#include <stdio.h>
#include <endian.h>
#include <cmath>
+#ifdef __SSE__
+#include <immintrin.h>
+#endif
#include "db.h"
#include "flags.h"
using namespace bmusb;
using namespace std;
+using namespace std::placeholders;
namespace {
}
}
+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<float> &in, vector<float> *out_l, vector<float> *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);
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);
+}
+
+AudioMixer::~AudioMixer()
+{
+ 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(DeviceSpec device_spec)
+
+void AudioMixer::reset_resampler(DeviceSpec device_spec)
{
- lock_guard<mutex> lock(audio_mutex);
- reset_device_mutex_held(device_spec);
+ lock_guard<timed_mutex> 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 {
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);
+
+ 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);
- lock_guard<mutex> lock(audio_mutex);
+ unique_lock<timed_mutex> 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<float> audio;
- audio.resize(num_samples * num_channels);
+ unique_ptr<float[]> 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) {
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);
// 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<mutex> lock(audio_mutex);
+ unique_lock<timed_mutex> 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();
// 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);
return nullptr;
}
-void AudioMixer::find_sample_src_from_device(const vector<float> *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<DeviceSpec, vector<float>> &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) {
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<float> *samples_card, const InputMapping::Bus &bus, unsigned num_samples, float *output)
+void AudioMixer::fill_audio_bus(const map<DeviceSpec, vector<float>> &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;
vector<float> AudioMixer::get_output(double pts, unsigned num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy)
{
- vector<float> samples_card[MAX_CARDS]; // TODO: Needs room for other kinds of capture cards.
+ map<DeviceSpec, vector<float>> samples_card;
vector<float> samples_bus;
- lock_guard<mutex> lock(audio_mutex);
+ lock_guard<timed_mutex> 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<float> samples_out;
+ vector<float> 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<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
- // (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<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
+ // (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<mutex> 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.
// 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
// 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);
final_makeup_gain = m;
}
+ update_meters(samples_out);
+
return samples_out;
}
-vector<string> AudioMixer::get_names() const
+void AudioMixer::measure_bus_levels(unsigned bus_index, const vector<float> &left, const vector<float> &right)
+{
+ const float *ptrs[] = { left.data(), right.data() };
+ {
+ lock_guard<mutex> lock(audio_measure_mutex);
+ bus_r128[bus_index]->process(left.size(), const_cast<float **>(ptrs));
+ }
+}
+
+void AudioMixer::update_meters(const vector<float> &samples)
+{
+ // Upsample 4x to find interpolated peak.
+ peak_resampler.inp_data = const_cast<float *>(samples.data());
+ peak_resampler.inp_count = samples.size() / 2;
+
+ vector<float> interpolated_samples;
+ interpolated_samples.resize(samples.size());
+ {
+ lock_guard<mutex> 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<float>(peak, find_peak(interpolated_samples.data(), out_stereo_samples * 2));
+ peak_resampler.out_data = nullptr;
+ }
+ }
+
+ // Find R128 levels and L/R correlation.
+ vector<float> left, right;
+ deinterleave_samples(samples, &left, &right);
+ float *ptrs[] = { left.data(), right.data() };
+ {
+ lock_guard<mutex> lock(audio_measure_mutex);
+ r128.process(left.size(), ptrs);
+ correlation.process_samples(samples);
+ }
+
+ send_audio_level_callback();
+}
+
+void AudioMixer::reset_meters()
+{
+ lock_guard<mutex> 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<mutex> 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<BusLevel> bus_levels;
+ bus_levels.resize(input_mapping.buses.size());
+ {
+ lock_guard<mutex> 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<DeviceSpec, DeviceInfo> AudioMixer::get_devices() const
+{
+ lock_guard<timed_mutex> lock(audio_mutex);
+ return get_devices_mutex_held();
+}
+
+map<DeviceSpec, DeviceInfo> AudioMixer::get_devices_mutex_held() const
{
- lock_guard<mutex> lock(audio_mutex);
- vector<string> names;
+ map<DeviceSpec, DeviceInfo> 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));
}
- return names;
+ 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 devices;
}
void AudioMixer::set_name(DeviceSpec device_spec, const string &name)
{
AudioDevice *device = find_audio_device(device_spec);
- lock_guard<mutex> lock(audio_mutex);
+ lock_guard<timed_mutex> lock(audio_mutex);
device->name = name;
}
void AudioMixer::set_input_mapping(const InputMapping &new_input_mapping)
{
- lock_guard<mutex> lock(audio_mutex);
+ lock_guard<timed_mutex> lock(audio_mutex);
- // FIXME: This needs to be keyed on DeviceSpec.
- map<unsigned, set<unsigned>> interesting_channels;
+ map<DeviceSpec, set<unsigned>> 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<mutex> 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);
}
}
InputMapping AudioMixer::get_input_mapping() const
{
- lock_guard<mutex> lock(audio_mutex);
+ lock_guard<timed_mutex> lock(audio_mutex);
return input_mapping;
}