mirror of
https://github.com/FFmpeg/FFmpeg.git
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1a4315f24d
* commit '8e97a8c69162afce47abea96c8c0914f3550e212':
build: Remove check for gsm/gsm.h for libgsm
Also includes fix from 010baac12a
Merged-by: James Almer <jamrial@gmail.com>
139 lines
4.4 KiB
C
139 lines
4.4 KiB
C
/*
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* Interface to libgsm for GSM encoding
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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
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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 <gsm.h>
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#include "libavutil/common.h"
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#include "avcodec.h"
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#include "internal.h"
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#include "gsm.h"
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static av_cold int libgsm_encode_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 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 %"PRId64"bps\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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if (!avctx->priv_data)
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goto error;
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switch(avctx->codec_id) {
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case AV_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 AV_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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error:
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libgsm_encode_close(avctx);
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return -1;
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}
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static int libgsm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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const AVFrame *frame, int *got_packet_ptr)
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{
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int ret;
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gsm_signal *samples = (gsm_signal *)frame->data[0];
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struct gsm_state *state = avctx->priv_data;
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if ((ret = ff_alloc_packet2(avctx, avpkt, avctx->block_align, 0)) < 0)
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return ret;
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switch(avctx->codec_id) {
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case AV_CODEC_ID_GSM:
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gsm_encode(state, samples, avpkt->data);
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break;
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case AV_CODEC_ID_GSM_MS:
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gsm_encode(state, samples, avpkt->data);
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gsm_encode(state, samples + GSM_FRAME_SIZE, avpkt->data + 32);
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}
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*got_packet_ptr = 1;
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return 0;
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}
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#if CONFIG_LIBGSM_ENCODER
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AVCodec ff_libgsm_encoder = {
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.name = "libgsm",
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.long_name = NULL_IF_CONFIG_SMALL("libgsm GSM"),
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.type = AVMEDIA_TYPE_AUDIO,
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.id = AV_CODEC_ID_GSM,
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.init = libgsm_encode_init,
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.encode2 = libgsm_encode_frame,
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.close = libgsm_encode_close,
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.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
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AV_SAMPLE_FMT_NONE },
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};
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#endif
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#if CONFIG_LIBGSM_MS_ENCODER
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AVCodec ff_libgsm_ms_encoder = {
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.name = "libgsm_ms",
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.long_name = NULL_IF_CONFIG_SMALL("libgsm GSM Microsoft variant"),
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.type = AVMEDIA_TYPE_AUDIO,
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.id = AV_CODEC_ID_GSM_MS,
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.init = libgsm_encode_init,
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.encode2 = libgsm_encode_frame,
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.close = libgsm_encode_close,
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.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
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AV_SAMPLE_FMT_NONE },
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};
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#endif
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