]> git.sesse.net Git - nageru/blobdiff - audio_mixer.cpp
Communicate number of channels to the input mapping dialog.
[nageru] / audio_mixer.cpp
index 7a36b0de255e53daeeb96e0fcc2de5521b411364..b61a3c300009eda219179863617914325517ba1e 100644 (file)
@@ -4,6 +4,7 @@
 #include <endian.h>
 #include <bmusb/bmusb.h>
 #include <stdio.h>
+#include <endian.h>
 #include <cmath>
 
 #include "db.h"
@@ -15,33 +16,63 @@ using namespace std;
 
 namespace {
 
-// TODO: If these prove to be a bottleneck, they can be SSSE3-optimized.
+// TODO: If these prove to be a bottleneck, they can be SSSE3-optimized
+// (usually including multiple channels at a time).
 
-void convert_fixed24_to_fp32(float *dst, size_t out_channels, const uint8_t *src, size_t in_channels, size_t num_samples)
+void convert_fixed16_to_fp32(float *dst, size_t out_channel, size_t out_num_channels,
+                             const uint8_t *src, size_t in_channel, size_t in_num_channels,
+                             size_t num_samples)
 {
-       assert(in_channels >= out_channels);
+       assert(in_channel < in_num_channels);
+       assert(out_channel < out_num_channels);
+       src += in_channel * 2;
+       dst += out_channel;
+
        for (size_t i = 0; i < num_samples; ++i) {
-               for (size_t j = 0; j < out_channels; ++j) {
-                       uint32_t s1 = *src++;
-                       uint32_t s2 = *src++;
-                       uint32_t s3 = *src++;
-                       uint32_t s = s1 | (s1 << 8) | (s2 << 16) | (s3 << 24);
-                       dst[i * out_channels + j] = int(s) * (1.0f / 2147483648.0f);
-               }
-               src += 3 * (in_channels - out_channels);
+               int16_t s = le16toh(*(int16_t *)src);
+               *dst = s * (1.0f / 32768.0f);
+
+               src += 2 * in_num_channels;
+               dst += out_num_channels;
        }
 }
 
-void convert_fixed32_to_fp32(float *dst, size_t out_channels, const uint8_t *src, size_t in_channels, size_t num_samples)
+void convert_fixed24_to_fp32(float *dst, size_t out_channel, size_t out_num_channels,
+                             const uint8_t *src, size_t in_channel, size_t in_num_channels,
+                             size_t num_samples)
 {
-       assert(in_channels >= out_channels);
+       assert(in_channel < in_num_channels);
+       assert(out_channel < out_num_channels);
+       src += in_channel * 3;
+       dst += out_channel;
+
        for (size_t i = 0; i < num_samples; ++i) {
-               for (size_t j = 0; j < out_channels; ++j) {
-                       int32_t s = le32toh(*(int32_t *)src);
-                       dst[i * out_channels + j] = s * (1.0f / 2147483648.0f);
-                       src += 4;
-               }
-               src += 4 * (in_channels - out_channels);
+               uint32_t s1 = src[0];
+               uint32_t s2 = src[1];
+               uint32_t s3 = src[2];
+               uint32_t s = s1 | (s1 << 8) | (s2 << 16) | (s3 << 24);
+               *dst = int(s) * (1.0f / 2147483648.0f);
+
+               src += 3 * in_num_channels;
+               dst += out_num_channels;
+       }
+}
+
+void convert_fixed32_to_fp32(float *dst, size_t out_channel, size_t out_num_channels,
+                             const uint8_t *src, size_t in_channel, size_t in_num_channels,
+                             size_t num_samples)
+{
+       assert(in_channel < in_num_channels);
+       assert(out_channel < out_num_channels);
+       src += in_channel * 4;
+       dst += out_channel;
+
+       for (size_t i = 0; i < num_samples; ++i) {
+               int32_t s = le32toh(*(int32_t *)src);
+               *dst = s * (1.0f / 2147483648.0f);
+
+               src += 4 * in_num_channels;
+               dst += out_num_channels;
        }
 }
 
@@ -63,123 +94,172 @@ AudioMixer::AudioMixer(unsigned num_cards)
        set_final_makeup_gain_auto(global_flags.final_makeup_gain_auto);
 
        // Generate a very simple, default input mapping.
-       InputMapping::Input input;
+       InputMapping::Bus input;
        input.name = "Main";
-       input.input_source_type = InputSourceType::CAPTURE_CARD;
-       input.input_source_index = 0;
+       input.device.type = InputSourceType::CAPTURE_CARD;
+       input.device.index = 0;
        input.source_channel[0] = 0;
        input.source_channel[1] = 1;
 
        InputMapping new_input_mapping;
-       new_input_mapping.inputs.push_back(input);
+       new_input_mapping.buses.push_back(input);
        set_input_mapping(new_input_mapping);
 }
 
-void AudioMixer::reset_card(unsigned card_index)
+void AudioMixer::reset_device(DeviceSpec device_spec)
 {
        lock_guard<mutex> lock(audio_mutex);
-       reset_card_mutex_held(card_index);
+       reset_device_mutex_held(device_spec);
 }
 
-void AudioMixer::reset_card_mutex_held(unsigned card_index)
+void AudioMixer::reset_device_mutex_held(DeviceSpec device_spec)
 {
-       CaptureCard *card = &cards[card_index];
-       if (card->interesting_channels.empty()) {
-               card->resampling_queue.reset();
+       AudioDevice *device = find_audio_device(device_spec);
+       if (device->interesting_channels.empty()) {
+               device->resampling_queue.reset();
        } else {
-               card->resampling_queue.reset(new ResamplingQueue(card_index, OUTPUT_FREQUENCY, OUTPUT_FREQUENCY, card->interesting_channels.size()));
+               // TODO: ResamplingQueue should probably take the full device spec.
+               // (It's only used for console output, though.)
+               device->resampling_queue.reset(new ResamplingQueue(device_spec.index, device->capture_frequency, OUTPUT_FREQUENCY, device->interesting_channels.size()));
        }
-       card->next_local_pts = 0;
+       device->next_local_pts = 0;
 }
 
-void AudioMixer::add_audio(unsigned card_index, const uint8_t *data, unsigned num_samples, AudioFormat audio_format, int64_t frame_length)
+void AudioMixer::add_audio(DeviceSpec device_spec, const uint8_t *data, unsigned num_samples, AudioFormat audio_format, int64_t frame_length)
 {
-       lock_guard<mutex> lock(audio_mutex);
-       CaptureCard *card = &cards[card_index];
+       AudioDevice *device = find_audio_device(device_spec);
 
-       if (card->resampling_queue == nullptr) {
-               // No inputs use this card; throw it away.
+       lock_guard<mutex> lock(audio_mutex);
+       if (device->resampling_queue == nullptr) {
+               // No buses use this device; throw it away.
                return;
        }
 
-       unsigned num_channels = card->interesting_channels.size();
+       unsigned num_channels = device->interesting_channels.size();
        assert(num_channels > 0);
 
-       // Convert the audio to stereo fp32.
-       // FIXME: Pick out the right channels; this takes the first ones.
+       // Convert the audio to fp32.
        vector<float> audio;
        audio.resize(num_samples * num_channels);
-       switch (audio_format.bits_per_sample) {
-       case 0:
-               assert(num_samples == 0);
-               break;
-       case 24:
-               convert_fixed24_to_fp32(&audio[0], num_channels, data, audio_format.num_channels, num_samples);
-               break;
-       case 32:
-               convert_fixed32_to_fp32(&audio[0], num_channels, data, audio_format.num_channels, num_samples);
-               break;
-       default:
-               fprintf(stderr, "Cannot handle audio with %u bits per sample\n", audio_format.bits_per_sample);
-               assert(false);
+       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) {
+               case 0:
+                       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);
+                       break;
+               case 24:
+                       convert_fixed24_to_fp32(&audio[0], channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
+                       break;
+               case 32:
+                       convert_fixed32_to_fp32(&audio[0], channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
+                       break;
+               default:
+                       fprintf(stderr, "Cannot handle audio with %u bits per sample\n", audio_format.bits_per_sample);
+                       assert(false);
+               }
        }
 
        // Now add it.
-       int64_t local_pts = card->next_local_pts;
-       card->resampling_queue->add_input_samples(local_pts / double(TIMEBASE), audio.data(), num_samples);
-       card->next_local_pts = local_pts + frame_length;
+       int64_t local_pts = 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;
 }
 
-void AudioMixer::add_silence(unsigned card_index, unsigned samples_per_frame, unsigned num_frames, int64_t frame_length)
+void AudioMixer::add_silence(DeviceSpec device_spec, unsigned samples_per_frame, unsigned num_frames, int64_t frame_length)
 {
-       CaptureCard *card = &cards[card_index];
-       lock_guard<mutex> lock(audio_mutex);
+       AudioDevice *device = find_audio_device(device_spec);
 
-       if (card->resampling_queue == nullptr) {
-               // No inputs use this card; throw it away.
+       lock_guard<mutex> lock(audio_mutex);
+       if (device->resampling_queue == nullptr) {
+               // No buses use this device; throw it away.
                return;
        }
 
-       unsigned num_channels = card->interesting_channels.size();
+       unsigned num_channels = device->interesting_channels.size();
        assert(num_channels > 0);
 
        vector<float> silence(samples_per_frame * num_channels, 0.0f);
        for (unsigned i = 0; i < num_frames; ++i) {
-               card->resampling_queue->add_input_samples(card->next_local_pts / double(TIMEBASE), silence.data(), samples_per_frame);
+               device->resampling_queue->add_input_samples(device->next_local_pts / double(TIMEBASE), silence.data(), samples_per_frame);
                // Note that if the format changed in the meantime, we have
                // no way of detecting that; we just have to assume the frame length
                // is always the same.
-               card->next_local_pts += frame_length;
+               device->next_local_pts += frame_length;
+       }
+}
+
+AudioMixer::AudioDevice *AudioMixer::find_audio_device(DeviceSpec device)
+{
+       switch (device.type) {
+       case InputSourceType::CAPTURE_CARD:
+               return &cards[device.index];
+               break;
+       case InputSourceType::SILENCE:
+       default:
+               assert(false);
        }
+       return nullptr;
 }
 
-void AudioMixer::find_sample_src_from_capture_card(const vector<float> *samples_card, unsigned card_index, int source_channel, const float **srcptr, unsigned *stride)
+void AudioMixer::find_sample_src_from_device(const vector<float> *samples_card, DeviceSpec device_spec, int source_channel, const float **srcptr, unsigned *stride)
 {
        static float zero = 0.0f;
-       if (source_channel == -1) {
+       if (source_channel == -1 || device_spec.type == InputSourceType::SILENCE) {
                *srcptr = &zero;
                *stride = 0;
                return;
        }
-       // FIXME: map back through the interesting_channels squeeze map instead of using source_channel
-       // directly, which will be wrong (and might even overrun).
-       *srcptr = &samples_card[card_index][source_channel];
-       *stride = cards[card_index].interesting_channels.size();
+       AudioDevice *device = find_audio_device(device_spec);
+       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];
+       *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)
+{
+       if (bus.device.type == InputSourceType::SILENCE) {
+               memset(output, 0, num_samples * sizeof(*output));
+       } else {
+               assert(bus.device.type == InputSourceType::CAPTURE_CARD);
+               const float *lsrc, *rsrc;
+               unsigned lstride, rstride;
+               float *dptr = output;
+               find_sample_src_from_device(samples_card, bus.device, bus.source_channel[0], &lsrc, &lstride);
+               find_sample_src_from_device(samples_card, bus.device, bus.source_channel[1], &rsrc, &rstride);
+               for (unsigned i = 0; i < num_samples; ++i) {
+                       *dptr++ = *lsrc;
+                       *dptr++ = *rsrc;
+                       lsrc += lstride;
+                       rsrc += rstride;
+               }
+       }
 }
 
 vector<float> AudioMixer::get_output(double pts, unsigned num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy)
 {
-       vector<float> samples_card[MAX_CARDS];
+       vector<float> samples_card[MAX_CARDS];  // TODO: Needs room for other kinds of capture cards.
        vector<float> samples_bus;
 
        lock_guard<mutex> 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) {
-               CaptureCard *card = &cards[card_index];
-               if (!card->interesting_channels.empty()) {
-                       samples_card[card_index].resize(num_samples * card->interesting_channels.size());
-                       card->resampling_queue->get_output_samples(
+               AudioDevice *device = &cards[card_index];
+               if (!device->interesting_channels.empty()) {
+                       samples_card[card_index].resize(num_samples * device->interesting_channels.size());
+                       device->resampling_queue->get_output_samples(
                                pts,
                                &samples_card[card_index][0],
                                num_samples,
@@ -191,28 +271,11 @@ vector<float> AudioMixer::get_output(double pts, unsigned num_samples, Resamplin
        vector<float> samples_out;
        samples_out.resize(num_samples * 2);
        samples_bus.resize(num_samples * 2);
-       for (unsigned input_index = 0; input_index < input_mapping.inputs.size(); ++input_index) {
-               const InputMapping::Input &input = input_mapping.inputs[input_index];
-               if (input.input_source_type == InputSourceType::SILENCE) {
-                       memset(&samples_bus[0], 0, samples_bus.size() * sizeof(samples_bus[0]));
-               } else {
-                       // TODO: Move this into its own function. Can be SSSE3-optimized if need be.
-                       assert(input.input_source_type == InputSourceType::CAPTURE_CARD);
-                       const float *lsrc, *rsrc;
-                       unsigned lstride, rstride;
-                       float *dptr = &samples_bus[0];
-                       find_sample_src_from_capture_card(samples_card, input.input_source_index, input.source_channel[0], &lsrc, &lstride);
-                       find_sample_src_from_capture_card(samples_card, input.input_source_index, input.source_channel[1], &rsrc, &rstride);
-                       for (unsigned i = 0; i < num_samples; ++i) {
-                               *dptr++ = *lsrc;
-                               *dptr++ = *rsrc;
-                               lsrc += lstride;
-                               rsrc += rstride;
-                       }
-               }
+       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[input_index]);
-               if (input_index == 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;
                        }
@@ -336,34 +399,39 @@ vector<float> AudioMixer::get_output(double pts, unsigned num_samples, Resamplin
        return samples_out;
 }
 
-vector<string> AudioMixer::get_names() const
+map<DeviceSpec, DeviceInfo> AudioMixer::get_devices() 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 CaptureCard *card = &cards[card_index];
-               names.push_back(card->name);
+               const DeviceSpec spec{ InputSourceType::CAPTURE_CARD, card_index };
+               const AudioDevice *device = &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;
+       return devices;
 }
 
-void AudioMixer::set_name(unsigned card_index, const string &name)
+void AudioMixer::set_name(DeviceSpec device_spec, const string &name)
 {
+       AudioDevice *device = find_audio_device(device_spec);
+
        lock_guard<mutex> lock(audio_mutex);
-       CaptureCard *card = &cards[card_index];
-       card->name = name;
+       device->name = name;
 }
 
 void AudioMixer::set_input_mapping(const InputMapping &new_input_mapping)
 {
        lock_guard<mutex> lock(audio_mutex);
 
-       map<unsigned, set<unsigned>> interesting_channels;
-       for (const InputMapping::Input &input : new_input_mapping.inputs) {
-               if (input.input_source_type == InputSourceType::CAPTURE_CARD) {
+       map<DeviceSpec, set<unsigned>> interesting_channels;
+       for (const InputMapping::Bus &bus : new_input_mapping.buses) {
+               if (bus.device.type == InputSourceType::CAPTURE_CARD) {
                        for (unsigned channel = 0; channel < 2; ++channel) {
-                               if (input.source_channel[channel] != -1) {
-                                       interesting_channels[input.input_source_index].insert(input.source_channel[channel]);
+                               if (bus.source_channel[channel] != -1) {
+                                       interesting_channels[bus.device].insert(bus.source_channel[channel]);
                                }
                        }
                }
@@ -371,10 +439,11 @@ 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) {
-               CaptureCard *card = &cards[card_index];
-               if (card->interesting_channels != interesting_channels[card_index]) {
-                       card->interesting_channels = interesting_channels[card_index];
-                       reset_card_mutex_held(card_index);
+               DeviceSpec device_spec{InputSourceType::CAPTURE_CARD, card_index};
+               AudioDevice *device = &cards[card_index];
+               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});
                }
        }