On 1/5/2017 3:34 PM, Paul B Mahol wrote:
> diff --git a/libavcodec/qdmc.c b/libavcodec/qdmc.c
> new file mode 100644
> index 0000000..5559db3
> --- /dev/null
> +++ b/libavcodec/qdmc.c
> @@ -0,0 +1,817 @@
> +/*
> + * QDMC compatible decoder
> + * Copyright (c) 2017 Paul B Mahol
> + *
> + * 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 <math.h>
> +#include <stddef.h>
> +#include <stdio.h>
> +
> +#define BITSTREAM_READER_LE
> +
> +#include "libavutil/channel_layout.h"
> +
> +#include "avcodec.h"
> +#include "get_bits.h"
> +#include "internal.h"
> +#include "fft.h"
> +
> +typedef struct QDMCTone {
> + uint8_t mode;
> + uint8_t phase;
> + uint8_t offset;
> + int16_t freq;
> + int16_t amplitude;
> +} QDMCTone;
> +
> +typedef struct QDMCContext {
> + AVCodecContext *avctx;
> +
> + uint8_t frame_bits;
> + int band_index;
> + int frame_size;
> + int subframe_size;
> + int fft_offset;
> + int buffer_offset;
> + float *buffer_ptr;
> + int nb_channels;
> +
> + int group_size;
> + int checksum_size;
> +
> + uint8_t noise[2][19][16];
> + QDMCTone tones[5][8192];
> + int nb_tones[5];
> + int cur_tone[5];
> + float alt_sin[5][31];
> + float fft_buffer[4][8192 * 2];
> + float noise2_buffer[4096 * 2];
> + float noise_buffer[4096 * 2];
> + int rndval;
> +
> + DECLARE_ALIGNED(32, FFTComplex, cmplx)[2][512];
> + float buffer[2 * 32768];
> +
> + FFTContext fft_ctx;
> +} QDMCContext;
> +
> +static float sin_table[512];
> +static VLC vtable[6];
Why are these not part of QDMCContext?
[...]
> +
> +static av_cold int qdmc_init_static_data(QDMCContext *s)
> +{
> + static int done;
> + int i, ret;
> +
> + if (done)
> + return 0;
> +
> + ret = ff_init_vlc_sparse(&vtable[0], 12,
> FF_ARRAY_ELEMS(noise_value_bits),
> + noise_value_bits, 1, 1, noise_value_codes, 2,
> 2, noise_value_symbols, 1, 1, INIT_VLC_LE);
> + if (ret < 0)
> + return ret;
> + ret = ff_init_vlc_sparse(&vtable[1], 10,
> FF_ARRAY_ELEMS(noise_segment_length_bits),
> + noise_segment_length_bits, 1, 1,
> noise_segment_length_codes, 2, 2,
> + noise_segment_length_symbols, 1, 1,
> INIT_VLC_LE);
> + if (ret < 0)
> + return ret;
> + ret = ff_init_vlc_sparse(&vtable[2], 13, FF_ARRAY_ELEMS(amplitude_bits),
> + amplitude_bits, 1, 1, amplitude_codes, 2, 2,
> NULL, 0, 0, INIT_VLC_LE);
> + if (ret < 0)
> + return ret;
> + ret = ff_init_vlc_sparse(&vtable[3], 18, FF_ARRAY_ELEMS(freq_diff_bits),
> + freq_diff_bits, 1, 1, freq_diff_codes, 4, 4,
> NULL, 0, 0, INIT_VLC_LE);
> + if (ret < 0)
> + return ret;
> + ret = ff_init_vlc_sparse(&vtable[4], 8,
> FF_ARRAY_ELEMS(amplitude_diff_bits),
> + amplitude_diff_bits, 1, 1,
> amplitude_diff_codes, 1, 1, NULL, 0, 0, INIT_VLC_LE);
> + if (ret < 0)
> + return ret;
> + ret = ff_init_vlc_sparse(&vtable[5], 6, FF_ARRAY_ELEMS(phase_diff_bits),
> + phase_diff_bits, 1, 1, phase_diff_codes, 1, 1,
> NULL, 0, 0, INIT_VLC_LE);
> + if (ret < 0)
> + return ret;
> +
> + for (i = 0; i < 512; i++)
> + sin_table[i] = sin(2 * i * M_PI * 0.001953125);
These constants should be float.
> +
> + done = 1;
> +
> + return 0;
> +}
[...]
> +static av_cold int qdmc_decode_init(AVCodecContext *avctx)
> +{
> + QDMCContext *s = avctx->priv_data;
> + uint8_t *extradata;
> + int extradata_size, fft_size, fft_order, ret, size, g, j, x;
> +
> + if ((ret = qdmc_init_static_data(s)) < 0)
> + return ret;
> +
> + if (!avctx->extradata || (avctx->extradata_size < 48)) {
> + av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
> + return AVERROR_INVALIDDATA;
> + }
> +
> + extradata = avctx->extradata;
> + extradata_size = avctx->extradata_size;
> +
> + while (extradata_size > 8) {
> + if (!memcmp(extradata, "frmaQDMC", 8))
> + break;
> + extradata++;
> + extradata_size--;
Use bytestream.h instead. It will simplify this function a lot.
> + }
> +
> + if (extradata_size < 12) {
> + av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
> + extradata_size);
> + return AVERROR_INVALIDDATA;
> + }
> +
> + if (memcmp(extradata, "frmaQDMC", 8)) {
> + av_log(avctx, AV_LOG_ERROR, "invalid headers, QDMC not found\n");
> + return AVERROR_INVALIDDATA;
> + }
> +
> + extradata += 8;
> + extradata_size -= 8;
> +
> + size = AV_RB32(extradata);
> + extradata += 4;
> +
> + if (size > extradata_size) {
> + av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
> + extradata_size, size);
> + return AVERROR_INVALIDDATA;
> + }
> +
> + if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) {
> + av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
> + return AVERROR_INVALIDDATA;
> + }
> + extradata += 8;
> +
> + avctx->channels = s->nb_channels = AV_RB32(extradata);
> + extradata += 4;
> + if (s->nb_channels <= 0 || s->nb_channels > 2) {
> + av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");
> + return AVERROR_INVALIDDATA;
> + }
> + avctx->channel_layout = avctx->channels == 2 ? AV_CH_LAYOUT_STEREO :
> + AV_CH_LAYOUT_MONO;
> +
> + avctx->sample_rate = AV_RB32(extradata);
> + extradata += 4;
> +
> + avctx->bit_rate = AV_RB32(extradata);
> + extradata += 4;
> +
> + s->group_size = AV_RB32(extradata);
> + extradata += 4;
> +
> + fft_size = AV_RB32(extradata);
> + fft_order = av_log2(fft_size) + 1;
> + extradata += 4;
> +
> + s->checksum_size = AV_RB32(extradata);
> + if (s->checksum_size >= 1U << 28) {
> + av_log(avctx, AV_LOG_ERROR, "data block size too large (%u)\n",
> s->checksum_size);
> + return AVERROR_INVALIDDATA;
> + }
> +
> + if (avctx->sample_rate >= 32000) {
> + x = 28000;
> + s->frame_bits = 13;
> + } else if (avctx->sample_rate >= 16000) {
> + x = 20000;
> + s->frame_bits = 12;
> + } else {
> + x = 16000;
> + s->frame_bits = 11;
> + }
> + s->frame_size = 1 << s->frame_bits;
> + s->subframe_size = s->frame_size >> 5;
> +
> + if (avctx->channels == 2)
> + x = 3 * x / 2;
> + s->band_index = noise_bands_selector[FFMIN(6,
> llrint(floor(avctx->bit_rate * 3.0 / (double)x + 0.5)))];
> +
> + if ((fft_order < 7) || (fft_order > 9)) {
> + avpriv_request_sample(avctx, "Unknown FFT order %d", fft_order);
> + return AVERROR_PATCHWELCOME;
> + }
> +
> + if (fft_size != (1 << (fft_order - 1))) {
> + av_log(avctx, AV_LOG_ERROR, "FFT size %d not power of 2.\n",
> fft_size);
> + return AVERROR_INVALIDDATA;
> + }
> +
> + ff_fft_init(&s->fft_ctx, fft_order, 1);
> +
> + avctx->sample_fmt = AV_SAMPLE_FMT_S16;
> +
> + for (g = 5; g > 0; g--) {
> + for (j = 0; j < (1 << g) - 1; j++)
> + s->alt_sin[5-g][j] = sin_table[(((j+1) << (8 - g)) & 0x1FF)];
> + }
> +
> + make_noises(s);
> +
> + return 0;
> +}
[...]
> +static void add_noise(QDMCContext *s, int ch, int current_subframe)
> +{
> + int i, j, aindex;
> + float amplitude;
> + float *im = &s->fft_buffer[0 + ch][s->fft_offset + s->subframe_size *
> current_subframe];
> + float *re = &s->fft_buffer[2 + ch][s->fft_offset + s->subframe_size *
> current_subframe];
> +
> + memset(s->noise2_buffer, 0, 4 * s->subframe_size);
> +
> + for (i = 0; i < noise_bands_size[s->band_index]; i++) {
> + if (qdmc_nodes[i + 21 * s->band_index] > s->subframe_size - 1)
> + break;
> +
> + aindex = s->noise[ch][i][current_subframe/2];
> + amplitude = 0.0;
> + if (aindex > 0)
> + amplitude = real_amp(aindex);
> +
> + lin_calc(s, amplitude, qdmc_nodes[21 * s->band_index + i],
> + qdmc_nodes[21 * s->band_index + i + 2], i);
> + }
> +
> + for (j = 2; j < s->subframe_size - 1; j++) {
> + float rnd_re, rnd_im;
> +
> + s->rndval = 214013 * s->rndval + 2531011;
> + rnd_im = ((s->rndval & 0x7FFF) - 16384.0) * 0.000030517578 *
> s->noise2_buffer[j];
> + s->rndval = 214013 * s->rndval + 2531011;
> + rnd_re = ((s->rndval & 0x7FFF) - 16384.0) * 0.000030517578 *
> s->noise2_buffer[j];
Also float.
_______________________________________________
ffmpeg-devel mailing list
ffmpeg-devel@ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
