#include "db.h"
#include "flags.h"
+#include "mixer.h"
+#include "state.pb.h"
#include "timebase.h"
using namespace bmusb;
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);
locut_enabled[bus_index] = global_flags.locut_enabled;
+ eq[bus_index][EQ_BAND_BASS].init(FILTER_LOW_SHELF, 1);
+ // Note: EQ_BAND_MID isn't used (see comments in apply_eq()).
+ eq[bus_index][EQ_BAND_TREBLE].init(FILTER_HIGH_SHELF, 1);
+
gain_staging_db[bus_index] = global_flags.initial_gain_staging_db;
compressor[bus_index].reset(new StereoCompressor(OUTPUT_FREQUENCY));
compressor_threshold_dbfs[bus_index] = ref_level_dbfs - 12.0f; // -12 dB.
}
set_limiter_enabled(global_flags.limiter_enabled);
set_final_makeup_gain_auto(global_flags.final_makeup_gain_auto);
-
- // Generate a very simple, default input mapping.
- InputMapping::Bus input;
- input.name = "Main";
- input.device.type = InputSourceType::CAPTURE_CARD;
- input.device.index = 0;
- input.source_channel[0] = 0;
- input.source_channel[1] = 1;
+ alsa_pool.init();
InputMapping new_input_mapping;
- new_input_mapping.buses.push_back(input);
- set_input_mapping(new_input_mapping);
+ if (!global_flags.input_mapping_filename.empty()) {
+ if (!load_input_mapping_from_file(get_devices(),
+ global_flags.input_mapping_filename,
+ &new_input_mapping)) {
+ fprintf(stderr, "Failed to load input mapping from '%s', exiting.\n",
+ global_flags.input_mapping_filename.c_str());
+ exit(1);
+ }
+ } else {
+ // Generate a very simple, default input mapping.
+ InputMapping::Bus input;
+ input.name = "Main";
+ input.device.type = InputSourceType::CAPTURE_CARD;
+ input.device.index = 0;
+ input.source_channel[0] = 0;
+ input.source_channel[1] = 1;
- // Look for ALSA cards.
- available_alsa_cards = ALSAInput::enumerate_devices();
+ new_input_mapping.buses.push_back(input);
+ }
+ set_input_mapping(new_input_mapping);
r128.init(2, OUTPUT_FREQUENCY);
r128.integr_start();
peak_resampler.setup(OUTPUT_FREQUENCY, OUTPUT_FREQUENCY * 4, /*num_channels=*/2, /*hlen=*/16, /*frel=*/1.0);
}
-AudioMixer::~AudioMixer()
-{
- for (unsigned card_index = 0; card_index < available_alsa_cards.size(); ++card_index) {
- const AudioDevice &device = alsa_inputs[card_index];
- if (device.alsa_device != nullptr) {
- device.alsa_device->stop_capture_thread();
- }
- }
-}
-
-
void AudioMixer::reset_resampler(DeviceSpec device_spec)
{
lock_guard<timed_mutex> lock(audio_mutex);
device->next_local_pts = 0;
}
-void AudioMixer::reset_alsa_mutex_held(DeviceSpec device_spec)
-{
- assert(device_spec.type == InputSourceType::ALSA_INPUT);
- unsigned card_index = device_spec.index;
- AudioDevice *device = find_audio_device(device_spec);
-
- if (device->alsa_device != nullptr) {
- device->alsa_device->stop_capture_thread();
- }
- if (device->interesting_channels.empty()) {
- device->alsa_device.reset();
- } else {
- const ALSAInput::Device &alsa_dev = available_alsa_cards[card_index];
- device->alsa_device.reset(new ALSAInput(alsa_dev.address.c_str(), OUTPUT_FREQUENCY, alsa_dev.num_channels, bind(&AudioMixer::add_audio, this, device_spec, _1, _2, _3, _4)));
- device->capture_frequency = device->alsa_device->get_sample_rate();
- device->alsa_device->start_capture_thread();
- }
-}
-
bool AudioMixer::add_audio(DeviceSpec device_spec, const uint8_t *data, unsigned num_samples, AudioFormat audio_format, int64_t frame_length)
{
AudioDevice *device = find_audio_device(device_spec);
return true;
}
+bool AudioMixer::silence_card(DeviceSpec device_spec, bool silence)
+{
+ AudioDevice *device = find_audio_device(device_spec);
+
+ unique_lock<timed_mutex> lock(audio_mutex, defer_lock);
+ if (!lock.try_lock_for(chrono::milliseconds(10))) {
+ return false;
+ }
+
+ if (device->silenced && !silence) {
+ reset_resampler_mutex_held(device_spec);
+ }
+ device->silenced = silence;
+ return true;
+}
+
AudioMixer::AudioDevice *AudioMixer::find_audio_device(DeviceSpec device)
{
switch (device.type) {
}
}
+vector<DeviceSpec> AudioMixer::get_active_devices() const
+{
+ vector<DeviceSpec> ret;
+ for (unsigned card_index = 0; card_index < MAX_VIDEO_CARDS; ++card_index) {
+ const DeviceSpec device_spec{InputSourceType::CAPTURE_CARD, card_index};
+ if (!find_audio_device(device_spec)->interesting_channels.empty()) {
+ ret.push_back(device_spec);
+ }
+ }
+ for (unsigned card_index = 0; card_index < MAX_ALSA_CARDS; ++card_index) {
+ const DeviceSpec device_spec{InputSourceType::ALSA_INPUT, card_index};
+ if (!find_audio_device(device_spec)->interesting_channels.empty()) {
+ ret.push_back(device_spec);
+ }
+ }
+ return ret;
+}
+
vector<float> AudioMixer::get_output(double pts, unsigned num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy)
{
map<DeviceSpec, vector<float>> samples_card;
lock_guard<timed_mutex> lock(audio_mutex);
// Pick out all the interesting channels from all the cards.
- // TODO: If the card has been hotswapped, the number of channels
- // might have changed; if so, we need to do some sort of remapping
- // to silence.
- for (const auto &spec_and_info : get_devices_mutex_held()) {
- const DeviceSpec &device_spec = spec_and_info.first;
+ for (const DeviceSpec &device_spec : get_active_devices()) {
AudioDevice *device = find_audio_device(device_spec);
- if (!device->interesting_channels.empty()) {
- samples_card[device_spec].resize(num_samples * device->interesting_channels.size());
+ samples_card[device_spec].resize(num_samples * device->interesting_channels.size());
+ if (device->silenced) {
+ memset(&samples_card[device_spec][0], 0, samples_card[device_spec].size() * sizeof(float));
+ } else {
device->resampling_queue->get_output_samples(
pts,
&samples_card[device_spec][0],
samples_bus.resize(num_samples * 2);
for (unsigned bus_index = 0; bus_index < input_mapping.buses.size(); ++bus_index) {
fill_audio_bus(samples_card, input_mapping.buses[bus_index], num_samples, &samples_bus[0]);
-
- // Cut away everything under 120 Hz (or whatever the cutoff is);
- // we don't need it for voice, and it will reduce headroom
- // and confuse the compressor. (In particular, any hums at 50 or 60 Hz
- // should be dampened.)
- if (locut_enabled[bus_index]) {
- locut[bus_index].render(samples_bus.data(), samples_bus.size() / 2, locut_cutoff_hz * 2.0 * M_PI / OUTPUT_FREQUENCY, 0.5f);
- }
+ apply_eq(bus_index, &samples_bus);
{
lock_guard<mutex> lock(compressor_mutex);
}
}
- float volume = from_db(fader_volume_db[bus_index]);
- if (bus_index == 0) {
- for (unsigned i = 0; i < num_samples * 2; ++i) {
- samples_out[i] = samples_bus[i] * volume;
- }
- } else {
- for (unsigned i = 0; i < num_samples * 2; ++i) {
- samples_out[i] += samples_bus[i] * volume;
- }
- }
-
+ add_bus_to_master(bus_index, samples_bus, &samples_out);
deinterleave_samples(samples_bus, &left, &right);
- measure_bus_levels(bus_index, left, right, volume);
+ measure_bus_levels(bus_index, left, right);
}
{
return samples_out;
}
-void AudioMixer::measure_bus_levels(unsigned bus_index, const vector<float> &left, const vector<float> &right, float volume)
+void AudioMixer::apply_eq(unsigned bus_index, vector<float> *samples_bus)
+{
+ constexpr float bass_freq_hz = 200.0f;
+ constexpr float treble_freq_hz = 4700.0f;
+
+ // Cut away everything under 120 Hz (or whatever the cutoff is);
+ // we don't need it for voice, and it will reduce headroom
+ // and confuse the compressor. (In particular, any hums at 50 or 60 Hz
+ // should be dampened.)
+ if (locut_enabled[bus_index]) {
+ locut[bus_index].render(samples_bus->data(), samples_bus->size() / 2, locut_cutoff_hz * 2.0 * M_PI / OUTPUT_FREQUENCY, 0.5f);
+ }
+
+ // Apply the rest of the EQ. Since we only have a simple three-band EQ,
+ // we can implement it with two shelf filters. We use a simple gain to
+ // set the mid-level filter, and then offset the low and high bands
+ // from that if we need to. (We could perhaps have folded the gain into
+ // the next part, but it's so cheap that the trouble isn't worth it.)
+ if (fabs(eq_level_db[bus_index][EQ_BAND_MID]) > 0.01f) {
+ float g = from_db(eq_level_db[bus_index][EQ_BAND_MID]);
+ for (size_t i = 0; i < samples_bus->size(); ++i) {
+ (*samples_bus)[i] *= g;
+ }
+ }
+
+ float bass_adj_db = eq_level_db[bus_index][EQ_BAND_BASS] - eq_level_db[bus_index][EQ_BAND_MID];
+ if (fabs(bass_adj_db) > 0.01f) {
+ eq[bus_index][EQ_BAND_BASS].render(samples_bus->data(), samples_bus->size() / 2,
+ bass_freq_hz * 2.0 * M_PI / OUTPUT_FREQUENCY, 0.5f, bass_adj_db / 40.0f);
+ }
+
+ float treble_adj_db = eq_level_db[bus_index][EQ_BAND_TREBLE] - eq_level_db[bus_index][EQ_BAND_MID];
+ if (fabs(treble_adj_db) > 0.01f) {
+ eq[bus_index][EQ_BAND_TREBLE].render(samples_bus->data(), samples_bus->size() / 2,
+ treble_freq_hz * 2.0 * M_PI / OUTPUT_FREQUENCY, 0.5f, treble_adj_db / 40.0f);
+ }
+}
+
+void AudioMixer::add_bus_to_master(unsigned bus_index, const vector<float> &samples_bus, vector<float> *samples_out)
+{
+ assert(samples_bus.size() == samples_out->size());
+ assert(samples_bus.size() % 2 == 0);
+ unsigned num_samples = samples_bus.size() / 2;
+ if (fabs(fader_volume_db[bus_index] - last_fader_volume_db[bus_index]) > 1e-3) {
+ // The volume has changed; do a fade over the course of this frame.
+ // (We might have some numerical issues here, but it seems to sound OK.)
+ // For the purpose of fading here, the silence floor is set to -90 dB
+ // (the fader only goes to -84).
+ float old_volume = from_db(max<float>(last_fader_volume_db[bus_index], -90.0f));
+ float volume = from_db(max<float>(fader_volume_db[bus_index], -90.0f));
+
+ float volume_inc = pow(volume / old_volume, 1.0 / num_samples);
+ volume = old_volume;
+ if (bus_index == 0) {
+ for (unsigned i = 0; i < num_samples; ++i) {
+ (*samples_out)[i * 2 + 0] = samples_bus[i * 2 + 0] * volume;
+ (*samples_out)[i * 2 + 1] = samples_bus[i * 2 + 1] * volume;
+ volume *= volume_inc;
+ }
+ } else {
+ for (unsigned i = 0; i < num_samples; ++i) {
+ (*samples_out)[i * 2 + 0] += samples_bus[i * 2 + 0] * volume;
+ (*samples_out)[i * 2 + 1] += samples_bus[i * 2 + 1] * volume;
+ volume *= volume_inc;
+ }
+ }
+ } else {
+ float volume = from_db(fader_volume_db[bus_index]);
+ if (bus_index == 0) {
+ for (unsigned i = 0; i < num_samples; ++i) {
+ (*samples_out)[i * 2 + 0] = samples_bus[i * 2 + 0] * volume;
+ (*samples_out)[i * 2 + 1] = samples_bus[i * 2 + 1] * volume;
+ }
+ } else {
+ for (unsigned i = 0; i < num_samples; ++i) {
+ (*samples_out)[i * 2 + 0] += samples_bus[i * 2 + 0] * volume;
+ (*samples_out)[i * 2 + 1] += samples_bus[i * 2 + 1] * volume;
+ }
+ }
+ }
+
+ last_fader_volume_db[bus_index] = fader_volume_db[bus_index];
+}
+
+void AudioMixer::measure_bus_levels(unsigned bus_index, const vector<float> &left, const vector<float> &right)
{
assert(left.size() == right.size());
+ const float volume = from_db(fader_volume_db[bus_index]);
const float peak_levels[2] = {
find_peak(left.data(), left.size()) * volume,
find_peak(right.data(), right.size()) * volume
correlation.get_correlation());
}
-map<DeviceSpec, DeviceInfo> AudioMixer::get_devices() const
+map<DeviceSpec, DeviceInfo> AudioMixer::get_devices()
{
lock_guard<timed_mutex> lock(audio_mutex);
- return get_devices_mutex_held();
-}
-map<DeviceSpec, DeviceInfo> AudioMixer::get_devices_mutex_held() const
-{
map<DeviceSpec, DeviceInfo> devices;
for (unsigned card_index = 0; card_index < num_cards; ++card_index) {
const DeviceSpec spec{ InputSourceType::CAPTURE_CARD, card_index };
const AudioDevice *device = &video_cards[card_index];
DeviceInfo info;
- info.name = device->name;
- info.num_channels = 8; // FIXME: This is wrong for fake cards.
+ info.display_name = device->display_name;
+ info.num_channels = 8;
devices.insert(make_pair(spec, info));
}
- for (unsigned card_index = 0; card_index < available_alsa_cards.size(); ++card_index) {
+ vector<ALSAPool::Device> available_alsa_devices = alsa_pool.get_devices();
+ for (unsigned card_index = 0; card_index < available_alsa_devices.size(); ++card_index) {
const DeviceSpec spec{ InputSourceType::ALSA_INPUT, card_index };
- const ALSAInput::Device &device = available_alsa_cards[card_index];
+ const ALSAPool::Device &device = available_alsa_devices[card_index];
DeviceInfo info;
- info.name = device.name + " (" + device.info + ")";
+ info.display_name = device.display_name();
info.num_channels = device.num_channels;
+ info.alsa_name = device.name;
+ info.alsa_info = device.info;
+ info.alsa_address = device.address;
devices.insert(make_pair(spec, info));
}
return devices;
}
-void AudioMixer::set_name(DeviceSpec device_spec, const string &name)
+void AudioMixer::set_display_name(DeviceSpec device_spec, const string &name)
{
AudioDevice *device = find_audio_device(device_spec);
lock_guard<timed_mutex> lock(audio_mutex);
- device->name = name;
+ device->display_name = name;
+}
+
+void AudioMixer::serialize_device(DeviceSpec device_spec, DeviceSpecProto *device_spec_proto)
+{
+ lock_guard<timed_mutex> lock(audio_mutex);
+ switch (device_spec.type) {
+ case InputSourceType::SILENCE:
+ device_spec_proto->set_type(DeviceSpecProto::SILENCE);
+ break;
+ case InputSourceType::CAPTURE_CARD:
+ device_spec_proto->set_type(DeviceSpecProto::CAPTURE_CARD);
+ device_spec_proto->set_index(device_spec.index);
+ device_spec_proto->set_display_name(video_cards[device_spec.index].display_name);
+ break;
+ case InputSourceType::ALSA_INPUT:
+ alsa_pool.serialize_device(device_spec.index, device_spec_proto);
+ break;
+ }
}
void AudioMixer::set_input_mapping(const InputMapping &new_input_mapping)
}
// Reset resamplers for all cards that don't have the exact same state as before.
- for (const auto &spec_and_info : get_devices_mutex_held()) {
- const DeviceSpec &device_spec = spec_and_info.first;
+ for (unsigned card_index = 0; card_index < MAX_VIDEO_CARDS; ++card_index) {
+ const DeviceSpec device_spec{InputSourceType::CAPTURE_CARD, card_index};
AudioDevice *device = find_audio_device(device_spec);
if (device->interesting_channels != interesting_channels[device_spec]) {
device->interesting_channels = interesting_channels[device_spec];
- if (device_spec.type == InputSourceType::ALSA_INPUT) {
- reset_alsa_mutex_held(device_spec);
- }
+ reset_resampler_mutex_held(device_spec);
+ }
+ }
+ for (unsigned card_index = 0; card_index < MAX_ALSA_CARDS; ++card_index) {
+ const DeviceSpec device_spec{InputSourceType::ALSA_INPUT, card_index};
+ AudioDevice *device = find_audio_device(device_spec);
+ if (interesting_channels[device_spec].empty()) {
+ alsa_pool.release_device(card_index);
+ } else {
+ alsa_pool.hold_device(card_index);
+ }
+ if (device->interesting_channels != interesting_channels[device_spec]) {
+ device->interesting_channels = interesting_channels[device_spec];
+ alsa_pool.reset_device(device_spec.index);
reset_resampler_mutex_held(device_spec);
}
}
lock_guard<timed_mutex> lock(audio_mutex);
return input_mapping;
}
+
+void AudioMixer::reset_peak(unsigned bus_index)
+{
+ lock_guard<timed_mutex> lock(audio_mutex);
+ for (unsigned channel = 0; channel < 2; ++channel) {
+ PeakHistory &history = peak_history[bus_index][channel];
+ history.current_level = 0.0f;
+ history.historic_peak = 0.0f;
+ history.current_peak = 0.0f;
+ history.last_peak = 0.0f;
+ history.age_seconds = 0.0f;
+ }
+}
+
+AudioMixer *global_audio_mixer = nullptr;