#include <limits>
#include <utility>
-#include "db.h"
+#include "decibel.h"
+#include "delay_analyzer.h"
#include "flags.h"
-#include "metrics.h"
+#include "shared/metrics.h"
#include "state.pb.h"
-#include "timebase.h"
+#include "shared/timebase.h"
using namespace bmusb;
using namespace std;
}
}
+void convert_fixed16_to_fixed32(int32_t *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 * 2;
+ dst += out_channel;
+
+ for (size_t i = 0; i < num_samples; ++i) {
+ uint32_t s = uint32_t(uint16_t(le16toh(*(int16_t *)src))) << 16;
+
+ // Keep the sign bit in place, repeat the other 15 bits as far as they go.
+ *dst = s | ((s & 0x7fffffff) >> 15) | ((s & 0x7fffffff) >> 30);
+
+ src += 2 * in_num_channels;
+ dst += out_num_channels;
+ }
+}
+
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)
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);
+ uint32_t s = (s1 << 8) | (s2 << 16) | (s3 << 24); // Note: The bottom eight bits are zero; s3 includes the sign bit.
+ *dst = int(s) * (1.0f / (256.0f * 8388608.0f)); // 256 for signed down-shift by 8, then 2^23 for the actual conversion.
+
+ src += 3 * in_num_channels;
+ dst += out_num_channels;
+ }
+}
+
+void convert_fixed24_to_fixed32(int32_t *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 * 3;
+ dst += out_channel;
+
+ for (size_t i = 0; i < num_samples; ++i) {
+ uint32_t s1 = src[0];
+ uint32_t s2 = src[1];
+ uint32_t s3 = src[2];
+ uint32_t s = (s1 << 8) | (s2 << 16) | (s3 << 24);
+
+ // Keep the sign bit in place, repeat the other 23 bits as far as they go.
+ *dst = s | ((s & 0x7fffffff) >> 23);
src += 3 * in_num_channels;
dst += out_num_channels;
}
}
+// Basically just a reinterleave.
+void convert_fixed32_to_fixed32(int32_t *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;
+
+ src += 4 * in_num_channels;
+ dst += out_num_channels;
+ }
+}
+
float find_peak_plain(const float *samples, size_t num_samples) __attribute__((unused));
float find_peak_plain(const float *samples, size_t num_samples)
// Must happen after ALSAPool is initialized, as it needs to know the card list.
current_mapping_mode = MappingMode::MULTICHANNEL;
InputMapping new_input_mapping;
- if (!load_input_mapping_from_file(get_devices(),
+ if (!load_input_mapping_from_file(get_devices(HOLD_ALSA_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);
+ abort();
}
set_input_mapping(new_input_mapping);
} else {
if (device->interesting_channels.empty()) {
device->resampling_queue.reset();
} else {
+ // Make sure we never get negative delay. Even 1 ms is probably way less than we
+ // could ever hope to actually have; this is just a failsafe.
+ double delay_ms = max(global_flags.audio_queue_length_ms + device->extra_delay_ms, 1.0);
+
device->resampling_queue.reset(new ResamplingQueue(
device_spec, device->capture_frequency, OUTPUT_FREQUENCY, device->interesting_channels.size(),
- global_flags.audio_queue_length_ms * 0.001));
+ delay_ms * 0.001));
}
}
-bool AudioMixer::add_audio(DeviceSpec device_spec, const uint8_t *data, unsigned num_samples, AudioFormat audio_format, int64_t frame_length, steady_clock::time_point frame_time)
+bool AudioMixer::add_audio(DeviceSpec device_spec, const uint8_t *data, unsigned num_samples, AudioFormat audio_format, steady_clock::time_point frame_time)
{
+ if (delay_analyzer != nullptr && delay_analyzer->is_grabbing()) {
+ delay_analyzer->add_audio(device_spec, data, num_samples, audio_format, frame_time);
+ }
+
AudioDevice *device = find_audio_device(device_spec);
unique_lock<timed_mutex> lock(audio_mutex, defer_lock);
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) {
+ convert_audio_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, audio_format, num_samples);
+ }
+
+ // If we changed frequency since last frame, we'll need to reset the resampler.
+ if (audio_format.sample_rate != device->capture_frequency) {
+ device->capture_frequency = audio_format.sample_rate;
+ reset_resampler_mutex_held(device_spec);
+ }
+
+ // Now add it.
+ device->resampling_queue->add_input_samples(frame_time, audio.get(), num_samples, ResamplingQueue::ADJUST_RATE);
+ return true;
+}
+
+// Converts all channels.
+vector<int32_t> convert_audio_to_fixed32(const uint8_t *data, unsigned num_samples, bmusb::AudioFormat audio_format, unsigned num_channels)
+{
+ vector<int32_t> audio;
+
+ if (num_channels > audio_format.num_channels) {
+ audio.resize(num_samples * num_channels, 0);
+ } else {
+ audio.resize(num_samples * num_channels);
+ }
+ for (unsigned channel_index = 0; channel_index < num_channels && channel_index < audio_format.num_channels; ++channel_index) {
switch (audio_format.bits_per_sample) {
case 0:
assert(num_samples == 0);
break;
case 16:
- convert_fixed16_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
+ convert_fixed16_to_fixed32(&audio[0], channel_index, num_channels, data, channel_index, audio_format.num_channels, num_samples);
break;
case 24:
- convert_fixed24_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
+ convert_fixed24_to_fixed32(&audio[0], channel_index, num_channels, data, channel_index, audio_format.num_channels, num_samples);
break;
case 32:
- convert_fixed32_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
+ convert_fixed32_to_fixed32(&audio[0], channel_index, num_channels, data, channel_index, audio_format.num_channels, num_samples);
break;
default:
fprintf(stderr, "Cannot handle audio with %u bits per sample\n", audio_format.bits_per_sample);
}
}
- // If we changed frequency since last frame, we'll need to reset the resampler.
- if (audio_format.sample_rate != device->capture_frequency) {
- device->capture_frequency = audio_format.sample_rate;
- reset_resampler_mutex_held(device_spec);
- }
+ return audio;
+}
- // Now add it.
- device->resampling_queue->add_input_samples(frame_time, audio.get(), num_samples, ResamplingQueue::ADJUST_RATE);
- return true;
+// Converts only one channel.
+void convert_audio_to_fp32(float *dst, size_t out_channel, size_t out_num_channels,
+ const uint8_t *src, size_t in_channel, bmusb::AudioFormat in_audio_format,
+ size_t num_samples)
+{
+ switch (in_audio_format.bits_per_sample) {
+ case 0:
+ assert(num_samples == 0);
+ break;
+ case 16:
+ convert_fixed16_to_fp32(dst, out_channel, out_num_channels, src, in_channel, in_audio_format.num_channels, num_samples);
+ break;
+ case 24:
+ convert_fixed24_to_fp32(dst, out_channel, out_num_channels, src, in_channel, in_audio_format.num_channels, num_samples);
+ break;
+ case 32:
+ convert_fixed32_to_fp32(dst, out_channel, out_num_channels, src, in_channel, in_audio_format.num_channels, num_samples);
+ break;
+ default:
+ fprintf(stderr, "Cannot handle audio with %u bits per sample\n", in_audio_format.bits_per_sample);
+ assert(false);
+ }
}
-bool 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)
{
AudioDevice *device = find_audio_device(device_spec);
correlation.get_correlation());
}
-map<DeviceSpec, DeviceInfo> AudioMixer::get_devices()
+map<DeviceSpec, DeviceInfo> AudioMixer::get_devices(HoldDevices hold_devices)
{
lock_guard<timed_mutex> lock(audio_mutex);
info.num_channels = 8;
devices.insert(make_pair(spec, info));
}
- vector<ALSAPool::Device> available_alsa_devices = alsa_pool.get_devices();
+ vector<ALSAPool::Device> available_alsa_devices = alsa_pool.get_devices(hold_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];
new_input_mapping.buses.push_back(input);
+ // NOTE: Delay is implicitly at 0.0 ms, since none has been set in the mapping.
+
lock_guard<timed_mutex> lock(audio_mutex);
current_mapping_mode = MappingMode::SIMPLE;
set_input_mapping_lock_held(new_input_mapping);
}
// Reset resamplers for all cards that don't have the exact same state as before.
+ map<DeviceSpec, double> new_extra_delay_ms = new_input_mapping.extra_delay_ms; // Convenience so we can use [].
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]) {
+ if (device->interesting_channels != interesting_channels[device_spec] ||
+ device->extra_delay_ms != new_extra_delay_ms[device_spec]) {
device->interesting_channels = interesting_channels[device_spec];
+ device->extra_delay_ms = new_extra_delay_ms[device_spec];
reset_resampler_mutex_held(device_spec);
}
}
} else {
alsa_pool.hold_device(card_index);
}
- if (device->interesting_channels != interesting_channels[device_spec]) {
+ if (device->interesting_channels != interesting_channels[device_spec] ||
+ device->extra_delay_ms != new_extra_delay_ms[device_spec]) {
device->interesting_channels = interesting_channels[device_spec];
+ device->extra_delay_ms = new_extra_delay_ms[device_spec];
alsa_pool.reset_device(device_spec.index);
reset_resampler_mutex_held(device_spec);
}
for (unsigned card_index = 0; card_index < num_ffmpeg_inputs; ++card_index) {
const DeviceSpec device_spec{InputSourceType::FFMPEG_VIDEO_INPUT, card_index};
AudioDevice *device = find_audio_device(device_spec);
- if (device->interesting_channels != interesting_channels[device_spec]) {
+ if (device->interesting_channels != interesting_channels[device_spec] ||
+ device->extra_delay_ms != new_extra_delay_ms[device_spec]) {
device->interesting_channels = interesting_channels[device_spec];
+ device->extra_delay_ms = new_extra_delay_ms[device_spec];
reset_resampler_mutex_held(device_spec);
}
}
projects / nageru / blobdiff