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297 lines
8.3 KiB
C
297 lines
8.3 KiB
C
/* iSAC decoder
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*
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* Copyright (C) 2020 Collabora Ltd.
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* Author: Guillaume Desmottes <guillaume.desmottes@collabora.com>, Collabora Ltd.
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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 Free
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* Software Foundation, Inc., 51 Franklin Street, 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-isacdec
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* @title: isacdec
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* @short_description: iSAC audio decoder
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*
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* Since: 1.20
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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 "gstisacdec.h"
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#include "gstisacutils.h"
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#include <modules/audio_coding/codecs/isac/main/include/isac.h>
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GST_DEBUG_CATEGORY_STATIC (isacdec_debug);
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#define GST_CAT_DEFAULT isacdec_debug
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#define SAMPLE_SIZE 2 /* 16-bits samples */
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#define MAX_OUTPUT_SAMPLES 960 /* decoder produces max 960 samples */
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#define MAX_OUTPUT_SIZE (SAMPLE_SIZE * MAX_OUTPUT_SAMPLES)
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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/isac, "
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"rate = (int) { 16000, 32000 }, " "channels = (int) 1")
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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/x-raw, "
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"format = (string) " GST_AUDIO_NE (S16) ", "
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"rate = (int) { 16000, 32000 }, "
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"layout = (string) interleaved, " "channels = (int) 1")
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);
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struct _GstIsacDec
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{
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/*< private > */
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GstAudioDecoder parent;
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ISACStruct *isac;
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/* properties */
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};
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#define gst_isacdec_parent_class parent_class
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G_DEFINE_TYPE_WITH_CODE (GstIsacDec, gst_isacdec,
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GST_TYPE_AUDIO_DECODER,
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GST_DEBUG_CATEGORY_INIT (isacdec_debug, "isacdec", 0,
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"debug category for isacdec element"));
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GST_ELEMENT_REGISTER_DEFINE (isacdec, "isacdec", GST_RANK_PRIMARY,
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GST_TYPE_ISACDEC);
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static gboolean
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gst_isacdec_start (GstAudioDecoder * dec)
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{
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GstIsacDec *self = GST_ISACDEC (dec);
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gint16 ret;
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g_assert (!self->isac);
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ret = WebRtcIsac_Create (&self->isac);
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CHECK_ISAC_RET (ret, Create);
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return TRUE;
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}
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static gboolean
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gst_isacdec_stop (GstAudioDecoder * dec)
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{
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GstIsacDec *self = GST_ISACDEC (dec);
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if (self->isac) {
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gint16 ret;
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ret = WebRtcIsac_Free (self->isac);
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CHECK_ISAC_RET (ret, Free);
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self->isac = NULL;
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}
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return TRUE;
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}
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static gboolean
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gst_isacdec_set_format (GstAudioDecoder * dec, GstCaps * input_caps)
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{
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GstIsacDec *self = GST_ISACDEC (dec);
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GstAudioInfo output_format;
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gint16 ret;
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gboolean result;
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GstStructure *s;
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gint rate, channels;
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GstCaps *output_caps;
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GST_DEBUG_OBJECT (self, "input caps: %" GST_PTR_FORMAT, input_caps);
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s = gst_caps_get_structure (input_caps, 0);
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if (!s)
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return FALSE;
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if (!gst_structure_get_int (s, "rate", &rate)) {
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GST_ERROR_OBJECT (self, "'rate' missing in input caps: %" GST_PTR_FORMAT,
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input_caps);
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return FALSE;
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}
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if (!gst_structure_get_int (s, "channels", &channels)) {
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GST_ERROR_OBJECT (self,
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"'channels' missing in input caps: %" GST_PTR_FORMAT, input_caps);
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return FALSE;
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}
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gst_audio_info_set_format (&output_format, GST_AUDIO_FORMAT_S16LE, rate,
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channels, NULL);
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output_caps = gst_audio_info_to_caps (&output_format);
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GST_DEBUG_OBJECT (self, "output caps: %" GST_PTR_FORMAT, output_caps);
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gst_caps_unref (output_caps);
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ret = WebRtcIsac_SetDecSampRate (self->isac, rate);
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CHECK_ISAC_RET (ret, SetDecSampleRate);
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WebRtcIsac_DecoderInit (self->isac);
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result = gst_audio_decoder_set_output_format (dec, &output_format);
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gst_audio_decoder_set_plc_aware (dec, TRUE);
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return result;
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}
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static GstFlowReturn
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gst_isacdec_plc (GstIsacDec * self, GstClockTime duration)
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{
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GstAudioDecoder *dec = GST_AUDIO_DECODER (self);
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guint nb_plc_frames;
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GstBuffer *output;
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GstMapInfo map_write;
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size_t ret;
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/* Decoder produces 30 ms PLC frames */
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nb_plc_frames = duration / (30 * GST_MSECOND);
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GST_DEBUG_OBJECT (self,
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"GAP of %" GST_TIME_FORMAT " detected, request PLC for %d frames",
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GST_TIME_ARGS (duration), nb_plc_frames);
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output =
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gst_audio_decoder_allocate_output_buffer (dec,
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nb_plc_frames * MAX_OUTPUT_SIZE);
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if (!gst_buffer_map (output, &map_write, GST_MAP_WRITE)) {
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GST_ERROR_OBJECT (self, "Failed to map output buffer");
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gst_buffer_unref (output);
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return GST_FLOW_ERROR;
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}
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ret =
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WebRtcIsac_DecodePlc (self->isac, (gint16 *) map_write.data,
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nb_plc_frames);
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gst_buffer_unmap (output, &map_write);
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if (ret < 0) {
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/* error */
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gint16 code = WebRtcIsac_GetErrorCode (self->isac);
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GST_WARNING_OBJECT (self, "Failed to produce PLC: %s (%d)",
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isac_error_code_to_str (code), code);
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gst_buffer_unref (output);
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return GST_FLOW_ERROR;
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} else if (ret == 0) {
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GST_DEBUG_OBJECT (self, "Decoder didn't produce any PLC frame");
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gst_buffer_unref (output);
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return GST_FLOW_OK;
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}
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gst_buffer_set_size (output, ret * SAMPLE_SIZE);
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GST_LOG_OBJECT (self, "Produced %" G_GSIZE_FORMAT " PLC samples", ret);
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return gst_audio_decoder_finish_frame (dec, output, 1);
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}
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static GstFlowReturn
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gst_isacdec_handle_frame (GstAudioDecoder * dec, GstBuffer * input)
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{
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GstIsacDec *self = GST_ISACDEC (dec);
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GstMapInfo map_read, map_write;
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GstBuffer *output;
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gint16 ret, speech_type[1];
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gsize input_size;
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/* Can't drain the decoder */
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if (!input)
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return GST_FLOW_OK;
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if (!gst_buffer_get_size (input)) {
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/* Base class detected a gap in the stream, try to do PLC */
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return gst_isacdec_plc (self, GST_BUFFER_DURATION (input));
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}
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if (!gst_buffer_map (input, &map_read, GST_MAP_READ)) {
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GST_ELEMENT_ERROR (self, RESOURCE, READ, ("Failed to map input buffer"),
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(NULL));
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return GST_FLOW_ERROR;
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}
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input_size = map_read.size;
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output = gst_audio_decoder_allocate_output_buffer (dec, MAX_OUTPUT_SIZE);
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if (!gst_buffer_map (output, &map_write, GST_MAP_WRITE)) {
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GST_ELEMENT_ERROR (self, RESOURCE, WRITE, ("Failed to map output buffer"),
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(NULL));
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gst_buffer_unref (output);
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gst_buffer_unmap (input, &map_read);
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return GST_FLOW_ERROR;
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}
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ret = WebRtcIsac_Decode (self->isac, map_read.data, map_read.size,
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(gint16 *) map_write.data, speech_type);
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gst_buffer_unmap (input, &map_read);
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gst_buffer_unmap (output, &map_write);
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if (ret < 0) {
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/* error */
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gint16 code = WebRtcIsac_GetErrorCode (self->isac);
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GST_WARNING_OBJECT (self, "Failed to decode: %s (%d)",
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isac_error_code_to_str (code), code);
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gst_buffer_unref (output);
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/* Give a chance to decode next frames */
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return GST_FLOW_OK;
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} else if (ret == 0) {
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GST_DEBUG_OBJECT (self, "Decoder didn't produce any frame");
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gst_buffer_unref (output);
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output = NULL;
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} else {
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gst_buffer_set_size (output, ret * SAMPLE_SIZE);
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}
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GST_LOG_OBJECT (self, "Decoded %d samples from %" G_GSIZE_FORMAT " bytes",
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ret, input_size);
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return gst_audio_decoder_finish_frame (dec, output, 1);
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}
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static void
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gst_isacdec_class_init (GstIsacDecClass * klass)
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{
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GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
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GstAudioDecoderClass *base_class = GST_AUDIO_DECODER_CLASS (klass);
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base_class->start = GST_DEBUG_FUNCPTR (gst_isacdec_start);
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base_class->stop = GST_DEBUG_FUNCPTR (gst_isacdec_stop);
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base_class->set_format = GST_DEBUG_FUNCPTR (gst_isacdec_set_format);
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base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_isacdec_handle_frame);
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gst_element_class_set_static_metadata (gstelement_class, "iSAC decoder",
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"Codec/Decoder/Audio",
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"iSAC audio decoder",
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"Guillaume Desmottes <guillaume.desmottes@collabora.com>");
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gst_element_class_add_static_pad_template (gstelement_class, &sink_template);
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gst_element_class_add_static_pad_template (gstelement_class, &src_template);
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}
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static void
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gst_isacdec_init (GstIsacDec * self)
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{
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self->isac = NULL;
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}
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