#include <endian.h>
#include <bmusb/bmusb.h>
#include <stdio.h>
+#include <endian.h>
#include <cmath>
#include "db.h"
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_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) {
- 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);
+ 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_channels, const uint8_t *src, size_t in_channels, size_t num_samples)
+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_channels >= out_channels);
+ 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) {
- 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);
+ int32_t s = le32toh(*(int32_t *)src);
+ *dst = s * (1.0f / 2147483648.0f);
+
+ src += 4 * in_num_channels;
+ dst += out_num_channels;
}
}
assert(num_channels > 0);
// Convert the audio to stereo fp32.
- // FIXME: Pick out the right channels; this takes the first ones.
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 = card->interesting_channels.cbegin(); channel_it != card->interesting_channels.end(); ++channel_it, ++channel_index) {
+ switch (audio_format.bits_per_sample) {
+ case 0:
+ assert(num_samples == 0);
+ 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.
*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();
+ CaptureCard *card = &cards[card_index];
+ unsigned channel_index = 0;
+ for (int channel : card->interesting_channels) {
+ if (channel == source_channel) break;
+ ++channel_index;
+ }
+ assert(channel_index < card->interesting_channels.size());
+ *srcptr = &samples_card[card_index][channel_index];
+ *stride = card->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.input_source_type == InputSourceType::SILENCE) {
+ memset(output, 0, num_samples * sizeof(*output));
+ } else {
+ assert(bus.input_source_type == InputSourceType::CAPTURE_CARD);
+ const float *lsrc, *rsrc;
+ unsigned lstride, rstride;
+ float *dptr = output;
+ find_sample_src_from_capture_card(samples_card, bus.input_source_index, bus.source_channel[0], &lsrc, &lstride);
+ find_sample_src_from_capture_card(samples_card, bus.input_source_index, 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)
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) {
- const InputMapping::Bus &input = input_mapping.buses[bus_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;
- }
- }
+ 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) {