ext: Add LDAC encoder

LDAC is an audio coding technology developed by Sony that enables the transmission of High-Resolution (Hi-Res) audio contents over Bluetooth. Currently Adaptive Bit Rate (ABR) as supported by libldac encoder is not implemented. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1621>
2024-07-02 12:45:51 +00:00 · 2020-09-18 17:35:24 +05:30 · 2020-09-18 17:35:24 +05:30 · a576814553
parent 7cec64499d
commit a576814553
6 changed files with 743 additions and 0 deletions
--- a/ext/ldac/gstldacenc.c
+++ b/ext/ldac/gstldacenc.c
@ -0,0 +1,618 @@
+/*  GStreamer LDAC audio encoder
+ *  Copyright (C) 2020 Asymptotic <sanchayan@asymptotic.io>
+ *
+ *  This library is free software; you can redistribute it and/or
+ *  modify it under the terms of the GNU Lesser General Public
+ *  License as published by the Free Software Foundation; either
+ *  version 2.1 of the License, or (at your option) any later version.
+ *
+ *  This library is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ *  Lesser General Public License for more details.
+ *
+ *  You should have received a copy of the GNU Lesser General Public
+ *  License along with this library; if not, write to the Free Software
+ *  Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+ *
+ */
+
+/**
+ * SECTION:element-ldacenc
+ * @title: ldacenc
+ *
+ * This element encodes raw integer PCM audio into a Bluetooth LDAC audio.
+ *
+ * ## Example pipeline
+ * |[
+ * gst-launch-1.0 -v audiotestsrc ! ldacenc ! rtpldacpay mtu=679 ! avdtpsink
+ * ]| Encode a sine wave into LDAC, RTP payload it and send over bluetooth
+ *
+ * Since: 1.20
+ */
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <string.h>
+
+#include "gstldacenc.h"
+
+/*
+ * MTU size required for LDAC A2DP streaming. Required for initializing the
+ * encoder.
+ */
+#define GST_LDAC_MTU_REQUIRED    679
+
+GST_DEBUG_CATEGORY_STATIC (ldac_enc_debug);
+#define GST_CAT_DEFAULT ldac_enc_debug
+
+#define parent_class gst_ldac_enc_parent_class
+G_DEFINE_TYPE (GstLdacEnc, gst_ldac_enc, GST_TYPE_AUDIO_ENCODER);
+
+static GstStaticPadTemplate ldac_enc_sink_factory =
+GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS,
+    GST_STATIC_CAPS
+    ("audio/x-raw, format=(string) { S16LE, S24LE, S32LE, F32LE }, "
+        "rate = (int) { 44100, 48000, 88200, 96000 }, "
+        "channels = (int) [ 1, 2 ] "));
+
+static GstStaticPadTemplate ldac_enc_src_factory =
+    GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-ldac, "
+        "rate = (int) { 44100, 48000, 88200, 96000 }, "
+        "channels = (int) 1, channel-mode = (string)mono; "
+        "audio/x-ldac, "
+        "rate = (int) { 44100, 48000, 882000, 96000 }, "
+        "channels = (int) 2, " "channel-mode = (string) { dual, stereo }"));
+
+enum
+{
+  PROP_0,
+  PROP_EQMID
+};
+
+static void gst_ldac_enc_get_property (GObject * object,
+    guint property_id, GValue * value, GParamSpec * pspec);
+static void gst_ldac_enc_set_property (GObject * object,
+    guint property_id, const GValue * value, GParamSpec * pspec);
+
+static gboolean gst_ldac_enc_start (GstAudioEncoder * enc);
+static gboolean gst_ldac_enc_stop (GstAudioEncoder * enc);
+static gboolean gst_ldac_enc_set_format (GstAudioEncoder * enc,
+    GstAudioInfo * info);
+static gboolean gst_ldac_enc_negotiate (GstAudioEncoder * enc);
+static GstFlowReturn gst_ldac_enc_handle_frame (GstAudioEncoder * enc,
+    GstBuffer * buffer);
+static guint gst_ldac_enc_get_num_frames (guint eqmid, guint channels);
+static guint gst_ldac_enc_get_frame_length (guint eqmid, guint channels);
+static guint gst_ldac_enc_get_num_samples (guint rate);
+
+#define GST_LDAC_EQMID (gst_ldac_eqmid_get_type ())
+static GType
+gst_ldac_eqmid_get_type (void)
+{
+  static GType ldac_eqmid_type = 0;
+  static const GEnumValue eqmid_types[] = {
+    {GST_LDAC_EQMID_HQ, "HQ", "hq"},
+    {GST_LDAC_EQMID_SQ, "SQ", "sq"},
+    {GST_LDAC_EQMID_MQ, "MQ", "mq"},
+    {0, NULL, NULL}
+  };
+
+  if (!ldac_eqmid_type)
+    ldac_eqmid_type = g_enum_register_static ("GstLdacEqmid", eqmid_types);
+
+  return ldac_eqmid_type;
+}
+
+static void
+gst_ldac_enc_class_init (GstLdacEncClass * klass)
+{
+  GstAudioEncoderClass *encoder_class = GST_AUDIO_ENCODER_CLASS (klass);
+  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+  GObjectClass *gobject_class = (GObjectClass *) klass;
+
+  gobject_class->set_property = gst_ldac_enc_set_property;
+  gobject_class->get_property = gst_ldac_enc_get_property;
+
+  encoder_class->start = GST_DEBUG_FUNCPTR (gst_ldac_enc_start);
+  encoder_class->stop = GST_DEBUG_FUNCPTR (gst_ldac_enc_stop);
+  encoder_class->set_format = GST_DEBUG_FUNCPTR (gst_ldac_enc_set_format);
+  encoder_class->handle_frame = GST_DEBUG_FUNCPTR (gst_ldac_enc_handle_frame);
+  encoder_class->negotiate = GST_DEBUG_FUNCPTR (gst_ldac_enc_negotiate);
+
+  g_object_class_install_property (gobject_class, PROP_EQMID,
+      g_param_spec_enum ("eqmid", "Encode Quality Mode Index",
+          "Encode Quality Mode Index. 0: High Quality 1: Standard Quality "
+          "2: Mobile Use Quality", GST_LDAC_EQMID,
+          GST_LDAC_EQMID_SQ, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  gst_element_class_add_static_pad_template (element_class,
+      &ldac_enc_sink_factory);
+  gst_element_class_add_static_pad_template (element_class,
+      &ldac_enc_src_factory);
+
+  gst_element_class_set_static_metadata (element_class,
+      "Bluetooth LDAC audio encoder", "Codec/Encoder/Audio",
+      "Encode an LDAC audio stream",
+      "Sanchayan Maity <sanchayan@asymptotic.io>");
+
+  GST_DEBUG_CATEGORY_INIT (ldac_enc_debug, "ldacenc", 0,
+      "LDAC encoding element");
+}
+
+static void
+gst_ldac_enc_init (GstLdacEnc * self)
+{
+  GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_ENCODER_SINK_PAD (self));
+  self->eqmid = GST_LDAC_EQMID_SQ;
+  self->channel_mode = 0;
+  self->init_done = FALSE;
+}
+
+static void
+gst_ldac_enc_set_property (GObject * object, guint property_id,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstLdacEnc *self = GST_LDAC_ENC (object);
+
+  switch (property_id) {
+    case PROP_EQMID:
+      self->eqmid = g_value_get_enum (value);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_ldac_enc_get_property (GObject * object, guint property_id,
+    GValue * value, GParamSpec * pspec)
+{
+  GstLdacEnc *self = GST_LDAC_ENC (object);
+
+  switch (property_id) {
+    case PROP_EQMID:
+      g_value_set_enum (value, self->eqmid);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
+      break;
+  }
+}
+
+static GstCaps *
+gst_ldac_enc_do_negotiate (GstAudioEncoder * audio_enc)
+{
+  GstLdacEnc *enc = GST_LDAC_ENC (audio_enc);
+  GstCaps *caps, *filter_caps;
+  GstCaps *output_caps = NULL;
+  GstStructure *s;
+
+  /* Negotiate output format based on downstream caps restrictions */
+  caps = gst_pad_get_allowed_caps (GST_AUDIO_ENCODER_SRC_PAD (enc));
+  if (caps == GST_CAPS_NONE || gst_caps_is_empty (caps))
+    goto failure;
+
+  if (caps == NULL)
+    caps = gst_static_pad_template_get_caps (&ldac_enc_src_factory);
+
+  /* Fixate output caps */
+  filter_caps = gst_caps_new_simple ("audio/x-ldac", "rate", G_TYPE_INT,
+      enc->rate, "channels", G_TYPE_INT, enc->channels, NULL);
+  output_caps = gst_caps_intersect (caps, filter_caps);
+  gst_caps_unref (filter_caps);
+
+  if (output_caps == NULL || gst_caps_is_empty (output_caps)) {
+    GST_WARNING_OBJECT (enc, "Couldn't negotiate output caps with input rate "
+        "%d and input channels %d and allowed output caps %" GST_PTR_FORMAT,
+        enc->rate, enc->channels, caps);
+    goto failure;
+  }
+
+  gst_clear_caps (&caps);
+
+  GST_DEBUG_OBJECT (enc, "fixating caps %" GST_PTR_FORMAT, output_caps);
+  output_caps = gst_caps_truncate (output_caps);
+  s = gst_caps_get_structure (output_caps, 0);
+  if (enc->channels == 1)
+    gst_structure_fixate_field_string (s, "channel-mode", "mono");
+  else
+    gst_structure_fixate_field_string (s, "channel-mode", "stereo");
+  s = NULL;
+
+  /* In case there's anything else left to fixate */
+  output_caps = gst_caps_fixate (output_caps);
+  gst_caps_set_simple (output_caps, "framed", G_TYPE_BOOLEAN, TRUE, NULL);
+
+  GST_INFO_OBJECT (enc, "output caps %" GST_PTR_FORMAT, output_caps);
+
+  if (enc->channels == 1)
+    enc->channel_mode = LDACBT_CHANNEL_MODE_MONO;
+  else
+    enc->channel_mode = LDACBT_CHANNEL_MODE_STEREO;
+
+  return output_caps;
+
+failure:
+  if (output_caps)
+    gst_caps_unref (output_caps);
+  if (caps)
+    gst_caps_unref (caps);
+  return NULL;
+}
+
+static gboolean
+gst_ldac_enc_negotiate (GstAudioEncoder * audio_enc)
+{
+  GstLdacEnc *enc = GST_LDAC_ENC (audio_enc);
+  GstCaps *output_caps = NULL;
+
+  output_caps = gst_ldac_enc_do_negotiate (audio_enc);
+  if (output_caps == NULL) {
+    GST_ERROR_OBJECT (enc, "failed to negotiate");
+    return FALSE;
+  }
+
+  if (!gst_audio_encoder_set_output_format (audio_enc, output_caps)) {
+    GST_ERROR_OBJECT (enc, "failed to configure output caps on src pad");
+    gst_caps_unref (output_caps);
+    return FALSE;
+  }
+  gst_caps_unref (output_caps);
+
+  return GST_AUDIO_ENCODER_CLASS (parent_class)->negotiate (audio_enc);
+}
+
+static gboolean
+gst_ldac_enc_set_format (GstAudioEncoder * audio_enc, GstAudioInfo * info)
+{
+  GstLdacEnc *enc = GST_LDAC_ENC (audio_enc);
+  GstCaps *output_caps = NULL;
+  guint num_ldac_frames, num_samples;
+  gint ret = 0;
+
+  enc->rate = GST_AUDIO_INFO_RATE (info);
+  enc->channels = GST_AUDIO_INFO_CHANNELS (info);
+
+  switch (GST_AUDIO_INFO_FORMAT (info)) {
+    case GST_AUDIO_FORMAT_S16:
+      enc->ldac_fmt = LDACBT_SMPL_FMT_S16;
+      break;
+    case GST_AUDIO_FORMAT_S24:
+      enc->ldac_fmt = LDACBT_SMPL_FMT_S24;
+      break;
+    case GST_AUDIO_FORMAT_S32:
+      enc->ldac_fmt = LDACBT_SMPL_FMT_S32;
+      break;
+    case GST_AUDIO_FORMAT_F32:
+      enc->ldac_fmt = LDACBT_SMPL_FMT_F32;
+      break;
+    default:
+      GST_ERROR_OBJECT (enc, "Invalid audio format");
+      return FALSE;
+  }
+
+  output_caps = gst_ldac_enc_do_negotiate (audio_enc);
+  if (output_caps == NULL) {
+    GST_ERROR_OBJECT (enc, "failed to negotiate");
+    return FALSE;
+  }
+
+  if (!gst_audio_encoder_set_output_format (audio_enc, output_caps)) {
+    GST_ERROR_OBJECT (enc, "failed to configure output caps on src pad");
+    gst_caps_unref (output_caps);
+    return FALSE;
+  }
+  gst_caps_unref (output_caps);
+
+  num_samples = gst_ldac_enc_get_num_samples (enc->rate);
+  num_ldac_frames = gst_ldac_enc_get_num_frames (enc->eqmid, enc->channels);
+  gst_audio_encoder_set_frame_samples_min (audio_enc,
+      num_samples * num_ldac_frames);
+
+  /*
+   * If initialisation was already done means caps have changed, close the
+   * handle. Closed handle can be initialised and used again.
+   */
+  if (enc->init_done) {
+    ldacBT_close_handle (enc->ldac);
+    enc->init_done = FALSE;
+  }
+
+  /*
+   * libldac exposes a bluetooth centric API and emits multiple LDAC frames
+   * depending on the MTU. The MTU is required for LDAC A2DP streaming, is
+   * inclusive of the RTP header and is required by the encoder. The internal
+   * encoder API is not exposed in the public interface.
+   */
+  ret =
+      ldacBT_init_handle_encode (enc->ldac, GST_LDAC_MTU_REQUIRED, enc->eqmid,
+      enc->channel_mode, enc->ldac_fmt, enc->rate);
+  if (ret != 0) {
+    GST_ERROR_OBJECT (enc, "Failed to initialize LDAC handle, ret: %d", ret);
+    return FALSE;
+  }
+  enc->init_done = TRUE;
+
+  return TRUE;
+}
+
+static GstFlowReturn
+gst_ldac_enc_handle_frame (GstAudioEncoder * audio_enc, GstBuffer * buffer)
+{
+  GstLdacEnc *enc = GST_LDAC_ENC (audio_enc);
+  GstMapInfo in_map, out_map;
+  GstAudioInfo *info;
+  GstBuffer *outbuf;
+  const guint8 *in_data;
+  guint8 *out_data;
+  gint encoded, to_encode = 0;
+  gint samples_consumed = 0;
+  guint frames, frame_len;
+  guint ldac_enc_read = 0;
+  guint frame_count = 0;
+
+  if (buffer == NULL)
+    return GST_FLOW_OK;
+
+  if (!gst_buffer_map (buffer, &in_map, GST_MAP_READ)) {
+    GST_ELEMENT_ERROR (audio_enc, STREAM, FAILED, (NULL),
+        ("Failed to map data from input buffer"));
+    return GST_FLOW_ERROR;
+  }
+
+  info = gst_audio_encoder_get_audio_info (audio_enc);
+  ldac_enc_read = LDACBT_ENC_LSU * info->bpf;
+  /*
+   * We may produce extra frames at the end of encoding process (See below).
+   * Consider some additional frames while allocating output buffer if this
+   * happens.
+   */
+  frames = (in_map.size / ldac_enc_read) + 4;
+
+  frame_len = gst_ldac_enc_get_frame_length (enc->eqmid, info->channels);
+  outbuf = gst_audio_encoder_allocate_output_buffer (audio_enc,
+      frames * frame_len);
+  if (outbuf == NULL)
+    goto no_buffer;
+
+  gst_buffer_map (outbuf, &out_map, GST_MAP_WRITE);
+  in_data = in_map.data;
+  out_data = out_map.data;
+  to_encode = in_map.size;
+
+  /*
+   * ldacBT_encode does not generate an output frame each time it is called.
+   * For each invocation, it consumes number of sample * bpf bytes of data.
+   * Depending on the eqmid setting and channels, it will emit multiple frames
+   * only after the required number of frames are packed for payloading. Below
+   * for loop exists primarily to handle this.
+   */
+  for (;;) {
+    guint8 pcm[LDACBT_MAX_LSU * 4 /* bytes/sample */  * 2 /* ch */ ] = { 0 };
+    gint ldac_frame_num, written;
+    guint8 *inp_data = NULL;
+    gboolean done = FALSE;
+    gint ret;
+
+    /*
+     * Even with minimum frame samples specified in set_format with EOS,
+     * we may get a buffer which is not a multiple of LDACBT_ENC_LSU. LDAC
+     * encoder always reads a multiple of this and to handle this scenario
+     * we use local PCM array and in the last iteration when buffer bytes
+     * < LDACBT_ENC_LSU * bpf, we copy only to_encode bytes to prevent
+     * walking off the end of input buffer and the rest of the bytes in
+     * PCM buffer would be zero, so should be safe from encoding point of
+     * view.
+     */
+    if (to_encode < 0) {
+      /*
+       * We got < LDACBT_ENC_LSU * bpf for last iteration. Force the encoder
+       * to encode the remaining bytes in buffer by passing NULL to the input
+       * PCM buffer argument.
+       */
+      inp_data = NULL;
+      done = TRUE;
+    } else if (to_encode >= ldac_enc_read) {
+      memcpy (pcm, in_data, ldac_enc_read);
+      inp_data = &pcm[0];
+    } else if (to_encode > 0 && to_encode < ldac_enc_read) {
+      memcpy (pcm, in_data, to_encode);
+      inp_data = &pcm[0];
+    }
+
+    /*
+     * Note that while we do not explicitly pass length of data to library
+     * anywhere, based on the initialization considering eqmid and rate, the
+     * library will consume a fix number of samples per call. This combined
+     * with the previous step ensures that the library does not read outside
+     * of in_data and out_data.
+     */
+    ret = ldacBT_encode (enc->ldac, (void *) inp_data, &encoded,
+        (guint8 *) out_data, &written, &ldac_frame_num);
+    if (ret < 0) {
+      GST_ERROR_OBJECT (enc, "encoding error, ret = %d written = %d",
+          ret, ldac_frame_num);
+      goto encoding_error;
+    } 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_ERROR_OBJECT (enc, "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;
+}
--- a/ext/ldac/gstldacenc.h
+++ b/ext/ldac/gstldacenc.h
@ -0,0 +1,66 @@
+/*  GStreamer LDAC audio encoder
+ *  Copyright (C) 2020 Asymptotic <sanchayan@asymptotic.io>
+ *
+ *  This library is free software; you can redistribute it and/or
+ *  modify it under the terms of the GNU Lesser General Public
+ *  License as published by the Free Software Foundation; either
+ *  version 2.1 of the License, or (at your option) any later version.
+ *
+ *  This library is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ *  Lesser General Public License for more details.
+ *
+ *  You should have received a copy of the GNU Lesser General Public
+ *  License along with this library; if not, write to the Free Software
+ *  Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+ */
+
+#include <gst/gst.h>
+#include <gst/audio/audio.h>
+
+#include <ldac/ldacBT.h>
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_LDAC_ENC \
+	(gst_ldac_enc_get_type())
+#define GST_LDAC_ENC(obj) \
+	(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_LDAC_ENC,GstLdacEnc))
+#define GST_LDAC_ENC_CLASS(klass) \
+	(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_LDAC_ENC,GstLdacEncClass))
+#define GST_IS_LDAC_ENC(obj) \
+	(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_LDAC_ENC))
+#define GST_IS_LDAC_ENC_CLASS(obj) \
+	(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_LDAC_ENC))
+
+typedef struct _GstLdacEnc GstLdacEnc;
+typedef struct _GstLdacEncClass GstLdacEncClass;
+
+typedef enum
+{
+  GST_LDAC_EQMID_HQ = 0,
+  GST_LDAC_EQMID_SQ,
+  GST_LDAC_EQMID_MQ
+} GstLdacEqmid;
+
+struct _GstLdacEnc {
+    GstAudioEncoder audio_encoder;
+    GstLdacEqmid eqmid;
+
+    guint rate;
+    guint channels;
+    guint channel_mode;
+    gboolean init_done;
+
+    LDACBT_SMPL_FMT_T ldac_fmt;
+    HANDLE_LDAC_BT ldac;
+};
+
+struct _GstLdacEncClass {
+    GstAudioEncoderClass audio_encoder_class;
+};
+
+GType gst_ldac_enc_get_type (void);
+
+G_END_DECLS
--- a/ext/ldac/ldac-plugin.c
+++ b/ext/ldac/ldac-plugin.c
@ -0,0 +1,38 @@
+/* GStreamer LDAC audio plugin
+ * Copyright (C) 2020 Asymptotic <sanchayan@asymptotic.io>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include "gstldacenc.h"
+#include <string.h>
+
+static gboolean
+plugin_init (GstPlugin * plugin)
+{
+  gst_element_register (plugin, "ldacenc", GST_RANK_NONE, GST_TYPE_LDAC_ENC);
+  return TRUE;
+}
+
+GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
+    GST_VERSION_MINOR,
+    ldac,
+    "LDAC bluetooth audio support",
+    plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN);
--- a/ext/ldac/meson.build
+++ b/ext/ldac/meson.build
@ -0,0 +1,19 @@
+ldac_sources = [
+  'ldac-plugin.c',
+  'gstldacenc.c',
+]
+
+ldac_dep = cc.find_library('ldacBT_enc', required : get_option('ldac'))
+
+if ldac_dep.found()
+  gstldac = library('gstldac',
+    ldac_sources,
+    c_args : gst_plugins_bad_args,
+    include_directories : [configinc],
+    dependencies : [gstaudio_dep, ldac_dep],
+    install : true,
+    install_dir : plugins_install_dir,
+  )
+  pkgconfig.generate(gstldac, install_dir : plugins_pkgconfig_install_dir)
+  plugins += [gstldac]
+endif
--- a/ext/meson.build
+++ b/ext/meson.build
@ -24,6 +24,7 @@ subdir('iqa')
 subdir('isac')
 subdir('kate')
 subdir('ladspa')
+subdir('ldac')
 subdir('libde265')
 subdir('libmms')
 subdir('lv2')
--- a/meson_options.txt
+++ b/meson_options.txt
@ -113,6 +113,7 @@ option('iqa', type : 'feature', value : 'auto', description : 'Image quality ass
 option('kate', type : 'feature', value : 'auto', description : 'Kate subtitle parser, tagger, and codec plugin')
 option('kms', type : 'feature', value : 'auto', description : 'KMS video sink plugin')
 option('ladspa', type : 'feature', value : 'auto', description : 'LADSPA plugin bridge')
+option('ldac', type : 'feature', value : 'auto', description : 'LDAC bluetooth audio codec plugin')
 option('libde265', type : 'feature', value : 'auto', description : 'HEVC/H.265 video decoder plugin')
 option('libmms', type : 'feature', value : 'auto', description : 'Microsoft multimedia server network source plugin')
 option('lv2', type : 'feature', value : 'auto', description : 'LV2 audio plugin bridge')