| /* |
| * Copyright (c) 2012 Laurent Aimar |
| * |
| * 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 "libavutil/intreadwrite.h" |
| #include "avcodec.h" |
| #include "internal.h" |
| #include "dvaudio.h" |
| |
| typedef struct DVAudioContext { |
| int block_size; |
| int is_12bit; |
| int is_pal; |
| int16_t shuffle[2000]; |
| } DVAudioContext; |
| |
| static av_cold int decode_init(AVCodecContext *avctx) |
| { |
| DVAudioContext *s = avctx->priv_data; |
| int i; |
| |
| if (avctx->channels != 2) { |
| av_log(avctx, AV_LOG_ERROR, "invalid number of channels\n"); |
| return AVERROR(EINVAL); |
| } |
| |
| if (avctx->codec_tag == 0x0215) { |
| s->block_size = 7200; |
| } else if (avctx->codec_tag == 0x0216) { |
| s->block_size = 8640; |
| } else if (avctx->block_align == 7200 || |
| avctx->block_align == 8640) { |
| s->block_size = avctx->block_align; |
| } else { |
| return AVERROR(EINVAL); |
| } |
| |
| s->is_pal = s->block_size == 8640; |
| s->is_12bit = avctx->bits_per_raw_sample == 12; |
| avctx->sample_fmt = AV_SAMPLE_FMT_S16; |
| avctx->channel_layout = AV_CH_LAYOUT_STEREO; |
| |
| for (i = 0; i < FF_ARRAY_ELEMS(s->shuffle); i++) { |
| const unsigned a = s->is_pal ? 18 : 15; |
| const unsigned b = 3 * a; |
| |
| s->shuffle[i] = 80 * ((21 * (i % 3) + 9 * (i / 3) + ((i / a) % 3)) % b) + |
| (2 + s->is_12bit) * (i / b) + 8; |
| } |
| |
| return 0; |
| } |
| |
| static inline uint16_t dv_audio_12to16(uint16_t sample) |
| { |
| uint16_t shift, result; |
| |
| sample = (sample < 0x800) ? sample : sample | 0xf000; |
| shift = (sample & 0xf00) >> 8; |
| |
| if (shift < 0x2 || shift > 0xd) { |
| result = sample; |
| } else if (shift < 0x8) { |
| shift--; |
| result = (sample - (256 * shift)) << shift; |
| } else { |
| shift = 0xe - shift; |
| result = ((sample + ((256 * shift) + 1)) << shift) - 1; |
| } |
| |
| return result; |
| } |
| |
| static int decode_frame(AVCodecContext *avctx, void *data, |
| int *got_frame_ptr, AVPacket *pkt) |
| { |
| DVAudioContext *s = avctx->priv_data; |
| AVFrame *frame = data; |
| const uint8_t *src = pkt->data; |
| int16_t *dst; |
| int ret, i; |
| |
| if (pkt->size < s->block_size) |
| return AVERROR_INVALIDDATA; |
| |
| frame->nb_samples = dv_get_audio_sample_count(pkt->data + 244, s->is_pal); |
| if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) |
| return ret; |
| dst = (int16_t *)frame->data[0]; |
| |
| for (i = 0; i < frame->nb_samples; i++) { |
| const uint8_t *v = &src[s->shuffle[i]]; |
| |
| if (s->is_12bit) { |
| *dst++ = dv_audio_12to16((v[0] << 4) | ((v[2] >> 4) & 0x0f)); |
| *dst++ = dv_audio_12to16((v[1] << 4) | ((v[2] >> 0) & 0x0f)); |
| } else { |
| *dst++ = AV_RB16(&v[0]); |
| *dst++ = AV_RB16(&v[s->is_pal ? 4320 : 3600]); |
| } |
| } |
| |
| *got_frame_ptr = 1; |
| |
| return s->block_size; |
| } |
| |
| AVCodec ff_dvaudio_decoder = { |
| .name = "dvaudio", |
| .long_name = NULL_IF_CONFIG_SMALL("Ulead DV Audio"), |
| .type = AVMEDIA_TYPE_AUDIO, |
| .id = AV_CODEC_ID_DVAUDIO, |
| .init = decode_init, |
| .decode = decode_frame, |
| .capabilities = AV_CODEC_CAP_DR1, |
| .priv_data_size = sizeof(DVAudioContext), |
| }; |