Split up into a function that can take a functor to give samples.

1 files changed, 101 insertions, 50 deletions

diff --git a/src/leqm-nrt.cc b/src/leqm-nrt.cc
index 9914b12..a4fdf40 100644
--- a/src/leqm-nrt.cc
+++ b/src/leqm-nrt.cc
@@ -46,6 +46,7 @@
 #include <iostream>
 #include <thread>
 #include <mutex>
+#include <functional>
 
 // Version 0.0.18 (C) Luca Trisciani 2011-2013, 2017-2018
 // Tool from the DCP-Werkstatt Software Bundle
@@ -289,7 +290,7 @@ struct Result
 };
 
 
-Result calculate(
+Result calculate_file(
 	std::string sound_filename,
 	std::vector<double> channel_corrections,
 	int buffer_size_ms,
@@ -301,6 +302,23 @@ Result calculate(
 	);
 
 
+Result calculate_function(
+	std::function<int64_t (double*, int64_t)> get_audio_data,
+	int channels,
+	int sample_rate,
+	int frames,
+	int bits_per_sample,
+	std::string log_prefix,
+	std::vector<double> channel_corrections,
+	int buffer_size_ms,
+	int number_of_filter_interpolation_points,
+	int num_cpu,
+	bool enable_leqm_log,
+	bool enable_leqm10_log,
+	bool measure_timing
+	);
+
+
 int main(int argc, const char ** argv)
 {
 	int number_of_filter_interpolation_points = 64; // This value is low for precision. Calibration is done with 32768 point.
@@ -417,7 +435,7 @@ int main(int argc, const char ** argv)
 		}
 	}
 
-	auto result = calculate(sound_filename, channel_corrections, buffer_size_ms, number_of_filter_interpolation_points, num_cpu, enable_leqm_log, enable_leqm10_log, measure_timing);
+	auto result = calculate_file(sound_filename, channel_corrections, buffer_size_ms, number_of_filter_interpolation_points, num_cpu, enable_leqm_log, enable_leqm10_log, measure_timing);
 
 	if (display_leqnw) {
 		printf("Leq(nW): %.4f\n", result.leq_nw); // Leq(no Weighting)
@@ -427,7 +445,7 @@ int main(int argc, const char ** argv)
 	return result.status;
 }
 
-Result calculate(
+Result calculate_file(
 	std::string sound_filename,
 	std::vector<double> channel_corrections,
 	int buffer_size_ms,
@@ -438,10 +456,6 @@ Result calculate(
 	bool measure_timing
 	)
 {
-	struct timespec starttime;
-	double * shorttermaveragedarray = nullptr;
-	int constexpr origpoints = 21; //number of points in the standard CCIR filter
-
 	SF_INFO sf_info;
 	sf_info.format = 0;
 	SNDFILE* file = sf_open(sound_filename.c_str(), SFM_READ, &sf_info);
@@ -449,17 +463,81 @@ Result calculate(
 		return {-101};
 	}
 
+	int bitdepth = 0;
+	switch(sf_info.format & SF_FORMAT_SUBMASK) {
+		// all signed bitdepth
+		case 0x0001:
+			bitdepth = 8;
+			break;
+		case 0x0002:
+			bitdepth = 16;
+			break;
+		case 0x0003:
+			bitdepth = 24;
+			break;
+		case 0x0004:
+			bitdepth = 32;
+			break;
+		default:
+			printf("No known bitdepth! Exiting ...\n");
+			return -1;
+	}
+
+	auto result = calculate_function(
+		[file](double* buffer, int64_t samples) -> int64_t {
+			return sf_read_double(file, buffer, samples);
+		},
+		sf_info.channels,
+		sf_info.samplerate,
+		sf_info.frames,
+		bitdepth,
+		sound_filename,
+		channel_corrections,
+		buffer_size_ms,
+		number_of_filter_interpolation_points,
+		num_cpu,
+		enable_leqm_log,
+		enable_leqm10_log,
+		measure_timing
+		);
+
+	sf_close(file);
+
+	return result;
+}
+
+
+Result calculate_function(
+	std::function<int64_t (double*, int64_t)> get_audio_data,
+	int channels,
+	int sample_rate,
+	int frames,
+	int bits_per_sample,
+	std::string log_prefix,
+	std::vector<double> channel_corrections,
+	int buffer_size_ms,
+	int number_of_filter_interpolation_points,
+	int num_cpu,
+	bool enable_leqm_log,
+	bool enable_leqm10_log,
+	bool measure_timing
+	)
+{
+	struct timespec starttime;
+	double * shorttermaveragedarray = nullptr;
+	int constexpr origpoints = 21; //number of points in the standard CCIR filter
+
 	std::vector<double> channel_conf_cal;
 
 	//postprocessing parameters
-	if (static_cast<int>(channel_corrections.size()) == sf_info.channels) {
+	if (static_cast<int>(channel_corrections.size()) == channels) {
 		for (auto i: channel_corrections) {
 			channel_conf_cal.push_back(convloglin_single(i));
 
 		}
-	} else if (channel_corrections.empty() && sf_info.channels == 6) {
+	} else if (channel_corrections.empty() && channels == 6) {
 		double conf51[] = {0, 0, 0, 0, -3, -3};
-		for (auto cind = 0; cind < sf_info.channels; cind++) {
+		for (auto cind = 0; cind < channels; cind++) {
 			channel_conf_cal.push_back(convloglin_single(conf51[cind]));
 		}
 	} else {
@@ -468,14 +546,14 @@ Result calculate(
 
 	FILE *leqm10logfile = nullptr;
 	if (enable_leqm10_log) {
-		std::string log_filename = sound_filename + ".leqm10.txt";
+		std::string log_filename = log_prefix + ".leqm10.txt";
 		leqm10logfile = fopen(log_filename.c_str(), "w");
 		/* If that failed we just won't get any log */
 	}
 
 	FILE *leqmlogfile = nullptr;
 	if (enable_leqm_log) {
-		std::string log_filename = sound_filename + ".leqmlog.txt";
+		std::string log_filename = log_prefix + ".leqmlog.txt";
 		leqmlogfile = fopen(log_filename.c_str(), "w");
 		/* If that failed we just won't get any log */
 	}
@@ -484,28 +562,28 @@ Result calculate(
 		clock_gettime(CLOCK_MONOTONIC, &starttime);
 	}
 
-	if ((sf_info.samplerate * buffer_size_ms) % 1000) {
+	if ((sample_rate * buffer_size_ms) % 1000) {
 		return -102;
 	}
 
-	int buffer_size_samples = (sf_info.samplerate * sf_info.channels * buffer_size_ms) / 1000;
+	int buffer_size_samples = (sample_rate * channels * buffer_size_ms) / 1000;
 	std::vector<double> buffer(buffer_size_samples);
 
 	int numbershortperiods = 0;
 	if (enable_leqm10_log) {
 		//if duration < 10 mm exit
-		double featdursec = sf_info.frames / sf_info.samplerate;
+		double featdursec = frames / sample_rate;
 		if ((featdursec/60.0) < 10.0) {
 			printf("The audio file is too short to measure Leq(m10).\n");
 			return 0;
 		}
 
 		//how many short periods in overall duration
-		int remainder = sf_info.frames % (sf_info.samplerate*buffer_size_ms/1000);
+		int remainder = frames % (sample_rate*buffer_size_ms/1000);
 		if (remainder == 0) {
-			numbershortperiods = sf_info.frames/(sf_info.samplerate*buffer_size_ms/1000);
+			numbershortperiods = frames/(sample_rate*buffer_size_ms/1000);
 		} else {
-			numbershortperiods = sf_info.frames/(sf_info.samplerate*buffer_size_ms/1000) + 1;
+			numbershortperiods = frames/(sample_rate*buffer_size_ms/1000) + 1;
 		}
 
 		shorttermaveragedarray = (double *) malloc(sizeof(*shorttermaveragedarray)*numbershortperiods);
@@ -521,32 +599,7 @@ Result calculate(
 	std::vector<double> eqfreqresp(number_of_filter_interpolation_points);
 	std::vector<double> ir(number_of_filter_interpolation_points * 2);
 
-	// And what to do for floating point sample coding?
-
-	int bitdepth = 0;
-	switch(sf_info.format & SF_FORMAT_SUBMASK) {
-		// all signed bitdepth
-		case 0x0001:
-			bitdepth = 8;
-			break;
-		case 0x0002:
-			bitdepth = 16;
-			break;
-		case 0x0003:
-			bitdepth = 24;
-			break;
-		case 0x0004:
-			bitdepth = 32;
-			break;
-		default:
-			printf("No known bitdepth! Exiting ...\n");
-			return -1;
-	}
-
-
-
-
-	equalinterval2(freqsamples, freqresp_db, eqfreqsamples, eqfreqresp_db, number_of_filter_interpolation_points, sf_info.samplerate, origpoints, bitdepth);
+	equalinterval2(freqsamples, freqresp_db, eqfreqsamples, eqfreqresp_db, number_of_filter_interpolation_points, sample_rate, origpoints, bits_per_sample);
 	convloglin(eqfreqresp_db, eqfreqresp, number_of_filter_interpolation_points);
 
 #ifdef DEBUG
@@ -570,12 +623,12 @@ Result calculate(
 	int staindex = 0; //shorttermarrayindex
 
 
-	while((samples_read = sf_read_double(file, buffer.data(), buffer_size_samples)) > 0) {
+	while ((samples_read = get_audio_data(buffer.data(), buffer_size_samples)) > 0) {
 		worker_args.push_back(
 			std::make_shared<Worker>(
 				buffer,
 				samples_read,
-				sf_info.channels,
+				channels,
 				number_of_filter_interpolation_points,
 				ir,
 				&totsum,
@@ -593,7 +646,7 @@ Result calculate(
 			worker_args.clear();
 			//simply log here your measurement it will be a multiple of your threads and your buffer
 			if (leqmlogfile) {
-				logleqm(leqmlogfile, ((double) totsum.nsamples())/((double) sf_info.samplerate), totsum);
+				logleqm(leqmlogfile, ((double) totsum.nsamples())/((double) sample_rate), totsum);
 			} //endlog
 		}
 	}
@@ -603,7 +656,7 @@ Result calculate(
 			worker_args.clear();
 		}
 		if (leqmlogfile) {
-			logleqm(leqmlogfile, ((double) totsum.nsamples())/((double) sf_info.samplerate), totsum );
+			logleqm(leqmlogfile, ((double) totsum.nsamples())/((double) sample_rate), totsum );
 		}
 	}
 
@@ -669,8 +722,6 @@ Result calculate(
 		fclose(leqmlogfile);
 	}
 
-	sf_close(file);
-
 	return result;
 }