Make number of cards flexible at runtime.

[nageru] / nageru / resampling_queue.cpp

diff --git a/nageru/resampling_queue.cpp b/nageru/resampling_queue.cpp
index ef7b73511ff596f7e93894fd703038ad370c8841..045d5c7b0f28460db1700e47b10aff339bbbed66 100644 (file)
--- 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.