* Common code between the AC-3 and E-AC-3 decoders
* Copyright (c) 2007 Bartlomiej Wolowiec <bartek.wolowiec@gmail.com>
*
- * This file is part of Libav.
+ * This file is part of FFmpeg.
*
- * Libav is free software; you can redistribute it and/or
+ * 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.
*
- * Libav is distributed in the hope that it will be useful,
+ * 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 Libav; if not, write to the Free Software
+ * License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#define AVCODEC_AC3DEC_H
#include "libavutil/float_dsp.h"
+#include "libavutil/fixed_dsp.h"
#include "libavutil/lfg.h"
#include "ac3.h"
#include "ac3dsp.h"
int bitstream_mode; ///< bitstream mode (bsmod)
int channel_mode; ///< channel mode (acmod)
int lfe_on; ///< lfe channel in use
+ int dialog_normalization[2]; ///< dialog level in dBFS (dialnorm)
+ int compression_exists[2]; ///< compression field is valid for frame (compre)
+ int compression_gain[2]; ///< gain to apply for heavy compression (compr)
int channel_map; ///< custom channel map
int preferred_downmix; ///< Preferred 2-channel downmix mode (dmixmod)
int center_mix_level; ///< Center mix level index
int dolby_headphone_mode; ///< dolby headphone mode (dheadphonmod)
///@}
+ int preferred_stereo_downmix;
+ float ltrt_center_mix_level;
+ float ltrt_surround_mix_level;
+ float loro_center_mix_level;
+ float loro_surround_mix_level;
+ int target_level; ///< target level in dBFS
+ float level_gain[2];
+
///@name Frame syntax parameters
int snr_offset_strategy; ///< SNR offset strategy (snroffststr)
int block_switch_syntax; ///< block switch syntax enabled (blkswe)
int num_spx_bands; ///< number of spx bands (nspxbnds)
uint8_t spx_band_sizes[SPX_MAX_BANDS]; ///< number of bins in each spx band
uint8_t first_spx_coords[AC3_MAX_CHANNELS]; ///< first spx coordinates states (firstspxcos)
- float spx_noise_blend[AC3_MAX_CHANNELS][SPX_MAX_BANDS]; ///< spx noise blending factor (nblendfact)
- float spx_signal_blend[AC3_MAX_CHANNELS][SPX_MAX_BANDS];///< spx signal blending factor (sblendfact)
+ INTFLOAT spx_noise_blend[AC3_MAX_CHANNELS][SPX_MAX_BANDS]; ///< spx noise blending factor (nblendfact)
+ INTFLOAT spx_signal_blend[AC3_MAX_CHANNELS][SPX_MAX_BANDS];///< spx signal blending factor (sblendfact)
///@}
///@name Adaptive hybrid transform
int fbw_channels; ///< number of full-bandwidth channels
int channels; ///< number of total channels
int lfe_ch; ///< index of LFE channel
- float downmix_coeffs[AC3_MAX_CHANNELS][2]; ///< stereo downmix coefficients
+ SHORTFLOAT downmix_coeffs[AC3_MAX_CHANNELS][2]; ///< stereo downmix coefficients
int downmixed; ///< indicates if coeffs are currently downmixed
int output_mode; ///< output channel configuration
int out_channels; ///< number of output channels
///@}
///@name Dynamic range
- float dynamic_range[2]; ///< dynamic range
- float drc_scale; ///< percentage of dynamic range compression to be applied
+ INTFLOAT dynamic_range[2]; ///< dynamic range
+ INTFLOAT drc_scale; ///< percentage of dynamic range compression to be applied
+ int heavy_compression; ///< apply heavy compression
+ INTFLOAT heavy_dynamic_range[2]; ///< heavy dynamic range compression
///@}
///@name Bandwidth
///@name Optimization
BswapDSPContext bdsp;
- AVFloatDSPContext fdsp;
+#if USE_FIXED
+ AVFixedDSPContext *fdsp;
+#else
+ AVFloatDSPContext *fdsp;
+#endif
AC3DSPContext ac3dsp;
FmtConvertContext fmt_conv; ///< optimized conversion functions
///@}
- float *outptr[AC3_MAX_CHANNELS];
- float *xcfptr[AC3_MAX_CHANNELS];
- float *dlyptr[AC3_MAX_CHANNELS];
+ SHORTFLOAT *outptr[AC3_MAX_CHANNELS];
+ INTFLOAT *xcfptr[AC3_MAX_CHANNELS];
+ INTFLOAT *dlyptr[AC3_MAX_CHANNELS];
///@name Aligned arrays
- DECLARE_ALIGNED(16, int32_t, fixed_coeffs)[AC3_MAX_CHANNELS][AC3_MAX_COEFS]; ///< fixed-point transform coefficients
- DECLARE_ALIGNED(32, float, transform_coeffs)[AC3_MAX_CHANNELS][AC3_MAX_COEFS]; ///< transform coefficients
- DECLARE_ALIGNED(32, float, delay)[AC3_MAX_CHANNELS][AC3_BLOCK_SIZE]; ///< delay - added to the next block
- DECLARE_ALIGNED(32, float, window)[AC3_BLOCK_SIZE]; ///< window coefficients
- DECLARE_ALIGNED(32, float, tmp_output)[AC3_BLOCK_SIZE]; ///< temporary storage for output before windowing
- DECLARE_ALIGNED(32, float, output)[AC3_MAX_CHANNELS][AC3_BLOCK_SIZE]; ///< output after imdct transform and windowing
+ DECLARE_ALIGNED(16, int, fixed_coeffs)[AC3_MAX_CHANNELS][AC3_MAX_COEFS]; ///< fixed-point transform coefficients
+ DECLARE_ALIGNED(32, INTFLOAT, transform_coeffs)[AC3_MAX_CHANNELS][AC3_MAX_COEFS]; ///< transform coefficients
+ DECLARE_ALIGNED(32, INTFLOAT, delay)[AC3_MAX_CHANNELS][AC3_BLOCK_SIZE]; ///< delay - added to the next block
+ DECLARE_ALIGNED(32, INTFLOAT, window)[AC3_BLOCK_SIZE]; ///< window coefficients
+ DECLARE_ALIGNED(32, INTFLOAT, tmp_output)[AC3_BLOCK_SIZE]; ///< temporary storage for output before windowing
+ DECLARE_ALIGNED(32, SHORTFLOAT, output)[AC3_MAX_CHANNELS][AC3_BLOCK_SIZE]; ///< output after imdct transform and windowing
DECLARE_ALIGNED(32, uint8_t, input_buffer)[AC3_FRAME_BUFFER_SIZE + AV_INPUT_BUFFER_PADDING_SIZE]; ///< temp buffer to prevent overread
///@}
} AC3DecodeContext;
* Parse the E-AC-3 frame header.
* This parses both the bit stream info and audio frame header.
*/
-int ff_eac3_parse_header(AC3DecodeContext *s);
+static int ff_eac3_parse_header(AC3DecodeContext *s);
/**
* Decode mantissas in a single channel for the entire frame.
* This is used when AHT mode is enabled.
*/
-void ff_eac3_decode_transform_coeffs_aht_ch(AC3DecodeContext *s, int ch);
+static void ff_eac3_decode_transform_coeffs_aht_ch(AC3DecodeContext *s, int ch);
/**
* Apply spectral extension to each channel by copying lower frequency
* coefficients to higher frequency bins and applying side information to
* approximate the original high frequency signal.
*/
-void ff_eac3_apply_spectral_extension(AC3DecodeContext *s);
+static void ff_eac3_apply_spectral_extension(AC3DecodeContext *s);
#endif /* AVCODEC_AC3DEC_H */
projects / ffmpeg / blobdiff