| /* GStreamer Adaptive Multi-Rate Narrow-Band (AMR-NB) plugin |
| * Copyright (C) 2004 Ronald Bultje <rbultje@ronald.bitfreak.net> |
| * |
| * 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. |
| */ |
| |
| /** |
| * SECTION:element-amrwbdec |
| * @see_also: #GstAmrwbEnc |
| * |
| * AMR wideband decoder based on the |
| * <ulink url="http://sourceforge.net/projects/opencore-amr">opencore codec implementation</ulink>. |
| * |
| * <refsect2> |
| * <title>Example launch line</title> |
| * |[ |
| * gst-launch-1.0 filesrc location=abc.amr ! amrparse ! amrwbdec ! audioconvert ! audioresample ! autoaudiosink |
| * ]| |
| * </refsect2> |
| */ |
| |
| #ifdef HAVE_CONFIG_H |
| #include "config.h" |
| #endif |
| |
| #include <gst/audio/audio.h> |
| |
| #include "amrwbdec.h" |
| |
| static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink", |
| GST_PAD_SINK, |
| GST_PAD_ALWAYS, |
| GST_STATIC_CAPS ("audio/AMR-WB, " |
| "rate = (int) 16000, " "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) ", " |
| "layout = (string) interleaved, " |
| "rate = (int) 16000, " "channels = (int) 1") |
| ); |
| |
| GST_DEBUG_CATEGORY_STATIC (gst_amrwbdec_debug); |
| #define GST_CAT_DEFAULT gst_amrwbdec_debug |
| |
| #define L_FRAME16k 320 /* Frame size at 16kHz */ |
| |
| static const unsigned char block_size[16] = |
| { 18, 24, 33, 37, 41, 47, 51, 59, 61, |
| 6, 0, 0, 0, 0, 1, 1 |
| }; |
| |
| static gboolean gst_amrwbdec_start (GstAudioDecoder * dec); |
| static gboolean gst_amrwbdec_stop (GstAudioDecoder * dec); |
| static gboolean gst_amrwbdec_set_format (GstAudioDecoder * dec, GstCaps * caps); |
| static GstFlowReturn gst_amrwbdec_parse (GstAudioDecoder * dec, |
| GstAdapter * adapter, gint * offset, gint * length); |
| static GstFlowReturn gst_amrwbdec_handle_frame (GstAudioDecoder * dec, |
| GstBuffer * buffer); |
| |
| #define gst_amrwbdec_parent_class parent_class |
| G_DEFINE_TYPE (GstAmrwbDec, gst_amrwbdec, GST_TYPE_AUDIO_DECODER); |
| |
| static void |
| gst_amrwbdec_class_init (GstAmrwbDecClass * klass) |
| { |
| GstAudioDecoderClass *base_class = GST_AUDIO_DECODER_CLASS (klass); |
| GstElementClass *element_class = GST_ELEMENT_CLASS (klass); |
| |
| gst_element_class_add_static_pad_template (element_class, &sink_template); |
| gst_element_class_add_static_pad_template (element_class, &src_template); |
| |
| gst_element_class_set_static_metadata (element_class, "AMR-WB audio decoder", |
| "Codec/Decoder/Audio", |
| "Adaptive Multi-Rate Wideband audio decoder", |
| "Renato Araujo <renato.filho@indt.org.br>"); |
| |
| base_class->start = GST_DEBUG_FUNCPTR (gst_amrwbdec_start); |
| base_class->stop = GST_DEBUG_FUNCPTR (gst_amrwbdec_stop); |
| base_class->set_format = GST_DEBUG_FUNCPTR (gst_amrwbdec_set_format); |
| base_class->parse = GST_DEBUG_FUNCPTR (gst_amrwbdec_parse); |
| base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_amrwbdec_handle_frame); |
| |
| GST_DEBUG_CATEGORY_INIT (gst_amrwbdec_debug, "amrwbdec", 0, |
| "AMR-WB audio decoder"); |
| } |
| |
| static void |
| gst_amrwbdec_init (GstAmrwbDec * amrwbdec) |
| { |
| gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (amrwbdec), TRUE); |
| gst_audio_decoder_set_use_default_pad_acceptcaps (GST_AUDIO_DECODER_CAST |
| (amrwbdec), TRUE); |
| GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_DECODER_SINK_PAD (amrwbdec)); |
| } |
| |
| static gboolean |
| gst_amrwbdec_start (GstAudioDecoder * dec) |
| { |
| GstAmrwbDec *amrwbdec = GST_AMRWBDEC (dec); |
| |
| GST_DEBUG_OBJECT (dec, "start"); |
| if (!(amrwbdec->handle = D_IF_init ())) |
| return FALSE; |
| |
| amrwbdec->rate = 0; |
| amrwbdec->channels = 0; |
| |
| return TRUE; |
| } |
| |
| static gboolean |
| gst_amrwbdec_stop (GstAudioDecoder * dec) |
| { |
| GstAmrwbDec *amrwbdec = GST_AMRWBDEC (dec); |
| |
| GST_DEBUG_OBJECT (dec, "stop"); |
| D_IF_exit (amrwbdec->handle); |
| |
| return TRUE; |
| } |
| |
| static gboolean |
| gst_amrwbdec_set_format (GstAudioDecoder * dec, GstCaps * caps) |
| { |
| GstStructure *structure; |
| GstAmrwbDec *amrwbdec; |
| GstAudioInfo info; |
| |
| amrwbdec = GST_AMRWBDEC (dec); |
| |
| structure = gst_caps_get_structure (caps, 0); |
| |
| /* get channel count */ |
| gst_structure_get_int (structure, "channels", &amrwbdec->channels); |
| gst_structure_get_int (structure, "rate", &amrwbdec->rate); |
| |
| /* create reverse caps */ |
| gst_audio_info_init (&info); |
| gst_audio_info_set_format (&info, |
| GST_AUDIO_FORMAT_S16, amrwbdec->rate, amrwbdec->channels, NULL); |
| |
| gst_audio_decoder_set_output_format (dec, &info); |
| |
| return TRUE; |
| } |
| |
| static GstFlowReturn |
| gst_amrwbdec_parse (GstAudioDecoder * dec, GstAdapter * adapter, |
| gint * offset, gint * length) |
| { |
| GstAmrwbDec *amrwbdec = GST_AMRWBDEC (dec); |
| guint8 header[1]; |
| guint size; |
| gboolean sync, eos; |
| gint block, mode; |
| |
| size = gst_adapter_available (adapter); |
| if (size < 1) |
| return GST_FLOW_ERROR; |
| |
| gst_audio_decoder_get_parse_state (dec, &sync, &eos); |
| |
| /* need to peek data to get the size */ |
| gst_adapter_copy (adapter, header, 0, 1); |
| mode = (header[0] >> 3) & 0x0F; |
| block = block_size[mode]; |
| |
| GST_DEBUG_OBJECT (amrwbdec, "mode %d, block %d", mode, block); |
| |
| if (block) { |
| if (block > size) |
| return GST_FLOW_EOS; |
| *offset = 0; |
| *length = block; |
| } else { |
| /* no frame yet, skip one byte */ |
| GST_LOG_OBJECT (amrwbdec, "skipping byte"); |
| *offset = 1; |
| return GST_FLOW_EOS; |
| } |
| |
| return GST_FLOW_OK; |
| } |
| |
| static GstFlowReturn |
| gst_amrwbdec_handle_frame (GstAudioDecoder * dec, GstBuffer * buffer) |
| { |
| GstAmrwbDec *amrwbdec; |
| GstBuffer *out; |
| GstMapInfo inmap, outmap; |
| |
| amrwbdec = GST_AMRWBDEC (dec); |
| |
| /* no fancy flushing */ |
| if (!buffer || !gst_buffer_get_size (buffer)) |
| return GST_FLOW_OK; |
| |
| /* the library seems to write into the source data, hence the copy. */ |
| /* should be no problem */ |
| gst_buffer_map (buffer, &inmap, GST_MAP_READ); |
| |
| /* get output */ |
| out = gst_buffer_new_and_alloc (sizeof (gint16) * L_FRAME16k); |
| gst_buffer_map (out, &outmap, GST_MAP_WRITE); |
| |
| /* decode */ |
| D_IF_decode (amrwbdec->handle, (unsigned char *) inmap.data, |
| (short int *) outmap.data, _good_frame); |
| |
| gst_buffer_unmap (out, &outmap); |
| gst_buffer_unmap (buffer, &inmap); |
| |
| /* send out */ |
| return gst_audio_decoder_finish_frame (dec, out, 1); |
| } |