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7c9a315578
We set the eqmid in caps to be usable downstream by rtpldacpay for knowing the frame count. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1797>
625 lines
18 KiB
C
625 lines
18 KiB
C
/* GStreamer LDAC audio encoder
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* Copyright (C) 2020 Asymptotic <sanchayan@asymptotic.io>
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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 Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 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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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with this library; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
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*
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*/
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/**
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* SECTION:element-ldacenc
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* @title: ldacenc
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*
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* This element encodes raw integer PCM audio into a Bluetooth LDAC audio.
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*
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* ## Example pipeline
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* |[
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* gst-launch-1.0 -v audiotestsrc ! ldacenc ! rtpldacpay mtu=679 ! avdtpsink
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* ]| Encode a sine wave into LDAC, RTP payload it and send over bluetooth
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*
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* Since: 1.20
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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 "gstldacenc.h"
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/*
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* MTU size required for LDAC A2DP streaming. Required for initializing the
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* encoder.
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*/
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#define GST_LDAC_MTU_REQUIRED 679
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GST_DEBUG_CATEGORY_STATIC (ldac_enc_debug);
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#define GST_CAT_DEFAULT ldac_enc_debug
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#define parent_class gst_ldac_enc_parent_class
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G_DEFINE_TYPE (GstLdacEnc, gst_ldac_enc, GST_TYPE_AUDIO_ENCODER);
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GST_ELEMENT_REGISTER_DEFINE (ldacenc, "ldacenc", GST_RANK_NONE,
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GST_TYPE_LDAC_ENC);
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#define SAMPLE_RATES "44100, 48000, 88200, 96000"
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static GstStaticPadTemplate ldac_enc_sink_factory =
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GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS,
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GST_STATIC_CAPS
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("audio/x-raw, format=(string) { S16LE, S24LE, S32LE, F32LE }, "
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"rate = (int) { " SAMPLE_RATES " }, channels = (int) [ 1, 2 ] "));
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static GstStaticPadTemplate ldac_enc_src_factory =
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GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-ldac, "
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"rate = (int) { " SAMPLE_RATES " }, "
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"channels = (int) 1, channel-mode = (string)mono; "
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"audio/x-ldac, "
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"rate = (int) { " SAMPLE_RATES " }, "
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"channels = (int) 2, channel-mode = (string) { dual, stereo }"));
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enum
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{
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PROP_0,
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PROP_EQMID
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};
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static void gst_ldac_enc_get_property (GObject * object,
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guint property_id, GValue * value, GParamSpec * pspec);
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static void gst_ldac_enc_set_property (GObject * object,
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guint property_id, const GValue * value, GParamSpec * pspec);
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static gboolean gst_ldac_enc_start (GstAudioEncoder * enc);
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static gboolean gst_ldac_enc_stop (GstAudioEncoder * enc);
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static gboolean gst_ldac_enc_set_format (GstAudioEncoder * enc,
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GstAudioInfo * info);
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static gboolean gst_ldac_enc_negotiate (GstAudioEncoder * enc);
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static GstFlowReturn gst_ldac_enc_handle_frame (GstAudioEncoder * enc,
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GstBuffer * buffer);
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static guint gst_ldac_enc_get_num_frames (guint eqmid, guint channels);
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static guint gst_ldac_enc_get_frame_length (guint eqmid, guint channels);
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static guint gst_ldac_enc_get_num_samples (guint rate);
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#define GST_LDAC_EQMID (gst_ldac_eqmid_get_type ())
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static GType
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gst_ldac_eqmid_get_type (void)
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{
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static GType ldac_eqmid_type = 0;
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static const GEnumValue eqmid_types[] = {
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{GST_LDAC_EQMID_HQ, "HQ", "hq"},
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{GST_LDAC_EQMID_SQ, "SQ", "sq"},
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{GST_LDAC_EQMID_MQ, "MQ", "mq"},
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{0, NULL, NULL}
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};
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if (!ldac_eqmid_type)
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ldac_eqmid_type = g_enum_register_static ("GstLdacEqmid", eqmid_types);
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return ldac_eqmid_type;
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}
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static void
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gst_ldac_enc_class_init (GstLdacEncClass * klass)
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{
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GstAudioEncoderClass *encoder_class = GST_AUDIO_ENCODER_CLASS (klass);
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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GObjectClass *gobject_class = (GObjectClass *) klass;
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gobject_class->set_property = gst_ldac_enc_set_property;
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gobject_class->get_property = gst_ldac_enc_get_property;
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encoder_class->start = GST_DEBUG_FUNCPTR (gst_ldac_enc_start);
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encoder_class->stop = GST_DEBUG_FUNCPTR (gst_ldac_enc_stop);
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encoder_class->set_format = GST_DEBUG_FUNCPTR (gst_ldac_enc_set_format);
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encoder_class->handle_frame = GST_DEBUG_FUNCPTR (gst_ldac_enc_handle_frame);
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encoder_class->negotiate = GST_DEBUG_FUNCPTR (gst_ldac_enc_negotiate);
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g_object_class_install_property (gobject_class, PROP_EQMID,
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g_param_spec_enum ("eqmid", "Encode Quality Mode Index",
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"Encode Quality Mode Index. 0: High Quality 1: Standard Quality "
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"2: Mobile Use Quality", GST_LDAC_EQMID,
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GST_LDAC_EQMID_SQ, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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gst_element_class_add_static_pad_template (element_class,
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&ldac_enc_sink_factory);
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gst_element_class_add_static_pad_template (element_class,
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&ldac_enc_src_factory);
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gst_element_class_set_static_metadata (element_class,
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"Bluetooth LDAC audio encoder", "Codec/Encoder/Audio",
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"Encode an LDAC audio stream",
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"Sanchayan Maity <sanchayan@asymptotic.io>");
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GST_DEBUG_CATEGORY_INIT (ldac_enc_debug, "ldacenc", 0,
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"LDAC encoding element");
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}
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static void
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gst_ldac_enc_init (GstLdacEnc * self)
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{
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GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_ENCODER_SINK_PAD (self));
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self->eqmid = GST_LDAC_EQMID_SQ;
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self->channel_mode = 0;
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self->init_done = FALSE;
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}
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static void
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gst_ldac_enc_set_property (GObject * object, guint property_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstLdacEnc *self = GST_LDAC_ENC (object);
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switch (property_id) {
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case PROP_EQMID:
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self->eqmid = g_value_get_enum (value);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
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break;
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}
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}
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static void
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gst_ldac_enc_get_property (GObject * object, guint property_id,
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GValue * value, GParamSpec * pspec)
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{
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GstLdacEnc *self = GST_LDAC_ENC (object);
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switch (property_id) {
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case PROP_EQMID:
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g_value_set_enum (value, self->eqmid);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
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break;
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}
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}
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static GstCaps *
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gst_ldac_enc_do_negotiate (GstAudioEncoder * audio_enc)
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{
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GstLdacEnc *enc = GST_LDAC_ENC (audio_enc);
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GstCaps *caps, *filter_caps;
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GstCaps *output_caps = NULL;
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GstStructure *s;
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/* Negotiate output format based on downstream caps restrictions */
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caps = gst_pad_get_allowed_caps (GST_AUDIO_ENCODER_SRC_PAD (enc));
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if (caps == NULL)
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caps = gst_static_pad_template_get_caps (&ldac_enc_src_factory);
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else if (gst_caps_is_empty (caps))
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goto failure;
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/* Fixate output caps */
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filter_caps = gst_caps_new_simple ("audio/x-ldac", "rate", G_TYPE_INT,
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enc->rate, "channels", G_TYPE_INT, enc->channels, NULL);
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output_caps = gst_caps_intersect (caps, filter_caps);
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gst_caps_unref (filter_caps);
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if (output_caps == NULL || gst_caps_is_empty (output_caps)) {
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GST_WARNING_OBJECT (enc, "Couldn't negotiate output caps with input rate "
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"%d and input channels %d and allowed output caps %" GST_PTR_FORMAT,
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enc->rate, enc->channels, caps);
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goto failure;
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}
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gst_clear_caps (&caps);
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GST_DEBUG_OBJECT (enc, "fixating caps %" GST_PTR_FORMAT, output_caps);
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output_caps = gst_caps_truncate (output_caps);
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s = gst_caps_get_structure (output_caps, 0);
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if (enc->channels == 1)
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gst_structure_fixate_field_string (s, "channel-mode", "mono");
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else
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gst_structure_fixate_field_string (s, "channel-mode", "stereo");
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s = NULL;
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/* In case there's anything else left to fixate */
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output_caps = gst_caps_fixate (output_caps);
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gst_caps_set_simple (output_caps, "framed", G_TYPE_BOOLEAN, TRUE, NULL);
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/* Set EQMID in caps to be used downstream by rtpldacpay */
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gst_caps_set_simple (output_caps, "eqmid", G_TYPE_INT, enc->eqmid, NULL);
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GST_INFO_OBJECT (enc, "output caps %" GST_PTR_FORMAT, output_caps);
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if (enc->channels == 1)
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enc->channel_mode = LDACBT_CHANNEL_MODE_MONO;
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else
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enc->channel_mode = LDACBT_CHANNEL_MODE_STEREO;
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return output_caps;
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failure:
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if (output_caps)
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gst_caps_unref (output_caps);
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if (caps)
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gst_caps_unref (caps);
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return NULL;
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}
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static gboolean
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gst_ldac_enc_negotiate (GstAudioEncoder * audio_enc)
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{
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GstLdacEnc *enc = GST_LDAC_ENC (audio_enc);
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GstCaps *output_caps = NULL;
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output_caps = gst_ldac_enc_do_negotiate (audio_enc);
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if (output_caps == NULL) {
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GST_ERROR_OBJECT (enc, "failed to negotiate");
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return FALSE;
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}
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if (!gst_audio_encoder_set_output_format (audio_enc, output_caps)) {
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GST_ERROR_OBJECT (enc, "failed to configure output caps on src pad");
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gst_caps_unref (output_caps);
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return FALSE;
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}
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gst_caps_unref (output_caps);
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return GST_AUDIO_ENCODER_CLASS (parent_class)->negotiate (audio_enc);
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}
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static gboolean
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gst_ldac_enc_set_format (GstAudioEncoder * audio_enc, GstAudioInfo * info)
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{
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GstLdacEnc *enc = GST_LDAC_ENC (audio_enc);
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GstCaps *output_caps = NULL;
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guint num_ldac_frames, num_samples;
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gint ret = 0;
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enc->rate = GST_AUDIO_INFO_RATE (info);
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enc->channels = GST_AUDIO_INFO_CHANNELS (info);
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switch (GST_AUDIO_INFO_FORMAT (info)) {
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case GST_AUDIO_FORMAT_S16:
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enc->ldac_fmt = LDACBT_SMPL_FMT_S16;
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break;
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case GST_AUDIO_FORMAT_S24:
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enc->ldac_fmt = LDACBT_SMPL_FMT_S24;
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break;
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case GST_AUDIO_FORMAT_S32:
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enc->ldac_fmt = LDACBT_SMPL_FMT_S32;
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break;
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case GST_AUDIO_FORMAT_F32:
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enc->ldac_fmt = LDACBT_SMPL_FMT_F32;
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break;
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default:
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GST_ERROR_OBJECT (enc, "Invalid audio format");
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return FALSE;
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}
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output_caps = gst_ldac_enc_do_negotiate (audio_enc);
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if (output_caps == NULL) {
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GST_ERROR_OBJECT (enc, "failed to negotiate");
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return FALSE;
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}
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if (!gst_audio_encoder_set_output_format (audio_enc, output_caps)) {
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GST_ERROR_OBJECT (enc, "failed to configure output caps on src pad");
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gst_caps_unref (output_caps);
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return FALSE;
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}
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gst_caps_unref (output_caps);
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num_samples = gst_ldac_enc_get_num_samples (enc->rate);
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num_ldac_frames = gst_ldac_enc_get_num_frames (enc->eqmid, enc->channels);
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gst_audio_encoder_set_frame_samples_min (audio_enc,
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num_samples * num_ldac_frames);
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/*
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* If initialisation was already done means caps have changed, close the
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* handle. Closed handle can be initialised and used again.
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*/
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if (enc->init_done) {
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ldacBT_close_handle (enc->ldac);
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enc->init_done = FALSE;
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}
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/*
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* libldac exposes a bluetooth centric API and emits multiple LDAC frames
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* depending on the MTU. The MTU is required for LDAC A2DP streaming, is
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* inclusive of the RTP header and is required by the encoder. The internal
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* encoder API is not exposed in the public interface.
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*/
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ret =
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ldacBT_init_handle_encode (enc->ldac, GST_LDAC_MTU_REQUIRED, enc->eqmid,
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enc->channel_mode, enc->ldac_fmt, enc->rate);
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if (ret != 0) {
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GST_ERROR_OBJECT (enc, "Failed to initialize LDAC handle, ret: %d", ret);
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return FALSE;
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}
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enc->init_done = TRUE;
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return TRUE;
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}
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static GstFlowReturn
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gst_ldac_enc_handle_frame (GstAudioEncoder * audio_enc, GstBuffer * buffer)
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{
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GstLdacEnc *enc = GST_LDAC_ENC (audio_enc);
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GstMapInfo in_map, out_map;
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GstAudioInfo *info;
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GstBuffer *outbuf;
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const guint8 *in_data;
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guint8 *out_data;
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gint encoded, to_encode = 0;
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gint samples_consumed = 0;
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guint frames, frame_len;
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guint ldac_enc_read = 0;
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guint frame_count = 0;
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if (buffer == NULL)
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return GST_FLOW_OK;
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if (!gst_buffer_map (buffer, &in_map, GST_MAP_READ)) {
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GST_ELEMENT_ERROR (audio_enc, STREAM, FAILED, (NULL),
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("Failed to map data from input buffer"));
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return GST_FLOW_ERROR;
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}
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info = gst_audio_encoder_get_audio_info (audio_enc);
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ldac_enc_read = LDACBT_ENC_LSU * info->bpf;
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/*
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* We may produce extra frames at the end of encoding process (See below).
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* Consider some additional frames while allocating output buffer if this
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* happens.
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*/
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frames = (in_map.size / ldac_enc_read) + 4;
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frame_len = gst_ldac_enc_get_frame_length (enc->eqmid, info->channels);
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outbuf = gst_audio_encoder_allocate_output_buffer (audio_enc,
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frames * frame_len);
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if (outbuf == NULL)
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goto no_buffer;
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gst_buffer_map (outbuf, &out_map, GST_MAP_WRITE);
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in_data = in_map.data;
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out_data = out_map.data;
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to_encode = in_map.size;
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/*
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* ldacBT_encode does not generate an output frame each time it is called.
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* For each invocation, it consumes number of sample * bpf bytes of data.
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* Depending on the eqmid setting and channels, it will emit multiple frames
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* only after the required number of frames are packed for payloading. Below
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* for loop exists primarily to handle this.
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*/
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for (;;) {
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guint8 pcm[LDACBT_MAX_LSU * 4 /* bytes/sample */ * 2 /* ch */ ] = { 0 };
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gint ldac_frame_num, written;
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guint8 *inp_data = NULL;
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gboolean done = FALSE;
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gint ret;
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/*
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* Even with minimum frame samples specified in set_format with EOS,
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* we may get a buffer which is not a multiple of LDACBT_ENC_LSU. LDAC
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* encoder always reads a multiple of this and to handle this scenario
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* we use local PCM array and in the last iteration when buffer bytes
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* < LDACBT_ENC_LSU * bpf, we copy only to_encode bytes to prevent
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* walking off the end of input buffer and the rest of the bytes in
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* PCM buffer would be zero, so should be safe from encoding point of
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* view.
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*/
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if (to_encode < 0) {
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/*
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* We got < LDACBT_ENC_LSU * bpf for last iteration. Force the encoder
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* to encode the remaining bytes in buffer by passing NULL to the input
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* PCM buffer argument.
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*/
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inp_data = NULL;
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done = TRUE;
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} else if (to_encode >= ldac_enc_read) {
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memcpy (pcm, in_data, ldac_enc_read);
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inp_data = &pcm[0];
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} else if (to_encode > 0 && to_encode < ldac_enc_read) {
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memcpy (pcm, in_data, to_encode);
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inp_data = &pcm[0];
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}
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/*
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* Note that while we do not explicitly pass length of data to library
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* anywhere, based on the initialization considering eqmid and rate, the
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* library will consume a fix number of samples per call. This combined
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* with the previous step ensures that the library does not read outside
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* of in_data and out_data.
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*/
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ret = ldacBT_encode (enc->ldac, (void *) inp_data, &encoded,
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(guint8 *) out_data, &written, &ldac_frame_num);
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if (ret < 0) {
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GST_ELEMENT_ERROR (enc, STREAM, ENCODE, (NULL),
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("encoding error, ret = %d written = %d", ret, ldac_frame_num));
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goto encoding_error;
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} else {
|
|
to_encode -= encoded;
|
|
in_data = in_data + encoded;
|
|
out_data = out_data + written;
|
|
frame_count += ldac_frame_num;
|
|
|
|
GST_LOG_OBJECT (enc,
|
|
"To Encode: %d, Encoded: %d, Written: %d, LDAC Frames: %d", to_encode,
|
|
encoded, written, ldac_frame_num);
|
|
|
|
if (done || (to_encode == 0 && encoded == ldac_enc_read))
|
|
break;
|
|
}
|
|
}
|
|
|
|
gst_buffer_unmap (outbuf, &out_map);
|
|
|
|
if (frame_count > 0) {
|
|
samples_consumed = in_map.size / info->bpf;
|
|
gst_buffer_set_size (outbuf, frame_count * frame_len);
|
|
} else {
|
|
samples_consumed = 0;
|
|
gst_buffer_replace (&outbuf, NULL);
|
|
}
|
|
|
|
gst_buffer_unmap (buffer, &in_map);
|
|
|
|
return gst_audio_encoder_finish_frame (audio_enc, outbuf, samples_consumed);
|
|
|
|
no_buffer:
|
|
{
|
|
gst_buffer_unmap (buffer, &in_map);
|
|
|
|
GST_ELEMENT_ERROR (enc, STREAM, FAILED, (NULL),
|
|
("could not allocate output buffer"));
|
|
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
encoding_error:
|
|
{
|
|
gst_buffer_unmap (buffer, &in_map);
|
|
|
|
ldacBT_free_handle (enc->ldac);
|
|
|
|
enc->ldac = NULL;
|
|
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_ldac_enc_start (GstAudioEncoder * audio_enc)
|
|
{
|
|
GstLdacEnc *enc = GST_LDAC_ENC (audio_enc);
|
|
|
|
GST_INFO_OBJECT (enc, "Setup LDAC codec");
|
|
/* Note that this only allocates the LDAC handle */
|
|
enc->ldac = ldacBT_get_handle ();
|
|
if (enc->ldac == NULL) {
|
|
GST_ERROR_OBJECT (enc, "Failed to get LDAC handle");
|
|
return FALSE;
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_ldac_enc_stop (GstAudioEncoder * audio_enc)
|
|
{
|
|
GstLdacEnc *enc = GST_LDAC_ENC (audio_enc);
|
|
|
|
GST_INFO_OBJECT (enc, "Finish LDAC codec");
|
|
|
|
if (enc->ldac) {
|
|
ldacBT_free_handle (enc->ldac);
|
|
enc->ldac = NULL;
|
|
}
|
|
|
|
enc->eqmid = GST_LDAC_EQMID_SQ;
|
|
enc->channel_mode = 0;
|
|
enc->init_done = FALSE;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/**
|
|
* gst_ldac_enc_get_frame_length
|
|
* @eqmid: Encode Quality Mode Index
|
|
* @channels: Number of channels
|
|
*
|
|
* Returns: Frame length.
|
|
*/
|
|
static guint
|
|
gst_ldac_enc_get_frame_length (guint eqmid, guint channels)
|
|
{
|
|
g_assert (channels == 1 || channels == 2);
|
|
|
|
switch (eqmid) {
|
|
/* Encode setting for High Quality */
|
|
case GST_LDAC_EQMID_HQ:
|
|
return 165 * channels;
|
|
/* Encode setting for Standard Quality */
|
|
case GST_LDAC_EQMID_SQ:
|
|
return 110 * channels;
|
|
/* Encode setting for Mobile use Quality */
|
|
case GST_LDAC_EQMID_MQ:
|
|
return 55 * channels;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
g_assert_not_reached ();
|
|
|
|
/* If assertion gets compiled out */
|
|
return 110 * channels;
|
|
}
|
|
|
|
/**
|
|
* gst_ldac_enc_get_num_frames
|
|
* @eqmid: Encode Quality Mode Index
|
|
* @channels: Number of channels
|
|
*
|
|
* Returns: Number of LDAC frames per packet.
|
|
*/
|
|
static guint
|
|
gst_ldac_enc_get_num_frames (guint eqmid, guint channels)
|
|
{
|
|
g_assert (channels == 1 || channels == 2);
|
|
|
|
switch (eqmid) {
|
|
/* Encode setting for High Quality */
|
|
case GST_LDAC_EQMID_HQ:
|
|
return 4 / channels;
|
|
/* Encode setting for Standard Quality */
|
|
case GST_LDAC_EQMID_SQ:
|
|
return 6 / channels;
|
|
/* Encode setting for Mobile use Quality */
|
|
case GST_LDAC_EQMID_MQ:
|
|
return 12 / channels;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
g_assert_not_reached ();
|
|
|
|
/* If assertion gets compiled out */
|
|
return 6 / channels;
|
|
}
|
|
|
|
/**
|
|
* gst_ldac_enc_get_num_samples
|
|
* @rate: Sampling rate
|
|
*
|
|
* Number of samples in input PCM signal for encoding is fixed to
|
|
* LDACBT_ENC_LSU viz. 128 samples/channel and it is not affected
|
|
* by sampling frequency. However, frame size is 128 samples at 44.1
|
|
* and 48 KHz and 256 at 88.2 and 96 KHz.
|
|
*
|
|
* Returns: Number of samples / channel
|
|
*/
|
|
static guint
|
|
gst_ldac_enc_get_num_samples (guint rate)
|
|
{
|
|
switch (rate) {
|
|
case 44100:
|
|
case 48000:
|
|
return 128;
|
|
case 88200:
|
|
case 96000:
|
|
return 256;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
g_assert_not_reached ();
|
|
|
|
/* If assertion gets compiled out */
|
|
return 128;
|
|
}
|