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846 lines
26 KiB
C
846 lines
26 KiB
C
/* GStreamer Speex Encoder
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* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
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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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/**
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* SECTION:element-speexenc
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* @title: speexenc
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* @see_also: speexdec, oggmux
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*
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* This element encodes audio as a Speex stream.
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* [Speex](http://www.speex.org/) is a royalty-free
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* audio codec maintained by the [Xiph.org Foundation](http://www.xiph.org/).
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*
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* ## Example pipelines
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* |[
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* gst-launch-1.0 audiotestsrc num-buffers=100 ! speexenc ! oggmux ! filesink location=beep.ogg
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* ]| Encode an Ogg/Speex file.
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*
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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 <stdlib.h>
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#include <string.h>
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#include <time.h>
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#include <math.h>
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#include <speex/speex.h>
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#include <speex/speex_stereo.h>
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#include <gst/gsttagsetter.h>
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#include <gst/tag/tag.h>
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#include <gst/audio/audio.h>
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#include "gstspeexenc.h"
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GST_DEBUG_CATEGORY_STATIC (speexenc_debug);
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#define GST_CAT_DEFAULT speexenc_debug
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#define FORMAT_STR GST_AUDIO_NE(S16)
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static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw, "
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"format = (string) " FORMAT_STR ", "
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"layout = (string) interleaved, "
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"rate = (int) [ 6000, 48000 ], "
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"channels = (int) 1; "
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"audio/x-raw, "
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"format = (string) " FORMAT_STR ", "
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"layout = (string) interleaved, "
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"rate = (int) [ 6000, 48000 ], "
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"channels = (int) 2, " "channel-mask = (bitmask) 0x3")
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);
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static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-speex, "
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"rate = (int) [ 6000, 48000 ], " "channels = (int) [ 1, 2]")
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);
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#define DEFAULT_QUALITY 8.0
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#define DEFAULT_BITRATE 0
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#define DEFAULT_MODE GST_SPEEX_ENC_MODE_AUTO
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#define DEFAULT_VBR FALSE
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#define DEFAULT_ABR 0
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#define DEFAULT_VAD FALSE
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#define DEFAULT_DTX FALSE
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#define DEFAULT_COMPLEXITY 3
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#define DEFAULT_NFRAMES 1
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enum
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{
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PROP_0,
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PROP_QUALITY,
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PROP_BITRATE,
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PROP_MODE,
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PROP_VBR,
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PROP_ABR,
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PROP_VAD,
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PROP_DTX,
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PROP_COMPLEXITY,
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PROP_NFRAMES,
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PROP_LAST_MESSAGE
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};
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#define GST_TYPE_SPEEX_ENC_MODE (gst_speex_enc_mode_get_type())
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static GType
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gst_speex_enc_mode_get_type (void)
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{
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static GType speex_enc_mode_type = 0;
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static const GEnumValue speex_enc_modes[] = {
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{GST_SPEEX_ENC_MODE_AUTO, "Auto", "auto"},
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{GST_SPEEX_ENC_MODE_UWB, "Ultra Wide Band", "uwb"},
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{GST_SPEEX_ENC_MODE_WB, "Wide Band", "wb"},
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{GST_SPEEX_ENC_MODE_NB, "Narrow Band", "nb"},
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{0, NULL, NULL},
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};
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if (G_UNLIKELY (speex_enc_mode_type == 0)) {
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speex_enc_mode_type = g_enum_register_static ("GstSpeexEncMode",
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speex_enc_modes);
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}
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return speex_enc_mode_type;
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}
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static void gst_speex_enc_finalize (GObject * object);
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static gboolean gst_speex_enc_setup (GstSpeexEnc * enc);
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static void gst_speex_enc_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static void gst_speex_enc_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static GstFlowReturn gst_speex_enc_encode (GstSpeexEnc * enc, GstBuffer * buf);
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static gboolean gst_speex_enc_start (GstAudioEncoder * enc);
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static gboolean gst_speex_enc_stop (GstAudioEncoder * enc);
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static gboolean gst_speex_enc_set_format (GstAudioEncoder * enc,
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GstAudioInfo * info);
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static GstFlowReturn gst_speex_enc_handle_frame (GstAudioEncoder * enc,
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GstBuffer * in_buf);
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static gboolean gst_speex_enc_sink_event (GstAudioEncoder * enc,
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GstEvent * event);
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#define gst_speex_enc_parent_class parent_class
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G_DEFINE_TYPE_WITH_CODE (GstSpeexEnc, gst_speex_enc, GST_TYPE_AUDIO_ENCODER,
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G_IMPLEMENT_INTERFACE (GST_TYPE_TAG_SETTER, NULL);
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G_IMPLEMENT_INTERFACE (GST_TYPE_PRESET, NULL));
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static void
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gst_speex_enc_class_init (GstSpeexEncClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstAudioEncoderClass *base_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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base_class = (GstAudioEncoderClass *) klass;
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gobject_class->finalize = gst_speex_enc_finalize;
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gobject_class->set_property = gst_speex_enc_set_property;
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gobject_class->get_property = gst_speex_enc_get_property;
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base_class->start = GST_DEBUG_FUNCPTR (gst_speex_enc_start);
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base_class->stop = GST_DEBUG_FUNCPTR (gst_speex_enc_stop);
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base_class->set_format = GST_DEBUG_FUNCPTR (gst_speex_enc_set_format);
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base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_speex_enc_handle_frame);
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base_class->sink_event = GST_DEBUG_FUNCPTR (gst_speex_enc_sink_event);
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_QUALITY,
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g_param_spec_float ("quality", "Quality", "Encoding quality",
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0.0, 10.0, DEFAULT_QUALITY,
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G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_BITRATE,
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g_param_spec_int ("bitrate", "Encoding Bit-rate",
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"Specify an encoding bit-rate (in bps). (0 = automatic)",
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0, G_MAXINT, DEFAULT_BITRATE,
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G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_MODE,
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g_param_spec_enum ("mode", "Mode", "The encoding mode",
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GST_TYPE_SPEEX_ENC_MODE, GST_SPEEX_ENC_MODE_AUTO,
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G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_VBR,
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g_param_spec_boolean ("vbr", "VBR",
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"Enable variable bit-rate", DEFAULT_VBR,
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G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_ABR,
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g_param_spec_int ("abr", "ABR",
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"Enable average bit-rate (0 = disabled)",
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0, G_MAXINT, DEFAULT_ABR,
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G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_VAD,
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g_param_spec_boolean ("vad", "VAD",
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"Enable voice activity detection", DEFAULT_VAD,
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G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_DTX,
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g_param_spec_boolean ("dtx", "DTX",
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"Enable discontinuous transmission", DEFAULT_DTX,
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G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_COMPLEXITY,
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g_param_spec_int ("complexity", "Complexity",
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"Set encoding complexity",
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0, G_MAXINT, DEFAULT_COMPLEXITY,
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G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_NFRAMES,
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g_param_spec_int ("nframes", "NFrames",
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"Number of frames per buffer",
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0, G_MAXINT, DEFAULT_NFRAMES,
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G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_LAST_MESSAGE,
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g_param_spec_string ("last-message", "last-message",
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"The last status message", NULL,
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G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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gst_element_class_add_static_pad_template (gstelement_class, &src_factory);
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gst_element_class_add_static_pad_template (gstelement_class, &sink_factory);
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gst_element_class_set_static_metadata (gstelement_class,
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"Speex audio encoder", "Codec/Encoder/Audio",
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"Encodes audio in Speex format", "Wim Taymans <wim@fluendo.com>");
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GST_DEBUG_CATEGORY_INIT (speexenc_debug, "speexenc", 0, "Speex encoder");
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gst_type_mark_as_plugin_api (GST_TYPE_SPEEX_ENC_MODE, 0);
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}
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static void
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gst_speex_enc_finalize (GObject * object)
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{
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GstSpeexEnc *enc;
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enc = GST_SPEEX_ENC (object);
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g_free (enc->last_message);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static void
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gst_speex_enc_init (GstSpeexEnc * enc)
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{
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GstAudioEncoder *benc = GST_AUDIO_ENCODER (enc);
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/* arrange granulepos marking (and required perfect ts) */
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gst_audio_encoder_set_mark_granule (benc, TRUE);
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gst_audio_encoder_set_perfect_timestamp (benc, TRUE);
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GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_ENCODER_SINK_PAD (enc));
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}
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static gboolean
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gst_speex_enc_start (GstAudioEncoder * benc)
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{
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GstSpeexEnc *enc = GST_SPEEX_ENC (benc);
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GST_DEBUG_OBJECT (enc, "start");
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speex_bits_init (&enc->bits);
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enc->tags = gst_tag_list_new_empty ();
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enc->header_sent = FALSE;
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enc->encoded_samples = 0;
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return TRUE;
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}
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static gboolean
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gst_speex_enc_stop (GstAudioEncoder * benc)
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{
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GstSpeexEnc *enc = GST_SPEEX_ENC (benc);
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GST_DEBUG_OBJECT (enc, "stop");
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enc->header_sent = FALSE;
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if (enc->state) {
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speex_encoder_destroy (enc->state);
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enc->state = NULL;
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}
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speex_bits_destroy (&enc->bits);
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speex_bits_set_bit_buffer (&enc->bits, NULL, 0);
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gst_tag_list_unref (enc->tags);
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enc->tags = NULL;
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gst_tag_setter_reset_tags (GST_TAG_SETTER (enc));
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return TRUE;
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}
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static gint64
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gst_speex_enc_get_latency (GstSpeexEnc * enc)
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{
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/* See the Speex manual section "Latency and algorithmic delay" */
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if (enc->rate == 8000)
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return 30 * GST_MSECOND;
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else
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return 34 * GST_MSECOND;
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}
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static gboolean
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gst_speex_enc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
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{
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GstSpeexEnc *enc;
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enc = GST_SPEEX_ENC (benc);
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enc->channels = GST_AUDIO_INFO_CHANNELS (info);
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enc->rate = GST_AUDIO_INFO_RATE (info);
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/* handle reconfigure */
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if (enc->state) {
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speex_encoder_destroy (enc->state);
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enc->state = NULL;
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}
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if (!gst_speex_enc_setup (enc))
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return FALSE;
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/* feedback to base class */
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gst_audio_encoder_set_latency (benc,
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gst_speex_enc_get_latency (enc), gst_speex_enc_get_latency (enc));
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gst_audio_encoder_set_lookahead (benc, enc->lookahead);
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if (enc->nframes == 0) {
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/* as many frames as available input allows */
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gst_audio_encoder_set_frame_samples_min (benc, enc->frame_size);
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gst_audio_encoder_set_frame_samples_max (benc, enc->frame_size);
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gst_audio_encoder_set_frame_max (benc, 0);
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} else {
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/* exactly as many frames as configured */
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gst_audio_encoder_set_frame_samples_min (benc,
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enc->frame_size * enc->nframes);
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gst_audio_encoder_set_frame_samples_max (benc,
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enc->frame_size * enc->nframes);
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gst_audio_encoder_set_frame_max (benc, 1);
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}
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return TRUE;
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}
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static GstBuffer *
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gst_speex_enc_create_metadata_buffer (GstSpeexEnc * enc)
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{
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const GstTagList *user_tags;
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GstTagList *merged_tags;
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GstBuffer *comments = NULL;
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user_tags = gst_tag_setter_get_tag_list (GST_TAG_SETTER (enc));
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GST_DEBUG_OBJECT (enc, "upstream tags = %" GST_PTR_FORMAT, enc->tags);
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GST_DEBUG_OBJECT (enc, "user-set tags = %" GST_PTR_FORMAT, user_tags);
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/* gst_tag_list_merge() will handle NULL for either or both lists fine */
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merged_tags = gst_tag_list_merge (user_tags, enc->tags,
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gst_tag_setter_get_tag_merge_mode (GST_TAG_SETTER (enc)));
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if (merged_tags == NULL)
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merged_tags = gst_tag_list_new_empty ();
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GST_DEBUG_OBJECT (enc, "merged tags = %" GST_PTR_FORMAT, merged_tags);
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comments = gst_tag_list_to_vorbiscomment_buffer (merged_tags, NULL,
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0, "Encoded with GStreamer Speexenc");
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gst_tag_list_unref (merged_tags);
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GST_BUFFER_OFFSET (comments) = 0;
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GST_BUFFER_OFFSET_END (comments) = 0;
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return comments;
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}
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static void
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gst_speex_enc_set_last_msg (GstSpeexEnc * enc, const gchar * msg)
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{
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g_free (enc->last_message);
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enc->last_message = g_strdup (msg);
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GST_WARNING_OBJECT (enc, "%s", msg);
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g_object_notify (G_OBJECT (enc), "last-message");
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}
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static gboolean
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gst_speex_enc_setup (GstSpeexEnc * enc)
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{
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switch (enc->mode) {
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case GST_SPEEX_ENC_MODE_UWB:
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GST_LOG_OBJECT (enc, "configuring for requested UWB mode");
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enc->speex_mode = speex_lib_get_mode (SPEEX_MODEID_UWB);
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break;
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case GST_SPEEX_ENC_MODE_WB:
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GST_LOG_OBJECT (enc, "configuring for requested WB mode");
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enc->speex_mode = speex_lib_get_mode (SPEEX_MODEID_WB);
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break;
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case GST_SPEEX_ENC_MODE_NB:
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GST_LOG_OBJECT (enc, "configuring for requested NB mode");
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enc->speex_mode = speex_lib_get_mode (SPEEX_MODEID_NB);
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break;
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case GST_SPEEX_ENC_MODE_AUTO:
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/* fall through */
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GST_LOG_OBJECT (enc, "finding best mode");
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default:
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break;
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}
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if (enc->rate > 25000) {
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if (enc->mode == GST_SPEEX_ENC_MODE_AUTO) {
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GST_LOG_OBJECT (enc, "selected UWB mode for samplerate %d", enc->rate);
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enc->speex_mode = speex_lib_get_mode (SPEEX_MODEID_UWB);
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} else {
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if (enc->speex_mode != speex_lib_get_mode (SPEEX_MODEID_UWB)) {
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gst_speex_enc_set_last_msg (enc,
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"Warning: suggest to use ultra wide band mode for this rate");
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}
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}
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} else if (enc->rate > 12500) {
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if (enc->mode == GST_SPEEX_ENC_MODE_AUTO) {
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GST_LOG_OBJECT (enc, "selected WB mode for samplerate %d", enc->rate);
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enc->speex_mode = speex_lib_get_mode (SPEEX_MODEID_WB);
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} else {
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if (enc->speex_mode != speex_lib_get_mode (SPEEX_MODEID_WB)) {
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gst_speex_enc_set_last_msg (enc,
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"Warning: suggest to use wide band mode for this rate");
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}
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}
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} else {
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if (enc->mode == GST_SPEEX_ENC_MODE_AUTO) {
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GST_LOG_OBJECT (enc, "selected NB mode for samplerate %d", enc->rate);
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enc->speex_mode = speex_lib_get_mode (SPEEX_MODEID_NB);
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} else {
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if (enc->speex_mode != speex_lib_get_mode (SPEEX_MODEID_NB)) {
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gst_speex_enc_set_last_msg (enc,
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"Warning: suggest to use narrow band mode for this rate");
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}
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}
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}
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if (enc->rate != 8000 && enc->rate != 16000 && enc->rate != 32000) {
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gst_speex_enc_set_last_msg (enc,
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"Warning: speex is optimized for 8, 16 and 32 KHz");
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}
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speex_init_header (&enc->header, enc->rate, 1, enc->speex_mode);
|
|
enc->header.frames_per_packet = enc->nframes;
|
|
enc->header.vbr = enc->vbr;
|
|
enc->header.nb_channels = enc->channels;
|
|
|
|
/*Initialize Speex encoder */
|
|
enc->state = speex_encoder_init (enc->speex_mode);
|
|
|
|
speex_encoder_ctl (enc->state, SPEEX_GET_FRAME_SIZE, &enc->frame_size);
|
|
speex_encoder_ctl (enc->state, SPEEX_SET_COMPLEXITY, &enc->complexity);
|
|
speex_encoder_ctl (enc->state, SPEEX_SET_SAMPLING_RATE, &enc->rate);
|
|
|
|
if (enc->vbr)
|
|
speex_encoder_ctl (enc->state, SPEEX_SET_VBR_QUALITY, &enc->quality);
|
|
else {
|
|
gint tmp = floor (enc->quality);
|
|
|
|
speex_encoder_ctl (enc->state, SPEEX_SET_QUALITY, &tmp);
|
|
}
|
|
if (enc->bitrate) {
|
|
if (enc->quality >= 0.0 && enc->vbr) {
|
|
gst_speex_enc_set_last_msg (enc,
|
|
"Warning: bitrate option is overriding quality");
|
|
}
|
|
speex_encoder_ctl (enc->state, SPEEX_SET_BITRATE, &enc->bitrate);
|
|
}
|
|
if (enc->vbr) {
|
|
gint tmp = 1;
|
|
|
|
speex_encoder_ctl (enc->state, SPEEX_SET_VBR, &tmp);
|
|
} else if (enc->vad) {
|
|
gint tmp = 1;
|
|
|
|
speex_encoder_ctl (enc->state, SPEEX_SET_VAD, &tmp);
|
|
}
|
|
|
|
if (enc->dtx) {
|
|
gint tmp = 1;
|
|
|
|
speex_encoder_ctl (enc->state, SPEEX_SET_DTX, &tmp);
|
|
}
|
|
|
|
if (enc->dtx && !(enc->vbr || enc->abr || enc->vad)) {
|
|
gst_speex_enc_set_last_msg (enc,
|
|
"Warning: dtx is useless without vad, vbr or abr");
|
|
} else if ((enc->vbr || enc->abr) && (enc->vad)) {
|
|
gst_speex_enc_set_last_msg (enc,
|
|
"Warning: vad is already implied by vbr or abr");
|
|
}
|
|
|
|
if (enc->abr) {
|
|
speex_encoder_ctl (enc->state, SPEEX_SET_ABR, &enc->abr);
|
|
}
|
|
|
|
speex_encoder_ctl (enc->state, SPEEX_GET_LOOKAHEAD, &enc->lookahead);
|
|
|
|
GST_LOG_OBJECT (enc, "we have frame size %d, lookahead %d", enc->frame_size,
|
|
enc->lookahead);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_speex_enc_sink_event (GstAudioEncoder * benc, GstEvent * event)
|
|
{
|
|
GstSpeexEnc *enc;
|
|
|
|
enc = GST_SPEEX_ENC (benc);
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_TAG:
|
|
{
|
|
if (enc->tags) {
|
|
GstTagList *list;
|
|
|
|
gst_event_parse_tag (event, &list);
|
|
gst_tag_list_insert (enc->tags, list,
|
|
gst_tag_setter_get_tag_merge_mode (GST_TAG_SETTER (enc)));
|
|
} else {
|
|
g_assert_not_reached ();
|
|
}
|
|
break;
|
|
}
|
|
case GST_EVENT_SEGMENT:
|
|
enc->encoded_samples = 0;
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
/* we only peeked, let base class handle it */
|
|
return GST_AUDIO_ENCODER_CLASS (parent_class)->sink_event (benc, event);
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_speex_enc_encode (GstSpeexEnc * enc, GstBuffer * buf)
|
|
{
|
|
gint frame_size = enc->frame_size;
|
|
gint bytes = frame_size * 2 * enc->channels, samples;
|
|
gint outsize, written, dtx_ret = 0;
|
|
GstMapInfo map;
|
|
guint8 *data, *data0 = NULL, *bdata;
|
|
gsize bsize, size;
|
|
GstBuffer *outbuf;
|
|
GstFlowReturn ret = GST_FLOW_OK;
|
|
GstSegment *segment;
|
|
GstClockTime duration;
|
|
|
|
if (G_LIKELY (buf)) {
|
|
gst_buffer_map (buf, &map, GST_MAP_READ);
|
|
bdata = map.data;
|
|
bsize = map.size;
|
|
|
|
if (G_UNLIKELY (bsize % bytes)) {
|
|
GST_DEBUG_OBJECT (enc, "draining; adding silence samples");
|
|
|
|
/* If encoding part of a frame, and we have no set stop time on
|
|
* the output segment, we update the segment stop time to reflect
|
|
* the last sample. This will let oggmux set the last page's
|
|
* granpos to tell a decoder the dummy samples should be clipped.
|
|
*/
|
|
segment = &GST_AUDIO_ENCODER_OUTPUT_SEGMENT (enc);
|
|
GST_DEBUG_OBJECT (enc, "existing output segment %" GST_SEGMENT_FORMAT,
|
|
segment);
|
|
if (!GST_CLOCK_TIME_IS_VALID (segment->stop)) {
|
|
int input_samples = bsize / (enc->channels * 2);
|
|
GST_DEBUG_OBJECT (enc,
|
|
"No stop time and partial frame, updating segment");
|
|
duration =
|
|
gst_util_uint64_scale (enc->encoded_samples + input_samples,
|
|
GST_SECOND, enc->rate);
|
|
segment->stop = segment->start + duration;
|
|
GST_DEBUG_OBJECT (enc, "new output segment %" GST_SEGMENT_FORMAT,
|
|
segment);
|
|
gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (enc),
|
|
gst_event_new_segment (segment));
|
|
}
|
|
|
|
size = ((bsize / bytes) + 1) * bytes;
|
|
data0 = data = g_malloc0 (size);
|
|
memcpy (data, bdata, bsize);
|
|
gst_buffer_unmap (buf, &map);
|
|
bdata = NULL;
|
|
} else {
|
|
data = bdata;
|
|
size = bsize;
|
|
}
|
|
} else {
|
|
GST_DEBUG_OBJECT (enc, "nothing to drain");
|
|
goto done;
|
|
}
|
|
|
|
samples = size / (2 * enc->channels);
|
|
speex_bits_reset (&enc->bits);
|
|
|
|
/* FIXME what about dropped samples if DTS enabled ?? */
|
|
|
|
while (size) {
|
|
GST_DEBUG_OBJECT (enc, "encoding %d samples (%d bytes)", frame_size, bytes);
|
|
|
|
if (enc->channels == 2) {
|
|
speex_encode_stereo_int ((gint16 *) data, frame_size, &enc->bits);
|
|
}
|
|
dtx_ret += speex_encode_int (enc->state, (gint16 *) data, &enc->bits);
|
|
|
|
data += bytes;
|
|
size -= bytes;
|
|
}
|
|
|
|
speex_bits_insert_terminator (&enc->bits);
|
|
outsize = speex_bits_nbytes (&enc->bits);
|
|
|
|
if (bdata)
|
|
gst_buffer_unmap (buf, &map);
|
|
|
|
#if 0
|
|
ret = gst_pad_alloc_buffer_and_set_caps (GST_AUDIO_ENCODER_SRC_PAD (enc),
|
|
GST_BUFFER_OFFSET_NONE, outsize,
|
|
GST_PAD_CAPS (GST_AUDIO_ENCODER_SRC_PAD (enc)), &outbuf);
|
|
|
|
if ((GST_FLOW_OK != ret))
|
|
goto done;
|
|
#endif
|
|
outbuf = gst_buffer_new_allocate (NULL, outsize, NULL);
|
|
gst_buffer_map (outbuf, &map, GST_MAP_WRITE);
|
|
|
|
written = speex_bits_write (&enc->bits, (gchar *) map.data, outsize);
|
|
|
|
if (G_UNLIKELY (written < outsize)) {
|
|
GST_ERROR_OBJECT (enc, "short write: %d < %d bytes", written, outsize);
|
|
} else if (G_UNLIKELY (written > outsize)) {
|
|
GST_ERROR_OBJECT (enc, "overrun: %d > %d bytes", written, outsize);
|
|
written = outsize;
|
|
}
|
|
gst_buffer_unmap (outbuf, &map);
|
|
gst_buffer_resize (outbuf, 0, written);
|
|
|
|
if (!dtx_ret)
|
|
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_GAP);
|
|
|
|
ret = gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (enc),
|
|
outbuf, samples);
|
|
enc->encoded_samples += frame_size;
|
|
|
|
done:
|
|
g_free (data0);
|
|
return ret;
|
|
}
|
|
|
|
/*
|
|
* (really really) FIXME: move into core (dixit tpm)
|
|
*/
|
|
/*
|
|
* _gst_caps_set_buffer_array:
|
|
* @caps: (transfer full): a #GstCaps
|
|
* @field: field in caps to set
|
|
* @buf: header buffers
|
|
*
|
|
* Adds given buffers to an array of buffers set as the given @field
|
|
* on the given @caps. List of buffer arguments must be NULL-terminated.
|
|
*
|
|
* Returns: (transfer full): input caps with a streamheader field added, or NULL
|
|
* if some error occurred
|
|
*/
|
|
static GstCaps *
|
|
_gst_caps_set_buffer_array (GstCaps * caps, const gchar * field,
|
|
GstBuffer * buf, ...)
|
|
{
|
|
GstStructure *structure = NULL;
|
|
va_list va;
|
|
GValue array = { 0 };
|
|
GValue value = { 0 };
|
|
|
|
g_return_val_if_fail (caps != NULL, NULL);
|
|
g_return_val_if_fail (gst_caps_is_fixed (caps), NULL);
|
|
g_return_val_if_fail (field != NULL, NULL);
|
|
|
|
caps = gst_caps_make_writable (caps);
|
|
structure = gst_caps_get_structure (caps, 0);
|
|
|
|
g_value_init (&array, GST_TYPE_ARRAY);
|
|
|
|
va_start (va, buf);
|
|
/* put buffers in a fixed list */
|
|
while (buf) {
|
|
g_assert (gst_buffer_is_writable (buf));
|
|
|
|
/* mark buffer */
|
|
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_HEADER);
|
|
|
|
g_value_init (&value, GST_TYPE_BUFFER);
|
|
buf = gst_buffer_copy (buf);
|
|
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_HEADER);
|
|
gst_value_set_buffer (&value, buf);
|
|
gst_buffer_unref (buf);
|
|
gst_value_array_append_value (&array, &value);
|
|
g_value_unset (&value);
|
|
|
|
buf = va_arg (va, GstBuffer *);
|
|
}
|
|
va_end (va);
|
|
|
|
gst_structure_set_value (structure, field, &array);
|
|
g_value_unset (&array);
|
|
|
|
return caps;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_speex_enc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf)
|
|
{
|
|
GstSpeexEnc *enc;
|
|
GstFlowReturn ret = GST_FLOW_OK;
|
|
|
|
enc = GST_SPEEX_ENC (benc);
|
|
|
|
if (!enc->header_sent) {
|
|
/* Speex streams begin with two headers; the initial header (with
|
|
most of the codec setup parameters) which is mandated by the Ogg
|
|
bitstream spec. The second header holds any comment fields.
|
|
We merely need to make the headers, then pass them to libspeex
|
|
one at a time; libspeex handles the additional Ogg bitstream
|
|
constraints */
|
|
GstBuffer *buf1, *buf2;
|
|
GstCaps *caps;
|
|
guchar *data;
|
|
gint data_len;
|
|
GList *headers;
|
|
|
|
/* create header buffer */
|
|
data = (guint8 *) speex_header_to_packet (&enc->header, &data_len);
|
|
buf1 = gst_buffer_new_wrapped_full (0,
|
|
data, data_len, 0, data_len, data, (GDestroyNotify) speex_header_free);
|
|
GST_BUFFER_OFFSET_END (buf1) = 0;
|
|
GST_BUFFER_OFFSET (buf1) = 0;
|
|
|
|
/* create comment buffer */
|
|
buf2 = gst_speex_enc_create_metadata_buffer (enc);
|
|
|
|
/* mark and put on caps */
|
|
caps = gst_caps_new_simple ("audio/x-speex", "rate", G_TYPE_INT, enc->rate,
|
|
"channels", G_TYPE_INT, enc->channels, NULL);
|
|
caps = _gst_caps_set_buffer_array (caps, "streamheader", buf1, buf2, NULL);
|
|
|
|
/* negotiate with these caps */
|
|
GST_DEBUG_OBJECT (enc, "here are the caps: %" GST_PTR_FORMAT, caps);
|
|
|
|
gst_audio_encoder_set_output_format (GST_AUDIO_ENCODER (enc), caps);
|
|
gst_caps_unref (caps);
|
|
|
|
/* push out buffers */
|
|
/* store buffers for later pre_push sending */
|
|
headers = NULL;
|
|
GST_DEBUG_OBJECT (enc, "storing header buffers");
|
|
headers = g_list_prepend (headers, buf2);
|
|
headers = g_list_prepend (headers, buf1);
|
|
gst_audio_encoder_set_headers (benc, headers);
|
|
|
|
enc->header_sent = TRUE;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (enc, "received buffer %p of %" G_GSIZE_FORMAT " bytes", buf,
|
|
buf ? gst_buffer_get_size (buf) : 0);
|
|
|
|
ret = gst_speex_enc_encode (enc, buf);
|
|
|
|
return ret;
|
|
}
|
|
|
|
static void
|
|
gst_speex_enc_get_property (GObject * object, guint prop_id, GValue * value,
|
|
GParamSpec * pspec)
|
|
{
|
|
GstSpeexEnc *enc;
|
|
|
|
enc = GST_SPEEX_ENC (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_QUALITY:
|
|
g_value_set_float (value, enc->quality);
|
|
break;
|
|
case PROP_BITRATE:
|
|
g_value_set_int (value, enc->bitrate);
|
|
break;
|
|
case PROP_MODE:
|
|
g_value_set_enum (value, enc->mode);
|
|
break;
|
|
case PROP_VBR:
|
|
g_value_set_boolean (value, enc->vbr);
|
|
break;
|
|
case PROP_ABR:
|
|
g_value_set_int (value, enc->abr);
|
|
break;
|
|
case PROP_VAD:
|
|
g_value_set_boolean (value, enc->vad);
|
|
break;
|
|
case PROP_DTX:
|
|
g_value_set_boolean (value, enc->dtx);
|
|
break;
|
|
case PROP_COMPLEXITY:
|
|
g_value_set_int (value, enc->complexity);
|
|
break;
|
|
case PROP_NFRAMES:
|
|
g_value_set_int (value, enc->nframes);
|
|
break;
|
|
case PROP_LAST_MESSAGE:
|
|
g_value_set_string (value, enc->last_message);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_speex_enc_set_property (GObject * object, guint prop_id,
|
|
const GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstSpeexEnc *enc;
|
|
|
|
enc = GST_SPEEX_ENC (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_QUALITY:
|
|
enc->quality = g_value_get_float (value);
|
|
break;
|
|
case PROP_BITRATE:
|
|
enc->bitrate = g_value_get_int (value);
|
|
break;
|
|
case PROP_MODE:
|
|
enc->mode = g_value_get_enum (value);
|
|
break;
|
|
case PROP_VBR:
|
|
enc->vbr = g_value_get_boolean (value);
|
|
break;
|
|
case PROP_ABR:
|
|
enc->abr = g_value_get_int (value);
|
|
break;
|
|
case PROP_VAD:
|
|
enc->vad = g_value_get_boolean (value);
|
|
break;
|
|
case PROP_DTX:
|
|
enc->dtx = g_value_get_boolean (value);
|
|
break;
|
|
case PROP_COMPLEXITY:
|
|
enc->complexity = g_value_get_int (value);
|
|
break;
|
|
case PROP_NFRAMES:
|
|
enc->nframes = g_value_get_int (value);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|