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607 lines
17 KiB
C
607 lines
17 KiB
C
/* GStreamer AAC encoder plugin
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* Copyright (C) 2011 Kan Hu <kan.hu@linaro.org>
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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., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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/**
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* SECTION:element-voaacenc
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*
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* AAC audio encoder based on vo-aacenc library
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* <ulink url="http://sourceforge.net/projects/opencore-amr/files/vo-aacenc/">vo-aacenc library source file</ulink>.
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*
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* <refsect2>
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* <title>Example launch line</title>
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* |[
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* gst-launch filesrc location=abc.wav ! wavparse ! audioresample ! audioconvert ! voaacenc ! filesink location=abc.aac
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* ]|
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* </refsect2>
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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 <string.h>
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#include <gst/audio/multichannel.h>
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#include <gst/pbutils/codec-utils.h>
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#include "gstvoaacenc.h"
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#define VOAAC_ENC_DEFAULT_BITRATE (128000)
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#define VOAAC_ENC_DEFAULT_OUTPUTFORMAT (0) /* RAW */
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#define VOAAC_ENC_MPEGVERSION (4)
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#define VOAAC_ENC_CODECDATA_LEN (2)
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#define VOAAC_ENC_BITS_PER_SAMPLE (16)
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enum
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{
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PROP_0,
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PROP_BITRATE
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};
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#define SAMPLE_RATES " 8000, " \
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"11025, " \
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"12000, " \
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"16000, " \
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"22050, " \
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"24000, " \
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"32000, " \
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"44100, " \
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"48000, " \
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"64000, " \
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"88200, " \
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"96000"
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static GstStaticPadTemplate sink_template = 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-int, "
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"width = (int) 16, "
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"depth = (int) 16, "
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"signed = (boolean) TRUE, "
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"endianness = (int) BYTE_ORDER, "
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"rate = (int) { " SAMPLE_RATES " }, " "channels = (int) [1, 2]")
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);
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static GstStaticPadTemplate src_template = 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/mpeg, "
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"mpegversion = (int) 4, "
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"rate = (int) { " SAMPLE_RATES " }, "
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"channels = (int) [1, 2], "
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"stream-format = (string) { adts, raw }, " "base-profile = (string) lc")
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);
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GST_DEBUG_CATEGORY_STATIC (gst_voaacenc_debug);
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#define GST_CAT_DEFAULT gst_voaacenc_debug
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static gboolean voaacenc_core_init (GstVoAacEnc * voaacenc);
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static gboolean voaacenc_core_set_parameter (GstVoAacEnc * voaacenc);
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static void voaacenc_core_uninit (GstVoAacEnc * voaacenc);
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static gboolean gst_voaacenc_start (GstAudioEncoder * enc);
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static gboolean gst_voaacenc_stop (GstAudioEncoder * enc);
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static gboolean gst_voaacenc_set_format (GstAudioEncoder * enc,
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GstAudioInfo * info);
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static GstFlowReturn gst_voaacenc_handle_frame (GstAudioEncoder * enc,
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GstBuffer * in_buf);
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static GstCaps *gst_voaacenc_getcaps (GstAudioEncoder * enc);
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GST_BOILERPLATE (GstVoAacEnc, gst_voaacenc, GstAudioEncoder,
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GST_TYPE_AUDIO_ENCODER);
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static void
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gst_voaacenc_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstVoAacEnc *self = GST_VOAACENC (object);
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switch (prop_id) {
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case PROP_BITRATE:
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self->bitrate = g_value_get_int (value);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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return;
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}
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static void
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gst_voaacenc_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstVoAacEnc *self = GST_VOAACENC (object);
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switch (prop_id) {
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case PROP_BITRATE:
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g_value_set_int (value, self->bitrate);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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return;
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}
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static void
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gst_voaacenc_base_init (gpointer klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&sink_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&src_template));
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gst_element_class_set_details_simple (element_class, "AAC audio encoder",
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"Codec/Encoder/Audio", "AAC audio encoder", "Kan Hu <kan.hu@linaro.org>");
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}
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static void
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gst_voaacenc_class_init (GstVoAacEncClass * klass)
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{
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GObjectClass *object_class = G_OBJECT_CLASS (klass);
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GstAudioEncoderClass *base_class = GST_AUDIO_ENCODER_CLASS (klass);
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object_class->set_property = GST_DEBUG_FUNCPTR (gst_voaacenc_set_property);
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object_class->get_property = GST_DEBUG_FUNCPTR (gst_voaacenc_get_property);
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base_class->start = GST_DEBUG_FUNCPTR (gst_voaacenc_start);
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base_class->stop = GST_DEBUG_FUNCPTR (gst_voaacenc_stop);
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base_class->set_format = GST_DEBUG_FUNCPTR (gst_voaacenc_set_format);
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base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_voaacenc_handle_frame);
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base_class->getcaps = GST_DEBUG_FUNCPTR (gst_voaacenc_getcaps);
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g_object_class_install_property (object_class, PROP_BITRATE,
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g_param_spec_int ("bitrate",
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"Bitrate",
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"Target Audio Bitrate",
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0, G_MAXINT, VOAAC_ENC_DEFAULT_BITRATE,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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}
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static void
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gst_voaacenc_init (GstVoAacEnc * voaacenc, GstVoAacEncClass * klass)
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{
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voaacenc->bitrate = VOAAC_ENC_DEFAULT_BITRATE;
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voaacenc->output_format = VOAAC_ENC_DEFAULT_OUTPUTFORMAT;
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/* init rest */
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voaacenc->handle = NULL;
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}
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static gboolean
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gst_voaacenc_start (GstAudioEncoder * enc)
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{
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GstVoAacEnc *voaacenc = GST_VOAACENC (enc);
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GST_DEBUG_OBJECT (enc, "start");
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if (voaacenc_core_init (voaacenc) == FALSE)
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return FALSE;
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voaacenc->rate = 0;
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voaacenc->channels = 0;
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return TRUE;
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}
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static gboolean
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gst_voaacenc_stop (GstAudioEncoder * enc)
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{
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GstVoAacEnc *voaacenc = GST_VOAACENC (enc);
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GST_DEBUG_OBJECT (enc, "stop");
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voaacenc_core_uninit (voaacenc);
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return TRUE;
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}
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static gpointer
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gst_voaacenc_generate_sink_caps (gpointer data)
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{
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#define VOAAC_ENC_MAX_CHANNELS 6
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/* describe the channels position */
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static const GstAudioChannelPosition
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gst_voaacenc_channel_position[][VOAAC_ENC_MAX_CHANNELS] = {
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{ /* 1 ch: Mono */
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GST_AUDIO_CHANNEL_POSITION_FRONT_MONO},
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{ /* 2 ch: front left + front right (front stereo) */
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT},
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{ /* 3 ch: front center + front stereo */
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GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT},
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{ /* 4 ch: front center + front stereo + back center */
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GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_REAR_CENTER},
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{ /* 5 ch: front center + front stereo + back stereo */
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GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
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GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT},
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{ /* 6ch: front center + front stereo + back stereo + LFE */
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GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
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GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_LFE}
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};
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GstCaps *caps = gst_caps_new_empty ();
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gint i, c;
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static const int rates[] = {
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8000, 11025, 12000, 16000, 22050, 24000,
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32000, 44100, 48000, 64000, 88200, 96000
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};
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GValue rates_arr = { 0, };
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GValue tmp = { 0, };
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g_value_init (&rates_arr, GST_TYPE_LIST);
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g_value_init (&tmp, G_TYPE_INT);
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for (i = 0; i < G_N_ELEMENTS (rates); i++) {
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g_value_set_int (&tmp, rates[i]);
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gst_value_list_append_value (&rates_arr, &tmp);
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}
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g_value_unset (&tmp);
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for (i = 0; i < 2 /*VOAAC_ENC_MAX_CHANNELS */ ; i++) {
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GValue chanpos = { 0 };
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GValue pos = { 0 };
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GstStructure *structure;
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g_value_init (&chanpos, GST_TYPE_ARRAY);
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g_value_init (&pos, GST_TYPE_AUDIO_CHANNEL_POSITION);
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for (c = 0; c <= i; c++) {
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g_value_set_enum (&pos, gst_voaacenc_channel_position[i][c]);
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gst_value_array_append_value (&chanpos, &pos);
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}
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g_value_unset (&pos);
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structure = gst_structure_new ("audio/x-raw-int",
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"width", G_TYPE_INT, 16,
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"depth", G_TYPE_INT, 16,
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"signed", G_TYPE_BOOLEAN, TRUE,
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"endianness", G_TYPE_INT, G_BYTE_ORDER,
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"channels", G_TYPE_INT, i + 1, NULL);
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gst_structure_set_value (structure, "rate", &rates_arr);
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gst_structure_set_value (structure, "channel-positions", &chanpos);
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g_value_unset (&chanpos);
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gst_caps_append_structure (caps, structure);
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}
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g_value_unset (&rates_arr);
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GST_DEBUG ("generated sink caps: %" GST_PTR_FORMAT, caps);
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return caps;
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}
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static GstCaps *
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gst_voaacenc_get_sink_caps (void)
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{
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static GOnce g_once = G_ONCE_INIT;
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GstCaps *caps;
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g_once (&g_once, gst_voaacenc_generate_sink_caps, NULL);
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caps = g_once.retval;
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return caps;
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}
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static GstCaps *
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gst_voaacenc_getcaps (GstAudioEncoder * benc)
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{
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return gst_audio_encoder_proxy_getcaps (benc, gst_voaacenc_get_sink_caps ());
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}
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/* check downstream caps to configure format */
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static void
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gst_voaacenc_negotiate (GstVoAacEnc * voaacenc)
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{
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GstCaps *caps;
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caps = gst_pad_get_allowed_caps (GST_AUDIO_ENCODER_SRC_PAD (voaacenc));
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GST_DEBUG_OBJECT (voaacenc, "allowed caps: %" GST_PTR_FORMAT, caps);
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if (caps && gst_caps_get_size (caps) > 0) {
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GstStructure *s = gst_caps_get_structure (caps, 0);
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const gchar *str = NULL;
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if ((str = gst_structure_get_string (s, "stream-format"))) {
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if (strcmp (str, "adts") == 0) {
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GST_DEBUG_OBJECT (voaacenc, "use ADTS format for output");
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voaacenc->output_format = 1;
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} else if (strcmp (str, "raw") == 0) {
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GST_DEBUG_OBJECT (voaacenc, "use RAW format for output");
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voaacenc->output_format = 0;
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} else {
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GST_DEBUG_OBJECT (voaacenc, "unknown stream-format: %s", str);
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voaacenc->output_format = VOAAC_ENC_DEFAULT_OUTPUTFORMAT;
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}
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}
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}
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if (caps)
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gst_caps_unref (caps);
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}
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static gint
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gst_voaacenc_get_rate_index (gint rate)
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{
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static const gint rate_table[] = {
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96000, 88200, 64000, 48000, 44100, 32000,
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24000, 22050, 16000, 12000, 11025, 8000
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};
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gint i;
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for (i = 0; i < G_N_ELEMENTS (rate_table); ++i) {
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if (rate == rate_table[i]) {
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return i;
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}
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}
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return -1;
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}
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static GstCaps *
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gst_voaacenc_create_source_pad_caps (GstVoAacEnc * voaacenc)
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{
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GstCaps *caps = NULL;
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GstBuffer *codec_data;
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gint index;
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guint8 data[VOAAC_ENC_CODECDATA_LEN];
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if ((index = gst_voaacenc_get_rate_index (voaacenc->rate)) >= 0) {
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/* LC profile only */
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data[0] = ((0x02 << 3) | (index >> 1));
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data[1] = ((index & 0x01) << 7) | (voaacenc->channels << 3);
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caps = gst_caps_new_simple ("audio/mpeg",
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"mpegversion", G_TYPE_INT, VOAAC_ENC_MPEGVERSION,
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"channels", G_TYPE_INT, voaacenc->channels,
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"rate", G_TYPE_INT, voaacenc->rate,
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"stream-format", G_TYPE_STRING,
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(voaacenc->output_format ? "adts" : "raw")
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, NULL);
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gst_codec_utils_aac_caps_set_level_and_profile (caps, data, sizeof (data));
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if (!voaacenc->output_format) {
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codec_data = gst_buffer_new_and_alloc (VOAAC_ENC_CODECDATA_LEN);
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memcpy (GST_BUFFER_DATA (codec_data), data, sizeof (data));
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gst_caps_set_simple (caps, "codec_data", GST_TYPE_BUFFER, codec_data,
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NULL);
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gst_buffer_unref (codec_data);
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}
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}
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return caps;
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}
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static gboolean
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gst_voaacenc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
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{
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gboolean ret = FALSE;
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GstVoAacEnc *voaacenc;
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GstCaps *src_caps;
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voaacenc = GST_VOAACENC (benc);
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/* get channel count */
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voaacenc->channels = GST_AUDIO_INFO_CHANNELS (info);
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voaacenc->rate = GST_AUDIO_INFO_RATE (info);
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/* precalc buffer size as it's constant now */
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voaacenc->inbuf_size = voaacenc->channels * 2 * 1024;
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gst_voaacenc_negotiate (voaacenc);
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/* create reverse caps */
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src_caps = gst_voaacenc_create_source_pad_caps (voaacenc);
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if (src_caps) {
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gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (voaacenc), src_caps);
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gst_caps_unref (src_caps);
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ret = voaacenc_core_set_parameter (voaacenc);
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}
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/* report needs to base class */
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gst_audio_encoder_set_frame_samples_min (benc, 1024);
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gst_audio_encoder_set_frame_samples_max (benc, 1024);
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gst_audio_encoder_set_frame_max (benc, 1);
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return ret;
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}
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static GstFlowReturn
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gst_voaacenc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf)
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{
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GstVoAacEnc *voaacenc;
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GstFlowReturn ret = GST_FLOW_OK;
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GstBuffer *out;
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VO_AUDIO_OUTPUTINFO output_info = { {0} };
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VO_CODECBUFFER input = { 0 };
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VO_CODECBUFFER output = { 0 };
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voaacenc = GST_VOAACENC (benc);
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g_return_val_if_fail (voaacenc->handle, GST_FLOW_NOT_NEGOTIATED);
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/* we don't deal with squeezing remnants, so simply discard those */
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if (G_UNLIKELY (buf == NULL)) {
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GST_DEBUG_OBJECT (benc, "no data");
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goto exit;
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}
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if (G_UNLIKELY (GST_BUFFER_SIZE (buf) < voaacenc->inbuf_size)) {
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GST_DEBUG_OBJECT (voaacenc, "discarding trailing data %d",
|
|
buf ? GST_BUFFER_SIZE (buf) : 0);
|
|
ret = gst_audio_encoder_finish_frame (benc, NULL, -1);
|
|
goto exit;
|
|
}
|
|
|
|
/* max size */
|
|
if ((ret =
|
|
gst_pad_alloc_buffer_and_set_caps (GST_AUDIO_ENCODER_SRC_PAD
|
|
(voaacenc), 0, voaacenc->inbuf_size,
|
|
GST_PAD_CAPS (GST_AUDIO_ENCODER_SRC_PAD (voaacenc)),
|
|
&out)) != GST_FLOW_OK) {
|
|
goto exit;
|
|
}
|
|
|
|
output.Buffer = GST_BUFFER_DATA (out);
|
|
output.Length = voaacenc->inbuf_size;
|
|
|
|
g_assert (GST_BUFFER_SIZE (buf) == voaacenc->inbuf_size);
|
|
input.Buffer = GST_BUFFER_DATA (buf);
|
|
input.Length = voaacenc->inbuf_size;
|
|
voaacenc->codec_api.SetInputData (voaacenc->handle, &input);
|
|
|
|
/* encode */
|
|
if (voaacenc->codec_api.GetOutputData (voaacenc->handle, &output,
|
|
&output_info) != VO_ERR_NONE) {
|
|
gst_buffer_unref (out);
|
|
goto encode_failed;
|
|
}
|
|
|
|
GST_LOG_OBJECT (voaacenc, "encoded to %d bytes", output.Length);
|
|
GST_BUFFER_SIZE (out) = output.Length;
|
|
|
|
GST_LOG_OBJECT (voaacenc, "Pushing out buffer time: %" GST_TIME_FORMAT
|
|
" duration: %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (out)),
|
|
GST_TIME_ARGS (GST_BUFFER_DURATION (out)));
|
|
|
|
ret = gst_audio_encoder_finish_frame (benc, out, 1024);
|
|
|
|
exit:
|
|
return ret;
|
|
|
|
/* ERRORS */
|
|
encode_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (voaacenc, STREAM, ENCODE, (NULL), ("encode failed"));
|
|
ret = GST_FLOW_ERROR;
|
|
goto exit;
|
|
}
|
|
}
|
|
|
|
static VO_U32
|
|
voaacenc_core_mem_alloc (VO_S32 uID, VO_MEM_INFO * pMemInfo)
|
|
{
|
|
if (!pMemInfo)
|
|
return VO_ERR_INVALID_ARG;
|
|
|
|
pMemInfo->VBuffer = g_malloc (pMemInfo->Size);
|
|
return 0;
|
|
}
|
|
|
|
static VO_U32
|
|
voaacenc_core_mem_free (VO_S32 uID, VO_PTR pMem)
|
|
{
|
|
g_free (pMem);
|
|
return 0;
|
|
}
|
|
|
|
static VO_U32
|
|
voaacenc_core_mem_set (VO_S32 uID, VO_PTR pBuff, VO_U8 uValue, VO_U32 uSize)
|
|
{
|
|
memset (pBuff, uValue, uSize);
|
|
return 0;
|
|
}
|
|
|
|
static VO_U32
|
|
voaacenc_core_mem_copy (VO_S32 uID, VO_PTR pDest, VO_PTR pSource, VO_U32 uSize)
|
|
{
|
|
memcpy (pDest, pSource, uSize);
|
|
return 0;
|
|
}
|
|
|
|
static VO_U32
|
|
voaacenc_core_mem_check (VO_S32 uID, VO_PTR pBuffer, VO_U32 uSize)
|
|
{
|
|
return 0;
|
|
}
|
|
|
|
static gboolean
|
|
voaacenc_core_init (GstVoAacEnc * voaacenc)
|
|
{
|
|
VO_CODEC_INIT_USERDATA user_data = { 0 };
|
|
voGetAACEncAPI (&voaacenc->codec_api);
|
|
|
|
voaacenc->mem_operator.Alloc = voaacenc_core_mem_alloc;
|
|
voaacenc->mem_operator.Copy = voaacenc_core_mem_copy;
|
|
voaacenc->mem_operator.Free = voaacenc_core_mem_free;
|
|
voaacenc->mem_operator.Set = voaacenc_core_mem_set;
|
|
voaacenc->mem_operator.Check = voaacenc_core_mem_check;
|
|
user_data.memflag = VO_IMF_USERMEMOPERATOR;
|
|
user_data.memData = &voaacenc->mem_operator;
|
|
voaacenc->codec_api.Init (&voaacenc->handle, VO_AUDIO_CodingAAC, &user_data);
|
|
|
|
if (voaacenc->handle == NULL) {
|
|
return FALSE;
|
|
}
|
|
return TRUE;
|
|
|
|
}
|
|
|
|
static gboolean
|
|
voaacenc_core_set_parameter (GstVoAacEnc * voaacenc)
|
|
{
|
|
AACENC_PARAM params = { 0 };
|
|
guint32 ret;
|
|
|
|
params.sampleRate = voaacenc->rate;
|
|
params.bitRate = voaacenc->bitrate;
|
|
params.nChannels = voaacenc->channels;
|
|
if (voaacenc->output_format) {
|
|
params.adtsUsed = 1;
|
|
} else {
|
|
params.adtsUsed = 0;
|
|
}
|
|
|
|
ret =
|
|
voaacenc->codec_api.SetParam (voaacenc->handle, VO_PID_AAC_ENCPARAM,
|
|
¶ms);
|
|
if (ret != VO_ERR_NONE) {
|
|
GST_ERROR_OBJECT (voaacenc, "Failed to set encoder parameters");
|
|
return FALSE;
|
|
}
|
|
return TRUE;
|
|
}
|
|
|
|
static void
|
|
voaacenc_core_uninit (GstVoAacEnc * voaacenc)
|
|
{
|
|
if (voaacenc->handle) {
|
|
voaacenc->codec_api.Uninit (voaacenc->handle);
|
|
voaacenc->handle = NULL;
|
|
}
|
|
}
|