gstreamer/subprojects/gst-plugins-bad/ext/lc3/gstlc3dec.c

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/* GStreamer LC3 Bluetooth LE audio decoder
* Copyright (C) 2023 Asymptotic Inc. <taruntej@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 Street, Suite 500,
* Boston, MA 02110-1335, USA.
*/
/**
* SECTION:element-lc3dec
*
* The lc3dec decodes LC3 data into raw audio.
*
* ## Example pipeline
* |[
* gst-launch-1.0 -v filesrc location=encoded.lc3 blocksize=200 ! \
* audio/x-lc3,frame-bytes=100,frame-duration-us=10000,channels=2,rate=48000,channel-mask=\(bitmask\)0x00000000000000003 !\
* lc3dec ! wavenc ! filesink location=decoded.wav
* ]|
*
* Decodes the LC3 frames each with 100 bytes of size, converts it to raw audio and saves into a .wav file
*
* Since: 1.24
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/gst.h>
#include <gst/audio/gstaudiodecoder.h>
#include "gstlc3common.h"
#include "gstlc3dec.h"
GST_DEBUG_CATEGORY_STATIC (gst_lc3_dec_debug_category);
#define GST_CAT_DEFAULT gst_lc3_dec_debug_category
#define parent_class gst_lc3_dec_parent_class
G_DEFINE_TYPE (GstLc3Dec, gst_lc3_dec, GST_TYPE_AUDIO_DECODER);
GST_ELEMENT_REGISTER_DEFINE (lc3dec, "lc3dec", GST_RANK_NONE, GST_TYPE_LC3_DEC);
/* prototypes */
static gboolean gst_lc3_dec_start (GstAudioDecoder * decoder);
static gboolean gst_lc3_dec_stop (GstAudioDecoder * decoder);
static gboolean gst_lc3_dec_set_format (GstAudioDecoder * decoder,
GstCaps * caps);
static GstFlowReturn gst_lc3_dec_handle_frame (GstAudioDecoder * decoder,
GstBuffer * buffer);
/* pad templates */
static GstStaticPadTemplate gst_lc3_dec_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = " FORMAT ", layout=interleaved, "
"rate = { " SAMPLE_RATES " }, channels = [1,MAX]")
);
static GstStaticPadTemplate gst_lc3_dec_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-lc3, rate = { " SAMPLE_RATES " }, "
"channels = [1,MAX],"
"frame-bytes = (int) [" FRAME_BYTES_RANGE "], "
"frame-duration-us = (int) { " FRAME_DURATIONS " }, "
"framed=(boolean) true")
);
/* class initialization */
static void
gst_lc3_dec_class_init (GstLc3DecClass * klass)
{
GstAudioDecoderClass *audio_decoder_class = GST_AUDIO_DECODER_CLASS (klass);
gst_element_class_add_static_pad_template (GST_ELEMENT_CLASS (klass),
&gst_lc3_dec_src_template);
gst_element_class_add_static_pad_template (GST_ELEMENT_CLASS (klass),
&gst_lc3_dec_sink_template);
gst_element_class_set_static_metadata (GST_ELEMENT_CLASS (klass),
"LC3 Bluetooth Audio decoder", "Codec/Decoder/Audio",
"Decodes an LC3 Audio stream to raw audio",
"Taruntej Kanakamalla <taruntej@asymptotic.io>");
GST_DEBUG_CATEGORY_INIT (gst_lc3_dec_debug_category, "lc3dec", 0,
"debug category for lc3dec element");
audio_decoder_class->start = GST_DEBUG_FUNCPTR (gst_lc3_dec_start);
audio_decoder_class->stop = GST_DEBUG_FUNCPTR (gst_lc3_dec_stop);
audio_decoder_class->set_format = GST_DEBUG_FUNCPTR (gst_lc3_dec_set_format);
audio_decoder_class->handle_frame =
GST_DEBUG_FUNCPTR (gst_lc3_dec_handle_frame);
}
static void
gst_lc3_dec_init (GstLc3Dec * lc3_dec)
{
}
static gboolean
gst_lc3_dec_start (GstAudioDecoder * decoder)
{
/* let the baseclass convert the segment data
* from 'bytes' to 'time' format
*/
gst_audio_decoder_set_estimate_rate (decoder, TRUE);
/* Inform the base class that the LC3 lib can do PLC */
gst_audio_decoder_set_plc_aware (decoder, TRUE);
return TRUE;
}
static gboolean
gst_lc3_dec_stop (GstAudioDecoder * decoder)
{
GstLc3Dec *lc3_dec = GST_LC3_DEC (decoder);
if (lc3_dec->dec_ch != NULL) {
for (int ich = 0; ich < lc3_dec->channels; ich++) {
g_free (lc3_dec->dec_ch[ich]);
lc3_dec->dec_ch[ich] = NULL;
}
g_free (lc3_dec->dec_ch);
lc3_dec->dec_ch = NULL;
}
return TRUE;
}
static gboolean
gst_lc3_dec_set_format (GstAudioDecoder * decoder, GstCaps * caps)
{
GstLc3Dec *lc3_dec = GST_LC3_DEC (decoder);
GstAudioInfo info;
GstStructure *s;
GstAudioChannelPosition pos[64] = { GST_AUDIO_CHANNEL_POSITION_INVALID, };
gint in_ch, in_rate;
guint64 in_chmsk = 0;
GstClockTime latency;
GST_DEBUG_OBJECT (lc3_dec, "set_format");
GST_DEBUG_OBJECT (lc3_dec, "input caps %" GST_PTR_FORMAT, caps);
s = gst_caps_get_structure (caps, 0);
if (!gst_structure_get_int (s, "frame-duration-us",
&lc3_dec->frame_duration_us)) {
GST_ERROR_OBJECT (lc3_dec,
"sink caps does not contain 'frame-duration-us'");
return FALSE;
}
if (!gst_structure_get_int (s, "frame-bytes", &lc3_dec->frame_bytes)) {
GST_ERROR_OBJECT (lc3_dec, "sink caps does not contain 'frame-bytes'");
return FALSE;
}
/* use rate and channel from input caps to create filter caps */
gst_structure_get_int (s, "rate", &in_rate);
gst_structure_get_int (s, "channels", &in_ch);
if (!gst_structure_get (s, "channel-mask", GST_TYPE_BITMASK, &in_chmsk, NULL)) {
GST_INFO_OBJECT (lc3_dec,
"channel-mask not present in the sink caps, getting fallback mask");
in_chmsk = gst_audio_channel_get_fallback_mask (in_ch);
}
s = NULL;
gst_audio_channel_positions_from_mask (in_ch, in_chmsk, pos);
gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_S16LE, in_rate, in_ch,
pos);
/* get rate, format, channels from the output caps */
lc3_dec->rate = GST_AUDIO_INFO_RATE (&info);
lc3_dec->channels = GST_AUDIO_INFO_CHANNELS (&info);
switch (GST_AUDIO_INFO_FORMAT (&info)) {
case GST_AUDIO_FORMAT_S16LE:
lc3_dec->format = LC3_PCM_FORMAT_S16;
break;
case GST_AUDIO_FORMAT_S24LE:
lc3_dec->format = LC3_PCM_FORMAT_S24_3LE;
break;
case GST_AUDIO_FORMAT_F32:
lc3_dec->format = LC3_PCM_FORMAT_FLOAT;
break;
case GST_AUDIO_FORMAT_S24_32LE:
default:
lc3_dec->format = LC3_PCM_FORMAT_S24;
break;
}
GST_INFO_OBJECT (lc3_dec, "lc3dec params "
"rate: %" G_GINT32_FORMAT ", channels: %" G_GINT32_FORMAT
", lc3_pcm_format = %" G_GINT32_FORMAT " frame len: %" G_GINT32_FORMAT
", frame_duration " "%" G_GINT32_FORMAT, lc3_dec->rate, lc3_dec->channels,
lc3_dec->format, lc3_dec->frame_bytes, lc3_dec->frame_duration_us);
lc3_dec->frame_samples =
lc3_frame_samples (lc3_dec->frame_duration_us, lc3_dec->rate);
lc3_dec->bpf = GST_AUDIO_INFO_BPF (&info);
latency =
gst_util_uint64_scale_int (lc3_dec->frame_bytes, GST_SECOND,
lc3_dec->rate);
gst_audio_decoder_set_latency (decoder, latency, latency);
/* Setup and Init decoder handle */
if (lc3_dec->dec_ch != NULL) {
for (int ich = 0; ich < lc3_dec->channels; ich++) {
g_free (lc3_dec->dec_ch[ich]);
lc3_dec->dec_ch[ich] = NULL;
}
g_free (lc3_dec->dec_ch);
lc3_dec->dec_ch = NULL;
}
lc3_dec->dec_ch = g_new0 (lc3_decoder_t, lc3_dec->channels);
for (guint8 i = 0; i < lc3_dec->channels; i++) {
/* The decoder can resample for us. But we leave the resampling to before decoding
* explicitly for now. So pass the same sample rate for sr_hz and sr_pcm_hz
*/
lc3_dec->dec_ch[i] =
lc3_setup_decoder (lc3_dec->frame_duration_us, lc3_dec->rate,
lc3_dec->rate, g_malloc (lc3_decoder_size (lc3_dec->frame_duration_us,
lc3_dec->rate)));
if (lc3_dec->dec_ch[i] == NULL) {
GST_ERROR_OBJECT (lc3_dec,
"Failed to create decoder handle for channel %" G_GUINT32_FORMAT, i);
return FALSE;
}
}
gst_audio_decoder_set_output_format (decoder, &info);
return TRUE;
}
static GstFlowReturn
gst_lc3_dec_handle_frame (GstAudioDecoder * decoder, GstBuffer * inbuf)
{
GstLc3Dec *lc3_dec = GST_LC3_DEC (decoder);
GstBuffer *outbuf = NULL;
GstMapInfo out_map;
GstMapInfo in_map;
gssize output_size;
GstAudioClippingMeta *audio_meta;
gboolean do_plc = gst_audio_decoder_get_plc (decoder) &&
gst_audio_decoder_get_plc_aware (decoder);
/* no fancy draining */
if (G_UNLIKELY (inbuf == NULL))
return GST_FLOW_OK;
gst_buffer_map (inbuf, &in_map, GST_MAP_READ);
if (G_UNLIKELY (in_map.size == 0 && !do_plc)) {
GST_ERROR_OBJECT (lc3_dec,
"PLC handled by the base class, should not get a zero sized buffer");
return GST_FLOW_ERROR;
}
GST_LOG_OBJECT (lc3_dec, "received %lu bytes ", in_map.size);
/* we expect exactly one frame each time */
if (G_UNLIKELY (in_map.size == 0 && !do_plc) &&
(in_map.size != (lc3_dec->frame_bytes * lc3_dec->channels)))
goto mixed_frames;
output_size = lc3_dec->frame_samples * lc3_dec->bpf;
GST_LOG_OBJECT (lc3_dec, "allocating %lu bytes to output buffer",
output_size);
outbuf = gst_audio_decoder_allocate_output_buffer (decoder, output_size);
if (outbuf == NULL)
goto no_buffer;
gst_buffer_map (outbuf, &out_map, GST_MAP_WRITE);
for (guint c = 0; c < lc3_dec->channels; c++) {
gint ret = 0;
void *in = in_map.data ? in_map.data + (c * lc3_dec->frame_bytes) : NULL;
ret =
lc3_decode (lc3_dec->dec_ch[c], in, lc3_dec->frame_bytes,
lc3_dec->format, out_map.data + (c * lc3_dec->bpf / lc3_dec->channels),
lc3_dec->channels);
if (ret < 0) {
GST_ERROR_OBJECT (lc3_dec,
"Failed to decode frame for buffer %" GST_PTR_FORMAT, inbuf);
return GST_FLOW_ERROR;
} else if (ret == 1) {
GST_INFO_OBJECT (lc3_dec, "PLC operated for channel: %d", c + 1);
}
}
audio_meta = gst_buffer_get_audio_clipping_meta (inbuf);
if (audio_meta) {
switch (audio_meta->format) {
case GST_FORMAT_DEFAULT:
{
output_size =
output_size - (audio_meta->start * lc3_dec->bpf) -
(audio_meta->end * lc3_dec->bpf);
gst_buffer_resize (outbuf, (audio_meta->start * lc3_dec->bpf),
output_size);
}
break;
default:
GST_WARNING_OBJECT (lc3_dec, "audio meta format: %d not handled",
audio_meta->format);
break;
}
}
gst_buffer_unmap (outbuf, &out_map);
gst_buffer_unmap (inbuf, &in_map);
return gst_audio_decoder_finish_frame (decoder, outbuf, 1);
/* ERRORS */
mixed_frames:
{
GST_WARNING_OBJECT (lc3_dec,
"inconsistent input data/frames, Need to be %"
G_GINT32_FORMAT " bytes", lc3_dec->frame_bytes * lc3_dec->channels);
return GST_FLOW_ERROR;
}
no_buffer:
{
GST_ERROR_OBJECT (lc3_dec, "could not allocate output buffer");
return GST_FLOW_ERROR;
}
}