* Copyright (C) 2006 Benjamin Larsson
* Copyright (C) 2007 Konstantin Shishkov
*
- * 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
*/
#include "libavutil/common.h"
#include "libavutil/intmath.h"
#include "libavutil/intreadwrite.h"
-#include "libavcore/audioconvert.h"
+#include "libavutil/audioconvert.h"
#include "avcodec.h"
#include "dsputil.h"
#include "fft.h"
#include "dca.h"
#include "synth_filter.h"
#include "dcadsp.h"
+#include "fmtconvert.h"
//#define TRACE
DCA_EXT_EXSS_XLL = 0x200, ///< lossless extension in ExSS
};
+/* -1 are reserved or unknown */
+static const int dca_ext_audio_descr_mask[] = {
+ DCA_EXT_XCH,
+ -1,
+ DCA_EXT_X96,
+ DCA_EXT_XCH | DCA_EXT_X96,
+ -1,
+ -1,
+ DCA_EXT_XXCH,
+ -1,
+};
+
+/* extensions that reside in core substream */
+#define DCA_CORE_EXTS (DCA_EXT_XCH | DCA_EXT_XXCH | DCA_EXT_X96)
+
/* Tables for mapping dts channel configurations to libavcodec multichannel api.
* Some compromises have been made for special configurations. Most configurations
* are never used so complete accuracy is not needed.
* L = left, R = right, C = center, S = surround, F = front, R = rear, T = total, OV = overhead.
* S -> side, when both rear and back are configured move one of them to the side channel
* OV -> center back
- * All 2 channel configurations -> CH_LAYOUT_STEREO
+ * All 2 channel configurations -> AV_CH_LAYOUT_STEREO
*/
static const int64_t dca_core_channel_layout[] = {
AV_CH_LAYOUT_STEREO, ///< 2, LT +RT (left and right total)
AV_CH_LAYOUT_STEREO|AV_CH_FRONT_CENTER, ///< 3, C+L+R
AV_CH_LAYOUT_STEREO|AV_CH_BACK_CENTER, ///< 3, L+R+S
- AV_CH_LAYOUT_STEREO|AV_CH_FRONT_CENTER|CH_BACK_CENTER, ///< 4, C + L + R+ S
+ AV_CH_LAYOUT_STEREO|AV_CH_FRONT_CENTER|AV_CH_BACK_CENTER, ///< 4, C + L + R+ S
AV_CH_LAYOUT_STEREO|AV_CH_SIDE_LEFT|AV_CH_SIDE_RIGHT, ///< 4, L + R +SL+ SR
AV_CH_LAYOUT_STEREO|AV_CH_FRONT_CENTER|AV_CH_SIDE_LEFT|AV_CH_SIDE_RIGHT, ///< 5, C + L + R+ SL+SR
AV_CH_LAYOUT_STEREO|AV_CH_SIDE_LEFT|AV_CH_SIDE_RIGHT|AV_CH_FRONT_LEFT_OF_CENTER|AV_CH_FRONT_RIGHT_OF_CENTER, ///< 6, CL + CR + L + R + SL + SR
/* Subband samples history (for ADPCM) */
float subband_samples_hist[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][4];
- DECLARE_ALIGNED(16, float, subband_fir_hist)[DCA_PRIM_CHANNELS_MAX][512];
- DECLARE_ALIGNED(16, float, subband_fir_noidea)[DCA_PRIM_CHANNELS_MAX][32];
+ DECLARE_ALIGNED(32, float, subband_fir_hist)[DCA_PRIM_CHANNELS_MAX][512];
+ DECLARE_ALIGNED(32, float, subband_fir_noidea)[DCA_PRIM_CHANNELS_MAX][32];
int hist_index[DCA_PRIM_CHANNELS_MAX];
- DECLARE_ALIGNED(16, float, raXin)[32];
+ DECLARE_ALIGNED(32, float, raXin)[32];
int output; ///< type of output
- float add_bias; ///< output bias
float scale_bias; ///< output scale
- DECLARE_ALIGNED(16, float, subband_samples)[DCA_BLOCKS_MAX][DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][8];
- DECLARE_ALIGNED(16, float, samples)[(DCA_PRIM_CHANNELS_MAX+1)*256];
+ DECLARE_ALIGNED(32, float, subband_samples)[DCA_BLOCKS_MAX][DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][8];
+ DECLARE_ALIGNED(32, float, samples)[(DCA_PRIM_CHANNELS_MAX+1)*256];
const float *samples_chanptr[DCA_PRIM_CHANNELS_MAX+1];
uint8_t dca_buffer[DCA_MAX_FRAME_SIZE + DCA_MAX_EXSS_HEADER_SIZE + DCA_BUFFER_PADDING_SIZE];
int current_subframe;
int current_subsubframe;
+ int core_ext_mask; ///< present extensions in the core substream
+
/* XCh extension information */
- int xch_present;
+ int xch_present; ///< XCh extension present and valid
int xch_base_channel; ///< index of first (only) channel containing XCH data
- /* Other detected extensions in the core substream */
- int xxch_present;
- int x96_present;
-
/* ExSS header parser */
int static_fields; ///< static fields present
int mix_metadata; ///< mixing metadata present
FFTContext imdct;
SynthFilterContext synth;
DCADSPContext dcadsp;
+ FmtConvertContext fmt_conv;
} DCAContext;
static const uint16_t dca_vlc_offs[] = {
static void qmf_32_subbands(DCAContext * s, int chans,
float samples_in[32][8], float *samples_out,
- float scale, float bias)
+ float scale)
{
const float *prCoeff;
int i;
s->synth.synth_filter_float(&s->imdct,
s->subband_fir_hist[chans], &s->hist_index[chans],
s->subband_fir_noidea[chans], prCoeff,
- samples_out, s->raXin, scale, bias);
+ samples_out, s->raXin, scale);
samples_out+= 32;
}
static void lfe_interpolation_fir(DCAContext *s, int decimation_select,
int num_deci_sample, float *samples_in,
- float *samples_out, float scale,
- float bias)
+ float *samples_out, float scale)
{
/* samples_in: An array holding decimated samples.
* Samples in current subframe starts from samples_in[0],
/* Interpolation */
for (deciindex = 0; deciindex < num_deci_sample; deciindex++) {
s->dcadsp.lfe_fir(samples_out, samples_in, prCoeff, decifactor,
- scale, bias);
+ scale);
samples_in++;
samples_out += 2 * decifactor;
}
/* downmixing routines */
#define MIX_REAR1(samples, si1, rs, coef) \
- samples[i] += (samples[si1] - add_bias) * coef[rs][0]; \
- samples[i+256] += (samples[si1] - add_bias) * coef[rs][1];
+ samples[i] += samples[si1] * coef[rs][0]; \
+ samples[i+256] += samples[si1] * coef[rs][1];
#define MIX_REAR2(samples, si1, si2, rs, coef) \
- samples[i] += (samples[si1] - add_bias) * coef[rs][0] + (samples[si2] - add_bias) * coef[rs+1][0]; \
- samples[i+256] += (samples[si1] - add_bias) * coef[rs][1] + (samples[si2] - add_bias) * coef[rs+1][1];
+ samples[i] += samples[si1] * coef[rs][0] + samples[si2] * coef[rs+1][0]; \
+ samples[i+256] += samples[si1] * coef[rs][1] + samples[si2] * coef[rs+1][1];
#define MIX_FRONT3(samples, coef) \
- t = samples[i+c] - add_bias; \
- u = samples[i+l] - add_bias; \
- v = samples[i+r] - add_bias; \
- samples[i] = t * coef[0][0] + u * coef[1][0] + v * coef[2][0] + add_bias; \
- samples[i+256] = t * coef[0][1] + u * coef[1][1] + v * coef[2][1] + add_bias;
+ t = samples[i+c]; \
+ u = samples[i+l]; \
+ v = samples[i+r]; \
+ samples[i] = t * coef[0][0] + u * coef[1][0] + v * coef[2][0]; \
+ samples[i+256] = t * coef[0][1] + u * coef[1][1] + v * coef[2][1];
#define DOWNMIX_TO_STEREO(op1, op2) \
for (i = 0; i < 256; i++){ \
static void dca_downmix(float *samples, int srcfmt,
int downmix_coef[DCA_PRIM_CHANNELS_MAX][2],
- const int8_t *channel_mapping, float add_bias)
+ const int8_t *channel_mapping)
{
int c,l,r,sl,sr,s;
int i;
block[m] = get_bitalloc(&s->gb, &dca_smpl_bitalloc[abits], sel);
}
- s->dsp.int32_to_float_fmul_scalar(subband_samples[k][l],
+ s->fmt_conv.int32_to_float_fmul_scalar(subband_samples[k][l],
block, rscale, 8);
}
/* static float pcm_to_double[8] =
{32768.0, 32768.0, 524288.0, 524288.0, 0, 8388608.0, 8388608.0};*/
qmf_32_subbands(s, k, subband_samples[k], &s->samples[256 * s->channel_order_tab[k]],
- M_SQRT1_2*s->scale_bias /*pcm_to_double[s->source_pcm_res] */ ,
- s->add_bias );
+ M_SQRT1_2*s->scale_bias /*pcm_to_double[s->source_pcm_res] */ );
}
/* Down mixing */
if (s->avctx->request_channels == 2 && s->prim_channels > 2) {
- dca_downmix(s->samples, s->amode, s->downmix_coef, s->channel_order_tab, s->add_bias);
+ dca_downmix(s->samples, s->amode, s->downmix_coef, s->channel_order_tab);
}
/* Generate LFE samples for this subsubframe FIXME!!! */
lfe_interpolation_fir(s, s->lfe, 2 * s->lfe,
s->lfe_data + 2 * s->lfe * (block_index + 4),
&s->samples[256 * dca_lfe_index[s->amode]],
- (1.0/256.0)*s->scale_bias, s->add_bias);
+ (1.0/256.0)*s->scale_bias);
/* Outputs 20bits pcm samples */
}
PutBitContext pb;
if ((unsigned)src_size > (unsigned)max_size) {
-// av_log(NULL, AV_LOG_ERROR, "Input frame size larger then DCA_MAX_FRAME_SIZE!\n");
+// av_log(NULL, AV_LOG_ERROR, "Input frame size larger than DCA_MAX_FRAME_SIZE!\n");
// return -1;
src_size = max_size;
}
*/
static void dca_exss_skip_mix_coeffs(GetBitContext *gb, int channels, int out_ch)
{
- for (int i = 0; i < channels; i++) {
+ int i;
+
+ for (i = 0; i < channels; i++) {
int mix_map_mask = get_bits(gb, out_ch);
int num_coeffs = av_popcount(mix_map_mask);
skip_bits_long(gb, num_coeffs * 6);
if (!(extensions_mask & DCA_EXT_CORE))
av_log(s->avctx, AV_LOG_WARNING, "DTS core detection mismatch.\n");
- if (!!(extensions_mask & DCA_EXT_XCH) != s->xch_present)
- av_log(s->avctx, AV_LOG_WARNING, "DTS XCh detection mismatch.\n");
- if (!!(extensions_mask & DCA_EXT_XXCH) != s->xxch_present)
- av_log(s->avctx, AV_LOG_WARNING, "DTS XXCh detection mismatch.\n");
- if (!!(extensions_mask & DCA_EXT_X96) != s->x96_present)
- av_log(s->avctx, AV_LOG_WARNING, "DTS X96 detection mismatch.\n");
+ if ((extensions_mask & DCA_CORE_EXTS) != s->core_ext_mask)
+ av_log(s->avctx, AV_LOG_WARNING, "DTS extensions detection mismatch (%d, %d)\n",
+ extensions_mask & DCA_CORE_EXTS, s->core_ext_mask);
return 0;
}
{
int ss_index;
int blownup;
- int header_size;
- int hd_size;
int num_audiop = 1;
int num_assets = 1;
int active_ss_mask[8];
ss_index = get_bits(&s->gb, 2);
blownup = get_bits1(&s->gb);
- header_size = get_bits(&s->gb, 8 + 4 * blownup) + 1;
- hd_size = get_bits_long(&s->gb, 16 + 4 * blownup) + 1;
+ skip_bits(&s->gb, 8 + 4 * blownup); // header_size
+ skip_bits(&s->gb, 16 + 4 * blownup); // hd_size
s->static_fields = get_bits1(&s->gb);
if (s->static_fields) {
int lfe_samples;
int num_core_channels = 0;
int i;
- int16_t *samples = data;
+ float *samples_flt = data;
+ int16_t *samples_s16 = data;
+ int out_size;
DCAContext *s = avctx->priv_data;
int channels;
int core_ss_end;
s->xch_present = 0;
- s->x96_present = 0;
- s->xxch_present = 0;
s->dca_buffer_size = dca_convert_bitstream(buf, buf_size, s->dca_buffer,
DCA_MAX_FRAME_SIZE + DCA_MAX_EXSS_HEADER_SIZE);
//set AVCodec values with parsed data
avctx->sample_rate = s->sample_rate;
avctx->bit_rate = s->bit_rate;
+ avctx->frame_size = s->sample_blocks * 32;
s->profile = FF_PROFILE_DTS;
/* record number of core channels incase less than max channels are requested */
num_core_channels = s->prim_channels;
- /* extensions start at 32-bit boundaries into bitstream */
- skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);
+ if (s->ext_coding)
+ s->core_ext_mask = dca_ext_audio_descr_mask[s->ext_descr];
+ else
+ s->core_ext_mask = 0;
core_ss_end = FFMIN(s->frame_size, s->dca_buffer_size) * 8;
+ /* only scan for extensions if ext_descr was unknown or indicated a
+ * supported XCh extension */
+ if (s->core_ext_mask < 0 || s->core_ext_mask & DCA_EXT_XCH) {
+
+ /* if ext_descr was unknown, clear s->core_ext_mask so that the
+ * extensions scan can fill it up */
+ s->core_ext_mask = FFMAX(s->core_ext_mask, 0);
+
+ /* extensions start at 32-bit boundaries into bitstream */
+ skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);
+
while(core_ss_end - get_bits_count(&s->gb) >= 32) {
uint32_t bits = get_bits_long(&s->gb, 32);
/* skip length-to-end-of-frame field for the moment */
skip_bits(&s->gb, 10);
- s->profile = FFMAX(s->profile, FF_PROFILE_DTS_ES);
+ s->core_ext_mask |= DCA_EXT_XCH;
/* extension amode should == 1, number of channels in extension */
/* AFAIK XCh is not used for more channels */
/* XXCh: extended channels */
/* usually found either in core or HD part in DTS-HD HRA streams,
* but not in DTS-ES which contains XCh extensions instead */
- s->xxch_present = 1;
- s->profile = FFMAX(s->profile, FF_PROFILE_DTS_ES);
+ s->core_ext_mask |= DCA_EXT_XXCH;
break;
case 0x1d95f262: {
av_log(avctx, AV_LOG_DEBUG, "FSIZE96 = %d bytes\n", fsize96);
av_log(avctx, AV_LOG_DEBUG, "REVNO = %d\n", get_bits(&s->gb, 4));
- s->x96_present = 1;
- s->profile = FFMAX(s->profile, FF_PROFILE_DTS_96_24);
+ s->core_ext_mask |= DCA_EXT_X96;
break;
}
}
skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);
}
+ } else {
+ /* no supported extensions, skip the rest of the core substream */
+ skip_bits_long(&s->gb, core_ss_end - get_bits_count(&s->gb));
+ }
+
+ if (s->core_ext_mask & DCA_EXT_X96)
+ s->profile = FF_PROFILE_DTS_96_24;
+ else if (s->core_ext_mask & (DCA_EXT_XCH | DCA_EXT_XXCH))
+ s->profile = FF_PROFILE_DTS_ES;
+
/* check for ExSS (HD part) */
if (s->dca_buffer_size - s->frame_size > 32
&& get_bits_long(&s->gb, 32) == DCA_HD_MARKER)
/* There is nothing that prevents a dts frame to change channel configuration
- but FFmpeg doesn't support that so only set the channels if it is previously
+ but Libav doesn't support that so only set the channels if it is previously
unset. Ideally during the first probe for channels the crc should be checked
and only set avctx->channels when the crc is ok. Right now the decoder could
set the channels based on a broken first frame.*/
return -1;
}
- if (*data_size < (s->sample_blocks / 8) * 256 * sizeof(int16_t) * channels)
+ out_size = 256 / 8 * s->sample_blocks * channels *
+ av_get_bytes_per_sample(avctx->sample_fmt);
+ if (*data_size < out_size)
return -1;
- *data_size = 256 / 8 * s->sample_blocks * sizeof(int16_t) * channels;
+ *data_size = out_size;
/* filter to get final output */
for (i = 0; i < (s->sample_blocks / 8); i++) {
float* rt_chan = s->samples + s->channel_order_tab[s->xch_base_channel - 1] * 256;
int j;
for(j = 0; j < 256; ++j) {
- lt_chan[j] -= (back_chan[j] - s->add_bias) * M_SQRT1_2;
- rt_chan[j] -= (back_chan[j] - s->add_bias) * M_SQRT1_2;
+ lt_chan[j] -= back_chan[j] * M_SQRT1_2;
+ rt_chan[j] -= back_chan[j] * M_SQRT1_2;
}
}
- s->dsp.float_to_int16_interleave(samples, s->samples_chanptr, 256, channels);
- samples += 256 * channels;
+ if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT) {
+ s->fmt_conv.float_interleave(samples_flt, s->samples_chanptr, 256,
+ channels);
+ samples_flt += 256 * channels;
+ } else {
+ s->fmt_conv.float_to_int16_interleave(samples_s16,
+ s->samples_chanptr, 256,
+ channels);
+ samples_s16 += 256 * channels;
+ }
}
/* update lfe history */
ff_mdct_init(&s->imdct, 6, 1, 1.0);
ff_synth_filter_init(&s->synth);
ff_dcadsp_init(&s->dcadsp);
+ ff_fmt_convert_init(&s->fmt_conv, avctx);
for (i = 0; i < DCA_PRIM_CHANNELS_MAX+1; i++)
s->samples_chanptr[i] = s->samples + i * 256;
- avctx->sample_fmt = AV_SAMPLE_FMT_S16;
- if (s->dsp.float_to_int16_interleave == ff_float_to_int16_interleave_c) {
- s->add_bias = 385.0f;
+ if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT) {
+ avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
s->scale_bias = 1.0 / 32768.0;
} else {
- s->add_bias = 0.0f;
+ avctx->sample_fmt = AV_SAMPLE_FMT_S16;
s->scale_bias = 1.0;
-
- /* allow downmixing to stereo */
- if (avctx->channels > 0 && avctx->request_channels < avctx->channels &&
- avctx->request_channels == 2) {
- avctx->channels = avctx->request_channels;
- }
}
+ /* allow downmixing to stereo */
+ if (avctx->channels > 0 && avctx->request_channels < avctx->channels &&
+ avctx->request_channels == 2) {
+ avctx->channels = avctx->request_channels;
+ }
return 0;
}
return 0;
}
-AVCodec dca_decoder = {
+static const AVProfile profiles[] = {
+ { FF_PROFILE_DTS, "DTS" },
+ { FF_PROFILE_DTS_ES, "DTS-ES" },
+ { FF_PROFILE_DTS_96_24, "DTS 96/24" },
+ { FF_PROFILE_DTS_HD_HRA, "DTS-HD HRA" },
+ { FF_PROFILE_DTS_HD_MA, "DTS-HD MA" },
+ { FF_PROFILE_UNKNOWN },
+};
+
+AVCodec ff_dca_decoder = {
.name = "dca",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_DTS,
.close = dca_decode_end,
.long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
.capabilities = CODEC_CAP_CHANNEL_CONF,
+ .sample_fmts = (const enum AVSampleFormat[]) {
+ AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
+ },
+ .profiles = NULL_IF_CONFIG_SMALL(profiles),
};
projects / ffmpeg / blobdiff