X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=nageru%2Fresampling_queue.cpp;h=045d5c7b0f28460db1700e47b10aff339bbbed66;hb=ecaec75dd52d076ba53cafa1fed716ebc0d93da6;hp=ef7b73511ff596f7e93894fd703038ad370c8841;hpb=9b7d691b4cc5db7dbfc18c82e86c1207fcac4722;p=nageru

diff --git a/nageru/resampling_queue.cpp b/nageru/resampling_queue.cpp
index ef7b735..045d5c7 100644
--- a/nageru/resampling_queue.cpp
+++ b/nageru/resampling_queue.cpp
@@ -30,8 +30,8 @@
 using namespace std;
 using namespace std::chrono;
 
-ResamplingQueue::ResamplingQueue(DeviceSpec device_spec, unsigned freq_in, unsigned freq_out, unsigned num_channels, double expected_delay_seconds)
-	: device_spec(device_spec), freq_in(freq_in), freq_out(freq_out), num_channels(num_channels),
+ResamplingQueue::ResamplingQueue(const std::string &debug_description, unsigned freq_in, unsigned freq_out, unsigned num_channels, double expected_delay_seconds)
+	: debug_description(debug_description), freq_in(freq_in), freq_out(freq_out), num_channels(num_channels),
 	  current_estimated_freq_in(freq_in),
 	  ratio(double(freq_out) / double(freq_in)), expected_delay(expected_delay_seconds * OUTPUT_FREQUENCY)
 {
@@ -62,7 +62,7 @@ void ResamplingQueue::add_input_samples(steady_clock::time_point ts, const float
 		current_estimated_freq_in = (a1.input_samples_received - a0.input_samples_received) / duration<double>(a1.ts - a0.ts).count();
 		if (!(current_estimated_freq_in >= 0.0)) {
 			fprintf(stderr, "%s: PANIC: Input audio clock going backwards, ignoring.\n",
-				spec_to_string(device_spec).c_str());
+				debug_description.c_str());
 			current_estimated_freq_in = freq_in;
 		}
 
@@ -115,7 +115,7 @@ bool ResamplingQueue::get_output_samples(steady_clock::time_point ts, float *sam
 			// so that we don't need a long period to stabilize at the beginning.
 			if (err < 0.0) {
 				int delay_samples_to_add = lrintf(-err);
-				for (ssize_t i = 0; i < delay_samples_to_add * num_channels; ++i) {
+				for (ssize_t i = 0; i < delay_samples_to_add * int(num_channels); ++i) {
 					buffer.push_front(0.0f);
 				}
 				total_consumed_samples -= delay_samples_to_add;  // Equivalent to increasing input_samples_received on a0 and a1.
@@ -143,7 +143,7 @@ bool ResamplingQueue::get_output_samples(steady_clock::time_point ts, float *sam
 		// (we start ResamplingQueues also when we e.g. switch sound sources),
 		// but in general, a little bit of increased timing jitter is acceptable
 		// right after a setup change like this.
-		double loop_bandwidth_hz = (total_consumed_samples < 4 * freq_in) ? 0.2 : 0.02;
+		double loop_bandwidth_hz = (total_consumed_samples < 4 * int(freq_in)) ? 0.2 : 0.02;
 
 		// Set filters. The first filter much wider than the first one (20x as wide).
 		double w = (2.0 * M_PI) * loop_bandwidth_hz * num_samples / freq_out;
@@ -170,7 +170,7 @@ bool ResamplingQueue::get_output_samples(steady_clock::time_point ts, float *sam
 			// This should never happen unless delay is set way too low,
 			// or we're dropping a lot of data.
 			fprintf(stderr, "%s: PANIC: Out of input samples to resample, still need %d output samples! (correction factor is %f)\n",
-				spec_to_string(device_spec).c_str(), int(vresampler.out_count), rcorr);
+				debug_description.c_str(), int(vresampler.out_count), rcorr);
 			memset(vresampler.out_data, 0, vresampler.out_count * num_channels * sizeof(float));
 
 			// Reset the loop filter.