass->in_plugin) < 0)
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GST_DEBUG_OBJECT (ffmpegaudenc, "Failed to set context defaults");
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if ((oclass->in_plugin->capabilities & CODEC_CAP_EXPERIMENTAL) &&
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ffmpegaudenc->compliance != GST_FFMPEG_EXPERIMENTAL) {
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GST_ELEMENT_ERROR (ffmpegaudenc, LIBRARY, SETTINGS,
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("Codec is experimental, but settings don't allow encoders to "
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"produce output of experimental quality"),
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("This codec may not create output that is conformant to the specs "
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"or of good quality. If you must use it anyway, set the "
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"compliance property to experimental"));
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}
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return FALSE;
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}
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/* some codecs support more than one format, first auto-choose one */
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GST_DEBUG_OBJECT (ffmpegaudenc, "picking an output format ...");
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allowed_caps = gst_pad_get_allowed_caps (GST_AUDIO_ENCODER_SRC_PAD (encoder));
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if (!allowed_caps) {
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GST_DEBUG_OBJECT (ffmpegaudenc, "... but no peer, using template caps");
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/* we need to copy because get_allowed_caps returns a ref, and
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* get_pad_template_caps doesn't */
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allowed_caps =
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gst_pad_get_pad_template_caps (GST_AUDIO_ENCODER_SRC_PAD (encoder));
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}
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GST_DEBUG_OBJECT (ffmpegaudenc, "chose caps %" GST_PTR_FORMAT, allowed_caps);
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gst_ffmpeg_caps_with_codecid (oclass->in_plugin->id,
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oclass->in_plugin->type, allowed_caps, ffmpegaudenc->context);
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/* try to set this caps on the other side */
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other_caps = gst_ffmpeg_codecid_to_caps (oclass->in_plugin->id,
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ffmpegaudenc->context, TRUE);
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if (!other_caps) {
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gst_caps_unref (allowed_caps);
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gst_ffmpeg_avcodec_close (ffmpegaudenc->context);
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GST_DEBUG ("Unsupported codec - no caps found");
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if (avcodec_get_context_defaults3 (ffmpegaudenc->context,
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oclass->in_plugin) < 0)
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GST_DEBUG_OBJECT (ffmpegaudenc, "Failed to set context defaults");
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return FALSE;
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}
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icaps = gst_caps_intersect (allowed_caps, other_caps);
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gst_caps_unref (allowed_caps);
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gst_caps_unref (other_caps);
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if (gst_caps_is_empty (icaps)) {
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gst_caps_unref (icaps);
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return FALSE;
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}
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icaps = gst_caps_truncate (icaps);
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if (!gst_audio_encoder_set_output_format (GST_AUDIO_ENCODER (ffmpegaudenc),
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icaps)) {
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gst_ffmpeg_avcodec_close (ffmpegaudenc->context);
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gst_caps_unref (icaps);
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if (avcodec_get_context_defaults3 (ffmpegaudenc->context,
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oclass->in_plugin) < 0)
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GST_DEBUG_OBJECT (ffmpegaudenc, "Failed to set context defaults");
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return FALSE;
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}
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gst_caps_unref (icaps);
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frame_size = ffmpegaudenc->context->frame_size;
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if (frame_size > 1) {
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gst_audio_encoder_set_frame_samples_min (GST_AUDIO_ENCODER (ffmpegaudenc),
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frame_size);
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gst_audio_encoder_set_frame_samples_max (GST_AUDIO_ENCODER (ffmpegaudenc),
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frame_size);
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gst_audio_encoder_set_frame_max (GST_AUDIO_ENCODER (ffmpegaudenc), 1);
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} else {
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gst_audio_encoder_set_frame_samples_min (GST_AUDIO_ENCODER (ffmpegaudenc),
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0);
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gst_audio_encoder_set_frame_samples_max (GST_AUDIO_ENCODER (ffmpegaudenc),
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0);
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gst_audio_encoder_set_frame_max (GST_AUDIO_ENCODER (ffmpegaudenc), 0);
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}
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/* success! */
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ffmpegaudenc->opened = TRUE;
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return TRUE;
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}
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static GstFlowReturn
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gst_ffmpegaudenc_encode_audio (GstFFMpegAudEnc * ffmpegaudenc,
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guint8 * audio_in, guint in_size, gint * have_data)
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{
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GstAudioEncoder *enc;
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AVCodecContext *ctx;
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gint res;
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GstFlowReturn ret;
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GstAudioInfo *info;
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AVPacket pkt;
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AVFrame frame;
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gboolean planar;
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enc = GST_AUDIO_ENCODER (ffmpegaudenc);
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|
|
|
ctx = ffmpegaudenc->context;
|
|
|
|
GST_LOG_OBJECT (ffmpegaudenc, "encoding buffer %p size:%u", audio_in,
|
|
in_size);
|
|
|
|
memset (&pkt, 0, sizeof (pkt));
|
|
|
|
if (audio_in != NULL) {
|
|
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;
|
|
}
|
|
|
|
/* we have a frame to feed the encoder */
|
|
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);
|
|
|
|
} else {
|
|
/* flushing the encoder */
|
|
res = avcodec_encode_audio2 (ctx, &pkt, NULL, have_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;
|
|
case PROP_COMPLIANCE:
|
|
ffmpegaudenc->compliance = g_value_get_enum (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;
|
|
case PROP_COMPLIANCE:
|
|
g_value_set_enum (value, ffmpegaudenc->compliance);
|
|
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 >= AV_CODEC_ID_PCM_S16LE &&
|
|
in_plugin->id <= AV_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 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);
|
|
|
|
next:
|
|
in_plugin = av_codec_next (in_plugin);
|
|
}
|
|
|
|
GST_LOG ("Finished registering encoders");
|
|
|
|
return TRUE;
|
|
}
|