X-Git-Url: https://git.sesse.net/?p=nageru;a=blobdiff_plain;f=audio_mixer.cpp;h=e4d4cff4b86222e41d6b0d8181b8d519a1f1b6a1;hp=426125f888061a17546de06c4d3338b90098892a;hb=refs%2Fheads%2Fmultichannel_audio;hpb=99fe9fd44f57c9872eae24745da9c7a422ab0c98 diff --git a/audio_mixer.cpp b/audio_mixer.cpp index 426125f..e4d4cff 100644 --- a/audio_mixer.cpp +++ b/audio_mixer.cpp @@ -1,18 +1,31 @@ #include "audio_mixer.h" #include -#include #include -#include #include +#include +#ifdef __SSE2__ +#include +#endif +#include +#include +#include +#include +#include +#include #include +#include +#include +#include #include "db.h" #include "flags.h" +#include "state.pb.h" #include "timebase.h" using namespace bmusb; using namespace std; +using namespace std::placeholders; namespace { @@ -76,45 +89,139 @@ void convert_fixed32_to_fp32(float *dst, size_t out_channel, size_t out_num_chan } } +float find_peak_plain(const float *samples, size_t num_samples) __attribute__((unused)); + +float find_peak_plain(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; +} + +#ifdef __SSE__ +static inline float horizontal_max(__m128 m) +{ + __m128 tmp = _mm_shuffle_ps(m, m, _MM_SHUFFLE(1, 0, 3, 2)); + m = _mm_max_ps(m, tmp); + tmp = _mm_shuffle_ps(m, m, _MM_SHUFFLE(2, 3, 0, 1)); + m = _mm_max_ps(m, tmp); + return _mm_cvtss_f32(m); +} + +float find_peak(const float *samples, size_t num_samples) +{ + const __m128 abs_mask = _mm_castsi128_ps(_mm_set1_epi32(0x7fffffffu)); + __m128 m = _mm_setzero_ps(); + for (size_t i = 0; i < (num_samples & ~3); i += 4) { + __m128 x = _mm_loadu_ps(samples + i); + x = _mm_and_ps(x, abs_mask); + m = _mm_max_ps(m, x); + } + float result = horizontal_max(m); + + for (size_t i = (num_samples & ~3); i < num_samples; ++i) { + result = max(result, fabs(samples[i])); + } + +#if 0 + // Self-test. We should be bit-exact the same. + float reference_result = find_peak_plain(samples, num_samples); + if (result != reference_result) { + fprintf(stderr, "Error: Peak is %f [%f %f %f %f]; should be %f.\n", + result, + _mm_cvtss_f32(_mm_shuffle_ps(m, m, _MM_SHUFFLE(0, 0, 0, 0))), + _mm_cvtss_f32(_mm_shuffle_ps(m, m, _MM_SHUFFLE(1, 1, 1, 1))), + _mm_cvtss_f32(_mm_shuffle_ps(m, m, _MM_SHUFFLE(2, 2, 2, 2))), + _mm_cvtss_f32(_mm_shuffle_ps(m, m, _MM_SHUFFLE(3, 3, 3, 3))), + reference_result); + abort(); + } +#endif + return result; +} +#else +float find_peak(const float *samples, size_t num_samples) +{ + return find_peak_plain(samples, num_samples); +} +#endif + +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); + global_audio_mixer = this; - 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); + eq[bus_index][EQ_BAND_BASS].init(FILTER_LOW_SHELF, 1); + // Note: EQ_BAND_MID isn't used (see comments in apply_eq()). + eq[bus_index][EQ_BAND_TREBLE].init(FILTER_HIGH_SHELF, 1); + compressor[bus_index].reset(new StereoCompressor(OUTPUT_FREQUENCY)); + level_compressor[bus_index].reset(new StereoCompressor(OUTPUT_FREQUENCY)); + + set_bus_settings(bus_index, get_default_bus_settings()); + } set_limiter_enabled(global_flags.limiter_enabled); set_final_makeup_gain_auto(global_flags.final_makeup_gain_auto); + alsa_pool.init(); + + if (!global_flags.input_mapping_filename.empty()) { + current_mapping_mode = MappingMode::MULTICHANNEL; + InputMapping new_input_mapping; + if (!load_input_mapping_from_file(get_devices(), + global_flags.input_mapping_filename, + &new_input_mapping)) { + fprintf(stderr, "Failed to load input mapping from '%s', exiting.\n", + global_flags.input_mapping_filename.c_str()); + exit(1); + } + set_input_mapping(new_input_mapping); + } else { + set_simple_input(/*card_index=*/0); + if (global_flags.multichannel_mapping_mode) { + current_mapping_mode = MappingMode::MULTICHANNEL; + } + } - // Generate a very simple, default input mapping. - InputMapping::Bus input; - input.name = "Main"; - input.device.type = InputSourceType::CAPTURE_CARD; - input.device.index = 0; - input.source_channel[0] = 0; - input.source_channel[1] = 1; + r128.init(2, OUTPUT_FREQUENCY); + r128.integr_start(); - InputMapping new_input_mapping; - new_input_mapping.buses.push_back(input); - set_input_mapping(new_input_mapping); + // 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) +void AudioMixer::reset_resampler(DeviceSpec device_spec) { - lock_guard lock(audio_mutex); - reset_device_mutex_held(device_spec); + lock_guard lock(audio_mutex); + reset_resampler_mutex_held(device_spec); } -void AudioMixer::reset_device_mutex_held(DeviceSpec 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 { @@ -125,22 +232,24 @@ 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) +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); - 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(); assert(num_channels > 0); // Convert the audio to fp32. - vector audio; - audio.resize(num_samples * num_channels); + unique_ptr audio(new float[num_samples * num_channels]); unsigned channel_index = 0; for (auto channel_it = device->interesting_channels.cbegin(); channel_it != device->interesting_channels.end(); ++channel_it, ++channel_index) { switch (audio_format.bits_per_sample) { @@ -148,13 +257,13 @@ void AudioMixer::add_audio(DeviceSpec device_spec, const uint8_t *data, unsigned assert(num_samples == 0); break; case 16: - convert_fixed16_to_fp32(&audio[0], channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples); + convert_fixed16_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples); break; case 24: - convert_fixed24_to_fp32(&audio[0], channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples); + convert_fixed24_to_fp32(audio.get(), 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); + convert_fixed32_to_fp32(audio.get(), 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); @@ -164,18 +273,22 @@ void AudioMixer::add_audio(DeviceSpec device_spec, const uint8_t *data, unsigned // Now add it. int64_t local_pts = device->next_local_pts; - device->resampling_queue->add_input_samples(local_pts / double(TIMEBASE), audio.data(), num_samples); + device->resampling_queue->add_input_samples(local_pts / double(TIMEBASE), audio.get(), 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 +302,81 @@ void AudioMixer::add_silence(DeviceSpec device_spec, unsigned samples_per_frame, // is always the same. device->next_local_pts += frame_length; } + return true; +} + +bool AudioMixer::silence_card(DeviceSpec device_spec, bool silence) +{ + 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->silenced && !silence) { + reset_resampler_mutex_held(device_spec); + } + device->silenced = silence; + return true; +} + +AudioMixer::BusSettings AudioMixer::get_default_bus_settings() +{ + BusSettings settings; + settings.fader_volume_db = 0.0f; + settings.muted = false; + settings.locut_enabled = global_flags.locut_enabled; + for (unsigned band_index = 0; band_index < NUM_EQ_BANDS; ++band_index) { + settings.eq_level_db[band_index] = 0.0f; + } + settings.gain_staging_db = global_flags.initial_gain_staging_db; + settings.level_compressor_enabled = global_flags.gain_staging_auto; + settings.compressor_threshold_dbfs = ref_level_dbfs - 12.0f; // -12 dB. + settings.compressor_enabled = global_flags.compressor_enabled; + return settings; +} + +AudioMixer::BusSettings AudioMixer::get_bus_settings(unsigned bus_index) const +{ + lock_guard lock(audio_mutex); + BusSettings settings; + settings.fader_volume_db = fader_volume_db[bus_index]; + settings.muted = mute[bus_index]; + settings.locut_enabled = locut_enabled[bus_index]; + for (unsigned band_index = 0; band_index < NUM_EQ_BANDS; ++band_index) { + settings.eq_level_db[band_index] = eq_level_db[bus_index][band_index]; + } + settings.gain_staging_db = gain_staging_db[bus_index]; + settings.level_compressor_enabled = level_compressor_enabled[bus_index]; + settings.compressor_threshold_dbfs = compressor_threshold_dbfs[bus_index]; + settings.compressor_enabled = compressor_enabled[bus_index]; + return settings; +} + +void AudioMixer::set_bus_settings(unsigned bus_index, const AudioMixer::BusSettings &settings) +{ + lock_guard lock(audio_mutex); + fader_volume_db[bus_index] = settings.fader_volume_db; + mute[bus_index] = settings.muted; + locut_enabled[bus_index] = settings.locut_enabled; + for (unsigned band_index = 0; band_index < NUM_EQ_BANDS; ++band_index) { + eq_level_db[bus_index][band_index] = settings.eq_level_db[band_index]; + } + gain_staging_db[bus_index] = settings.gain_staging_db; + last_gain_staging_db[bus_index] = gain_staging_db[bus_index]; + level_compressor_enabled[bus_index] = settings.level_compressor_enabled; + compressor_threshold_dbfs[bus_index] = settings.compressor_threshold_dbfs; + compressor_enabled[bus_index] = settings.compressor_enabled; } 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 +384,9 @@ 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 +395,27 @@ 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)); + memset(output, 0, num_samples * 2 * 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; @@ -244,102 +430,129 @@ void AudioMixer::fill_audio_bus(const vector *samples_card, const InputMa } } +vector AudioMixer::get_active_devices() const +{ + vector ret; + for (unsigned card_index = 0; card_index < MAX_VIDEO_CARDS; ++card_index) { + const DeviceSpec device_spec{InputSourceType::CAPTURE_CARD, card_index}; + if (!find_audio_device(device_spec)->interesting_channels.empty()) { + ret.push_back(device_spec); + } + } + for (unsigned card_index = 0; card_index < MAX_ALSA_CARDS; ++card_index) { + const DeviceSpec device_spec{InputSourceType::ALSA_INPUT, card_index}; + if (!find_audio_device(device_spec)->interesting_channels.empty()) { + ret.push_back(device_spec); + } + } + return ret; +} + +namespace { + +void apply_gain(float db, float last_db, vector *samples) +{ + if (fabs(db - last_db) < 1e-3) { + // Constant over this frame. + const float gain = from_db(db); + for (size_t i = 0; i < samples->size(); ++i) { + (*samples)[i] *= gain; + } + } else { + // We need to do a fade. + unsigned num_samples = samples->size() / 2; + float gain = from_db(last_db); + const float gain_inc = pow(from_db(db - last_db), 1.0 / num_samples); + for (size_t i = 0; i < num_samples; ++i) { + (*samples)[i * 2 + 0] *= gain; + (*samples)[i * 2 + 1] *= gain; + gain *= gain_inc; + } + } +} + +} // namespace + 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. - // 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 (unsigned card_index = 0; card_index < num_cards; ++card_index) { - AudioDevice *device = &cards[card_index]; - if (!device->interesting_channels.empty()) { - samples_card[card_index].resize(num_samples * device->interesting_channels.size()); + for (const DeviceSpec &device_spec : get_active_devices()) { + AudioDevice *device = find_audio_device(device_spec); + samples_card[device_spec].resize(num_samples * device->interesting_channels.size()); + if (device->silenced) { + memset(&samples_card[device_spec][0], 0, samples_card[device_spec].size() * sizeof(float)); + } else { 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]); + apply_eq(bus_index, &samples_bus); - 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) { - 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 db = gain_staging_db[bus_index]; + float last_db = last_gain_staging_db[bus_index]; + apply_gain(db, last_db, &samples_bus); + } + last_gain_staging_db[bus_index] = gain_staging_db[bus_index]; + +#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(); - } + add_bus_to_master(bus_index, samples_bus, &samples_out); + deinterleave_samples(samples_bus, &left, &right); + measure_bus_levels(bus_index, left, right); + } + + { + 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. @@ -356,7 +569,8 @@ 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 @@ -368,14 +582,13 @@ 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 current_makeup_lu = to_db(final_makeup_gain); + double loudness_lu = r128.loudness_M() - ref_level_lufs; target_loudness_factor = final_makeup_gain * from_db(-loudness_lu); - // If we're outside +/- 5 LU uncorrected, we don't count it as + // If we're outside +/- 5 LU (after correction), we don't count it as // a normal signal (probably silence) and don't change the // correction factor; just apply what we already have. - if (fabs(loudness_lu - current_makeup_lu) >= 5.0 || !final_makeup_gain_auto) { + if (fabs(loudness_lu) >= 5.0 || !final_makeup_gain_auto) { alpha = 0.0; } else { // Formula adapted from @@ -396,36 +609,352 @@ vector AudioMixer::get_output(double pts, unsigned num_samples, Resamplin final_makeup_gain = m; } + update_meters(samples_out); + return samples_out; } -map AudioMixer::get_names() const +namespace { + +void apply_filter_fade(StereoFilter *filter, float *data, unsigned num_samples, float cutoff_hz, float db, float last_db) { - lock_guard lock(audio_mutex); - map names; + // A granularity of 32 samples is an okay tradeoff between speed and + // smoothness; recalculating the filters is pretty expensive, so it's + // good that we don't do this all the time. + static constexpr unsigned filter_granularity_samples = 32; + + const float cutoff_linear = cutoff_hz * 2.0 * M_PI / OUTPUT_FREQUENCY; + if (fabs(db - last_db) < 1e-3) { + // Constant over this frame. + if (fabs(db) > 0.01f) { + filter->render(data, num_samples, cutoff_linear, 0.5f, db / 40.0f); + } + } else { + // We need to do a fade. (Rounding up avoids division by zero.) + unsigned num_blocks = (num_samples + filter_granularity_samples - 1) / filter_granularity_samples; + const float inc_db_norm = (db - last_db) / 40.0f / num_blocks; + float db_norm = db / 40.0f; + for (size_t i = 0; i < num_samples; i += filter_granularity_samples) { + size_t samples_this_block = std::min(num_samples - i, filter_granularity_samples); + filter->render(data + i * 2, samples_this_block, cutoff_linear, 0.5f, db_norm); + db_norm += inc_db_norm; + } + } +} + +} // namespace + +void AudioMixer::apply_eq(unsigned bus_index, vector *samples_bus) +{ + constexpr float bass_freq_hz = 200.0f; + constexpr float treble_freq_hz = 4700.0f; + + // 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); + } + + // Apply the rest of the EQ. Since we only have a simple three-band EQ, + // we can implement it with two shelf filters. We use a simple gain to + // set the mid-level filter, and then offset the low and high bands + // from that if we need to. (We could perhaps have folded the gain into + // the next part, but it's so cheap that the trouble isn't worth it.) + // + // If any part of the EQ has changed appreciably since last frame, + // we fade smoothly during the course of this frame. + const float bass_db = eq_level_db[bus_index][EQ_BAND_BASS]; + const float mid_db = eq_level_db[bus_index][EQ_BAND_MID]; + const float treble_db = eq_level_db[bus_index][EQ_BAND_TREBLE]; + + const float last_bass_db = last_eq_level_db[bus_index][EQ_BAND_BASS]; + const float last_mid_db = last_eq_level_db[bus_index][EQ_BAND_MID]; + const float last_treble_db = last_eq_level_db[bus_index][EQ_BAND_TREBLE]; + + assert(samples_bus->size() % 2 == 0); + const unsigned num_samples = samples_bus->size() / 2; + + apply_gain(mid_db, last_mid_db, samples_bus); + + apply_filter_fade(&eq[bus_index][EQ_BAND_BASS], samples_bus->data(), num_samples, bass_freq_hz, bass_db - mid_db, last_bass_db - last_mid_db); + apply_filter_fade(&eq[bus_index][EQ_BAND_TREBLE], samples_bus->data(), num_samples, treble_freq_hz, treble_db - mid_db, last_treble_db - last_mid_db); + + last_eq_level_db[bus_index][EQ_BAND_BASS] = bass_db; + last_eq_level_db[bus_index][EQ_BAND_MID] = mid_db; + last_eq_level_db[bus_index][EQ_BAND_TREBLE] = treble_db; +} + +void AudioMixer::add_bus_to_master(unsigned bus_index, const vector &samples_bus, vector *samples_out) +{ + assert(samples_bus.size() == samples_out->size()); + assert(samples_bus.size() % 2 == 0); + unsigned num_samples = samples_bus.size() / 2; + const float new_volume_db = mute[bus_index] ? -90.0f : fader_volume_db[bus_index].load(); + if (fabs(new_volume_db - last_fader_volume_db[bus_index]) > 1e-3) { + // The volume has changed; do a fade over the course of this frame. + // (We might have some numerical issues here, but it seems to sound OK.) + // For the purpose of fading here, the silence floor is set to -90 dB + // (the fader only goes to -84). + float old_volume = from_db(max(last_fader_volume_db[bus_index], -90.0f)); + float volume = from_db(max(new_volume_db, -90.0f)); + + float volume_inc = pow(volume / old_volume, 1.0 / num_samples); + volume = old_volume; + if (bus_index == 0) { + for (unsigned i = 0; i < num_samples; ++i) { + (*samples_out)[i * 2 + 0] = samples_bus[i * 2 + 0] * volume; + (*samples_out)[i * 2 + 1] = samples_bus[i * 2 + 1] * volume; + volume *= volume_inc; + } + } else { + for (unsigned i = 0; i < num_samples; ++i) { + (*samples_out)[i * 2 + 0] += samples_bus[i * 2 + 0] * volume; + (*samples_out)[i * 2 + 1] += samples_bus[i * 2 + 1] * volume; + volume *= volume_inc; + } + } + } else if (new_volume_db > -90.0f) { + float volume = from_db(new_volume_db); + if (bus_index == 0) { + for (unsigned i = 0; i < num_samples; ++i) { + (*samples_out)[i * 2 + 0] = samples_bus[i * 2 + 0] * volume; + (*samples_out)[i * 2 + 1] = samples_bus[i * 2 + 1] * volume; + } + } else { + for (unsigned i = 0; i < num_samples; ++i) { + (*samples_out)[i * 2 + 0] += samples_bus[i * 2 + 0] * volume; + (*samples_out)[i * 2 + 1] += samples_bus[i * 2 + 1] * volume; + } + } + } + + last_fader_volume_db[bus_index] = new_volume_db; +} + +void AudioMixer::measure_bus_levels(unsigned bus_index, const vector &left, const vector &right) +{ + assert(left.size() == right.size()); + const float volume = mute[bus_index] ? 0.0f : from_db(fader_volume_db[bus_index]); + const float peak_levels[2] = { + find_peak(left.data(), left.size()) * volume, + find_peak(right.data(), right.size()) * volume + }; + for (unsigned channel = 0; channel < 2; ++channel) { + // Compute the current value, including hold and falloff. + // The constants are borrowed from zita-mu1 by Fons Adriaensen. + static constexpr float hold_sec = 0.5f; + static constexpr float falloff_db_sec = 15.0f; // dB/sec falloff after hold. + float current_peak; + PeakHistory &history = peak_history[bus_index][channel]; + history.historic_peak = max(history.historic_peak, peak_levels[channel]); + if (history.age_seconds < hold_sec) { + current_peak = history.last_peak; + } else { + current_peak = history.last_peak * from_db(-falloff_db_sec * (history.age_seconds - hold_sec)); + } + + // See if we have a new peak to replace the old (possibly falling) one. + if (peak_levels[channel] > current_peak) { + history.last_peak = peak_levels[channel]; + history.age_seconds = 0.0f; // Not 100% correct, but more than good enough given our frame sizes. + current_peak = peak_levels[channel]; + } else { + history.age_seconds += float(left.size()) / OUTPUT_FREQUENCY; + } + history.current_level = peak_levels[channel]; + history.current_peak = current_peak; + } +} + +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_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_levels.size(); ++bus_index) { + bus_levels[bus_index].current_level_dbfs[0] = to_db(peak_history[bus_index][0].current_level); + bus_levels[bus_index].current_level_dbfs[1] = to_db(peak_history[bus_index][1].current_level); + bus_levels[bus_index].peak_level_dbfs[0] = to_db(peak_history[bus_index][0].current_peak); + bus_levels[bus_index].peak_level_dbfs[1] = to_db(peak_history[bus_index][1].current_peak); + bus_levels[bus_index].historic_peak_dbfs = to_db( + max(peak_history[bus_index][0].historic_peak, + peak_history[bus_index][1].historic_peak)); + 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() +{ + lock_guard lock(audio_mutex); + + map devices; for (unsigned card_index = 0; card_index < num_cards; ++card_index) { const DeviceSpec spec{ InputSourceType::CAPTURE_CARD, card_index }; - const AudioDevice *device = &cards[card_index]; - names.insert(make_pair(spec, device->name)); + const AudioDevice *device = &video_cards[card_index]; + DeviceInfo info; + info.display_name = device->display_name; + info.num_channels = 8; + devices.insert(make_pair(spec, info)); + } + vector available_alsa_devices = alsa_pool.get_devices(); + for (unsigned card_index = 0; card_index < available_alsa_devices.size(); ++card_index) { + const DeviceSpec spec{ InputSourceType::ALSA_INPUT, card_index }; + const ALSAPool::Device &device = available_alsa_devices[card_index]; + DeviceInfo info; + info.display_name = device.display_name(); + info.num_channels = device.num_channels; + info.alsa_name = device.name; + info.alsa_info = device.info; + info.alsa_address = device.address; + devices.insert(make_pair(spec, info)); } - return names; + return devices; } -void AudioMixer::set_name(DeviceSpec device_spec, const string &name) +void AudioMixer::set_display_name(DeviceSpec device_spec, const string &name) { AudioDevice *device = find_audio_device(device_spec); - lock_guard lock(audio_mutex); - device->name = name; + lock_guard lock(audio_mutex); + device->display_name = name; +} + +void AudioMixer::serialize_device(DeviceSpec device_spec, DeviceSpecProto *device_spec_proto) +{ + lock_guard lock(audio_mutex); + switch (device_spec.type) { + case InputSourceType::SILENCE: + device_spec_proto->set_type(DeviceSpecProto::SILENCE); + break; + case InputSourceType::CAPTURE_CARD: + device_spec_proto->set_type(DeviceSpecProto::CAPTURE_CARD); + device_spec_proto->set_index(device_spec.index); + device_spec_proto->set_display_name(video_cards[device_spec.index].display_name); + break; + case InputSourceType::ALSA_INPUT: + alsa_pool.serialize_device(device_spec.index, device_spec_proto); + break; + } +} + +void AudioMixer::set_simple_input(unsigned card_index) +{ + InputMapping new_input_mapping; + InputMapping::Bus input; + input.name = "Main"; + input.device.type = InputSourceType::CAPTURE_CARD; + input.device.index = card_index; + input.source_channel[0] = 0; + input.source_channel[1] = 1; + + new_input_mapping.buses.push_back(input); + + lock_guard lock(audio_mutex); + current_mapping_mode = MappingMode::SIMPLE; + set_input_mapping_lock_held(new_input_mapping); + fader_volume_db[0] = 0.0f; +} + +unsigned AudioMixer::get_simple_input() const +{ + lock_guard lock(audio_mutex); + if (input_mapping.buses.size() == 1 && + input_mapping.buses[0].device.type == InputSourceType::CAPTURE_CARD && + input_mapping.buses[0].source_channel[0] == 0 && + input_mapping.buses[0].source_channel[1] == 1) { + return input_mapping.buses[0].device.index; + } else { + return numeric_limits::max(); + } } void AudioMixer::set_input_mapping(const InputMapping &new_input_mapping) { - lock_guard lock(audio_mutex); + lock_guard lock(audio_mutex); + set_input_mapping_lock_held(new_input_mapping); + current_mapping_mode = MappingMode::MULTICHANNEL; +} +AudioMixer::MappingMode AudioMixer::get_mapping_mode() const +{ + lock_guard lock(audio_mutex); + return current_mapping_mode; +} + +void AudioMixer::set_input_mapping_lock_held(const InputMapping &new_input_mapping) +{ 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].insert(bus.source_channel[channel]); @@ -435,12 +964,26 @@ void AudioMixer::set_input_mapping(const InputMapping &new_input_mapping) } // 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) { - DeviceSpec device_spec{InputSourceType::CAPTURE_CARD, card_index}; - AudioDevice *device = &cards[card_index]; + for (unsigned card_index = 0; card_index < MAX_VIDEO_CARDS; ++card_index) { + const DeviceSpec device_spec{InputSourceType::CAPTURE_CARD, card_index}; + AudioDevice *device = find_audio_device(device_spec); if (device->interesting_channels != interesting_channels[device_spec]) { device->interesting_channels = interesting_channels[device_spec]; - reset_device_mutex_held(DeviceSpec{InputSourceType::CAPTURE_CARD, card_index}); + reset_resampler_mutex_held(device_spec); + } + } + for (unsigned card_index = 0; card_index < MAX_ALSA_CARDS; ++card_index) { + const DeviceSpec device_spec{InputSourceType::ALSA_INPUT, card_index}; + AudioDevice *device = find_audio_device(device_spec); + if (interesting_channels[device_spec].empty()) { + alsa_pool.release_device(card_index); + } else { + alsa_pool.hold_device(card_index); + } + if (device->interesting_channels != interesting_channels[device_spec]) { + device->interesting_channels = interesting_channels[device_spec]; + alsa_pool.reset_device(device_spec.index); + reset_resampler_mutex_held(device_spec); } } @@ -449,6 +992,27 @@ 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; } + +unsigned AudioMixer::num_buses() const +{ + lock_guard lock(audio_mutex); + return input_mapping.buses.size(); +} + +void AudioMixer::reset_peak(unsigned bus_index) +{ + lock_guard lock(audio_mutex); + for (unsigned channel = 0; channel < 2; ++channel) { + PeakHistory &history = peak_history[bus_index][channel]; + history.current_level = 0.0f; + history.historic_peak = 0.0f; + history.current_peak = 0.0f; + history.last_peak = 0.0f; + history.age_seconds = 0.0f; + } +} + +AudioMixer *global_audio_mixer = nullptr;