| /* GStreamer SBC audio decoder |
| * |
| * Copyright (C) 2004-2010 Marcel Holtmann <marcel@holtmann.org> |
| * Copyright (C) 2013 Tim-Philipp Müller <tim centricular net> |
| * |
| * 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-sbdec |
| * @title: sbdec |
| * |
| * This element decodes a Bluetooth SBC audio streams to raw integer PCM audio |
| * |
| * ## Example pipelines |
| * |[ |
| * gst-launch-1.0 -v filesrc location=audio.sbc ! sbcparse ! sbcdec ! audioconvert ! audioresample ! autoaudiosink |
| * ]| Decode a raw SBC file. |
| * |
| */ |
| |
| #ifdef HAVE_CONFIG_H |
| #include <config.h> |
| #endif |
| |
| #include <string.h> |
| |
| #include "gstsbcdec.h" |
| |
| /* FIXME: where does this come from? how is it derived? */ |
| #define BUF_SIZE 8192 |
| |
| GST_DEBUG_CATEGORY_STATIC (sbc_dec_debug); |
| #define GST_CAT_DEFAULT sbc_dec_debug |
| |
| #define parent_class gst_sbc_dec_parent_class |
| G_DEFINE_TYPE (GstSbcDec, gst_sbc_dec, GST_TYPE_AUDIO_DECODER); |
| |
| static GstStaticPadTemplate sbc_dec_sink_factory = |
| GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, |
| GST_STATIC_CAPS ("audio/x-sbc, channels = (int) [ 1, 2 ], " |
| "rate = (int) { 16000, 32000, 44100, 48000 }, " |
| "parsed = (boolean) true")); |
| |
| static GstStaticPadTemplate sbc_dec_src_factory = |
| GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, |
| GST_STATIC_CAPS ("audio/x-raw, format=" GST_AUDIO_NE (S16) ", " |
| "rate = (int) { 16000, 32000, 44100, 48000 }, " |
| "channels = (int) [ 1, 2 ], layout=interleaved")); |
| |
| static GstFlowReturn |
| gst_sbc_dec_handle_frame (GstAudioDecoder * audio_dec, GstBuffer * buf) |
| { |
| GstSbcDec *dec = GST_SBC_DEC (audio_dec); |
| GstBuffer *outbuf = NULL; |
| GstMapInfo out_map; |
| GstMapInfo in_map; |
| gsize output_size; |
| guint num_frames, i; |
| |
| /* no fancy draining */ |
| if (G_UNLIKELY (buf == NULL)) |
| return GST_FLOW_OK; |
| |
| gst_buffer_map (buf, &in_map, GST_MAP_READ); |
| |
| if (G_UNLIKELY (in_map.size == 0)) |
| goto done; |
| |
| /* we assume all frames are of the same size, this is implied by the |
| * input caps applying to the whole input buffer, and the parser should |
| * also have made sure of that */ |
| if (G_UNLIKELY (in_map.size % dec->frame_len != 0)) |
| goto mixed_frames; |
| |
| num_frames = in_map.size / dec->frame_len; |
| output_size = num_frames * dec->samples_per_frame * sizeof (gint16); |
| |
| outbuf = gst_audio_decoder_allocate_output_buffer (audio_dec, output_size); |
| |
| if (outbuf == NULL) |
| goto no_buffer; |
| |
| gst_buffer_map (outbuf, &out_map, GST_MAP_WRITE); |
| |
| for (i = 0; i < num_frames; ++i) { |
| gssize ret; |
| gsize written; |
| |
| ret = sbc_decode (&dec->sbc, in_map.data + (i * dec->frame_len), |
| dec->frame_len, out_map.data + (i * dec->samples_per_frame * 2), |
| dec->samples_per_frame * 2, &written); |
| |
| if (ret <= 0 || written != (dec->samples_per_frame * 2)) { |
| GST_WARNING_OBJECT (dec, "decoding error, ret = %" G_GSSIZE_FORMAT ", " |
| "written = %" G_GSSIZE_FORMAT, ret, written); |
| break; |
| } |
| } |
| |
| gst_buffer_unmap (outbuf, &out_map); |
| |
| if (i > 0) |
| gst_buffer_set_size (outbuf, i * dec->samples_per_frame * 2); |
| else |
| gst_buffer_replace (&outbuf, NULL); |
| |
| done: |
| |
| gst_buffer_unmap (buf, &in_map); |
| |
| return gst_audio_decoder_finish_frame (audio_dec, outbuf, 1); |
| |
| /* ERRORS */ |
| mixed_frames: |
| { |
| GST_WARNING_OBJECT (dec, "inconsistent input data/frames, skipping"); |
| goto done; |
| } |
| no_buffer: |
| { |
| GST_ERROR_OBJECT (dec, "could not allocate output buffer"); |
| goto done; |
| } |
| } |
| |
| static gboolean |
| gst_sbc_dec_set_format (GstAudioDecoder * audio_dec, GstCaps * caps) |
| { |
| GstSbcDec *dec = GST_SBC_DEC (audio_dec); |
| const gchar *channel_mode; |
| GstAudioInfo info; |
| GstStructure *s; |
| gint channels, rate, subbands, blocks, bitpool; |
| |
| s = gst_caps_get_structure (caps, 0); |
| gst_structure_get_int (s, "channels", &channels); |
| gst_structure_get_int (s, "rate", &rate); |
| |
| /* save input format */ |
| channel_mode = gst_structure_get_string (s, "channel-mode"); |
| if (channel_mode == NULL || |
| !gst_structure_get_int (s, "subbands", &subbands) || |
| !gst_structure_get_int (s, "blocks", &blocks) || |
| !gst_structure_get_int (s, "bitpool", &bitpool)) |
| return FALSE; |
| |
| if (strcmp (channel_mode, "mono") == 0) { |
| dec->frame_len = 4 + (subbands * 1) / 2 + ((blocks * 1 * bitpool) + 7) / 8; |
| } else if (strcmp (channel_mode, "dual") == 0) { |
| dec->frame_len = 4 + (subbands * 2) / 2 + ((blocks * 2 * bitpool) + 7) / 8; |
| } else if (strcmp (channel_mode, "stereo") == 0) { |
| dec->frame_len = 4 + (subbands * 2) / 2 + ((blocks * bitpool) + 7) / 8; |
| } else if (strcmp (channel_mode, "joint") == 0) { |
| dec->frame_len = |
| 4 + (subbands * 2) / 2 + ((subbands + blocks * bitpool) + 7) / 8; |
| } else { |
| return FALSE; |
| } |
| |
| dec->samples_per_frame = channels * blocks * subbands; |
| |
| GST_INFO_OBJECT (dec, "frame len: %" G_GSIZE_FORMAT ", samples per frame " |
| "%" G_GSIZE_FORMAT, dec->frame_len, dec->samples_per_frame); |
| |
| /* set up output format */ |
| gst_audio_info_init (&info); |
| gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_S16, rate, channels, NULL); |
| gst_audio_decoder_set_output_format (audio_dec, &info); |
| |
| return TRUE; |
| } |
| |
| static gboolean |
| gst_sbc_dec_start (GstAudioDecoder * dec) |
| { |
| GstSbcDec *sbcdec = GST_SBC_DEC (dec); |
| |
| GST_INFO_OBJECT (dec, "Setup subband codec"); |
| sbc_init (&sbcdec->sbc, 0); |
| |
| return TRUE; |
| } |
| |
| static gboolean |
| gst_sbc_dec_stop (GstAudioDecoder * dec) |
| { |
| GstSbcDec *sbcdec = GST_SBC_DEC (dec); |
| |
| GST_INFO_OBJECT (sbcdec, "Finish subband codec"); |
| sbc_finish (&sbcdec->sbc); |
| sbcdec->samples_per_frame = 0; |
| sbcdec->frame_len = 0; |
| |
| return TRUE; |
| } |
| |
| static void |
| gst_sbc_dec_class_init (GstSbcDecClass * klass) |
| { |
| GstAudioDecoderClass *audio_decoder_class = (GstAudioDecoderClass *) klass; |
| GstElementClass *element_class = (GstElementClass *) klass; |
| |
| audio_decoder_class->start = GST_DEBUG_FUNCPTR (gst_sbc_dec_start); |
| audio_decoder_class->stop = GST_DEBUG_FUNCPTR (gst_sbc_dec_stop); |
| audio_decoder_class->set_format = GST_DEBUG_FUNCPTR (gst_sbc_dec_set_format); |
| audio_decoder_class->handle_frame = |
| GST_DEBUG_FUNCPTR (gst_sbc_dec_handle_frame); |
| |
| gst_element_class_add_static_pad_template (element_class, |
| &sbc_dec_sink_factory); |
| gst_element_class_add_static_pad_template (element_class, |
| &sbc_dec_src_factory); |
| |
| gst_element_class_set_static_metadata (element_class, |
| "Bluetooth SBC audio decoder", "Codec/Decoder/Audio", |
| "Decode an SBC audio stream", "Marcel Holtmann <marcel@holtmann.org>"); |
| |
| GST_DEBUG_CATEGORY_INIT (sbc_dec_debug, "sbcdec", 0, "SBC decoding element"); |
| } |
| |
| static void |
| gst_sbc_dec_init (GstSbcDec * dec) |
| { |
| gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (dec), TRUE); |
| gst_audio_decoder_set_use_default_pad_acceptcaps (GST_AUDIO_DECODER_CAST |
| (dec), TRUE); |
| GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_DECODER_SINK_PAD (dec)); |
| |
| dec->samples_per_frame = 0; |
| dec->frame_len = 0; |
| } |