/* iSAC decoder * * Copyright (C) 2020 Collabora Ltd. * Author: Guillaume Desmottes , Collabora Ltd. * * 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, Fifth Floor, * Boston, MA 02110-1301 USA. */ /** * SECTION:element-isacdec * @title: isacdec * @short_description: iSAC audio decoder * * Since: 1.20 * */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include "gstisacdec.h" #include "gstisacutils.h" #include GST_DEBUG_CATEGORY_STATIC (isacdec_debug); #define GST_CAT_DEFAULT isacdec_debug #define SAMPLE_SIZE 2 /* 16-bits samples */ #define MAX_OUTPUT_SAMPLES 960 /* decoder produces max 960 samples */ #define MAX_OUTPUT_SIZE (SAMPLE_SIZE * MAX_OUTPUT_SAMPLES) static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/isac, " "rate = (int) { 16000, 32000 }, " "channels = (int) 1") ); static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw, " "format = (string) " GST_AUDIO_NE (S16) ", " "rate = (int) { 16000, 32000 }, " "layout = (string) interleaved, " "channels = (int) 1") ); struct _GstIsacDec { /*< private > */ GstAudioDecoder parent; ISACStruct *isac; /* properties */ }; #define gst_isacdec_parent_class parent_class G_DEFINE_TYPE_WITH_CODE (GstIsacDec, gst_isacdec, GST_TYPE_AUDIO_DECODER, GST_DEBUG_CATEGORY_INIT (isacdec_debug, "isacdec", 0, "debug category for isacdec element")); GST_ELEMENT_REGISTER_DEFINE (isacdec, "isacdec", GST_RANK_PRIMARY, GST_TYPE_ISACDEC); static gboolean gst_isacdec_start (GstAudioDecoder * dec) { GstIsacDec *self = GST_ISACDEC (dec); gint16 ret; g_assert (!self->isac); ret = WebRtcIsac_Create (&self->isac); CHECK_ISAC_RET (ret, Create); return TRUE; } static gboolean gst_isacdec_stop (GstAudioDecoder * dec) { GstIsacDec *self = GST_ISACDEC (dec); if (self->isac) { gint16 ret; ret = WebRtcIsac_Free (self->isac); CHECK_ISAC_RET (ret, Free); self->isac = NULL; } return TRUE; } static gboolean gst_isacdec_set_format (GstAudioDecoder * dec, GstCaps * input_caps) { GstIsacDec *self = GST_ISACDEC (dec); GstAudioInfo output_format; gint16 ret; gboolean result; GstStructure *s; gint rate, channels; GstCaps *output_caps; GST_DEBUG_OBJECT (self, "input caps: %" GST_PTR_FORMAT, input_caps); s = gst_caps_get_structure (input_caps, 0); if (!s) return FALSE; if (!gst_structure_get_int (s, "rate", &rate)) { GST_ERROR_OBJECT (self, "'rate' missing in input caps: %" GST_PTR_FORMAT, input_caps); return FALSE; } if (!gst_structure_get_int (s, "channels", &channels)) { GST_ERROR_OBJECT (self, "'channels' missing in input caps: %" GST_PTR_FORMAT, input_caps); return FALSE; } gst_audio_info_set_format (&output_format, GST_AUDIO_FORMAT_S16LE, rate, channels, NULL); output_caps = gst_audio_info_to_caps (&output_format); GST_DEBUG_OBJECT (self, "output caps: %" GST_PTR_FORMAT, output_caps); gst_caps_unref (output_caps); ret = WebRtcIsac_SetDecSampRate (self->isac, rate); CHECK_ISAC_RET (ret, SetDecSampleRate); WebRtcIsac_DecoderInit (self->isac); result = gst_audio_decoder_set_output_format (dec, &output_format); gst_audio_decoder_set_plc_aware (dec, TRUE); return result; } static GstFlowReturn gst_isacdec_plc (GstIsacDec * self, GstClockTime duration) { GstAudioDecoder *dec = GST_AUDIO_DECODER (self); guint nb_plc_frames; GstBuffer *output; GstMapInfo map_write; size_t ret; /* Decoder produces 30 ms PLC frames */ nb_plc_frames = duration / (30 * GST_MSECOND); GST_DEBUG_OBJECT (self, "GAP of %" GST_TIME_FORMAT " detected, request PLC for %d frames", GST_TIME_ARGS (duration), nb_plc_frames); output = gst_audio_decoder_allocate_output_buffer (dec, nb_plc_frames * MAX_OUTPUT_SIZE); if (!gst_buffer_map (output, &map_write, GST_MAP_WRITE)) { GST_ERROR_OBJECT (self, "Failed to map output buffer"); gst_buffer_unref (output); return GST_FLOW_ERROR; } ret = WebRtcIsac_DecodePlc (self->isac, (gint16 *) map_write.data, nb_plc_frames); gst_buffer_unmap (output, &map_write); if (ret < 0) { /* error */ gint16 code = WebRtcIsac_GetErrorCode (self->isac); GST_WARNING_OBJECT (self, "Failed to produce PLC: %s (%d)", isac_error_code_to_str (code), code); gst_buffer_unref (output); return GST_FLOW_ERROR; } else if (ret == 0) { GST_DEBUG_OBJECT (self, "Decoder didn't produce any PLC frame"); gst_buffer_unref (output); return GST_FLOW_OK; } gst_buffer_set_size (output, ret * SAMPLE_SIZE); GST_LOG_OBJECT (self, "Produced %" G_GSIZE_FORMAT " PLC samples", ret); return gst_audio_decoder_finish_frame (dec, output, 1); } static GstFlowReturn gst_isacdec_handle_frame (GstAudioDecoder * dec, GstBuffer * input) { GstIsacDec *self = GST_ISACDEC (dec); GstMapInfo map_read, map_write; GstBuffer *output; gint16 ret, speech_type[1]; gsize input_size; /* Can't drain the decoder */ if (!input) return GST_FLOW_OK; if (!gst_buffer_get_size (input)) { /* Base class detected a gap in the stream, try to do PLC */ return gst_isacdec_plc (self, GST_BUFFER_DURATION (input)); } if (!gst_buffer_map (input, &map_read, GST_MAP_READ)) { GST_ELEMENT_ERROR (self, RESOURCE, READ, ("Failed to map input buffer"), (NULL)); return GST_FLOW_ERROR; } input_size = map_read.size; output = gst_audio_decoder_allocate_output_buffer (dec, MAX_OUTPUT_SIZE); if (!gst_buffer_map (output, &map_write, GST_MAP_WRITE)) { GST_ELEMENT_ERROR (self, RESOURCE, WRITE, ("Failed to map output buffer"), (NULL)); gst_buffer_unref (output); gst_buffer_unmap (input, &map_read); return GST_FLOW_ERROR; } ret = WebRtcIsac_Decode (self->isac, map_read.data, map_read.size, (gint16 *) map_write.data, speech_type); gst_buffer_unmap (input, &map_read); gst_buffer_unmap (output, &map_write); if (ret < 0) { /* error */ gint16 code = WebRtcIsac_GetErrorCode (self->isac); GST_WARNING_OBJECT (self, "Failed to decode: %s (%d)", isac_error_code_to_str (code), code); gst_buffer_unref (output); /* Give a chance to decode next frames */ return GST_FLOW_OK; } else if (ret == 0) { GST_DEBUG_OBJECT (self, "Decoder didn't produce any frame"); gst_buffer_unref (output); output = NULL; } else { gst_buffer_set_size (output, ret * SAMPLE_SIZE); } GST_LOG_OBJECT (self, "Decoded %d samples from %" G_GSIZE_FORMAT " bytes", ret, input_size); return gst_audio_decoder_finish_frame (dec, output, 1); } static void gst_isacdec_class_init (GstIsacDecClass * klass) { GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass); GstAudioDecoderClass *base_class = GST_AUDIO_DECODER_CLASS (klass); base_class->start = GST_DEBUG_FUNCPTR (gst_isacdec_start); base_class->stop = GST_DEBUG_FUNCPTR (gst_isacdec_stop); base_class->set_format = GST_DEBUG_FUNCPTR (gst_isacdec_set_format); base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_isacdec_handle_frame); gst_element_class_set_static_metadata (gstelement_class, "iSAC decoder", "Codec/Decoder/Audio", "iSAC audio decoder", "Guillaume Desmottes "); gst_element_class_add_static_pad_template (gstelement_class, &sink_template); gst_element_class_add_static_pad_template (gstelement_class, &src_template); } static void gst_isacdec_init (GstIsacDec * self) { self->isac = NULL; }