📄 replaygain.c
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/* grabbag - Convenience lib for various routines common to several tools * Copyright (C) 2002,2003,2004,2005 Josh Coalson * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License * as published by the Free Software Foundation; either version 2 * of the License, or (at your option) any later version. * * This program 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 General Public License for more details. * * You should have received a copy of the GNU General Public License * along with this program; if not, write to the Free Software * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. */#include "share/grabbag.h"#include "share/replaygain_analysis.h"#include "FLAC/assert.h"#include "FLAC/file_decoder.h"#include "FLAC/metadata.h"#include <locale.h>#include <math.h>#include <stdio.h>#include <stdlib.h>#include <string.h>#if defined _MSC_VER || defined __MINGW32__#include <io.h> /* for chmod() */#endif#include <sys/stat.h> /* for stat(), maybe chmod() */#ifdef local_min#undef local_min#endif#define local_min(a,b) ((a)<(b)?(a):(b))#ifdef local_max#undef local_max#endif#define local_max(a,b) ((a)>(b)?(a):(b))static const FLAC__byte *tag_title_gain_ = "REPLAYGAIN_TRACK_GAIN";static const FLAC__byte *tag_title_peak_ = "REPLAYGAIN_TRACK_PEAK";static const FLAC__byte *tag_album_gain_ = "REPLAYGAIN_ALBUM_GAIN";static const FLAC__byte *tag_album_peak_ = "REPLAYGAIN_ALBUM_PEAK";static const char *peak_format_ = "%s=%1.8f";static const char *gain_format_ = "%s=%+2.2f dB";static double album_peak_, title_peak_;const unsigned GRABBAG__REPLAYGAIN_MAX_TAG_SPACE_REQUIRED = 148;/* FLAC__STREAM_METADATA_VORBIS_COMMENT_ENTRY_LENGTH_LEN/8 + 21 + 1 + 10 + FLAC__STREAM_METADATA_VORBIS_COMMENT_ENTRY_LENGTH_LEN/8 + 21 + 1 + 12 + FLAC__STREAM_METADATA_VORBIS_COMMENT_ENTRY_LENGTH_LEN/8 + 21 + 1 + 10 + FLAC__STREAM_METADATA_VORBIS_COMMENT_ENTRY_LENGTH_LEN/8 + 21 + 1 + 12*/static FLAC__bool get_file_stats_(const char *filename, struct stat *stats){ FLAC__ASSERT(0 != filename); FLAC__ASSERT(0 != stats); return (0 == stat(filename, stats));}static void set_file_stats_(const char *filename, struct stat *stats){ FLAC__ASSERT(0 != filename); FLAC__ASSERT(0 != stats); (void)chmod(filename, stats->st_mode);}static FLAC__bool append_tag_(FLAC__StreamMetadata *block, const char *format, const FLAC__byte *name, float value){ char buffer[256]; char *saved_locale; FLAC__StreamMetadata_VorbisComment_Entry entry; FLAC__ASSERT(0 != block); FLAC__ASSERT(block->type == FLAC__METADATA_TYPE_VORBIS_COMMENT); FLAC__ASSERT(0 != name); FLAC__ASSERT(0 != value); buffer[sizeof(buffer)-1] = '\0'; /* * We need to save the old locale and switch to "C" because the locale * influences the formatting of %f and we want it a certain way. */ saved_locale = setlocale(LC_ALL, 0); setlocale(LC_ALL, "C");#if defined _MSC_VER || defined __MINGW32__ _snprintf(buffer, sizeof(buffer)-1, format, name, value);#else snprintf(buffer, sizeof(buffer)-1, format, name, value);#endif setlocale(LC_ALL, saved_locale); entry.entry = (FLAC__byte *)buffer; entry.length = strlen(buffer); return FLAC__metadata_object_vorbiscomment_append_comment(block, entry, /*copy=*/true);}FLAC__bool grabbag__replaygain_is_valid_sample_frequency(unsigned sample_frequency){ static const unsigned valid_sample_rates[] = { 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000 }; static const unsigned n_valid_sample_rates = sizeof(valid_sample_rates) / sizeof(valid_sample_rates[0]); unsigned i; for(i = 0; i < n_valid_sample_rates; i++) if(sample_frequency == valid_sample_rates[i]) return true; return false;}FLAC__bool grabbag__replaygain_init(unsigned sample_frequency){ title_peak_ = album_peak_ = 0.0; return InitGainAnalysis((long)sample_frequency) == INIT_GAIN_ANALYSIS_OK;}FLAC__bool grabbag__replaygain_analyze(const FLAC__int32 * const input[], FLAC__bool is_stereo, unsigned bps, unsigned samples){ /* using a small buffer improves data locality; we'd like it to fit easily in the dcache */ static Float_t lbuffer[2048], rbuffer[2048]; static const unsigned nbuffer = sizeof(lbuffer) / sizeof(lbuffer[0]); FLAC__int32 block_peak = 0, s; unsigned i, j; FLAC__ASSERT(bps >= 4 && bps <= FLAC__REFERENCE_CODEC_MAX_BITS_PER_SAMPLE); FLAC__ASSERT(FLAC__MIN_BITS_PER_SAMPLE == 4); /* * We use abs() on a FLAC__int32 which is undefined for the most negative value. * If the reference codec ever handles 32bps we will have to write a special * case here. */ FLAC__ASSERT(FLAC__REFERENCE_CODEC_MAX_BITS_PER_SAMPLE < 32); if(bps == 16) { if(is_stereo) { j = 0; while(samples > 0) { const unsigned n = local_min(samples, nbuffer); for(i = 0; i < n; i++, j++) { s = input[0][j]; lbuffer[i] = (Float_t)s; s = abs(s); block_peak = local_max(block_peak, s); s = input[1][j]; rbuffer[i] = (Float_t)s; s = abs(s); block_peak = local_max(block_peak, s); } samples -= n; if(AnalyzeSamples(lbuffer, rbuffer, n, 2) != GAIN_ANALYSIS_OK) return false; } } else { j = 0; while(samples > 0) { const unsigned n = local_min(samples, nbuffer); for(i = 0; i < n; i++, j++) { s = input[0][j]; lbuffer[i] = (Float_t)s; s = abs(s); block_peak = local_max(block_peak, s); } samples -= n; if(AnalyzeSamples(lbuffer, 0, n, 1) != GAIN_ANALYSIS_OK) return false; } } } else { /* bps must be < 32 according to above assertion */ const double scale = ( (bps > 16)? (double)1. / (double)(1u << (bps - 16)) : (double)(1u << (16 - bps)) ); if(is_stereo) { j = 0; while(samples > 0) { const unsigned n = local_min(samples, nbuffer); for(i = 0; i < n; i++, j++) { s = input[0][j]; lbuffer[i] = (Float_t)(scale * (double)s); s = abs(s); block_peak = local_max(block_peak, s); s = input[1][j]; rbuffer[i] = (Float_t)(scale * (double)s); s = abs(s); block_peak = local_max(block_peak, s); } samples -= n; if(AnalyzeSamples(lbuffer, rbuffer, n, 2) != GAIN_ANALYSIS_OK) return false; } } else { j = 0; while(samples > 0) { const unsigned n = local_min(samples, nbuffer); for(i = 0; i < n; i++, j++) { s = input[0][j]; lbuffer[i] = (Float_t)(scale * (double)s); s = abs(s); block_peak = local_max(block_peak, s); } samples -= n; if(AnalyzeSamples(lbuffer, 0, n, 1) != GAIN_ANALYSIS_OK) return false; } } } { const double peak_scale = (double)(1u << (bps - 1)); double peak = (double)block_peak / peak_scale; if(peak > title_peak_) title_peak_ = peak; if(peak > album_peak_) album_peak_ = peak; } return true;}void grabbag__replaygain_get_album(float *gain, float *peak){ *gain = (float)GetAlbumGain(); *peak = (float)album_peak_; album_peak_ = 0.0;}void grabbag__replaygain_get_title(float *gain, float *peak){ *gain = (float)GetTitleGain(); *peak = (float)title_peak_; title_peak_ = 0.0;}typedef struct { unsigned channels; unsigned bits_per_sample; unsigned sample_rate; FLAC__bool error;} DecoderInstance;static FLAC__StreamDecoderWriteStatus write_callback_(const FLAC__FileDecoder *decoder, const FLAC__Frame *frame, const FLAC__int32 * const buffer[], void *client_data){ DecoderInstance *instance = (DecoderInstance*)client_data; const unsigned bits_per_sample = frame->header.bits_per_sample; const unsigned channels = frame->header.channels; const unsigned sample_rate = frame->header.sample_rate; const unsigned samples = frame->header.blocksize; (void)decoder; if( !instance->error && (channels == 2 || channels == 1) && bits_per_sample == instance->bits_per_sample && channels == instance->channels && sample_rate == instance->sample_rate ) { instance->error = !grabbag__replaygain_analyze(buffer, channels==2, bits_per_sample, samples); } else { instance->error = true; } if(!instance->error) return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE; else return FLAC__STREAM_DECODER_WRITE_STATUS_ABORT;}static void metadata_callback_(const FLAC__FileDecoder *decoder, const FLAC__StreamMetadata *metadata, void *client_data){ DecoderInstance *instance = (DecoderInstance*)client_data; (void)decoder; if(metadata->type == FLAC__METADATA_TYPE_STREAMINFO) { instance->bits_per_sample = metadata->data.stream_info.bits_per_sample; instance->channels = metadata->data.stream_info.channels; instance->sample_rate = metadata->data.stream_info.sample_rate;
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