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f095391a14
* qatar/master: (31 commits) cdxl demux: do not create packets with uninitialized data at EOF. Replace computations of remaining bits with calls to get_bits_left(). amrnb/amrwb: Remove get_bits usage. cosmetics: reindent avformat: do not require a pixel/sample format if there is no decoder avformat: do not fill-in audio packet duration in compute_pkt_fields() lavf: Use av_get_audio_frame_duration() in get_audio_frame_size() dca_parser: parse the sample rate and frame durations libspeexdec: do not set AVCodecContext.frame_size libopencore-amr: do not set AVCodecContext.frame_size alsdec: do not set AVCodecContext.frame_size siff: do not set AVCodecContext.frame_size amr demuxer: do not set AVCodecContext.frame_size. aiffdec: do not set AVCodecContext.frame_size mov: do not set AVCodecContext.frame_size ape: do not set AVCodecContext.frame_size. rdt: remove workaround for infinite loop with aac avformat: do not require frame_size in avformat_find_stream_info() for CELT avformat: do not require frame_size in avformat_find_stream_info() for MP1/2/3 avformat: do not require frame_size in avformat_find_stream_info() for AAC ... Conflicts: doc/APIchanges libavcodec/Makefile libavcodec/avcodec.h libavcodec/h264.c libavcodec/h264_ps.c libavcodec/utils.c libavcodec/version.h libavcodec/x86/dsputil_mmx.c libavformat/utils.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
172 lines
5.2 KiB
C
172 lines
5.2 KiB
C
/*
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* Copyright (C) 2008 David Conrad
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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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#include <speex/speex.h>
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#include <speex/speex_header.h>
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#include <speex/speex_stereo.h>
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#include <speex/speex_callbacks.h>
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#include "avcodec.h"
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typedef struct {
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AVFrame frame;
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SpeexBits bits;
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SpeexStereoState stereo;
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void *dec_state;
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SpeexHeader *header;
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int frame_size;
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} LibSpeexContext;
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static av_cold int libspeex_decode_init(AVCodecContext *avctx)
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{
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LibSpeexContext *s = avctx->priv_data;
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const SpeexMode *mode;
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// defaults in the case of a missing header
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if (avctx->sample_rate <= 8000)
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mode = &speex_nb_mode;
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else if (avctx->sample_rate <= 16000)
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mode = &speex_wb_mode;
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else
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mode = &speex_uwb_mode;
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if (avctx->extradata_size >= 80)
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s->header = speex_packet_to_header(avctx->extradata, avctx->extradata_size);
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avctx->sample_fmt = AV_SAMPLE_FMT_S16;
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if (s->header) {
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avctx->sample_rate = s->header->rate;
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avctx->channels = s->header->nb_channels;
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mode = speex_lib_get_mode(s->header->mode);
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if (!mode) {
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av_log(avctx, AV_LOG_ERROR, "Unknown Speex mode %d", s->header->mode);
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return AVERROR_INVALIDDATA;
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}
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} else
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av_log(avctx, AV_LOG_INFO, "Missing Speex header, assuming defaults.\n");
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if (avctx->channels > 2) {
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av_log(avctx, AV_LOG_ERROR, "Only stereo and mono are supported.\n");
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return AVERROR(EINVAL);
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}
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speex_bits_init(&s->bits);
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s->dec_state = speex_decoder_init(mode);
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if (!s->dec_state) {
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av_log(avctx, AV_LOG_ERROR, "Error initializing libspeex decoder.\n");
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return -1;
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}
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if (!s->header) {
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speex_decoder_ctl(s->dec_state, SPEEX_GET_FRAME_SIZE, &s->frame_size);
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}
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if (avctx->channels == 2) {
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SpeexCallback callback;
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callback.callback_id = SPEEX_INBAND_STEREO;
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callback.func = speex_std_stereo_request_handler;
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callback.data = &s->stereo;
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s->stereo = (SpeexStereoState)SPEEX_STEREO_STATE_INIT;
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speex_decoder_ctl(s->dec_state, SPEEX_SET_HANDLER, &callback);
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}
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avcodec_get_frame_defaults(&s->frame);
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avctx->coded_frame = &s->frame;
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return 0;
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}
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static int libspeex_decode_frame(AVCodecContext *avctx, void *data,
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int *got_frame_ptr, AVPacket *avpkt)
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{
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uint8_t *buf = avpkt->data;
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int buf_size = avpkt->size;
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LibSpeexContext *s = avctx->priv_data;
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int16_t *output;
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int ret, consumed = 0;
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/* get output buffer */
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s->frame.nb_samples = s->frame_size;
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if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
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av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
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return ret;
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}
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output = (int16_t *)s->frame.data[0];
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/* if there is not enough data left for the smallest possible frame,
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reset the libspeex buffer using the current packet, otherwise ignore
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the current packet and keep decoding frames from the libspeex buffer. */
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if (speex_bits_remaining(&s->bits) < 43) {
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/* check for flush packet */
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if (!buf || !buf_size) {
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*got_frame_ptr = 0;
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return buf_size;
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}
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/* set new buffer */
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speex_bits_read_from(&s->bits, buf, buf_size);
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consumed = buf_size;
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}
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/* decode a single frame */
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ret = speex_decode_int(s->dec_state, &s->bits, output);
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if (ret <= -2) {
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av_log(avctx, AV_LOG_ERROR, "Error decoding Speex frame.\n");
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return AVERROR_INVALIDDATA;
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}
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if (avctx->channels == 2)
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speex_decode_stereo_int(output, s->frame_size, &s->stereo);
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*got_frame_ptr = 1;
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*(AVFrame *)data = s->frame;
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return consumed;
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}
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static av_cold int libspeex_decode_close(AVCodecContext *avctx)
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{
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LibSpeexContext *s = avctx->priv_data;
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speex_header_free(s->header);
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speex_bits_destroy(&s->bits);
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speex_decoder_destroy(s->dec_state);
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return 0;
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}
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static av_cold void libspeex_decode_flush(AVCodecContext *avctx)
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{
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LibSpeexContext *s = avctx->priv_data;
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speex_bits_reset(&s->bits);
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}
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AVCodec ff_libspeex_decoder = {
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.name = "libspeex",
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.type = AVMEDIA_TYPE_AUDIO,
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.id = CODEC_ID_SPEEX,
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.priv_data_size = sizeof(LibSpeexContext),
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.init = libspeex_decode_init,
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.close = libspeex_decode_close,
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.decode = libspeex_decode_frame,
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.flush = libspeex_decode_flush,
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.capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DELAY | CODEC_CAP_DR1,
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.long_name = NULL_IF_CONFIG_SMALL("libspeex Speex"),
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};
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