mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-12-23 12:43:46 +02:00
189 lines
7.2 KiB
C
189 lines
7.2 KiB
C
/*
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* SMPTE 302M encoder
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* Copyright (c) 2010 Google, Inc.
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* Copyright (c) 2013 Darryl Wallace <wallacdj@gmail.com>
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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 "avcodec.h"
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#include "internal.h"
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#include "mathops.h"
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#include "put_bits.h"
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#define AES3_HEADER_LEN 4
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typedef struct S302MEncContext {
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uint8_t framing_index; /* Set for even channels on multiple of 192 samples */
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} S302MEncContext;
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static av_cold int s302m_encode_init(AVCodecContext *avctx)
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{
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S302MEncContext *s = avctx->priv_data;
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if (avctx->channels & 1 || avctx->channels > 8) {
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av_log(avctx, AV_LOG_ERROR,
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"Encoding %d channel(s) is not allowed. Only 2, 4, 6 and 8 channels are supported.\n",
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avctx->channels);
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return AVERROR(EINVAL);
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}
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switch (avctx->sample_fmt) {
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case AV_SAMPLE_FMT_S16:
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avctx->bits_per_raw_sample = 16;
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break;
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case AV_SAMPLE_FMT_S32:
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if (avctx->bits_per_raw_sample > 20) {
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if (avctx->bits_per_raw_sample > 24)
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av_log(avctx, AV_LOG_WARNING, "encoding as 24 bits-per-sample\n");
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avctx->bits_per_raw_sample = 24;
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} else if (!avctx->bits_per_raw_sample) {
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avctx->bits_per_raw_sample = 24;
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} else if (avctx->bits_per_raw_sample <= 20) {
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avctx->bits_per_raw_sample = 20;
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}
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}
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avctx->frame_size = 0;
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avctx->bit_rate = 48000 * avctx->channels *
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(avctx->bits_per_raw_sample + 4);
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s->framing_index = 0;
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return 0;
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}
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static int s302m_encode2_frame(AVCodecContext *avctx, AVPacket *avpkt,
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const AVFrame *frame, int *got_packet_ptr)
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{
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S302MEncContext *s = avctx->priv_data;
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const int buf_size = AES3_HEADER_LEN +
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(frame->nb_samples *
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avctx->channels *
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(avctx->bits_per_raw_sample + 4)) / 8;
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int ret, c, channels;
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uint8_t *o;
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PutBitContext pb;
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if (buf_size - AES3_HEADER_LEN > UINT16_MAX) {
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av_log(avctx, AV_LOG_ERROR, "number of samples in frame too big\n");
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return AVERROR(EINVAL);
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}
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if ((ret = ff_alloc_packet2(avctx, avpkt, buf_size, 0)) < 0)
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return ret;
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o = avpkt->data;
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init_put_bits(&pb, o, buf_size);
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put_bits(&pb, 16, buf_size - AES3_HEADER_LEN);
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put_bits(&pb, 2, (avctx->channels - 2) >> 1); // number of channels
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put_bits(&pb, 8, 0); // channel ID
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put_bits(&pb, 2, (avctx->bits_per_raw_sample - 16) / 4); // bits per samples (0 = 16bit, 1 = 20bit, 2 = 24bit)
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put_bits(&pb, 4, 0); // alignments
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flush_put_bits(&pb);
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o += AES3_HEADER_LEN;
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if (avctx->bits_per_raw_sample == 24) {
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const uint32_t *samples = (uint32_t *)frame->data[0];
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for (c = 0; c < frame->nb_samples; c++) {
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uint8_t vucf = s->framing_index == 0 ? 0x10: 0;
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for (channels = 0; channels < avctx->channels; channels += 2) {
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o[0] = ff_reverse[(samples[0] & 0x0000FF00) >> 8];
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o[1] = ff_reverse[(samples[0] & 0x00FF0000) >> 16];
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o[2] = ff_reverse[(samples[0] & 0xFF000000) >> 24];
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o[3] = ff_reverse[(samples[1] & 0x00000F00) >> 4] | vucf;
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o[4] = ff_reverse[(samples[1] & 0x000FF000) >> 12];
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o[5] = ff_reverse[(samples[1] & 0x0FF00000) >> 20];
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o[6] = ff_reverse[(samples[1] & 0xF0000000) >> 28];
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o += 7;
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samples += 2;
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}
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s->framing_index++;
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if (s->framing_index >= 192)
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s->framing_index = 0;
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}
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} else if (avctx->bits_per_raw_sample == 20) {
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const uint32_t *samples = (uint32_t *)frame->data[0];
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for (c = 0; c < frame->nb_samples; c++) {
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uint8_t vucf = s->framing_index == 0 ? 0x80: 0;
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for (channels = 0; channels < avctx->channels; channels += 2) {
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o[0] = ff_reverse[ (samples[0] & 0x000FF000) >> 12];
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o[1] = ff_reverse[ (samples[0] & 0x0FF00000) >> 20];
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o[2] = ff_reverse[((samples[0] & 0xF0000000) >> 28) | vucf];
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o[3] = ff_reverse[ (samples[1] & 0x000FF000) >> 12];
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o[4] = ff_reverse[ (samples[1] & 0x0FF00000) >> 20];
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o[5] = ff_reverse[ (samples[1] & 0xF0000000) >> 28];
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o += 6;
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samples += 2;
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}
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s->framing_index++;
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if (s->framing_index >= 192)
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s->framing_index = 0;
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}
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} else if (avctx->bits_per_raw_sample == 16) {
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const uint16_t *samples = (uint16_t *)frame->data[0];
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for (c = 0; c < frame->nb_samples; c++) {
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uint8_t vucf = s->framing_index == 0 ? 0x10 : 0;
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for (channels = 0; channels < avctx->channels; channels += 2) {
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o[0] = ff_reverse[ samples[0] & 0xFF];
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o[1] = ff_reverse[(samples[0] & 0xFF00) >> 8];
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o[2] = ff_reverse[(samples[1] & 0x0F) << 4] | vucf;
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o[3] = ff_reverse[(samples[1] & 0x0FF0) >> 4];
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o[4] = ff_reverse[(samples[1] & 0xF000) >> 12];
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o += 5;
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samples += 2;
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}
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s->framing_index++;
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if (s->framing_index >= 192)
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s->framing_index = 0;
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}
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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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AVCodec ff_s302m_encoder = {
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.name = "s302m",
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.long_name = NULL_IF_CONFIG_SMALL("SMPTE 302M"),
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.type = AVMEDIA_TYPE_AUDIO,
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.id = AV_CODEC_ID_S302M,
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.priv_data_size = sizeof(S302MEncContext),
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.init = s302m_encode_init,
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.encode2 = s302m_encode2_frame,
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.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S32,
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AV_SAMPLE_FMT_S16,
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AV_SAMPLE_FMT_NONE },
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.capabilities = AV_CODEC_CAP_VARIABLE_FRAME_SIZE | AV_CODEC_CAP_EXPERIMENTAL,
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.supported_samplerates = (const int[]) { 48000, 0 },
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/* .channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_STEREO,
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AV_CH_LAYOUT_QUAD,
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AV_CH_LAYOUT_5POINT1_BACK,
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AV_CH_LAYOUT_5POINT1_BACK | AV_CH_LAYOUT_STEREO_DOWNMIX,
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0 }, */
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};
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