pineapple-src/externals/ffmpeg/libavcodec/s302menc.c
2021-02-09 04:25:58 +01:00

188 lines
7.2 KiB
C
Executable file

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