X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=audio_mixer.cpp;h=8d09ad3bfa9a279f5d74af38613503ce835311e8;hb=1fc8b5d92fce7ac0b7e87ed3087e393774fd4b3c;hp=88ca8c617a8559ce771e21f1e337133bba51296c;hpb=455786e07d6ea91bc8af639723f64ae367d6b4c6;p=nageru diff --git a/audio_mixer.cpp b/audio_mixer.cpp index 88ca8c6..8d09ad3 100644 --- a/audio_mixer.cpp +++ b/audio_mixer.cpp @@ -13,6 +13,7 @@ using namespace bmusb; using namespace std; +using namespace std::placeholders; namespace { @@ -76,20 +77,47 @@ void convert_fixed32_to_fp32(float *dst, size_t out_channel, size_t out_num_chan } } +float find_peak(const float *samples, size_t num_samples) +{ + float m = fabs(samples[0]); + for (size_t i = 1; i < num_samples; ++i) { + m = max(m, fabs(samples[i])); + } + return m; +} + +void deinterleave_samples(const vector &in, vector *out_l, vector *out_r) +{ + size_t num_samples = in.size() / 2; + out_l->resize(num_samples); + out_r->resize(num_samples); + + const float *inptr = in.data(); + float *lptr = &(*out_l)[0]; + float *rptr = &(*out_r)[0]; + for (size_t i = 0; i < num_samples; ++i) { + *lptr++ = *inptr++; + *rptr++ = *inptr++; + } +} + } // namespace AudioMixer::AudioMixer(unsigned num_cards) : num_cards(num_cards), - level_compressor(OUTPUT_FREQUENCY), limiter(OUTPUT_FREQUENCY), - compressor(OUTPUT_FREQUENCY) + correlation(OUTPUT_FREQUENCY) { - locut.init(FILTER_HPF, 2); - - set_locut_enabled(global_flags.locut_enabled); - set_gain_staging_db(global_flags.initial_gain_staging_db); - set_gain_staging_auto(global_flags.gain_staging_auto); - set_compressor_enabled(global_flags.compressor_enabled); + 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; + 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. + compressor_enabled[bus_index] = global_flags.compressor_enabled; + level_compressor[bus_index].reset(new StereoCompressor(OUTPUT_FREQUENCY)); + level_compressor_enabled[bus_index] = global_flags.gain_staging_auto; + } set_limiter_enabled(global_flags.limiter_enabled); set_final_makeup_gain_auto(global_flags.final_makeup_gain_auto); @@ -104,17 +132,39 @@ AudioMixer::AudioMixer(unsigned num_cards) InputMapping new_input_mapping; new_input_mapping.buses.push_back(input); set_input_mapping(new_input_mapping); + + // Look for ALSA cards. + available_alsa_cards = ALSAInput::enumerate_devices(); + + r128.init(2, OUTPUT_FREQUENCY); + r128.integr_start(); + + // hlen=16 is pretty low quality, but we use quite a bit of CPU otherwise, + // and there's a limit to how important the peak meter is. + peak_resampler.setup(OUTPUT_FREQUENCY, OUTPUT_FREQUENCY * 4, /*num_channels=*/2, /*hlen=*/16, /*frel=*/1.0); } -void AudioMixer::reset_device(DeviceSpec device_spec) +AudioMixer::~AudioMixer() { - lock_guard lock(audio_mutex); - reset_device_mutex_held(device_spec); + 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_device_mutex_held(DeviceSpec device_spec) + +void AudioMixer::reset_resampler(DeviceSpec device_spec) +{ + lock_guard lock(audio_mutex); + reset_resampler_mutex_held(device_spec); +} + +void AudioMixer::reset_resampler_mutex_held(DeviceSpec device_spec) { AudioDevice *device = find_audio_device(device_spec); + if (device->interesting_channels.empty()) { device->resampling_queue.reset(); } else { @@ -125,14 +175,36 @@ void AudioMixer::reset_device_mutex_held(DeviceSpec device_spec) device->next_local_pts = 0; } -void AudioMixer::add_audio(DeviceSpec device_spec, const uint8_t *data, unsigned num_samples, AudioFormat audio_format, int64_t frame_length) +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); - lock_guard lock(audio_mutex); + 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); + + unique_lock lock(audio_mutex, defer_lock); + if (!lock.try_lock_for(chrono::milliseconds(10))) { + return false; + } if (device->resampling_queue == nullptr) { // No buses use this device; throw it away. - return; + return true; } unsigned num_channels = device->interesting_channels.size(); @@ -166,16 +238,20 @@ void AudioMixer::add_audio(DeviceSpec device_spec, const uint8_t *data, unsigned int64_t local_pts = device->next_local_pts; device->resampling_queue->add_input_samples(local_pts / double(TIMEBASE), audio.data(), num_samples); device->next_local_pts = local_pts + frame_length; + return true; } -void AudioMixer::add_silence(DeviceSpec device_spec, unsigned samples_per_frame, unsigned num_frames, int64_t frame_length) +bool AudioMixer::add_silence(DeviceSpec device_spec, unsigned samples_per_frame, unsigned num_frames, int64_t frame_length) { AudioDevice *device = find_audio_device(device_spec); - lock_guard lock(audio_mutex); + unique_lock lock(audio_mutex, defer_lock); + if (!lock.try_lock_for(chrono::milliseconds(10))) { + return false; + } if (device->resampling_queue == nullptr) { // No buses use this device; throw it away. - return; + return true; } unsigned num_channels = device->interesting_channels.size(); @@ -189,14 +265,16 @@ void AudioMixer::add_silence(DeviceSpec device_spec, unsigned samples_per_frame, // is always the same. device->next_local_pts += frame_length; } + return true; } AudioMixer::AudioDevice *AudioMixer::find_audio_device(DeviceSpec device) { switch (device.type) { case InputSourceType::CAPTURE_CARD: - return &cards[device.index]; - break; + return &video_cards[device.index]; + case InputSourceType::ALSA_INPUT: + return &alsa_inputs[device.index]; case InputSourceType::SILENCE: default: assert(false); @@ -204,7 +282,9 @@ AudioMixer::AudioDevice *AudioMixer::find_audio_device(DeviceSpec device) return nullptr; } -void AudioMixer::find_sample_src_from_device(const vector *samples_card, DeviceSpec device_spec, int source_channel, const float **srcptr, unsigned *stride) +// Get a pointer to the given channel from the given device. +// The channel must be picked out earlier and resampled. +void AudioMixer::find_sample_src_from_device(const map> &samples_card, DeviceSpec device_spec, int source_channel, const float **srcptr, unsigned *stride) { static float zero = 0.0f; if (source_channel == -1 || device_spec.type == InputSourceType::SILENCE) { @@ -213,23 +293,27 @@ void AudioMixer::find_sample_src_from_device(const vector *samples_card, return; } AudioDevice *device = find_audio_device(device_spec); + assert(device->interesting_channels.count(source_channel) != 0); unsigned channel_index = 0; for (int channel : device->interesting_channels) { if (channel == source_channel) break; ++channel_index; } assert(channel_index < device->interesting_channels.size()); - *srcptr = &samples_card[device_spec.index][channel_index]; + const auto it = samples_card.find(device_spec); + assert(it != samples_card.end()); + *srcptr = &(it->second)[channel_index]; *stride = device->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) +void AudioMixer::fill_audio_bus(const map> &samples_card, const InputMapping::Bus &bus, unsigned num_samples, float *output) { if (bus.device.type == InputSourceType::SILENCE) { memset(output, 0, num_samples * sizeof(*output)); } else { - assert(bus.device.type == InputSourceType::CAPTURE_CARD); + assert(bus.device.type == InputSourceType::CAPTURE_CARD || + bus.device.type == InputSourceType::ALSA_INPUT); const float *lsrc, *rsrc; unsigned lstride, rstride; float *dptr = output; @@ -246,97 +330,104 @@ void AudioMixer::fill_audio_bus(const vector *samples_card, const InputMa vector AudioMixer::get_output(double pts, unsigned num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy) { - vector samples_card[MAX_CARDS]; // TODO: Needs room for other kinds of capture cards. + map> samples_card; vector samples_bus; - lock_guard lock(audio_mutex); + 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) { - AudioDevice *device = &cards[card_index]; + // 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; + AudioDevice *device = find_audio_device(device_spec); if (!device->interesting_channels.empty()) { - samples_card[card_index].resize(num_samples * device->interesting_channels.size()); + samples_card[device_spec].resize(num_samples * device->interesting_channels.size()); device->resampling_queue->get_output_samples( pts, - &samples_card[card_index][0], + &samples_card[device_spec][0], num_samples, rate_adjustment_policy); } } - // TODO: Move lo-cut etc. into each bus. - vector samples_out; + vector samples_out, left, right; 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_bus[i] * volume; - } - } else { - for (unsigned i = 0; i < num_samples * 2; ++i) { - samples_out[i] += samples_bus[i] * volume; - } + // 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); } - } - - // 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) { - locut.render(samples_out.data(), samples_out.size() / 2, locut_cutoff_hz * 2.0 * M_PI / OUTPUT_FREQUENCY, 0.5f); - } - - { - 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 - // (or more precisely, near it, since we don't use infinite ratio), - // then apply a makeup gain to get it to -14 dBFS. -14 dBFS is, of course, - // entirely arbitrary, but from practical tests with speech, it seems to - // put ut around -23 LUFS, so it's a reasonable starting point for later use. { - if (level_compressor_enabled) { + 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 + // (or more precisely, near it, since we don't use infinite ratio), + // then apply a makeup gain to get it to -14 dBFS. -14 dBFS is, of course, + // entirely arbitrary, but from practical tests with speech, it seems to + // put ut around -23 LUFS, so it's a reasonable starting point for later use. + if (level_compressor_enabled[bus_index]) { float threshold = 0.01f; // -40 dBFS. float ratio = 20.0f; float attack_time = 0.5f; float release_time = 20.0f; float makeup_gain = from_db(ref_level_dbfs - (-40.0f)); // +26 dB. - level_compressor.process(samples_out.data(), samples_out.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain); - gain_staging_db = to_db(level_compressor.get_attenuation() * makeup_gain); + level_compressor[bus_index]->process(samples_bus.data(), samples_bus.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain); + gain_staging_db[bus_index] = to_db(level_compressor[bus_index]->get_attenuation() * makeup_gain); } else { // Just apply the gain we already had. - float g = from_db(gain_staging_db); - for (size_t i = 0; i < samples_out.size(); ++i) { - samples_out[i] *= g; + float g = from_db(gain_staging_db[bus_index]); + for (size_t i = 0; i < samples_bus.size(); ++i) { + samples_bus[i] *= g; } } + +#if 0 + printf("level=%f (%+5.2f dBFS) attenuation=%f (%+5.2f dB) end_result=%+5.2f dB\n", + level_compressor.get_level(), to_db(level_compressor.get_level()), + level_compressor.get_attenuation(), to_db(level_compressor.get_attenuation()), + to_db(level_compressor.get_level() * level_compressor.get_attenuation() * makeup_gain)); +#endif + + // The real compressor. + if (compressor_enabled[bus_index]) { + float threshold = from_db(compressor_threshold_dbfs[bus_index]); + float ratio = 20.0f; + float attack_time = 0.005f; + float release_time = 0.040f; + float makeup_gain = 2.0f; // +6 dB. + compressor[bus_index]->process(samples_bus.data(), samples_bus.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain); + // compressor_att = compressor.get_attenuation(); + } } - #if 0 - printf("level=%f (%+5.2f dBFS) attenuation=%f (%+5.2f dB) end_result=%+5.2f dB\n", - level_compressor.get_level(), to_db(level_compressor.get_level()), - level_compressor.get_attenuation(), to_db(level_compressor.get_attenuation()), - to_db(level_compressor.get_level() * level_compressor.get_attenuation() * makeup_gain)); - #endif - - // float limiter_att, compressor_att; - - // The real compressor. - if (compressor_enabled) { - float threshold = from_db(compressor_threshold_dbfs); - float ratio = 20.0f; - float attack_time = 0.005f; - float release_time = 0.040f; - float makeup_gain = 2.0f; // +6 dB. - compressor.process(samples_out.data(), samples_out.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain); - // compressor_att = compressor.get_attenuation(); + // TODO: We should measure post-fader. + deinterleave_samples(samples_bus, &left, &right); + measure_bus_levels(bus_index, left, right); + + 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; + } } + } + + { + lock_guard lock(compressor_mutex); // Finally a limiter at -4 dB (so, -10 dBFS) to take out the worst peaks only. // Note that since ratio is not infinite, we could go slightly higher than this. @@ -353,7 +444,8 @@ vector AudioMixer::get_output(double pts, unsigned num_samples, Resamplin // printf("limiter=%+5.1f compressor=%+5.1f\n", to_db(limiter_att), to_db(compressor_att)); } - // At this point, we are most likely close to +0 LU, but all of our + // At this point, we are most likely close to +0 LU (at least if the + // faders sum to 0 dB and the compressors are on), but all of our // measurements have been on raw sample values, not R128 values. // So we have a final makeup gain to get us to +0 LU; the gain // adjustments required should be relatively small, and also, the @@ -365,7 +457,7 @@ vector AudioMixer::get_output(double pts, unsigned num_samples, Resamplin // Note that there's a feedback loop here, so we choose a very slow filter // (half-time of 30 seconds). double target_loudness_factor, alpha; - double loudness_lu = loudness_lufs - ref_level_lufs; + double loudness_lu = r128.loudness_M() - ref_level_lufs; double current_makeup_lu = to_db(final_makeup_gain); target_loudness_factor = final_makeup_gain * from_db(-loudness_lu); @@ -393,50 +485,170 @@ vector AudioMixer::get_output(double pts, unsigned num_samples, Resamplin final_makeup_gain = m; } + update_meters(samples_out); + return samples_out; } -vector AudioMixer::get_names() const +void AudioMixer::measure_bus_levels(unsigned bus_index, const vector &left, const vector &right) +{ + const float *ptrs[] = { left.data(), right.data() }; + { + lock_guard lock(audio_measure_mutex); + bus_r128[bus_index]->process(left.size(), const_cast(ptrs)); + } +} + +void AudioMixer::update_meters(const vector &samples) +{ + // Upsample 4x to find interpolated peak. + peak_resampler.inp_data = const_cast(samples.data()); + peak_resampler.inp_count = samples.size() / 2; + + vector interpolated_samples; + interpolated_samples.resize(samples.size()); + { + lock_guard lock(audio_measure_mutex); + + while (peak_resampler.inp_count > 0) { // About four iterations. + peak_resampler.out_data = &interpolated_samples[0]; + peak_resampler.out_count = interpolated_samples.size() / 2; + peak_resampler.process(); + size_t out_stereo_samples = interpolated_samples.size() / 2 - peak_resampler.out_count; + peak = max(peak, find_peak(interpolated_samples.data(), out_stereo_samples * 2)); + peak_resampler.out_data = nullptr; + } + } + + // Find R128 levels and L/R correlation. + vector left, right; + deinterleave_samples(samples, &left, &right); + float *ptrs[] = { left.data(), right.data() }; + { + lock_guard lock(audio_measure_mutex); + r128.process(left.size(), ptrs); + correlation.process_samples(samples); + } + + send_audio_level_callback(); +} + +void AudioMixer::reset_meters() { - lock_guard lock(audio_mutex); - vector names; + lock_guard lock(audio_measure_mutex); + peak_resampler.reset(); + peak = 0.0f; + r128.reset(); + r128.integr_start(); + correlation.reset(); +} + +void AudioMixer::send_audio_level_callback() +{ + if (audio_level_callback == nullptr) { + return; + } + + lock_guard lock(audio_measure_mutex); + double loudness_s = r128.loudness_S(); + double loudness_i = r128.integrated(); + double loudness_range_low = r128.range_min(); + double loudness_range_high = r128.range_max(); + + vector bus_levels; + bus_levels.resize(input_mapping.buses.size()); + { + lock_guard lock(compressor_mutex); + for (unsigned bus_index = 0; bus_index < bus_r128.size(); ++bus_index) { + bus_levels[bus_index].loudness_lufs = bus_r128[bus_index]->loudness_S(); + bus_levels[bus_index].gain_staging_db = gain_staging_db[bus_index]; + if (compressor_enabled[bus_index]) { + bus_levels[bus_index].compressor_attenuation_db = -to_db(compressor[bus_index]->get_attenuation()); + } else { + bus_levels[bus_index].compressor_attenuation_db = 0.0; + } + } + } + + audio_level_callback(loudness_s, to_db(peak), bus_levels, + loudness_i, loudness_range_low, loudness_range_high, + to_db(final_makeup_gain), + correlation.get_correlation()); +} + +map AudioMixer::get_devices() const +{ + lock_guard lock(audio_mutex); + return get_devices_mutex_held(); +} + +map AudioMixer::get_devices_mutex_held() const +{ + map devices; for (unsigned card_index = 0; card_index < num_cards; ++card_index) { - const AudioDevice *device = &cards[card_index]; - names.push_back(device->name); + 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. + devices.insert(make_pair(spec, info)); + } + for (unsigned card_index = 0; card_index < available_alsa_cards.size(); ++card_index) { + const DeviceSpec spec{ InputSourceType::ALSA_INPUT, card_index }; + const ALSAInput::Device &device = available_alsa_cards[card_index]; + DeviceInfo info; + info.name = device.name + " (" + device.info + ")"; + info.num_channels = device.num_channels; + devices.insert(make_pair(spec, info)); } - return names; + return devices; } void AudioMixer::set_name(DeviceSpec device_spec, const string &name) { AudioDevice *device = find_audio_device(device_spec); - lock_guard lock(audio_mutex); + lock_guard lock(audio_mutex); device->name = name; } void AudioMixer::set_input_mapping(const InputMapping &new_input_mapping) { - lock_guard lock(audio_mutex); + lock_guard lock(audio_mutex); - // FIXME: This needs to be keyed on DeviceSpec. - map> interesting_channels; + map> interesting_channels; for (const InputMapping::Bus &bus : new_input_mapping.buses) { - if (bus.device.type == InputSourceType::CAPTURE_CARD) { + if (bus.device.type == InputSourceType::CAPTURE_CARD || + bus.device.type == InputSourceType::ALSA_INPUT) { for (unsigned channel = 0; channel < 2; ++channel) { if (bus.source_channel[channel] != -1) { - interesting_channels[bus.device.index].insert(bus.source_channel[channel]); + interesting_channels[bus.device].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) { - AudioDevice *device = &cards[card_index]; - if (device->interesting_channels != interesting_channels[card_index]) { - device->interesting_channels = interesting_channels[card_index]; - reset_device_mutex_held(DeviceSpec{InputSourceType::CAPTURE_CARD, card_index}); + for (const auto &spec_and_info : get_devices_mutex_held()) { + const DeviceSpec &device_spec = spec_and_info.first; + 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); + } + } + + { + lock_guard lock(audio_measure_mutex); + bus_r128.resize(new_input_mapping.buses.size()); + for (unsigned bus_index = 0; bus_index < bus_r128.size(); ++bus_index) { + if (bus_r128[bus_index] == nullptr) { + bus_r128[bus_index].reset(new Ebu_r128_proc); + } + bus_r128[bus_index]->init(2, OUTPUT_FREQUENCY); } } @@ -445,6 +657,6 @@ void AudioMixer::set_input_mapping(const InputMapping &new_input_mapping) InputMapping AudioMixer::get_input_mapping() const { - lock_guard lock(audio_mutex); + lock_guard lock(audio_mutex); return input_mapping; }