X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=audio_mixer.cpp;h=7baa218cd403589c8452e60a65871f78a75dce87;hb=41c91a56e836c4e81fcee2f3728ca09c0cc2a7bd;hp=06c7f870fcae75203eb75749ebf8c568e44d5bf9;hpb=35e2c9dbbd5899d6dd3b2f926f95eeba72d038ba;p=nageru diff --git a/audio_mixer.cpp b/audio_mixer.cpp index 06c7f87..7baa218 100644 --- a/audio_mixer.cpp +++ b/audio_mixer.cpp @@ -15,6 +15,8 @@ using namespace std; 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); @@ -59,33 +61,63 @@ AudioMixer::AudioMixer(unsigned num_cards) 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 lock(audio_mutex); + reset_card_mutex_held(card_index); +} - unique_lock 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 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 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); @@ -93,21 +125,25 @@ void AudioMixer::add_audio(unsigned card_index, const uint8_t *data, unsigned nu } // Now add it. - { - unique_lock 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 lock(card->audio_mutex); + lock_guard lock(audio_mutex); - vector 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 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 @@ -117,32 +153,72 @@ void AudioMixer::add_silence(unsigned card_index, unsigned samples_per_frame, un } } +void AudioMixer::find_sample_src_from_capture_card(const vector *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 AudioMixer::get_output(double pts, unsigned num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy) { - vector samples_card; - vector samples_out; - samples_out.resize(num_samples * 2); + vector samples_card[MAX_CARDS]; + vector samples_bus; - // TODO: Allow more flexible input mapping. + lock_guard lock(audio_mutex); + + // 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 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 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(cards[card_index].fader_volume_db); - if (card_index == 0) { + 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] * 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; } } } @@ -156,7 +232,7 @@ vector AudioMixer::get_output(double pts, unsigned num_samples, Resamplin } { - unique_lock lock(compressor_mutex); + lock_guard 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 @@ -227,11 +303,11 @@ vector AudioMixer::get_output(double pts, unsigned num_samples, Resamplin // 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 @@ -241,13 +317,13 @@ vector AudioMixer::get_output(double pts, unsigned num_samples, Resamplin } 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 lock(compressor_mutex); + lock_guard lock(compressor_mutex); double m = final_makeup_gain; for (size_t i = 0; i < samples_out.size(); i += 2) { samples_out[i + 0] *= m; @@ -259,3 +335,54 @@ vector AudioMixer::get_output(double pts, unsigned num_samples, Resamplin return samples_out; } + +vector AudioMixer::get_names() const +{ + lock_guard lock(audio_mutex); + vector 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 lock(audio_mutex); + CaptureCard *card = &cards[card_index]; + card->name = name; +} + +void AudioMixer::set_input_mapping(const InputMapping &new_input_mapping) +{ + lock_guard lock(audio_mutex); + + map> 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 lock(audio_mutex); + return input_mapping; +}