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542 lines
16 KiB
C
542 lines
16 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., 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-voaacenc
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* @title: voaacenc
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*
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* AAC audio encoder based on vo-aacenc library.
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*
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* [vo-aacenc library source file](http://sourceforge.net/projects/opencore-amr/files/vo-aacenc/)
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*
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* ## Example launch line
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* |[
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* gst-launch-1.0 filesrc location=abc.wav ! wavparse ! audioresample ! audioconvert ! voaacenc ! filesink location=abc.aac
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* ]|
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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 <string.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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/* voaacenc only supports 1 or 2 channels */
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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, "
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"format = (string) " GST_AUDIO_NE (S16) ", "
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"layout = (string) interleaved, "
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"rate = (int) { " SAMPLE_RATES " }, " "channels = (int) 1;"
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"audio/x-raw, "
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"format = (string) " GST_AUDIO_NE (S16) ", "
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"layout = (string) interleaved, "
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"rate = (int) { " SAMPLE_RATES " }, " "channels = (int) 2, "
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"channel-mask=(bitmask)0x3")
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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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G_DEFINE_TYPE (GstVoAacEnc, gst_voaacenc, GST_TYPE_AUDIO_ENCODER);
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GST_ELEMENT_REGISTER_DEFINE (voaacenc, "voaacenc",
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GST_RANK_SECONDARY, GST_TYPE_VOAACENC);
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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_class_init (GstVoAacEncClass * klass)
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{
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GObjectClass *object_class = G_OBJECT_CLASS (klass);
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GstElementClass *element_class = GST_ELEMENT_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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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 (bits per second)",
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0, 320000, VOAAC_ENC_DEFAULT_BITRATE,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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gst_element_class_add_static_pad_template (element_class, &sink_template);
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gst_element_class_add_static_pad_template (element_class, &src_template);
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gst_element_class_set_static_metadata (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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GST_DEBUG_CATEGORY_INIT (gst_voaacenc_debug, "voaacenc", 0, "voaac encoder");
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}
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static void
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gst_voaacenc_init (GstVoAacEnc * voaacenc)
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{
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GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_ENCODER_SINK_PAD (voaacenc));
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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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#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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aac_channel_positions[][VOAAC_ENC_MAX_CHANNELS] = {
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{ /* 1 ch: Mono */
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GST_AUDIO_CHANNEL_POSITION_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_LFE1}
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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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gint index;
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GstBuffer *codec_data;
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GstMapInfo map;
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if ((index = gst_voaacenc_get_rate_index (voaacenc->rate)) >= 0) {
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codec_data = gst_buffer_new_and_alloc (VOAAC_ENC_CODECDATA_LEN);
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gst_buffer_map (codec_data, &map, GST_MAP_WRITE);
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/* LC profile only */
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map.data[0] = ((0x02 << 3) | (index >> 1));
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map.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, NULL);
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gst_codec_utils_aac_caps_set_level_and_profile (caps, map.data,
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VOAAC_ENC_CODECDATA_LEN);
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gst_buffer_unmap (codec_data, &map);
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if (!voaacenc->output_format) {
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gst_caps_set_simple (caps,
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"stream-format", G_TYPE_STRING, "raw",
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"codec_data", GST_TYPE_BUFFER, codec_data, NULL);
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} else {
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gst_caps_set_simple (caps,
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"stream-format", G_TYPE_STRING, "adts",
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"framed", G_TYPE_BOOLEAN, TRUE, NULL);
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}
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gst_buffer_unref (codec_data);
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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_audio_encoder_set_output_format (GST_AUDIO_ENCODER (voaacenc),
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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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GstMapInfo map, omap;
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GstAudioInfo *info = gst_audio_encoder_get_audio_info (benc);
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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 (memcmp (info->position, aac_channel_positions[info->channels - 1],
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sizeof (GstAudioChannelPosition) * info->channels) != 0) {
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buf = gst_buffer_make_writable (buf);
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gst_audio_buffer_reorder_channels (buf, info->finfo->format,
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info->channels, info->position,
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aac_channel_positions[info->channels - 1]);
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}
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gst_buffer_map (buf, &map, GST_MAP_READ);
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if (G_UNLIKELY (map.size < voaacenc->inbuf_size)) {
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gst_buffer_unmap (buf, &map);
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GST_DEBUG_OBJECT (voaacenc, "discarding trailing data %d", (gint) map.size);
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ret = gst_audio_encoder_finish_frame (benc, NULL, -1);
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goto exit;
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}
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/* max size */
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out = gst_buffer_new_and_alloc (voaacenc->inbuf_size);
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gst_buffer_map (out, &omap, GST_MAP_WRITE);
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output.Buffer = omap.data;
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output.Length = voaacenc->inbuf_size;
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g_assert (map.size == voaacenc->inbuf_size);
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input.Buffer = map.data;
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input.Length = voaacenc->inbuf_size;
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voaacenc->codec_api.SetInputData (voaacenc->handle, &input);
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/* encode */
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if (voaacenc->codec_api.GetOutputData (voaacenc->handle, &output,
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&output_info) != VO_ERR_NONE) {
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gst_buffer_unmap (buf, &map);
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gst_buffer_unmap (out, &omap);
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gst_buffer_unref (out);
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goto encode_failed;
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}
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GST_LOG_OBJECT (voaacenc, "encoded to %lu bytes", output.Length);
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gst_buffer_unmap (buf, &map);
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gst_buffer_unmap (out, &omap);
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gst_buffer_resize (out, 0, output.Length);
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ret = gst_audio_encoder_finish_frame (benc, out, 1024);
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exit:
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return ret;
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/* ERRORS */
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encode_failed:
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{
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GST_ELEMENT_ERROR (voaacenc, STREAM, ENCODE, (NULL), ("encode failed"));
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ret = GST_FLOW_ERROR;
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goto exit;
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}
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}
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static VO_U32
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voaacenc_core_mem_alloc (VO_S32 uID, VO_MEM_INFO * pMemInfo)
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{
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if (!pMemInfo)
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return VO_ERR_INVALID_ARG;
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pMemInfo->VBuffer = g_malloc (pMemInfo->Size);
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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;
|
|
}
|
|
}
|