X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=audio_mixer.cpp;h=9844b88aa8ea5e8f624682c934e4f3713554ec74;hb=44add3f0cdb832399a59e50ca417d3ced1c5064e;hp=06c7f870fcae75203eb75749ebf8c568e44d5bf9;hpb=35e2c9dbbd5899d6dd3b2f926f95eeba72d038ba;p=nageru diff --git a/audio_mixer.cpp b/audio_mixer.cpp index 06c7f87..9844b88 100644 --- a/audio_mixer.cpp +++ b/audio_mixer.cpp @@ -4,6 +4,7 @@ #include #include #include +#include #include #include "db.h" @@ -15,31 +16,45 @@ using namespace std; namespace { -void convert_fixed24_to_fp32(float *dst, size_t out_channels, const uint8_t *src, size_t in_channels, size_t num_samples) +// TODO: If these prove to be a bottleneck, they can be SSSE3-optimized +// (usually including multiple channels at a time). + +void convert_fixed24_to_fp32(float *dst, size_t out_channel, size_t out_num_channels, + const uint8_t *src, size_t in_channel, size_t in_num_channels, + size_t num_samples) { - assert(in_channels >= out_channels); + assert(in_channel < in_num_channels); + assert(out_channel < out_num_channels); + src += in_channel * 3; + dst += out_channel; + for (size_t i = 0; i < num_samples; ++i) { - for (size_t j = 0; j < out_channels; ++j) { - uint32_t s1 = *src++; - uint32_t s2 = *src++; - uint32_t s3 = *src++; - uint32_t s = s1 | (s1 << 8) | (s2 << 16) | (s3 << 24); - dst[i * out_channels + j] = int(s) * (1.0f / 2147483648.0f); - } - src += 3 * (in_channels - out_channels); + uint32_t s1 = src[0]; + uint32_t s2 = src[1]; + uint32_t s3 = src[2]; + uint32_t s = s1 | (s1 << 8) | (s2 << 16) | (s3 << 24); + *dst = int(s) * (1.0f / 2147483648.0f); + + src += 3 * in_num_channels; + dst += out_num_channels; } } -void convert_fixed32_to_fp32(float *dst, size_t out_channels, const uint8_t *src, size_t in_channels, size_t num_samples) +void convert_fixed32_to_fp32(float *dst, size_t out_channel, size_t out_num_channels, + const uint8_t *src, size_t in_channel, size_t in_num_channels, + size_t num_samples) { - assert(in_channels >= out_channels); + assert(in_channel < in_num_channels); + assert(out_channel < out_num_channels); + src += in_channel * 4; + dst += out_channel; + for (size_t i = 0; i < num_samples; ++i) { - for (size_t j = 0; j < out_channels; ++j) { - int32_t s = le32toh(*(int32_t *)src); - dst[i * out_channels + j] = s * (1.0f / 2147483648.0f); - src += 4; - } - src += 4 * (in_channels - out_channels); + int32_t s = le32toh(*(int32_t *)src); + *dst = s * (1.0f / 2147483648.0f); + + src += 4 * in_num_channels; + dst += out_num_channels; } } @@ -59,55 +74,91 @@ 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. vector audio; - audio.resize(num_samples * 2); - 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); - break; - case 32: - convert_fixed32_to_fp32(&audio[0], 2, 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); - assert(false); + audio.resize(num_samples * num_channels); + unsigned channel_index = 0; + for (auto channel_it = card->interesting_channels.cbegin(); channel_it != card->interesting_channels.end(); ++channel_it, ++channel_index) { + switch (audio_format.bits_per_sample) { + case 0: + assert(num_samples == 0); + break; + case 24: + convert_fixed24_to_fp32(&audio[0], channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples); + break; + case 32: + convert_fixed32_to_fp32(&audio[0], channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples); + break; + default: + fprintf(stderr, "Cannot handle audio with %u bits per sample\n", audio_format.bits_per_sample); + assert(false); + } } // 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 +168,81 @@ 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; + } + CaptureCard *card = &cards[card_index]; + unsigned channel_index = 0; + for (int channel : card->interesting_channels) { + if (channel == source_channel) break; + ++channel_index; + } + assert(channel_index < card->interesting_channels.size()); + *srcptr = &samples_card[card_index][channel_index]; + *stride = card->interesting_channels.size(); +} + +// TODO: Can be SSSE3-optimized if need be. +void AudioMixer::fill_audio_bus(const vector *samples_card, const InputMapping::Bus &bus, unsigned num_samples, float *output) +{ + if (bus.input_source_type == InputSourceType::SILENCE) { + memset(output, 0, num_samples * sizeof(*output)); + } else { + assert(bus.input_source_type == InputSourceType::CAPTURE_CARD); + const float *lsrc, *rsrc; + unsigned lstride, rstride; + float *dptr = output; + find_sample_src_from_capture_card(samples_card, bus.input_source_index, bus.source_channel[0], &lsrc, &lstride); + find_sample_src_from_capture_card(samples_card, bus.input_source_index, bus.source_channel[1], &rsrc, &rstride); + for (unsigned i = 0; i < num_samples; ++i) { + *dptr++ = *lsrc; + *dptr++ = *rsrc; + lsrc += lstride; + rsrc += rstride; + } + } +} + 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; + + lock_guard 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 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); } + } - float volume = from_db(cards[card_index].fader_volume_db); - if (card_index == 0) { + // 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) { + fill_audio_bus(samples_card, input_mapping.buses[bus_index], num_samples, &samples_bus[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 +256,7 @@ 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 +327,11 @@ 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 +341,13 @@ 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 +359,54 @@ 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; +}