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179 lines
5.5 KiB
C
179 lines
5.5 KiB
C
/*
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mkvmerge -- utility for splicing together matroska files
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from component media subtypes
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dts_common.h
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Written by Moritz Bunkus <moritz@bunkus.org>
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Distributed under the GPL
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see the file COPYING for details
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or visit http://www.gnu.org/copyleft/gpl.html
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*/
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/*!
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\file
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\version \$Id: dts_common.h,v 1.6 2003/05/20 06:30:24 mosu Exp $
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\brief definitions and helper functions for DTS data
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\author Peter Niemayer <niemayer@isg.de>
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\author Moritz Bunkus <moritz@bunkus.org>
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*/
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#ifndef __DTSCOMMON_H
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#define __DTSCOMMON_H
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static const long long max_dts_packet_size = 15384;
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/* The following code looks a little odd as it was written in C++
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but with the possibility in mind to make this structure and the
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functions below it later available in C
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*/
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typedef struct dts_header_s {
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// ---------------------------------------------------
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// ---------------------------------------------------
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enum {
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// Used to extremely precisely specify the end-of-stream (single PCM
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// sample resolution).
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frametype_termination = 0,
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frametype_normal
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} frametype;
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// 0 for normal frames, 1 to 30 for termination frames. Number of PCM
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// samples the frame is shorter than normal.
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unsigned int deficit_sample_count;
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// If true, a CRC-sum is included in the data.
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bool crc_present;
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// number of PCM core sample blocks in this frame. Each PCM core sample block
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// consists of 32 samples. Notice that "core samples" means "samples
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// after the input decimator", so at sampling frequencies >48kHz, one core
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// sample represents 2 (or 4 for frequencies >96kHz) output samples.
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unsigned int num_pcm_sample_blocks;
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// Number of bytes this frame occupies (range: 95 to 16 383).
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unsigned int frame_byte_size;
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// Number of audio channels, -1 for "unknown".
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int audio_channels;
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// String describing the audio channel arrangement
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const char *audio_channel_arrangement;
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// -1 for "invalid"
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unsigned int core_sampling_frequency;
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// in bit per second, or -1 == "open", -2 == "variable", -3 == "lossless"
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int transmission_bitrate;
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// if true, sub-frames contain coefficients for downmixing to stereo
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bool embedded_down_mix;
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// if true, sub-frames contain coefficients for dynamic range correction
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bool embedded_dynamic_range;
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// if true, a time stamp is embedded at the end of the core audio data
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bool embedded_time_stamp;
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// if true, auxiliary data is appended at the end of the core audio data
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bool auxiliary_data;
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// if true, the source material was mastered in HDCD format
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bool hdcd_master;
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enum extension_audio_descriptor_enum {
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extension_xch = 0, // channel extension
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extension_unknown1,
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extension_x96k, // frequency extension
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extension_xch_x96k, // both channel and frequency extension
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extension_unknown4,
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extension_unknown5,
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extension_unknown6,
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extension_unknown7
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} extension_audio_descriptor; // significant only if extended_coding == true
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// if true, extended coding data is placed after the core audio data
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bool extended_coding;
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// if true, audio data check words are placed in each sub-sub-frame
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// rather than in each sub-frame, only
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bool audio_sync_word_in_sub_sub;
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enum lfe_type_enum {
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lfe_none,
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lfe_128, // 128 indicates the interpolation factor to reconstruct the
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// LFE channel
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lfe_64, // 64 indicates the interpolation factor to reconstruct the
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// LFE channel
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lfe_invalid
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} lfe_type;
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// if true, past frames will be used to predict ADPCM values for the
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// current one. This means, if this flag is false, the current frame is
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// better suited as an audio-jump-point (like an "I-frame" in video-coding).
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bool predictor_history_flag;
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// which FIR coefficients to use for sub-band reconstruction
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enum multirate_interpolator_enum {
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mi_non_perfect,
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mi_perfect
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} multirate_interpolator;
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// 0 to 15
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unsigned int encoder_software_revision;
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// 0 to 3 - "top-secret" bits indicating the "copy history" of the material
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unsigned int copy_history;
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// 16, 20 or 24 bits per sample, or -1 == invalid
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int source_pcm_resolution;
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// if true, source surround channels are mastered in DTS-ES
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bool source_surround_in_es;
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// if true, left and right front channels are encoded as
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// sum and difference (L = L + R, R = L - R)
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bool front_sum_difference;
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// same as front_sum_difference for surround left and right channels
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bool surround_sum_difference;
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// gain in dB to apply for dialog normalization
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int dialog_normalization_gain;
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} dts_header_t;
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int find_dts_header(const unsigned char *buf, unsigned int size,
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struct dts_header_s *dts_header);
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void print_dts_header(const struct dts_header_s *dts_header);
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inline int get_dts_packet_length_in_core_samples(const struct dts_header_s
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*dts_header) {
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// computes the length (in time, not size) of the packet in "samples".
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int r;
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r = dts_header->num_pcm_sample_blocks * 32;
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if (dts_header->frametype == dts_header_s::frametype_termination)
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r -= dts_header->deficit_sample_count;
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return r;
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}
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inline double get_dts_packet_length_in_milliseconds(const struct dts_header_s
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*dts_header) {
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// computes the length (in time, not size) of the packet in "samples".
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int samples = get_dts_packet_length_in_core_samples(dts_header);
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double t = ((double)samples*1000.0) /
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((double)dts_header->core_sampling_frequency);
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return t;
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}
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#endif // __DTSCOMMON_H
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