X-Git-Url: https://git.sesse.net/?p=nageru;a=blobdiff_plain;f=audio_mixer.cpp;h=769dabfa74a0831cb6678a52f15d245d2c4ca6f2;hp=93d59f4cc6f4017d9d43bc5ab45ff09b4710fe9f;hb=d9babea9e8b67a7ccbfa931c16b2c7ca38b84c5a;hpb=0f4b5fde73be7c8606d5812b8007cb23b8083bb6

diff --git a/audio_mixer.cpp b/audio_mixer.cpp
index 93d59f4..769dabf 100644
--- a/audio_mixer.cpp
+++ b/audio_mixer.cpp
@@ -13,6 +13,7 @@
 
 using namespace bmusb;
 using namespace std;
+using namespace std::placeholders;
 
 namespace {
 
@@ -76,13 +77,38 @@ 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<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)
+	  compressor(OUTPUT_FREQUENCY),
+	  correlation(OUTPUT_FREQUENCY)
 {
 	locut.init(FILTER_HPF, 2);
 
@@ -104,17 +130,39 @@ 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()
+{
+	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_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 {
@@ -125,14 +173,36 @@ 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);
+
+	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();
@@ -166,16 +236,20 @@ 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<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();
@@ -189,14 +263,16 @@ 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 +280,9 @@ AudioMixer::AudioDevice *AudioMixer::find_audio_device(DeviceSpec device)
 	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) {
@@ -213,23 +291,27 @@ void AudioMixer::find_sample_src_from_device(const vector<float> *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<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;
@@ -246,31 +328,39 @@ void AudioMixer::fill_audio_bus(const vector<float> *samples_card, const InputMa
 
 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]);
 
+		// 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) {
@@ -365,7 +455,7 @@ vector<float> 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);
 
@@ -393,36 +483,133 @@ vector<float> AudioMixer::get_output(double pts, unsigned num_samples, Resamplin
 		final_makeup_gain = m;
 	}
 
+	update_meters(samples_out);
+
 	return samples_out;
 }
 
-map<DeviceSpec, string> AudioMixer::get_names() const
+void AudioMixer::measure_bus_levels(unsigned bus_index, const vector<float> &left, const vector<float> &right)
 {
-	lock_guard<mutex> lock(audio_mutex);
-	map<DeviceSpec, string> names;
+	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<float> bus_loudness;
+	bus_loudness.resize(input_mapping.buses.size());
+	for (unsigned bus_index = 0; bus_index < bus_r128.size(); ++bus_index) {
+		bus_loudness[bus_index] = bus_r128[bus_index]->loudness_S();
+	}
+
+	audio_level_callback(loudness_s, to_db(peak), bus_loudness,
+		loudness_i, loudness_range_low, loudness_range_high,
+		gain_staging_db,
+		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
+{
+	map<DeviceSpec, DeviceInfo> 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.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<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);
 
 	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].insert(bus.source_channel[channel]);
@@ -432,12 +619,26 @@ 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 (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];
-			reset_device_mutex_held(DeviceSpec{InputSourceType::CAPTURE_CARD, card_index});
+			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);
 		}
 	}
 
@@ -446,6 +647,6 @@ void AudioMixer::set_input_mapping(const InputMapping &new_input_mapping)
 
 InputMapping AudioMixer::get_input_mapping() const
 {
-	lock_guard<mutex> lock(audio_mutex);
+	lock_guard<timed_mutex> lock(audio_mutex);
 	return input_mapping;
 }