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);
input.source_channel[0] = 0;
input.source_channel[1] = 1;
- input_mapping.inputs.push_back(input);
+ InputMapping new_input_mapping;
+ new_input_mapping.inputs.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 inputs 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);
+
+ if (card->resampling_queue == nullptr) {
+ // No inputs use this card; throw it away.
+ return;
+ }
- vector<float> silence(samples_per_frame * 2, 0.0f);
+ 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);
}
+ }
+
+ // 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 input_index = 0; input_index < input_mapping.inputs.size(); ++input_index) {
+ const InputMapping::Input &input = input_mapping.inputs[input_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(cards[card_index].fader_volume_db);
- if (card_index == 0) {
+ float volume = from_db(fader_volume_db[input_index]);
+ if (input_index == 0) {
for (unsigned i = 0; i < num_samples * 2; ++i) {
- samples_out[i] = samples_card[i] * volume;
+ samples_out[i] = samples_bus[i] * volume;
}
} else {
for (unsigned i = 0; i < num_samples * 2; ++i) {
- samples_out[i] += samples_card[i] * volume;
+ 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
}
{
- 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;
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];
- unique_lock<mutex> lock(card->audio_mutex);
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];
- unique_lock<mutex> lock(card->audio_mutex);
card->name = name;
}
-void AudioMixer::set_input_mapping(const InputMapping &input_mapping)
+void AudioMixer::set_input_mapping(const InputMapping &new_input_mapping)
{
- lock_guard<mutex> lock(mapping_mutex);
- this->input_mapping = input_mapping;
+ lock_guard<mutex> lock(audio_mutex);
+
+ map<unsigned, set<unsigned>> interesting_channels;
+ for (const InputMapping::Input &input : new_input_mapping.inputs) {
+ if (input.input_source_type == InputSourceType::CAPTURE_CARD) {
+ for (unsigned channel = 0; channel < 2; ++channel) {
+ if (input.source_channel[channel] != -1) {
+ interesting_channels[input.input_source_index].insert(input.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(mapping_mutex);
+ lock_guard<mutex> lock(audio_mutex);
return input_mapping;
}
#include <math.h>
#include <stdint.h>
#include <atomic>
+#include <map>
#include <memory>
#include <mutex>
+#include <set>
#include <vector>
#include "bmusb/bmusb.h"
enum class InputSourceType { SILENCE, CAPTURE_CARD };
struct InputMapping {
- struct Input {
+ struct Input { // TODO: rename to Bus?
std::string name;
InputSourceType input_source_type;
unsigned input_source_index;
// See comments inside get_output().
void set_current_loudness(double level_lufs) { loudness_lufs = level_lufs; }
- void set_fader_volume(unsigned card_index, float level_db) { cards[card_index].fader_volume_db = level_db; }
+ void set_fader_volume(unsigned input_index, float level_db) { fader_volume_db[input_index] = level_db; }
std::vector<std::string> get_names() const;
void set_name(unsigned card_index, const std::string &name);
}
private:
+ void find_sample_src_from_capture_card(const std::vector<float> *samples_card, unsigned card_index, int source_channel, const float **srcptr, unsigned *stride);
+ void reset_card_mutex_held(unsigned card_index);
+
unsigned num_cards;
- struct CaptureCard {
- std::atomic<float> fader_volume_db{0.0f};
+ mutable std::mutex audio_mutex;
- // Everything below audio_mutex is protected by it.
- mutable std::mutex audio_mutex;
+ struct CaptureCard {
std::unique_ptr<ResamplingQueue> resampling_queue;
int64_t next_local_pts = 0;
std::string name;
+ // Which channels we consider interesting (ie., are part of some input_mapping).
+ std::set<unsigned> interesting_channels;
};
- CaptureCard cards[MAX_CARDS];
+ CaptureCard cards[MAX_CARDS]; // Under audio_mutex.
StereoFilter locut; // Default cutoff 120 Hz, 24 dB/oct.
std::atomic<float> locut_cutoff_hz;
double final_makeup_gain = 1.0; // Under compressor_mutex. Read/write by the user. Note: Not in dB, we want the numeric precision so that we can change it slowly.
bool final_makeup_gain_auto = true; // Under compressor_mutex.
- mutable std::mutex mapping_mutex;
- InputMapping input_mapping; // Under mapping_mutex.
+ InputMapping input_mapping; // Under audio_mutex.
+ std::atomic<float> fader_volume_db[MAX_BUSES] {{ 0.0f }};
};
#endif // !defined(_AUDIO_MIXER_H)
projects / nageru / commitdiff