diff options
Diffstat (limited to 'src/audio_dec/gsm610.c')
-rw-r--r-- | src/audio_dec/gsm610.c | 281 |
1 files changed, 281 insertions, 0 deletions
diff --git a/src/audio_dec/gsm610.c b/src/audio_dec/gsm610.c new file mode 100644 index 000000000..be98da798 --- /dev/null +++ b/src/audio_dec/gsm610.c @@ -0,0 +1,281 @@ +/* + * Copyright (C) 2000-2003 the xine project + * + * This file is part of xine, a free video player. + * + * xine is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * xine 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 General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110, USA + * + * GSM 6.10 Audio Decoder + * This decoder is based on the GSM 6.10 codec library found at: + * http://kbs.cs.tu-berlin.de/~jutta/toast.html + * Additionally, here is an article regarding the software that appeared + * in Dr. Dobbs Journal: + * http://www.ddj.com/documents/s=1012/ddj9412b/9412b.htm + * + * This is the notice that comes with the software: + * -------------------------------------------------------------------- + * Copyright 1992, 1993, 1994 by Jutta Degener and Carsten Bormann, + * Technische Universitaet Berlin + * + * Any use of this software is permitted provided that this notice is not + * removed and that neither the authors nor the Technische Universitaet Berlin + * are deemed to have made any representations as to the suitability of this + * software for any purpose nor are held responsible for any defects of + * this software. THERE IS ABSOLUTELY NO WARRANTY FOR THIS SOFTWARE. + * + * As a matter of courtesy, the authors request to be informed about uses + * this software has found, about bugs in this software, and about any + * improvements that may be of general interest. + * + * Berlin, 28.11.1994 + * Jutta Degener + * Carsten Bormann + * -------------------------------------------------------------------- + */ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include <stdlib.h> +#include <string.h> + +#include <xine/xine_internal.h> +#include <xine/audio_out.h> +#include <xine/buffer.h> +#include <xine/xineutils.h> +#include "bswap.h" + +#include "private.h" +#include "gsm.h" + +#define AUDIOBUFSIZE 128*1024 + +#define GSM610_SAMPLE_SIZE 16 +#define GSM610_BLOCK_SIZE 160 + +typedef struct { + audio_decoder_class_t decoder_class; +} gsm610_class_t; + +typedef struct gsm610_decoder_s { + audio_decoder_t audio_decoder; + + xine_stream_t *stream; + + unsigned int buf_type; + int output_open; + int sample_rate; + + unsigned char *buf; + int bufsize; + int size; + + gsm gsm_state; + +} gsm610_decoder_t; + +/************************************************************************** + * xine audio plugin functions + *************************************************************************/ + +static void gsm610_decode_data (audio_decoder_t *this_gen, buf_element_t *buf) { + + gsm610_decoder_t *this = (gsm610_decoder_t *) this_gen; + audio_buffer_t *audio_buffer; + int in_ptr; + + if (buf->decoder_flags & BUF_FLAG_STDHEADER) { + this->sample_rate = buf->decoder_info[1]; + + this->buf = calloc(1, AUDIOBUFSIZE); + this->bufsize = AUDIOBUFSIZE; + this->size = 0; + + /* stream/meta info */ + _x_meta_info_set_utf8(this->stream, XINE_META_INFO_AUDIOCODEC, "GSM 6.10"); + + return; + } + + if (!this->output_open) { + + this->gsm_state = gsm_create(); + this->buf_type = buf->type; + + this->output_open = (this->stream->audio_out->open) (this->stream->audio_out, + this->stream, GSM610_SAMPLE_SIZE, this->sample_rate, AO_CAP_MODE_MONO); + } + + /* if the audio still isn't open, bail */ + if (!this->output_open) + return; + + if( this->size + buf->size > this->bufsize ) { + this->bufsize = this->size + 2 * buf->size; + xprintf(this->stream->xine, XINE_VERBOSITY_DEBUG, + "gsm610: increasing source buffer to %d to avoid overflow.\n", this->bufsize); + this->buf = realloc( this->buf, this->bufsize ); + } + + xine_fast_memcpy (&this->buf[this->size], buf->content, buf->size); + this->size += buf->size; + + if (buf->decoder_flags & BUF_FLAG_FRAME_END) { /* time to decode a frame */ + int16_t decode_buffer[GSM610_BLOCK_SIZE]; + + /* handle the Microsoft variant of GSM data */ + if (this->buf_type == BUF_AUDIO_MSGSM) { + + this->gsm_state->wav_fmt = 1; + + /* the data should line up on a 65-byte boundary */ + if ((buf->size % 65) != 0) { + xprintf (this->stream->xine, XINE_VERBOSITY_DEBUG, + "gsm610: received MS GSM block that does not line up\n"); + this->size = 0; + return; + } + + in_ptr = 0; + while (this->size) { + gsm_decode(this->gsm_state, &this->buf[in_ptr], decode_buffer); + if ((in_ptr % 65) == 0) { + in_ptr += 33; + this->size -= 33; + } else { + in_ptr += 32; + this->size -= 32; + } + + /* dispatch the decoded audio; assume that the audio buffer will + * always contain at least 160 samples */ + audio_buffer = this->stream->audio_out->get_buffer (this->stream->audio_out); + + xine_fast_memcpy(audio_buffer->mem, decode_buffer, + GSM610_BLOCK_SIZE * 2); + audio_buffer->num_frames = GSM610_BLOCK_SIZE; + + audio_buffer->vpts = buf->pts; + buf->pts = 0; /* only first buffer gets the real pts */ + this->stream->audio_out->put_buffer (this->stream->audio_out, audio_buffer, this->stream); + } + } else { + + /* handle the other variant, which consists of 33-byte blocks */ + this->gsm_state->wav_fmt = 0; + + /* the data should line up on a 33-byte boundary */ + if ((buf->size % 33) != 0) { + xprintf (this->stream->xine, XINE_VERBOSITY_DEBUG, "gsm610: received GSM block that does not line up\n"); + this->size = 0; + return; + } + + in_ptr = 0; + while (this->size) { + gsm_decode(this->gsm_state, &this->buf[in_ptr], decode_buffer); + in_ptr += 33; + this->size -= 33; + + /* dispatch the decoded audio; assume that the audio buffer will + * always contain at least 160 samples */ + audio_buffer = this->stream->audio_out->get_buffer (this->stream->audio_out); + + xine_fast_memcpy(audio_buffer->mem, decode_buffer, + GSM610_BLOCK_SIZE * 2); + audio_buffer->num_frames = GSM610_BLOCK_SIZE; + + audio_buffer->vpts = buf->pts; + buf->pts = 0; /* only first buffer gets the real pts */ + this->stream->audio_out->put_buffer (this->stream->audio_out, audio_buffer, this->stream); + } + } + } +} + +static void gsm610_reset (audio_decoder_t *this_gen) { +} + +static void gsm610_discontinuity (audio_decoder_t *this_gen) { +} + +static void gsm610_dispose (audio_decoder_t *this_gen) { + + gsm610_decoder_t *this = (gsm610_decoder_t *) this_gen; + + if (this->gsm_state) + gsm_destroy(this->gsm_state); + + if (this->output_open) + this->stream->audio_out->close (this->stream->audio_out, this->stream); + this->output_open = 0; + + if (this->buf) + free(this->buf); + + free (this_gen); +} + +static audio_decoder_t *open_plugin (audio_decoder_class_t *class_gen, xine_stream_t *stream) { + + gsm610_decoder_t *this ; + + this = (gsm610_decoder_t *) calloc(1, sizeof(gsm610_decoder_t)); + + this->audio_decoder.decode_data = gsm610_decode_data; + this->audio_decoder.reset = gsm610_reset; + this->audio_decoder.discontinuity = gsm610_discontinuity; + this->audio_decoder.dispose = gsm610_dispose; + + this->output_open = 0; + this->sample_rate = 0; + this->stream = stream; + this->buf = NULL; + this->size = 0; + + return &this->audio_decoder; +} + +static void *init_plugin (xine_t *xine, void *data) { + + gsm610_class_t *this ; + + this = (gsm610_class_t *) calloc(1, sizeof(gsm610_class_t)); + + this->decoder_class.open_plugin = open_plugin; + this->decoder_class.identifier = "GSM 6.10"; + this->decoder_class.description = N_("GSM 6.10 audio decoder plugin"); + this->decoder_class.dispose = default_audio_decoder_class_dispose; + + return this; +} + +static const uint32_t audio_types[] = { + BUF_AUDIO_MSGSM, + BUF_AUDIO_GSM610, + 0 +}; + +static const decoder_info_t dec_info_audio = { + audio_types, /* supported types */ + 9 /* priority */ +}; + +const plugin_info_t xine_plugin_info[] EXPORTED = { + /* type, API, "name", version, special_info, init_function */ + { PLUGIN_AUDIO_DECODER, 16, "gsm610", XINE_VERSION_CODE, &dec_info_audio, init_plugin }, + { PLUGIN_NONE, 0, "", 0, NULL, NULL } +}; |