namespace {
+// TODO: If these prove to be a bottleneck, they can be SSSE3-optimized.
+
void convert_fixed24_to_fp32(float *dst, size_t out_channels, const uint8_t *src, size_t in_channels, size_t num_samples)
{
assert(in_channels >= out_channels);
set_compressor_enabled(global_flags.compressor_enabled);
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.input_source_type = InputSourceType::CAPTURE_CARD;
+ input.input_source_index = 0;
+ input.source_channel[0] = 0;
+ input.source_channel[1] = 1;
+
+ InputMapping new_input_mapping;
+ new_input_mapping.buses.push_back(input);
+ set_input_mapping(new_input_mapping);
}
void AudioMixer::reset_card(unsigned card_index)
{
- CaptureCard *card = &cards[card_index];
+ lock_guard<mutex> lock(audio_mutex);
+ reset_card_mutex_held(card_index);
+}
- unique_lock<mutex> lock(card->audio_mutex);
- card->resampling_queue.reset(new ResamplingQueue(card_index, OUTPUT_FREQUENCY, OUTPUT_FREQUENCY, 2));
+void AudioMixer::reset_card_mutex_held(unsigned card_index)
+{
+ CaptureCard *card = &cards[card_index];
+ if (card->interesting_channels.empty()) {
+ card->resampling_queue.reset();
+ } else {
+ card->resampling_queue.reset(new ResamplingQueue(card_index, OUTPUT_FREQUENCY, OUTPUT_FREQUENCY, card->interesting_channels.size()));
+ }
card->next_local_pts = 0;
}
void AudioMixer::add_audio(unsigned card_index, const uint8_t *data, unsigned num_samples, AudioFormat audio_format, int64_t frame_length)
{
+ lock_guard<mutex> lock(audio_mutex);
CaptureCard *card = &cards[card_index];
+ if (card->resampling_queue == nullptr) {
+ // No buses use this card; throw it away.
+ return;
+ }
+
+ unsigned num_channels = card->interesting_channels.size();
+ assert(num_channels > 0);
+
// Convert the audio to stereo fp32.
+ // FIXME: Pick out the right channels; this takes the first ones.
vector<float> audio;
- audio.resize(num_samples * 2);
+ audio.resize(num_samples * num_channels);
switch (audio_format.bits_per_sample) {
case 0:
assert(num_samples == 0);
break;
case 24:
- convert_fixed24_to_fp32(&audio[0], 2, data, audio_format.num_channels, num_samples);
+ convert_fixed24_to_fp32(&audio[0], num_channels, data, audio_format.num_channels, num_samples);
break;
case 32:
- convert_fixed32_to_fp32(&audio[0], 2, data, audio_format.num_channels, num_samples);
+ convert_fixed32_to_fp32(&audio[0], num_channels, data, audio_format.num_channels, num_samples);
break;
default:
fprintf(stderr, "Cannot handle audio with %u bits per sample\n", audio_format.bits_per_sample);
}
// Now add it.
- {
- unique_lock<mutex> lock(card->audio_mutex);
-
- int64_t local_pts = card->next_local_pts;
- card->resampling_queue->add_input_samples(local_pts / double(TIMEBASE), audio.data(), num_samples);
- card->next_local_pts = local_pts + frame_length;
- }
+ int64_t local_pts = card->next_local_pts;
+ card->resampling_queue->add_input_samples(local_pts / double(TIMEBASE), audio.data(), num_samples);
+ card->next_local_pts = local_pts + frame_length;
}
void AudioMixer::add_silence(unsigned card_index, unsigned samples_per_frame, unsigned num_frames, int64_t frame_length)
{
CaptureCard *card = &cards[card_index];
- unique_lock<mutex> lock(card->audio_mutex);
+ lock_guard<mutex> lock(audio_mutex);
- vector<float> silence(samples_per_frame * 2, 0.0f);
+ if (card->resampling_queue == nullptr) {
+ // No buses use this card; throw it away.
+ return;
+ }
+
+ unsigned num_channels = card->interesting_channels.size();
+ assert(num_channels > 0);
+
+ vector<float> silence(samples_per_frame * num_channels, 0.0f);
for (unsigned i = 0; i < num_frames; ++i) {
card->resampling_queue->add_input_samples(card->next_local_pts / double(TIMEBASE), silence.data(), samples_per_frame);
// Note that if the format changed in the meantime, we have
}
}
+void AudioMixer::find_sample_src_from_capture_card(const vector<float> *samples_card, unsigned card_index, int source_channel, const float **srcptr, unsigned *stride)
+{
+ static float zero = 0.0f;
+ if (source_channel == -1) {
+ *srcptr = &zero;
+ *stride = 0;
+ return;
+ }
+ // FIXME: map back through the interesting_channels squeeze map instead of using source_channel
+ // directly, which will be wrong (and might even overrun).
+ *srcptr = &samples_card[card_index][source_channel];
+ *stride = cards[card_index].interesting_channels.size();
+}
+
vector<float> AudioMixer::get_output(double pts, unsigned num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy)
{
- vector<float> samples_card;
- vector<float> samples_out;
- samples_out.resize(num_samples * 2);
+ vector<float> samples_card[MAX_CARDS];
+ vector<float> samples_bus;
+
+ lock_guard<mutex> lock(audio_mutex);
- // TODO: Allow more flexible input mapping.
+ // Pick out all the interesting channels from all the cards.
for (unsigned card_index = 0; card_index < num_cards; ++card_index) {
- samples_card.resize(num_samples * 2);
- {
- unique_lock<mutex> lock(cards[card_index].audio_mutex);
- cards[card_index].resampling_queue->get_output_samples(
+ CaptureCard *card = &cards[card_index];
+ if (!card->interesting_channels.empty()) {
+ samples_card[card_index].resize(num_samples * card->interesting_channels.size());
+ card->resampling_queue->get_output_samples(
pts,
- &samples_card[0],
+ &samples_card[card_index][0],
num_samples,
rate_adjustment_policy);
}
- if (card_index == 0) {
+ }
+
+ // TODO: Move lo-cut etc. into each bus.
+ vector<float> samples_out;
+ samples_out.resize(num_samples * 2);
+ samples_bus.resize(num_samples * 2);
+ for (unsigned bus_index = 0; bus_index < input_mapping.buses.size(); ++bus_index) {
+ const InputMapping::Bus &input = input_mapping.buses[bus_index];
+ if (input.input_source_type == InputSourceType::SILENCE) {
+ memset(&samples_bus[0], 0, samples_bus.size() * sizeof(samples_bus[0]));
+ } else {
+ // TODO: Move this into its own function. Can be SSSE3-optimized if need be.
+ assert(input.input_source_type == InputSourceType::CAPTURE_CARD);
+ const float *lsrc, *rsrc;
+ unsigned lstride, rstride;
+ float *dptr = &samples_bus[0];
+ find_sample_src_from_capture_card(samples_card, input.input_source_index, input.source_channel[0], &lsrc, &lstride);
+ find_sample_src_from_capture_card(samples_card, input.input_source_index, input.source_channel[1], &rsrc, &rstride);
+ for (unsigned i = 0; i < num_samples; ++i) {
+ *dptr++ = *lsrc;
+ *dptr++ = *rsrc;
+ lsrc += lstride;
+ rsrc += rstride;
+ }
+ }
+
+ 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_card[i];
+ samples_out[i] = samples_bus[i] * volume;
}
} else {
for (unsigned i = 0; i < num_samples * 2; ++i) {
- samples_out[i] += samples_card[i];
+ samples_out[i] += samples_bus[i] * volume;
}
}
}
}
{
- unique_lock<mutex> lock(compressor_mutex);
+ lock_guard<mutex> lock(compressor_mutex);
// Apply a level compressor to get the general level right.
// Basically, if it's over about -40 dBFS, we squeeze it down to that level
// something we get out per-sample.
//
// Note that there's a feedback loop here, so we choose a very slow filter
- // (half-time of 100 seconds).
+ // (half-time of 30 seconds).
double target_loudness_factor, alpha;
double loudness_lu = loudness_lufs - ref_level_lufs;
double current_makeup_lu = to_db(final_makeup_gain);
- target_loudness_factor = from_db(-loudness_lu);
+ target_loudness_factor = final_makeup_gain * from_db(-loudness_lu);
// If we're outside +/- 5 LU uncorrected, we don't count it as
// a normal signal (probably silence) and don't change the
} else {
// Formula adapted from
// https://en.wikipedia.org/wiki/Low-pass_filter#Simple_infinite_impulse_response_filter.
- const double half_time_s = 100.0;
+ const double half_time_s = 30.0;
const double fc_mul_2pi_delta_t = 1.0 / (half_time_s * OUTPUT_FREQUENCY);
alpha = fc_mul_2pi_delta_t / (fc_mul_2pi_delta_t + 1.0);
}
{
- unique_lock<mutex> lock(compressor_mutex);
+ lock_guard<mutex> lock(compressor_mutex);
double m = final_makeup_gain;
for (size_t i = 0; i < samples_out.size(); i += 2) {
samples_out[i + 0] *= m;
return samples_out;
}
+
+vector<string> AudioMixer::get_names() const
+{
+ lock_guard<mutex> lock(audio_mutex);
+ vector<string> names;
+ for (unsigned card_index = 0; card_index < num_cards; ++card_index) {
+ const CaptureCard *card = &cards[card_index];
+ names.push_back(card->name);
+ }
+ return names;
+}
+
+void AudioMixer::set_name(unsigned card_index, const string &name)
+{
+ lock_guard<mutex> lock(audio_mutex);
+ CaptureCard *card = &cards[card_index];
+ card->name = name;
+}
+
+void AudioMixer::set_input_mapping(const InputMapping &new_input_mapping)
+{
+ lock_guard<mutex> lock(audio_mutex);
+
+ map<unsigned, set<unsigned>> interesting_channels;
+ for (const InputMapping::Bus &bus : new_input_mapping.buses) {
+ if (bus.input_source_type == InputSourceType::CAPTURE_CARD) {
+ for (unsigned channel = 0; channel < 2; ++channel) {
+ if (bus.source_channel[channel] != -1) {
+ interesting_channels[bus.input_source_index].insert(bus.source_channel[channel]);
+ }
+ }
+ }
+ }
+
+ // Reset resamplers for all cards that don't have the exact same state as before.
+ for (unsigned card_index = 0; card_index < num_cards; ++card_index) {
+ CaptureCard *card = &cards[card_index];
+ if (card->interesting_channels != interesting_channels[card_index]) {
+ card->interesting_channels = interesting_channels[card_index];
+ reset_card_mutex_held(card_index);
+ }
+ }
+
+ input_mapping = new_input_mapping;
+}
+
+InputMapping AudioMixer::get_input_mapping() const
+{
+ lock_guard<mutex> lock(audio_mutex);
+ return input_mapping;
+}
projects / nageru / blobdiff