using namespace bmusb;
using namespace std;
+using namespace std::chrono;
using namespace std::placeholders;
namespace {
limiter(OUTPUT_FREQUENCY),
correlation(OUTPUT_FREQUENCY)
{
- global_audio_mixer = this;
-
for (unsigned bus_index = 0; bus_index < MAX_BUSES; ++bus_index) {
locut[bus_index].init(FILTER_HPF, 2);
eq[bus_index][EQ_BAND_BASS].init(FILTER_LOW_SHELF, 1);
}
set_limiter_enabled(global_flags.limiter_enabled);
set_final_makeup_gain_auto(global_flags.final_makeup_gain_auto);
+
+ r128.init(2, OUTPUT_FREQUENCY);
+ r128.integr_start();
+
+ // hlen=16 is pretty low quality, but we use quite a bit of CPU otherwise,
+ // and there's a limit to how important the peak meter is.
+ peak_resampler.setup(OUTPUT_FREQUENCY, OUTPUT_FREQUENCY * 4, /*num_channels=*/2, /*hlen=*/16, /*frel=*/1.0);
+
+ global_audio_mixer = this;
alsa_pool.init();
if (!global_flags.input_mapping_filename.empty()) {
+ // Must happen after ALSAPool is initialized, as it needs to know the card list.
current_mapping_mode = MappingMode::MULTICHANNEL;
InputMapping new_input_mapping;
if (!load_input_mapping_from_file(get_devices(),
current_mapping_mode = MappingMode::MULTICHANNEL;
}
}
-
- r128.init(2, OUTPUT_FREQUENCY);
- r128.integr_start();
-
- // hlen=16 is pretty low quality, but we use quite a bit of CPU otherwise,
- // and there's a limit to how important the peak meter is.
- peak_resampler.setup(OUTPUT_FREQUENCY, OUTPUT_FREQUENCY * 4, /*num_channels=*/2, /*hlen=*/16, /*frel=*/1.0);
}
void AudioMixer::reset_resampler(DeviceSpec device_spec)
} else {
// TODO: ResamplingQueue should probably take the full device spec.
// (It's only used for console output, though.)
- device->resampling_queue.reset(new ResamplingQueue(device_spec.index, device->capture_frequency, OUTPUT_FREQUENCY, device->interesting_channels.size()));
+ device->resampling_queue.reset(new ResamplingQueue(
+ device_spec.index, device->capture_frequency, OUTPUT_FREQUENCY, device->interesting_channels.size(),
+ global_flags.audio_queue_length_ms * 0.001));
}
- device->next_local_pts = 0;
}
-bool AudioMixer::add_audio(DeviceSpec device_spec, const uint8_t *data, unsigned num_samples, AudioFormat audio_format, int64_t frame_length)
+bool AudioMixer::add_audio(DeviceSpec device_spec, const uint8_t *data, unsigned num_samples, AudioFormat audio_format, int64_t frame_length, steady_clock::time_point frame_time)
{
AudioDevice *device = find_audio_device(device_spec);
}
// Now add it.
- int64_t local_pts = device->next_local_pts;
- device->resampling_queue->add_input_samples(local_pts / double(TIMEBASE), audio.get(), num_samples);
- device->next_local_pts = local_pts + frame_length;
+ device->resampling_queue->add_input_samples(frame_time, audio.get(), num_samples, ResamplingQueue::ADJUST_RATE);
return true;
}
vector<float> silence(samples_per_frame * num_channels, 0.0f);
for (unsigned i = 0; i < num_frames; ++i) {
- device->resampling_queue->add_input_samples(device->next_local_pts / double(TIMEBASE), silence.data(), samples_per_frame);
- // Note that if the format changed in the meantime, we have
- // no way of detecting that; we just have to assume the frame length
- // is always the same.
- device->next_local_pts += frame_length;
+ device->resampling_queue->add_input_samples(steady_clock::now(), silence.data(), samples_per_frame, ResamplingQueue::DO_NOT_ADJUST_RATE);
}
return true;
}
} // namespace
-vector<float> AudioMixer::get_output(double pts, unsigned num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy)
+vector<float> AudioMixer::get_output(steady_clock::time_point ts, unsigned num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy)
{
map<DeviceSpec, vector<float>> samples_card;
vector<float> samples_bus;
memset(&samples_card[device_spec][0], 0, samples_card[device_spec].size() * sizeof(float));
} else {
device->resampling_queue->get_output_samples(
- pts,
+ ts,
&samples_card[device_spec][0],
num_samples,
rate_adjustment_policy);