X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=resampling_queue.cpp;h=06dc6fbf417bb1a13aae39f4cbf527e7740e5b6f;hb=327534a3031a332423411c9599c741f2f81657df;hp=9e8fb7d58da6d85049c3a895780c9bc2eed8578b;hpb=c0bf9deb26205bf35758d49f587961f19bdb15b8;p=nageru diff --git a/resampling_queue.cpp b/resampling_queue.cpp index 9e8fb7d..06dc6fb 100644 --- a/resampling_queue.cpp +++ b/resampling_queue.cpp @@ -20,15 +20,20 @@ #include "resampling_queue.h" #include -#include -#include #include +#include #include #include +#include +#include -ResamplingQueue::ResamplingQueue(unsigned card_num, unsigned freq_in, unsigned freq_out, unsigned num_channels) +using namespace std; +using namespace std::chrono; + +ResamplingQueue::ResamplingQueue(unsigned card_num, unsigned freq_in, unsigned freq_out, unsigned num_channels, double expected_delay_seconds) : card_num(card_num), freq_in(freq_in), freq_out(freq_out), num_channels(num_channels), - ratio(double(freq_out) / double(freq_in)) + current_estimated_freq_in(freq_in), + ratio(double(freq_out) / double(freq_in)), expected_delay(expected_delay_seconds * OUTPUT_FREQUENCY) { vresampler.setup(ratio, num_channels, /*hlen=*/32); @@ -38,91 +43,122 @@ ResamplingQueue::ResamplingQueue(unsigned card_num, unsigned freq_in, unsigned f vresampler.process (); } -void ResamplingQueue::add_input_samples(double pts, const float *samples, ssize_t num_samples) +void ResamplingQueue::add_input_samples(steady_clock::time_point ts, const float *samples, ssize_t num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy) { if (num_samples == 0) { return; } - if (first_input) { - // Synthesize a fake length. - last_input_len = double(num_samples) / freq_in; - first_input = false; - } else { - last_input_len = pts - last_input_pts; - } - - last_input_pts = pts; - k_a0 = k_a1; - k_a1 += num_samples; + assert(duration(ts.time_since_epoch()).count() >= 0.0); - for (ssize_t i = 0; i < num_samples * num_channels; ++i) { - buffer.push_back(samples[i]); + bool good_sample = (rate_adjustment_policy == ADJUST_RATE); + if (good_sample && a1.good_sample) { + a0 = a1; } + a1.ts = ts; + a1.input_samples_received += num_samples; + a1.good_sample = good_sample; + if (a0.good_sample && a1.good_sample) { + current_estimated_freq_in = (a1.input_samples_received - a0.input_samples_received) / duration(a1.ts - a0.ts).count(); + assert(current_estimated_freq_in >= 0.0); + + // Bound the frequency, so that a single wild result won't throw the filter off guard. + current_estimated_freq_in = min(current_estimated_freq_in, 1.2 * freq_in); + current_estimated_freq_in = max(current_estimated_freq_in, 0.8 * freq_in); + } + + buffer.insert(buffer.end(), samples, samples + num_samples * num_channels); } -bool ResamplingQueue::get_output_samples(double pts, float *samples, ssize_t num_samples) +bool ResamplingQueue::get_output_samples(steady_clock::time_point ts, float *samples, ssize_t num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy) { - if (first_input) { + assert(num_samples > 0); + if (a1.input_samples_received == 0) { // No data yet, just return zeros. - memset(samples, 0, num_samples * 2 * sizeof(float)); + memset(samples, 0, num_samples * num_channels * sizeof(float)); return true; } - double last_output_len; - if (first_output) { - // Synthesize a fake length. - last_output_len = double(num_samples) / freq_out; - } else { - last_output_len = pts - last_output_pts; + // This can happen when we get dropped frames on the master card. + if (duration(ts.time_since_epoch()).count() <= 0.0) { + rate_adjustment_policy = DO_NOT_ADJUST_RATE; } - last_output_pts = pts; - - // Using the time point since just before the last call to add_input_samples() as a base, - // estimate actual delay based on activity since then, measured in number of input samples: - double actual_delay = 0.0; - assert(last_input_len != 0); - actual_delay += (k_a1 - k_a0) * last_output_len / last_input_len; // Inserted samples since k_a0, rescaled for the different time periods. - actual_delay += k_a0 - total_consumed_samples; // Samples inserted before k_a0 but not consumed yet. - actual_delay += vresampler.inpdist(); // Delay in the resampler itself. - double err = actual_delay - expected_delay; - if (first_output && err < 0.0) { - // Before the very first block, insert artificial delay based on our initial estimate, - // so that we don't need a long period to stabilize at the beginning. - int delay_samples_to_add = lrintf(-err); - for (ssize_t i = 0; i < delay_samples_to_add * num_channels; ++i) { - buffer.push_front(0.0f); + + if (rate_adjustment_policy == ADJUST_RATE && (a0.good_sample || a1.good_sample)) { + // Estimate the current number of input samples produced at + // this instant in time, by extrapolating from the last known + // good point. Note that we could be extrapolating backward or + // forward, depending on the timing of the calls. + const InputPoint &base_point = a1.good_sample ? a1 : a0; + assert(duration(base_point.ts.time_since_epoch()).count() >= 0.0); + + // NOTE: Due to extrapolation, input_samples_received can + // actually go negative here the few first calls (ie., we asked + // about a timestamp where we hadn't actually started producing + // samples yet), but that is harmless. + const double input_samples_received = base_point.input_samples_received + + current_estimated_freq_in * duration(ts - base_point.ts).count(); + + // Estimate the number of input samples _consumed_ after we've run the resampler. + const double input_samples_consumed = total_consumed_samples + + num_samples / (ratio * rcorr); + + double actual_delay = input_samples_received - input_samples_consumed; + actual_delay += vresampler.inpdist(); // Delay in the resampler itself. + double err = actual_delay - expected_delay; + if (first_output) { + // Before the very first block, insert artificial delay based on our initial estimate, + // 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) { + buffer.push_front(0.0f); + } + total_consumed_samples -= delay_samples_to_add; // Equivalent to increasing input_samples_received on a0 and a1. + err += delay_samples_to_add; + } else if (err > 0.0) { + int delay_samples_to_remove = min(lrintf(err), buffer.size() / num_channels); + buffer.erase(buffer.begin(), buffer.begin() + delay_samples_to_remove * num_channels); + total_consumed_samples += delay_samples_to_remove; + err -= delay_samples_to_remove; + } } - total_consumed_samples -= delay_samples_to_add; // Equivalent to increasing k_a0 and k_a1. - err += delay_samples_to_add; first_output = false; + + // Compute loop filter coefficients for the two filters. We need to compute them + // every time, since they depend on the number of samples the user asked for. + // + // The loop bandwidth is at 0.02 Hz; our jitter is pretty large + // since none of the threads involved run at real-time priority. + // However, the first four seconds, we use a larger loop bandwidth (2 Hz), + // because there's a lot going on during startup, and thus the + // initial estimate might be tainted by jitter during that phase, + // and we want to converge faster. + // + // NOTE: The above logic might only hold during Nageru startup + // (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; + + // 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; + double w0 = 1.0 - exp(-20.0 * w); + double w1 = w * 1.5 / num_samples / ratio; + double w2 = w / 1.5; + + // Filter through the loop filter to find the correction ratio. + z1 += w0 * (w1 * err - z1); + z2 += w0 * (z1 - z2); + z3 += w2 * z2; + rcorr = 1.0 - z2 - z3; + if (rcorr > 1.05) rcorr = 1.05; + if (rcorr < 0.95) rcorr = 0.95; + assert(!isnan(rcorr)); + vresampler.set_rratio(rcorr); } - // Compute loop filter coefficients for the two filters. We need to compute them - // every time, since they depend on the number of samples the user asked for. - // - // The loop bandwidth is at 0.02 Hz; we trust the initial estimate quite well, - // and our jitter is pretty large since none of the threads involved run at - // real-time priority. - double loop_bandwidth_hz = 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; - double w0 = 1.0 - exp(-20.0 * w); - double w1 = w * 1.5 / num_samples / ratio; - double w2 = w / 1.5; - - // Filter through the loop filter to find the correction ratio. - z1 += w0 * (w1 * err - z1); - z2 += w0 * (z1 - z2); - z3 += w2 * z2; - double rcorr = 1.0 - z2 - z3; - if (rcorr > 1.05) rcorr = 1.05; - if (rcorr < 0.95) rcorr = 0.95; - assert(!isnan(rcorr)); - vresampler.set_rratio(rcorr); - - // Finally actually resample, consuming exactly output samples. + // Finally actually resample, producing exactly output samples. vresampler.out_data = samples; vresampler.out_count = num_samples; while (vresampler.out_count > 0) { @@ -131,7 +167,11 @@ bool ResamplingQueue::get_output_samples(double pts, float *samples, ssize_t num // or we're dropping a lot of data. fprintf(stderr, "Card %u: PANIC: Out of input samples to resample, still need %d output samples! (correction factor is %f)\n", card_num, int(vresampler.out_count), rcorr); - memset(vresampler.out_data, 0, vresampler.out_count * 2 * sizeof(float)); + memset(vresampler.out_data, 0, vresampler.out_count * num_channels * sizeof(float)); + + // Reset the loop filter. + z1 = z2 = z3 = 0.0; + return false; } @@ -140,14 +180,13 @@ bool ResamplingQueue::get_output_samples(double pts, float *samples, ssize_t num if (num_input_samples * num_channels > buffer.size()) { num_input_samples = buffer.size() / num_channels; } - for (size_t i = 0; i < num_input_samples * num_channels; ++i) { - inbuf[i] = buffer[i]; - } + copy(buffer.begin(), buffer.begin() + num_input_samples * num_channels, inbuf); vresampler.inp_count = num_input_samples; vresampler.inp_data = inbuf; - vresampler.process(); + int err = vresampler.process(); + assert(err == 0); size_t consumed_samples = num_input_samples - vresampler.inp_count; total_consumed_samples += consumed_samples;