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📄 decode.c

📁 这是著名的TCPMP播放器在WINDWOWS,和WINCE下编译通过的源程序.笔者对其中的LIBMAD库做了针对ARM MPU的优化. 并增加了词幕功能.
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			return FLAC__STREAM_DECODER_WRITE_STATUS_ABORT;
		}
	}

	if(wide_samples > 0) {
		decoder_session->samples_processed += wide_samples;
		decoder_session->frame_counter++;

		if(!(decoder_session->frame_counter & 0x3f))
			print_stats(decoder_session);

		if(decoder_session->analysis_mode) {
			flac__analyze_frame(frame, decoder_session->frame_counter-1, decoder_session->aopts, fout);
		}
		else if(!decoder_session->test_only) {
			if (decoder_session->replaygain.apply) {
				bytes_to_write = FLAC__replaygain_synthesis__apply_gain(
					u8buffer,
					!is_big_endian,
					is_unsigned_samples,
					buffer,
					wide_samples,
					channels,
					bps, /* source_bps */
					bps, /* target_bps */
					decoder_session->replaygain.scale,
					decoder_session->replaygain.spec.limiter == RGSS_LIMIT__HARD, /* hard_limit */
					decoder_session->replaygain.spec.noise_shaping != NOISE_SHAPING_NONE, /* do_dithering */
					&decoder_session->replaygain.dither_context
				);
			}
			else if(bps == 8) {
				if(is_unsigned_samples) {
					for(sample = wide_sample = 0; wide_sample < wide_samples; wide_sample++)
						for(channel = 0; channel < channels; channel++, sample++)
							u8buffer[sample] = (FLAC__uint8)(buffer[channel][wide_sample] + 0x80);
				}
				else {
					for(sample = wide_sample = 0; wide_sample < wide_samples; wide_sample++)
						for(channel = 0; channel < channels; channel++, sample++)
							s8buffer[sample] = (FLAC__int8)(buffer[channel][wide_sample]);
				}
				bytes_to_write = sample;
			}
			else if(bps == 16) {
				if(is_unsigned_samples) {
					for(sample = wide_sample = 0; wide_sample < wide_samples; wide_sample++)
						for(channel = 0; channel < channels; channel++, sample++)
							u16buffer[sample] = (FLAC__uint16)(buffer[channel][wide_sample] + 0x8000);
				}
				else {
					for(sample = wide_sample = 0; wide_sample < wide_samples; wide_sample++)
						for(channel = 0; channel < channels; channel++, sample++)
							s16buffer[sample] = (FLAC__int16)(buffer[channel][wide_sample]);
				}
				if(is_big_endian != is_big_endian_host_) {
					unsigned char tmp;
					const unsigned bytes = sample * 2;
					for(byte = 0; byte < bytes; byte += 2) {
						tmp = u8buffer[byte];
						u8buffer[byte] = u8buffer[byte+1];
						u8buffer[byte+1] = tmp;
					}
				}
				bytes_to_write = 2 * sample;
			}
			else if(bps == 24) {
				if(is_unsigned_samples) {
					for(sample = wide_sample = 0; wide_sample < wide_samples; wide_sample++)
						for(channel = 0; channel < channels; channel++, sample++)
							u32buffer[sample] = buffer[channel][wide_sample] + 0x800000;
				}
				else {
					for(sample = wide_sample = 0; wide_sample < wide_samples; wide_sample++)
						for(channel = 0; channel < channels; channel++, sample++)
							s32buffer[sample] = buffer[channel][wide_sample];
				}
				if(is_big_endian != is_big_endian_host_) {
					unsigned char tmp;
					const unsigned bytes = sample * 4;
					for(byte = 0; byte < bytes; byte += 4) {
						tmp = u8buffer[byte];
						u8buffer[byte] = u8buffer[byte+3];
						u8buffer[byte+3] = tmp;
						tmp = u8buffer[byte+1];
						u8buffer[byte+1] = u8buffer[byte+2];
						u8buffer[byte+2] = tmp;
					}
				}
				if(is_big_endian) {
					unsigned lbyte;
					const unsigned bytes = sample * 4;
					for(lbyte = byte = 0; byte < bytes; ) {
						byte++;
						u8buffer[lbyte++] = u8buffer[byte++];
						u8buffer[lbyte++] = u8buffer[byte++];
						u8buffer[lbyte++] = u8buffer[byte++];
					}
				}
				else {
					unsigned lbyte;
					const unsigned bytes = sample * 4;
					for(lbyte = byte = 0; byte < bytes; ) {
						u8buffer[lbyte++] = u8buffer[byte++];
						u8buffer[lbyte++] = u8buffer[byte++];
						u8buffer[lbyte++] = u8buffer[byte++];
						byte++;
					}
				}
				bytes_to_write = 3 * sample;
			}
			else {
				FLAC__ASSERT(0);
			}
		}
	}
	if(bytes_to_write > 0) {
		if(flac__utils_fwrite(u8buffer, 1, bytes_to_write, fout) != bytes_to_write) {
			/* if a pipe closed when writing to stdout, we let it go without an error message */
			if(errno == EPIPE && decoder_session->fout == stdout)
				decoder_session->aborting_due_to_until = true;
			decoder_session->abort_flag = true;
			return FLAC__STREAM_DECODER_WRITE_STATUS_ABORT;
		}
	}
	return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE;
}

void metadata_callback(const void *decoder, const FLAC__StreamMetadata *metadata, void *client_data)
{
	DecoderSession *decoder_session = (DecoderSession*)client_data;
	(void)decoder;
	if(metadata->type == FLAC__METADATA_TYPE_STREAMINFO) {
		FLAC__uint64 skip, until;
		decoder_session->bps = metadata->data.stream_info.bits_per_sample;
		decoder_session->channels = metadata->data.stream_info.channels;
		decoder_session->sample_rate = metadata->data.stream_info.sample_rate;

		flac__utils_canonicalize_skip_until_specification(decoder_session->skip_specification, decoder_session->sample_rate);
		FLAC__ASSERT(decoder_session->skip_specification->value.samples >= 0);
		skip = (FLAC__uint64)decoder_session->skip_specification->value.samples;

		/* remember, metadata->data.stream_info.total_samples can be 0, meaning 'unknown' */
		if(metadata->data.stream_info.total_samples > 0 && skip >= metadata->data.stream_info.total_samples) {
			flac__utils_printf(stderr, 1, "%s: ERROR trying to --skip more samples than in stream\n", decoder_session->inbasefilename);
			decoder_session->abort_flag = true;
			return;
		}
		else if(metadata->data.stream_info.total_samples == 0 && skip > 0) {
			flac__utils_printf(stderr, 1, "%s: ERROR, can't --skip when FLAC metadata has total sample count of 0\n", decoder_session->inbasefilename);
			decoder_session->abort_flag = true;
			return;
		}
		FLAC__ASSERT(skip == 0 || 0 == decoder_session->cue_specification);
		decoder_session->total_samples = metadata->data.stream_info.total_samples - skip;

		/* note that we use metadata->data.stream_info.total_samples instead of decoder_session->total_samples */
		if(!canonicalize_until_specification(decoder_session->until_specification, decoder_session->inbasefilename, decoder_session->sample_rate, skip, metadata->data.stream_info.total_samples)) {
			decoder_session->abort_flag = true;
			return;
		}
		FLAC__ASSERT(decoder_session->until_specification->value.samples >= 0);
		until = (FLAC__uint64)decoder_session->until_specification->value.samples;

		if(until > 0) {
			FLAC__ASSERT(decoder_session->total_samples != 0);
			FLAC__ASSERT(0 == decoder_session->cue_specification);
			decoder_session->total_samples -= (metadata->data.stream_info.total_samples - until);
		}

		if(decoder_session->bps != 8 && decoder_session->bps != 16 && decoder_session->bps != 24) {
			flac__utils_printf(stderr, 1, "%s: ERROR: bits per sample is not 8/16/24\n", decoder_session->inbasefilename);
			decoder_session->abort_flag = true;
			return;
		}
	}
	else if(metadata->type == FLAC__METADATA_TYPE_CUESHEET) {
		/* remember, at this point, decoder_session->total_samples can be 0, meaning 'unknown' */
		if(decoder_session->total_samples == 0) {
			flac__utils_printf(stderr, 1, "%s: ERROR can't use --cue when FLAC metadata has total sample count of 0\n", decoder_session->inbasefilename);
			decoder_session->abort_flag = true;
			return;
		}

		flac__utils_canonicalize_cue_specification(decoder_session->cue_specification, &metadata->data.cue_sheet, decoder_session->total_samples, decoder_session->skip_specification, decoder_session->until_specification);

		FLAC__ASSERT(!decoder_session->skip_specification->is_relative);
		FLAC__ASSERT(decoder_session->skip_specification->value_is_samples);

		FLAC__ASSERT(!decoder_session->until_specification->is_relative);
		FLAC__ASSERT(decoder_session->until_specification->value_is_samples);

		FLAC__ASSERT(decoder_session->skip_specification->value.samples >= 0);
		FLAC__ASSERT(decoder_session->until_specification->value.samples >= 0);
		FLAC__ASSERT((FLAC__uint64)decoder_session->until_specification->value.samples <= decoder_session->total_samples);
		FLAC__ASSERT(decoder_session->skip_specification->value.samples <= decoder_session->until_specification->value.samples);

		decoder_session->total_samples = decoder_session->until_specification->value.samples - decoder_session->skip_specification->value.samples;
	}
	else if(metadata->type == FLAC__METADATA_TYPE_VORBIS_COMMENT) {
		if (decoder_session->replaygain.spec.apply) {
			double gain, peak;
			if (!(decoder_session->replaygain.apply = grabbag__replaygain_load_from_vorbiscomment(metadata, decoder_session->replaygain.spec.use_album_gain, &gain, &peak))) {
				flac__utils_printf(stderr, 1, "%s: WARNING: can't get %s ReplayGain tag\n", decoder_session->inbasefilename, decoder_session->replaygain.spec.use_album_gain? "album":"track");
			}
			else {
				const char *ls[] = { "no", "peak", "hard" };
				const char *ns[] = { "no", "low", "medium", "high" };
				decoder_session->replaygain.scale = grabbag__replaygain_compute_scale_factor(peak, gain, decoder_session->replaygain.spec.preamp, decoder_session->replaygain.spec.limiter == RGSS_LIMIT__PEAK);
				FLAC__ASSERT(decoder_session->bps > 0 && decoder_session->bps <= 32);
				FLAC__replaygain_synthesis__init_dither_context(&decoder_session->replaygain.dither_context, decoder_session->bps, decoder_session->replaygain.spec.noise_shaping);
				flac__utils_printf(stderr, 1, "%s: INFO: applying %s ReplayGain (gain=%0.2fdB+preamp=%0.1fdB, %s noise shaping, %s limiting) to output\n", decoder_session->inbasefilename, decoder_session->replaygain.spec.use_album_gain? "album":"track", gain, decoder_session->replaygain.spec.preamp, ns[decoder_session->replaygain.spec.noise_shaping], ls[decoder_session->replaygain.spec.limiter]);
				flac__utils_printf(stderr, 1, "%s: WARNING: applying ReplayGain is not lossless\n", decoder_session->inbasefilename);
			}
		}
	}
}

void error_callback(const void *decoder, FLAC__StreamDecoderErrorStatus status, void *client_data)
{
	DecoderSession *decoder_session = (DecoderSession*)client_data;
	(void)decoder;
	flac__utils_printf(stderr, 1, "%s: *** Got error code %d:%s\n", decoder_session->inbasefilename, status, FLAC__StreamDecoderErrorStatusString[status]);
	if(!decoder_session->continue_through_decode_errors)
		decoder_session->abort_flag = true;
}

void print_error_with_state(const DecoderSession *d, const char *message)
{
	const int ilen = strlen(d->inbasefilename) + 1;
	const char *state_string;

	flac__utils_printf(stderr, 1, "\n%s: %s\n", d->inbasefilename, message);

#ifdef FLAC__HAS_OGG
	if(d->is_ogg) {
		state_string = OggFLAC__file_decoder_get_resolved_state_string(d->decoder.ogg.file);
	}
	else
#endif
	{
		state_string = FLAC__file_decoder_get_resolved_state_string(d->decoder.flac.file);
	}

	flac__utils_printf(stderr, 1, "%*s state = %s\n", ilen, "", state_string);

	/* print out some more info for some errors: */
	if (0 == strcmp(state_string, FLAC__StreamDecoderStateString[FLAC__STREAM_DECODER_UNPARSEABLE_STREAM])) {
		flac__utils_printf(stderr, 1,
			"\n"
			"The FLAC stream may have been created by a more advanced encoder.  Try\n"
			"  metaflac --show-vendor-tag %s\n"
			"If the version number is greater than %s, this decoder is probably\n"
			"not able to decode the file.  If the version number is not, the file\n"
			"may be corrupted, or you may have found a bug.  In this case please\n"
			"submit a bug report to\n"
			"    http://sourceforge.net/bugs/?func=addbug&group_id=13478\n"
			"Make sure to include an email contact in the comment and/or use the\n"
			"\"Monitor\" feature to monitor the bug status.\n",
			d->inbasefilename, FLAC__VERSION_STRING
		);
	}
	else if (
		0 == strcmp(state_string, FLAC__FileDecoderStateString[FLAC__FILE_DECODER_ERROR_OPENING_FILE])
#ifdef FLAC__HAS_OGG
		|| 0 == strcmp(state_string, OggFLAC__FileDecoderStateString[OggFLAC__FILE_DECODER_ERROR_OPENING_FILE])
#endif
	) {
		flac__utils_printf(stderr, 1,
			"\n"
			"An error occurred opening the input file; it is likely that it does not exist\n"
			"or is not readable.\n"
		);
	}
}

void print_stats(const DecoderSession *decoder_session)
{
	if(flac__utils_verbosity_ >= 2) {
#if defined _MSC_VER || defined __MINGW32__
		/* with MSVC you have to spoon feed it the casting */
		const double progress = (double)(FLAC__int64)decoder_session->samples_processed / (double)(FLAC__int64)decoder_session->total_samples * 100.0;
#else
		const double progress = (double)decoder_session->samples_processed / (double)decoder_session->total_samples * 100.0;
#endif
		if(decoder_session->total_samples > 0) {
			fprintf(stderr, "\r%s: %s%u%% complete",
				decoder_session->inbasefilename,
				decoder_session->test_only? "testing, " : decoder_session->analysis_mode? "analyzing, " : "",
				(unsigned)floor(progress + 0.5)
			);
		}
		else {
			fprintf(stderr, "\r%s: %s %u samples",
				decoder_session->inbasefilename,
				decoder_session->test_only? "tested" : decoder_session->analysis_mode? "analyzed" : "wrote",
				(unsigned)decoder_session->samples_processed
			);
		}
	}
}

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