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![Michael Niedermayer](/assets/img/avatar_default.png)
* qatar/master: (44 commits) replacement Indeo 3 decoder gsm demuxer: do not allocate packet twice. flvenc: use first packet delay as global delay. ac3enc: doxygen update. imc: return error codes instead of 0 for error conditions. imc: return meaningful error codes instead of -1 imc: do not set channel layout for stereo imc: validate channel count imc: check for ff_fft_init() failure imc: check output buffer size before decoding imc: use DSPContext.bswap16_buf() to byte-swap packet data rtsp: add allowed_media_types option libgsm: add flush function to reset the decoder state when seeking libgsm: simplify decoding by using a loop gsm: log error message when packet is too small libgsmdec: do not needlessly set *data_size to 0 gsmdec: do not needlessly set *data_size to 0 gsmdec: add flush function to reset the decoder state when seeking libgsmdec: check output buffer size before decoding gsmdec: log error message when output buffer is too small. ... Conflicts: Changelog ffplay.c libavcodec/indeo3.c libavcodec/mjpeg_parser.c libavcodec/vp3.c libavformat/cutils.c libavformat/id3v2.c libavutil/parseutils.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
223 lines
6.9 KiB
C
223 lines
6.9 KiB
C
/*
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* Interface to libgsm for gsm encoding/decoding
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* Copyright (c) 2005 Alban Bedel <albeu@free.fr>
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* Copyright (c) 2006, 2007 Michel Bardiaux <mbardiaux@mediaxim.be>
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg 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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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* Interface to libgsm for gsm encoding/decoding
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*/
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// The idiosyncrasies of GSM-in-WAV are explained at http://kbs.cs.tu-berlin.de/~jutta/toast.html
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#include "avcodec.h"
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#include <gsm/gsm.h>
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// gsm.h misses some essential constants
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#define GSM_BLOCK_SIZE 33
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#define GSM_MS_BLOCK_SIZE 65
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#define GSM_FRAME_SIZE 160
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static av_cold int libgsm_encode_init(AVCodecContext *avctx) {
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if (avctx->channels > 1) {
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av_log(avctx, AV_LOG_ERROR, "Mono required for GSM, got %d channels\n",
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avctx->channels);
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return -1;
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}
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if (avctx->sample_rate != 8000) {
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av_log(avctx, AV_LOG_ERROR, "Sample rate 8000Hz required for GSM, got %dHz\n",
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avctx->sample_rate);
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if (avctx->strict_std_compliance > FF_COMPLIANCE_UNOFFICIAL)
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return -1;
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}
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if (avctx->bit_rate != 13000 /* Official */ &&
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avctx->bit_rate != 13200 /* Very common */ &&
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avctx->bit_rate != 0 /* Unknown; a.o. mov does not set bitrate when decoding */ ) {
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av_log(avctx, AV_LOG_ERROR, "Bitrate 13000bps required for GSM, got %dbps\n",
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avctx->bit_rate);
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if (avctx->strict_std_compliance > FF_COMPLIANCE_UNOFFICIAL)
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return -1;
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}
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avctx->priv_data = gsm_create();
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switch(avctx->codec_id) {
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case CODEC_ID_GSM:
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avctx->frame_size = GSM_FRAME_SIZE;
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avctx->block_align = GSM_BLOCK_SIZE;
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break;
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case CODEC_ID_GSM_MS: {
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int one = 1;
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gsm_option(avctx->priv_data, GSM_OPT_WAV49, &one);
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avctx->frame_size = 2*GSM_FRAME_SIZE;
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avctx->block_align = GSM_MS_BLOCK_SIZE;
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}
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}
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avctx->coded_frame= avcodec_alloc_frame();
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avctx->coded_frame->key_frame= 1;
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return 0;
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}
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static av_cold int libgsm_encode_close(AVCodecContext *avctx) {
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av_freep(&avctx->coded_frame);
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gsm_destroy(avctx->priv_data);
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avctx->priv_data = NULL;
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return 0;
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}
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static int libgsm_encode_frame(AVCodecContext *avctx,
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unsigned char *frame, int buf_size, void *data) {
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// we need a full block
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if(buf_size < avctx->block_align) return 0;
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switch(avctx->codec_id) {
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case CODEC_ID_GSM:
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gsm_encode(avctx->priv_data,data,frame);
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break;
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case CODEC_ID_GSM_MS:
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gsm_encode(avctx->priv_data,data,frame);
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gsm_encode(avctx->priv_data,((short*)data)+GSM_FRAME_SIZE,frame+32);
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}
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return avctx->block_align;
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}
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AVCodec ff_libgsm_encoder = {
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.name = "libgsm",
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.type = AVMEDIA_TYPE_AUDIO,
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.id = CODEC_ID_GSM,
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.init = libgsm_encode_init,
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.encode = libgsm_encode_frame,
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.close = libgsm_encode_close,
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.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
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.long_name = NULL_IF_CONFIG_SMALL("libgsm GSM"),
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};
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AVCodec ff_libgsm_ms_encoder = {
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.name = "libgsm_ms",
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.type = AVMEDIA_TYPE_AUDIO,
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.id = CODEC_ID_GSM_MS,
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.init = libgsm_encode_init,
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.encode = libgsm_encode_frame,
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.close = libgsm_encode_close,
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.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
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.long_name = NULL_IF_CONFIG_SMALL("libgsm GSM Microsoft variant"),
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};
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static av_cold int libgsm_decode_init(AVCodecContext *avctx) {
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if (avctx->channels > 1) {
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av_log(avctx, AV_LOG_ERROR, "Mono required for GSM, got %d channels\n",
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avctx->channels);
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return -1;
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}
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if (!avctx->channels)
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avctx->channels = 1;
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if (!avctx->sample_rate)
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avctx->sample_rate = 8000;
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avctx->sample_fmt = AV_SAMPLE_FMT_S16;
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avctx->priv_data = gsm_create();
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switch(avctx->codec_id) {
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case CODEC_ID_GSM:
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avctx->frame_size = GSM_FRAME_SIZE;
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avctx->block_align = GSM_BLOCK_SIZE;
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break;
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case CODEC_ID_GSM_MS: {
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int one = 1;
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gsm_option(avctx->priv_data, GSM_OPT_WAV49, &one);
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avctx->frame_size = 2 * GSM_FRAME_SIZE;
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avctx->block_align = GSM_MS_BLOCK_SIZE;
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}
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}
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return 0;
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}
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static av_cold int libgsm_decode_close(AVCodecContext *avctx) {
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gsm_destroy(avctx->priv_data);
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avctx->priv_data = NULL;
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return 0;
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}
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static int libgsm_decode_frame(AVCodecContext *avctx,
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void *data, int *data_size,
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AVPacket *avpkt) {
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int i, ret;
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struct gsm_state *s = avctx->priv_data;
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uint8_t *buf = avpkt->data;
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int buf_size = avpkt->size;
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int16_t *samples = data;
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int out_size = avctx->frame_size * av_get_bytes_per_sample(avctx->sample_fmt);
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if (*data_size < out_size) {
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av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n");
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return AVERROR(EINVAL);
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}
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if (buf_size < avctx->block_align) {
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av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
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return AVERROR_INVALIDDATA;
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}
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for (i = 0; i < avctx->frame_size / GSM_FRAME_SIZE; i++) {
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if ((ret = gsm_decode(s, buf, samples)) < 0)
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return -1;
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buf += GSM_BLOCK_SIZE;
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samples += GSM_FRAME_SIZE;
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}
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*data_size = out_size;
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return avctx->block_align;
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}
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static void libgsm_flush(AVCodecContext *avctx) {
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gsm_destroy(avctx->priv_data);
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avctx->priv_data = gsm_create();
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}
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AVCodec ff_libgsm_decoder = {
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.name = "libgsm",
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.type = AVMEDIA_TYPE_AUDIO,
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.id = CODEC_ID_GSM,
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.init = libgsm_decode_init,
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.close = libgsm_decode_close,
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.decode = libgsm_decode_frame,
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.flush = libgsm_flush,
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.long_name = NULL_IF_CONFIG_SMALL("libgsm GSM"),
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};
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AVCodec ff_libgsm_ms_decoder = {
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.name = "libgsm_ms",
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.type = AVMEDIA_TYPE_AUDIO,
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.id = CODEC_ID_GSM_MS,
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.init = libgsm_decode_init,
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.close = libgsm_decode_close,
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.decode = libgsm_decode_frame,
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.flush = libgsm_flush,
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.long_name = NULL_IF_CONFIG_SMALL("libgsm GSM Microsoft variant"),
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};
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