X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Fqdm2.c;h=6acb7d8362527e2a8ee023b49de363139042add6;hb=64fe3eaeb351582787cbef75a2fe160253663363;hp=33d4824e7b9fa91062ab5caa354ed4d28e67c0cb;hpb=72415b2adb2c25f95ceede49001bb97ed9247dbb;p=ffmpeg diff --git a/libavcodec/qdm2.c b/libavcodec/qdm2.c index 33d4824e7b9..6acb7d83625 100644 --- a/libavcodec/qdm2.c +++ b/libavcodec/qdm2.c @@ -5,27 +5,28 @@ * Copyright (c) 2005 Alex Beregszaszi * Copyright (c) 2005 Roberto Togni * - * This file is part of FFmpeg. + * This file is part of Libav. * - * FFmpeg is free software; you can redistribute it and/or + * Libav 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, + * Libav 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 + * License along with Libav; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ /** - * @file libavcodec/qdm2.c + * @file * QDM2 decoder * @author Ewald Snel, Benjamin Larsson, Alex Beregszaszi, Roberto Togni + * * The decoder is not perfect yet, there are still some distortions * especially on files encoded with 16 or 8 subbands. */ @@ -34,11 +35,12 @@ #include #include -#define ALT_BITSTREAM_READER_LE +#define BITSTREAM_READER_LE #include "avcodec.h" #include "get_bits.h" #include "dsputil.h" -#include "fft.h" +#include "rdft.h" +#include "mpegaudiodsp.h" #include "mpegaudio.h" #include "qdm2data.h" @@ -75,6 +77,7 @@ do { \ #define SAMPLES_NEEDED_2(why) \ av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why); +#define QDM2_MAX_FRAME_SIZE 512 typedef int8_t sb_int8_array[2][30][64]; @@ -120,13 +123,15 @@ typedef struct { } FFTCoefficient; typedef struct { - DECLARE_ALIGNED(16, QDM2Complex, complex)[MPA_MAX_CHANNELS][256]; + DECLARE_ALIGNED(32, QDM2Complex, complex)[MPA_MAX_CHANNELS][256]; } QDM2FFT; /** * QDM2 decoder context */ typedef struct { + AVFrame frame; + /// Parameters from codec header, do not change during playback int nb_channels; ///< number of channels int channels; ///< number of channels @@ -167,12 +172,14 @@ typedef struct { /// I/O data const uint8_t *compressed_data; int compressed_size; - float output_buffer[1024]; + float output_buffer[QDM2_MAX_FRAME_SIZE * 2]; /// Synthesis filter - DECLARE_ALIGNED(16, MPA_INT, synth_buf)[MPA_MAX_CHANNELS][512*2]; + MPADSPContext mpadsp; + DECLARE_ALIGNED(32, float, synth_buf)[MPA_MAX_CHANNELS][512*2]; int synth_buf_offset[MPA_MAX_CHANNELS]; - DECLARE_ALIGNED(16, int32_t, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT]; + DECLARE_ALIGNED(32, float, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT]; + DECLARE_ALIGNED(32, float, samples)[MPA_MAX_CHANNELS * MPA_FRAME_SIZE]; /// Mixed temporary data used in decoding float tone_level[MPA_MAX_CHANNELS][30][64]; @@ -329,11 +336,6 @@ static av_cold void qdm2_init_vlc(void) } } - -/* for floating point to fixed point conversion */ -static const float f2i_scale = (float) (1 << (FRAC_BITS - 15)); - - static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth) { int value; @@ -385,7 +387,7 @@ static uint16_t qdm2_packet_checksum (const uint8_t *data, int length, int value /** - * Fills a QDM2SubPacket structure with packet type, size, and data pointer. + * Fill a QDM2SubPacket structure with packet type, size, and data pointer. * * @param gb bitreader context * @param sub_packet packet under analysis @@ -436,7 +438,7 @@ static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int /** - * Replaces 8 elements with their average value. + * Replace 8 elements with their average value. * Called by qdm2_decode_superblock before starting subblock decoding. * * @param q context @@ -482,8 +484,8 @@ static void build_sb_samples_from_noise (QDM2Context *q, int sb) for (ch = 0; ch < q->nb_channels; ch++) for (j = 0; j < 64; j++) { - q->sb_samples[ch][j * 2][sb] = (int32_t)(f2i_scale * SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j] + .5); - q->sb_samples[ch][j * 2 + 1][sb] = (int32_t)(f2i_scale * SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j] + .5); + q->sb_samples[ch][j * 2][sb] = SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j]; + q->sb_samples[ch][j * 2 + 1][sb] = SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j]; } } @@ -923,11 +925,11 @@ static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int l for (chs = 0; chs < q->nb_channels; chs++) for (k = 0; k < run; k++) if ((j + k) < 128) - q->sb_samples[chs][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs] + .5); + q->sb_samples[chs][j + k][sb] = q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs]; } else { for (k = 0; k < run; k++) if ((j + k) < 128) - q->sb_samples[ch][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[ch][sb][(j + k)/2] * samples[k] + .5); + q->sb_samples[ch][j + k][sb] = q->tone_level[ch][sb][(j + k)/2] * samples[k]; } j += run; @@ -942,7 +944,6 @@ static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int l * This is similar to process_subpacket_9, but for a single channel and for element [0] * same VLC tables as process_subpacket_9 are used. * - * @param q context * @param quantized_coeffs pointer to quantized_coeffs[ch][0] * @param gb bitreader context * @param length packet length in bits @@ -1210,7 +1211,8 @@ static void qdm2_decode_super_block (QDM2Context *q) init_get_bits(&gb, header.data, header.size*8); if (header.type == 2 || header.type == 4 || header.type == 5) { - int csum = 257 * get_bits(&gb, 8) + 2 * get_bits(&gb, 8); + int csum = 257 * get_bits(&gb, 8); + csum += 2 * get_bits(&gb, 8); csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum); @@ -1355,6 +1357,8 @@ static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext * return; local_int_14 = (offset >> local_int_8); + if (local_int_14 >= FF_ARRAY_ELEMS(fft_level_index_table)) + return; if (q->nb_channels > 1) { channel = get_bits1(gb); @@ -1588,7 +1592,7 @@ static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet) int i; q->fft.complex[channel][0].re *= 2.0f; q->fft.complex[channel][0].im = 0.0f; - ff_rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]); + q->rdft_ctx.rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]); /* add samples to output buffer */ for (i = 0; i < ((q->fft_frame_size + 15) & ~15); i++) q->output_buffer[q->channels * i + channel] += ((float *) q->fft.complex[channel])[i] * gain; @@ -1601,7 +1605,6 @@ static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet) */ static void qdm2_synthesis_filter (QDM2Context *q, int index) { - OUT_INT samples[MPA_MAX_CHANNELS * MPA_FRAME_SIZE]; int i, k, ch, sb_used, sub_sampling, dither_state = 0; /* copy sb_samples */ @@ -1613,11 +1616,12 @@ static void qdm2_synthesis_filter (QDM2Context *q, int index) q->sb_samples[ch][(8 * index) + i][k] = 0; for (ch = 0; ch < q->nb_channels; ch++) { - OUT_INT *samples_ptr = samples + ch; + float *samples_ptr = q->samples + ch; for (i = 0; i < 8; i++) { - ff_mpa_synth_filter(q->synth_buf[ch], &(q->synth_buf_offset[ch]), - ff_mpa_synth_window, &dither_state, + ff_mpa_synth_filter_float(&q->mpadsp, + q->synth_buf[ch], &(q->synth_buf_offset[ch]), + ff_mpa_synth_window_float, &dither_state, samples_ptr, q->nb_channels, q->sb_samples[ch][(8 * index) + i]); samples_ptr += 32 * q->nb_channels; @@ -1629,7 +1633,7 @@ static void qdm2_synthesis_filter (QDM2Context *q, int index) for (ch = 0; ch < q->channels; ch++) for (i = 0; i < q->frame_size; i++) - q->output_buffer[q->channels * i + ch] += (float)(samples[q->nb_channels * sub_sampling * i + ch] >> (sizeof(OUT_INT)*8-16)); + q->output_buffer[q->channels * i + ch] += (1 << 23) * q->samples[q->nb_channels * sub_sampling * i + ch]; } @@ -1646,7 +1650,7 @@ static av_cold void qdm2_init(QDM2Context *q) { initialized = 1; qdm2_init_vlc(); - ff_mpa_synth_init(ff_mpa_synth_window); + ff_mpa_synth_init_float(ff_mpa_synth_window_float); softclip_table_init(); rnd_table_init(); init_noise_samples(); @@ -1799,6 +1803,8 @@ static av_cold int qdm2_decode_init(AVCodecContext *avctx) avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata); extradata += 4; + if (s->channels > MPA_MAX_CHANNELS) + return AVERROR_INVALIDDATA; avctx->sample_rate = AV_RB32(extradata); extradata += 4; @@ -1813,6 +1819,10 @@ static av_cold int qdm2_decode_init(AVCodecContext *avctx) 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; + } s->fft_order = av_log2(s->fft_size) + 1; s->fft_frame_size = 2 * s->fft_size; // complex has two floats @@ -1820,6 +1830,8 @@ static av_cold int qdm2_decode_init(AVCodecContext *avctx) // something like max decodable tones s->group_order = av_log2(s->group_size) + 1; s->frame_size = s->group_size / 16; // 16 iterations per super block + if (s->frame_size > QDM2_MAX_FRAME_SIZE) + return AVERROR_INVALIDDATA; s->sub_sampling = s->fft_order - 7; s->frequency_range = 255 / (1 << (2 - s->sub_sampling)); @@ -1863,10 +1875,14 @@ static av_cold int qdm2_decode_init(AVCodecContext *avctx) } ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R); + ff_mpadsp_init(&s->mpadsp); qdm2_init(s); - avctx->sample_fmt = SAMPLE_FMT_S16; + avctx->sample_fmt = AV_SAMPLE_FMT_S16; + + avcodec_get_frame_defaults(&s->frame); + avctx->coded_frame = &s->frame; // dump_context(s); return 0; @@ -1883,7 +1899,7 @@ static av_cold int qdm2_decode_close(AVCodecContext *avctx) } -static void qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out) +static int qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out) { int ch, i; const int frame_size = (q->frame_size * q->channels); @@ -1919,7 +1935,7 @@ static void qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out) if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) { SAMPLES_NEEDED_2("has errors, and C list is not empty") - return; + return -1; } } @@ -1940,38 +1956,46 @@ static void qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out) out[i] = value; } + + return 0; } -static int qdm2_decode_frame(AVCodecContext *avctx, - void *data, int *data_size, - AVPacket *avpkt) +static int qdm2_decode_frame(AVCodecContext *avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; QDM2Context *s = avctx->priv_data; + int16_t *out; + int i, ret; if(!buf) return 0; if(buf_size < s->checksum_size) return -1; - *data_size = s->channels * s->frame_size * sizeof(int16_t); - - av_log(avctx, AV_LOG_DEBUG, "decode(%d): %p[%d] -> %p[%d]\n", - buf_size, buf, s->checksum_size, data, *data_size); - - qdm2_decode(s, buf, data); + /* get output buffer */ + s->frame.nb_samples = 16 * s->frame_size; + if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; + } + out = (int16_t *)s->frame.data[0]; - // reading only when next superblock found - if (s->sub_packet == 0) { - return s->checksum_size; + for (i = 0; i < 16; i++) { + if (qdm2_decode(s, buf, out) < 0) + return -1; + out += s->channels * s->frame_size; } - return 0; + *got_frame_ptr = 1; + *(AVFrame *)data = s->frame; + + return s->checksum_size; } -AVCodec qdm2_decoder = +AVCodec ff_qdm2_decoder = { .name = "qdm2", .type = AVMEDIA_TYPE_AUDIO, @@ -1980,5 +2004,6 @@ AVCodec qdm2_decoder = .init = qdm2_decode_init, .close = qdm2_decode_close, .decode = qdm2_decode_frame, + .capabilities = CODEC_CAP_DR1, .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"), };