gstreamer/ext/ldac/gstldacenc.c

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/* 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);
#define SAMPLE_RATES "44100, 48000, 88200, 96000"
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) { " SAMPLE_RATES " }, 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) { " SAMPLE_RATES " }, "
"channels = (int) 1, channel-mode = (string)mono; "
"audio/x-ldac, "
"rate = (int) { " SAMPLE_RATES " }, "
"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;
}