/* * Direct Stream Digital (DSD) encoder * * This file is part of FFmpeg. * * FFmpeg 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. * * FFmpeg 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 FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ /** * @file * DSD (Direct Stream Digital) "encoder": AV_SAMPLE_FMT_DSD is * already the bitstream, so packets are a plain copy of the samples. */ #include #include "avcodec.h" #include "codec_internal.h" #include "encode.h" static av_cold int dsd_encode_init(AVCodecContext *avctx) { avctx->bits_per_coded_sample = 8; avctx->block_align = avctx->ch_layout.nb_channels; avctx->bit_rate = 8LL * avctx->block_align * avctx->sample_rate; return 0; } static int dsd_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr) { int64_t size = frame->nb_samples * (int64_t)avctx->ch_layout.nb_channels; int ret; if ((ret = ff_get_encode_buffer(avctx, avpkt, size, 0)) < 0) return ret; memcpy(avpkt->data, frame->data[0], size); *got_packet_ptr = 1; return 0; } const FFCodec ff_dsd_msbf_encoder = { .p.name = "dsd_msbf", CODEC_LONG_NAME("DSD (Direct Stream Digital), most significant bit first"), .p.type = AVMEDIA_TYPE_AUDIO, .p.id = AV_CODEC_ID_DSD_MSBF, .p.capabilities = AV_CODEC_CAP_VARIABLE_FRAME_SIZE, CODEC_SAMPLEFMTS(AV_SAMPLE_FMT_DSD), .init = dsd_encode_init, FF_CODEC_ENCODE_CB(dsd_encode_frame), };