Do our own resampling

This commit is contained in:
jacqueline
2023-08-04 20:07:44 +10:00
parent 3b240d1cd5
commit 60f7677132
113 changed files with 877 additions and 380595 deletions
+1 -1
View File
@@ -5,7 +5,7 @@
idf_component_register(
SRCS "audio_task.cpp" "chunk.cpp" "fatfs_audio_input.cpp"
"stream_message.cpp" "i2s_audio_output.cpp" "stream_buffer.cpp" "track_queue.cpp"
"stream_event.cpp" "stream_info.cpp" "audio_fsm.cpp" "sink_mixer.cpp"
"stream_event.cpp" "stream_info.cpp" "audio_fsm.cpp" "sink_mixer.cpp" "resample.cpp"
INCLUDE_DIRS "include"
REQUIRES "codecs" "drivers" "cbor" "result" "tasks" "span" "memory" "tinyfsm" "database" "system_fsm" "playlist" "libsamplerate")
+7 -5
View File
@@ -34,6 +34,7 @@
#include "freertos/queue.h"
#include "freertos/ringbuf.h"
#include "pipeline.hpp"
#include "sample.hpp"
#include "sink_mixer.hpp"
#include "span.hpp"
@@ -225,7 +226,7 @@ auto AudioTask::BeginDecoding(InputStream& stream) -> bool {
codecs::ICodec::OutputFormat format = res.second.value();
StreamInfo::Pcm new_format{
.channels = format.num_channels,
.bits_per_sample = format.bits_per_sample,
.bits_per_sample = 32,
.sample_rate = format.sample_rate_hz,
};
@@ -255,7 +256,8 @@ auto AudioTask::ContinueDecoding(InputStream& stream) -> bool {
while (!stream.data().empty()) {
OutputStream writer{codec_buffer_.get()};
auto res = codec_->ContinueStream(stream.data(), writer.data());
auto res =
codec_->ContinueStream(stream.data(), writer.data_as<sample::Sample>());
stream.consume(res.first);
@@ -266,7 +268,7 @@ auto AudioTask::ContinueDecoding(InputStream& stream) -> bool {
return false;
}
} else {
writer.add(res.second->bytes_written);
writer.add(res.second->samples_written * sizeof(sample::Sample));
InputStream reader{codec_buffer_.get()};
SendToSink(reader);
@@ -295,12 +297,12 @@ auto AudioTask::FinishDecoding(InputStream& stream) -> void {
InputStream padded_stream{mad_buffer.get()};
OutputStream writer{codec_buffer_.get()};
auto res = codec_->ContinueStream(stream.data(), writer.data());
auto res = codec_->ContinueStream(stream.data(), writer.data_as<sample::Sample>());
if (res.second.has_error()) {
return;
}
writer.add(res.second->bytes_written);
writer.add(res.second->samples_written * sizeof(sample::Sample));
InputStream reader{codec_buffer_.get()};
SendToSink(reader);
+7
View File
@@ -117,11 +117,18 @@ auto I2SAudioOutput::AdjustVolumeDown() -> bool {
auto I2SAudioOutput::PrepareFormat(const StreamInfo::Pcm& orig)
-> StreamInfo::Pcm {
/*
return StreamInfo::Pcm{
.channels = std::min<uint8_t>(orig.channels, 2),
.bits_per_sample = std::clamp<uint8_t>(orig.bits_per_sample, 16, 32),
.sample_rate = std::clamp<uint32_t>(orig.sample_rate, 8000, 96000),
};
*/
return StreamInfo::Pcm{
.channels = std::min<uint8_t>(orig.channels, 2),
.bits_per_sample = 16,
.sample_rate = 48000,
};
}
auto I2SAudioOutput::Configure(const StreamInfo::Pcm& pcm) -> void {
+131
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@@ -0,0 +1,131 @@
/*
* FIR filter coefficients from resample-1.x smallfilter.h
* see Digital Audio Resampling Home Page located at
* http://ccrma.stanford.edu/~jos/resample/
*/
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+41
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@@ -0,0 +1,41 @@
#pragma once
#include <sys/_stdint.h>
#include <vector>
#include "span.hpp"
#include "sample.hpp"
namespace audio {
class Channel;
class Resampler {
public:
Resampler(uint32_t source_sample_rate,
uint32_t target_sample_rate,
uint8_t num_channels);
~Resampler();
auto source_sample_rate() -> uint32_t { return source_sample_rate_; }
auto target_sample_rate() -> uint32_t { return target_sample_rate_; }
auto channels() -> uint_fast8_t { return num_channels_; }
auto Process(cpp::span<const sample::Sample> input,
cpp::span<sample::Sample> output,
bool end_of_data) -> std::pair<size_t,size_t>;
private:
auto ApplyDither(cpp::span<sample::Sample>) -> void;
uint32_t source_sample_rate_;
uint32_t target_sample_rate_;
uint32_t factor_;
uint8_t num_channels_;
std::vector<Channel> channels_;
};
} // namespace audio
+7 -24
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@@ -10,6 +10,8 @@
#include <cstdint>
#include <memory>
#include "resample.hpp"
#include "sample.hpp"
#include "samplerate.h"
#include "audio_decoder.hpp"
@@ -38,12 +40,10 @@ class SinkMixer {
auto SetTargetFormat(const StreamInfo::Pcm& format) -> void;
auto HandleBytes() -> void;
template <typename T>
auto ConvertFixedToFloating(InputStream&, OutputStream&) -> void;
auto Resample(float, int, InputStream&, OutputStream&) -> void;
template <typename T>
auto Quantise(InputStream&) -> std::size_t;
auto Resample(InputStream&, OutputStream&) -> bool;
auto ApplyDither(cpp::span<sample::Sample> samples, uint_fast8_t bits) -> void;
auto Downscale(cpp::span<sample::Sample>, cpp::span<int16_t>) -> void;
enum class Command {
kReadBytes,
kSetSourceFormat,
@@ -58,31 +58,14 @@ class SinkMixer {
QueueHandle_t commands_;
SemaphoreHandle_t is_idle_;
SRC_STATE* resampler_;
std::unique_ptr<Resampler> resampler_;
std::unique_ptr<RawStream> input_stream_;
std::unique_ptr<RawStream> floating_point_stream_;
std::unique_ptr<RawStream> resampled_stream_;
cpp::span<std::byte> quantisation_buffer_;
cpp::span<short> quantisation_buffer_as_shorts_;
cpp::span<int> quantisation_buffer_as_ints_;
StreamInfo::Pcm target_format_;
StreamBufferHandle_t source_;
StreamBufferHandle_t sink_;
};
template <>
auto SinkMixer::ConvertFixedToFloating<short>(InputStream&, OutputStream&)
-> void;
template <>
auto SinkMixer::ConvertFixedToFloating<int>(InputStream&, OutputStream&)
-> void;
template <>
auto SinkMixer::Quantise<short>(InputStream&) -> std::size_t;
template <>
auto SinkMixer::Quantise<int>(InputStream&) -> std::size_t;
} // namespace audio
+1 -1
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@@ -77,7 +77,7 @@ class StreamInfo {
// The sample rate.
uint32_t sample_rate;
auto real_bytes_per_sample() const -> uint8_t {
auto bytes_per_sample() const -> uint8_t {
return bits_per_sample == 16 ? 2 : 4;
}
+260
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@@ -0,0 +1,260 @@
#include "resample.hpp"
#include <stdint.h>
#include <stdlib.h>
#include <string.h>
#include <algorithm>
#include <numeric>
#include "esp_log.h"
#include "sample.hpp"
#include "stream_info.hpp"
namespace audio {
static constexpr size_t kFilterSize = 1536;
constexpr auto calc_deltas(const std::array<int32_t, kFilterSize>& filter)
-> std::array<int32_t, kFilterSize> {
std::array<int32_t, kFilterSize> deltas;
for (size_t n = 0; n < kFilterSize - 1; n++)
deltas[n] = filter[n + 1] - filter[n];
return deltas;
}
static const std::array<int32_t, kFilterSize> kFilter{
#include "fir.h"
};
static const std::array<int32_t, kFilterSize> kFilterDeltas =
calc_deltas(kFilter);
class Channel {
public:
Channel(uint32_t src_rate,
uint32_t dest_rate,
size_t chunk_size,
size_t skip);
~Channel();
auto output_chunk_size() -> size_t { return output_chunk_size_; }
auto FlushSamples(cpp::span<sample::Sample> out) -> size_t;
auto AddSample(sample::Sample, cpp::span<sample::Sample> out) -> std::size_t;
auto ApplyFilter() -> sample::Sample;
private:
size_t output_chunk_size_;
size_t skip_;
uint32_t factor_; /* factor */
uint32_t time_; /* time */
uint32_t time_per_filter_iteration_; /* output step */
uint32_t filter_step_; /* filter step */
uint32_t filter_end_; /* filter end */
int32_t unity_scale_; /* unity scale */
int32_t samples_per_filter_wing_; /* extra samples */
int32_t latest_sample_; /* buffer index */
cpp::span<int32_t> sample_buffer_; /* the buffer */
};
enum {
Nl = 8, /* 2^Nl samples per zero crossing in fir */
= 8, /* phase bits for filter interpolation */
kPhaseBits = Nl + , /* phase bits (fract of fixed point) */
One = 1 << kPhaseBits,
};
Channel::Channel(uint32_t irate, uint32_t orate, size_t count, size_t skip)
: skip_(skip) {
factor_ = ((uint64_t)orate << kPhaseBits) / irate;
if (factor_ != One) {
time_per_filter_iteration_ = ((uint64_t)irate << kPhaseBits) / orate;
filter_step_ = 1 << (Nl + );
filter_end_ = kFilterSize << ;
samples_per_filter_wing_ = 1 + (filter_end_ / filter_step_);
unity_scale_ = 13128; /* unity scale factor for fir */
if (factor_ < One) {
unity_scale_ *= factor_;
unity_scale_ >>= kPhaseBits;
filter_step_ *= factor_;
filter_step_ >>= kPhaseBits;
samples_per_filter_wing_ *= time_per_filter_iteration_;
samples_per_filter_wing_ >>= kPhaseBits;
}
latest_sample_ = samples_per_filter_wing_;
time_ = latest_sample_ << kPhaseBits;
size_t buf_size = samples_per_filter_wing_ * 2 + count;
int32_t* buf = new int32_t[buf_size];
sample_buffer_ = {buf, buf_size};
count += buf_size; /* account for buffer accumulation */
}
output_chunk_size_ = ((uint64_t)count * factor_) >> kPhaseBits;
}
Channel::~Channel() {
delete sample_buffer_.data();
}
auto Channel::ApplyFilter() -> sample::Sample {
uint32_t iteration, p, i;
int32_t *sample, a;
int64_t value = 0;
// I did my best, but I'll be honest with you I've no idea about any of this
// maths stuff.
// Left wing of the filter.
sample = &sample_buffer_[time_ >> kPhaseBits];
p = time_ & ((1 << kPhaseBits) - 1);
iteration = factor_ < One ? (factor_ * p) >> kPhaseBits : p;
while (iteration < filter_end_) {
i = iteration >> ;
a = iteration & ((1 << ) - 1);
iteration += filter_step_;
a *= kFilterDeltas[i];
a >>= ;
a += kFilter[i];
value += static_cast<int64_t>(*--sample) * a;
}
// Right wing of the filter.
sample = &sample_buffer_[time_ >> kPhaseBits];
p = (One - p) & ((1 << kPhaseBits) - 1);
iteration = factor_ < One ? (factor_ * p) >> kPhaseBits : p;
if (p == 0) /* skip h[0] as it was already been summed above if p == 0 */
iteration += filter_step_;
while (iteration < filter_end_) {
i = iteration >> ;
a = iteration & ((1 << ) - 1);
iteration += filter_step_;
a *= kFilterDeltas[i];
a >>= ;
a += kFilter[i];
value += static_cast<int64_t>(*sample++) * a;
}
/* scale */
value >>= 2;
value *= unity_scale_;
value >>= 27;
return sample::Clip(value);
}
auto Channel::FlushSamples(cpp::span<sample::Sample> out) -> size_t {
size_t zeroes_needed = (2 * samples_per_filter_wing_) - latest_sample_;
size_t produced = 0;
while (zeroes_needed > 0) {
produced += AddSample(0, out.subspan(produced));
zeroes_needed--;
}
return produced;
}
auto Channel::AddSample(sample::Sample in, cpp::span<sample::Sample> out)
-> size_t {
// Add the latest sample to our working buffer.
sample_buffer_[latest_sample_++] = in;
// If we don't have enough samples to run the filter, then bail out and wait
// for more.
if (latest_sample_ < 2 * samples_per_filter_wing_) {
return 0;
}
// Apply the filter to the buffered samples. First, we work out how long (in
// samples) we can run the filter for before running out. This isn't as
// trivial as it might look; e.g. depending on the resampling factor we might
// be doubling the number of samples, or halving them.
uint32_t max_time = (latest_sample_ - samples_per_filter_wing_) << kPhaseBits;
size_t samples_output = 0;
while (time_ < max_time) {
out[skip_ * samples_output++] = ApplyFilter();
time_ += time_per_filter_iteration_;
}
// If we are approaching the end of our buffer, we need to shift all the data
// in it down to the front to make room for more samples.
int32_t current_sample = time_ >> kPhaseBits;
if (current_sample >= (sample_buffer_.size() - samples_per_filter_wing_)) {
// NB: bit shifting back and forth means we're only modifying `time` by
// whole samples.
time_ -= current_sample << kPhaseBits;
time_ += samples_per_filter_wing_ << kPhaseBits;
int32_t new_current_sample = time_ >> kPhaseBits;
new_current_sample -= samples_per_filter_wing_;
current_sample -= samples_per_filter_wing_;
int32_t samples_to_move = latest_sample_ - current_sample;
if (samples_to_move > 0) {
auto samples = sample_buffer_.subspan(current_sample, samples_to_move);
std::copy_backward(samples.begin(), samples.end(),
sample_buffer_.first(new_current_sample).end());
latest_sample_ = new_current_sample + samples_to_move;
} else {
latest_sample_ = new_current_sample;
}
}
return samples_output;
}
static const size_t kChunkSizeSamples = 256;
Resampler::Resampler(uint32_t source_sample_rate,
uint32_t target_sample_rate,
uint8_t num_channels)
: source_sample_rate_(source_sample_rate),
target_sample_rate_(target_sample_rate),
factor_(((uint64_t)target_sample_rate << kPhaseBits) /
source_sample_rate),
num_channels_(num_channels),
channels_() {
for (int i = 0; i < num_channels; i++) {
channels_.emplace_back(source_sample_rate, target_sample_rate,
kChunkSizeSamples, num_channels);
}
}
Resampler::~Resampler() {}
auto Resampler::Process(cpp::span<const sample::Sample> input,
cpp::span<sample::Sample> output,
bool end_of_data) -> std::pair<size_t, size_t> {
size_t samples_used = 0;
std::vector<size_t> samples_produced = {num_channels_, 0};
size_t total_samples_produced = 0;
size_t slop = (factor_ >> kPhaseBits) + 1;
uint_fast8_t cur_channel = 0;
while (input.size() > samples_used &&
output.size() > total_samples_produced + slop) {
// Work out where the next set of samples should be placed.
size_t next_output_index =
(samples_produced[cur_channel] * num_channels_) + cur_channel;
// Generate the next samples
size_t new_samples = channels_[cur_channel].AddSample(
input[samples_used++], output.subspan(next_output_index));
samples_produced[cur_channel] += new_samples;
total_samples_produced += new_samples;
cur_channel = (cur_channel + 1) % num_channels_;
}
return {samples_used, total_samples_produced};
}
} // namespace audio
+69 -152
View File
@@ -13,6 +13,8 @@
#include "esp_log.h"
#include "freertos/portmacro.h"
#include "freertos/projdefs.h"
#include "resample.hpp"
#include "sample.hpp"
#include "samplerate.h"
#include "stream_info.hpp"
@@ -21,10 +23,7 @@
static constexpr char kTag[] = "mixer";
static constexpr std::size_t kSourceBufferLength = 2 * 1024;
static constexpr std::size_t kInputBufferLength = 2 * 1024;
static constexpr std::size_t kReformatBufferLength = 8 * 1024;
static constexpr std::size_t kResampleBufferLength = kReformatBufferLength;
static constexpr std::size_t kQuantisedBufferLength = 1 * 1024;
static constexpr std::size_t kSampleBufferLength = 4 * 1024;
namespace audio {
@@ -34,20 +33,8 @@ SinkMixer::SinkMixer(StreamBufferHandle_t dest)
resampler_(nullptr),
source_(xStreamBufferCreate(kSourceBufferLength, 1)),
sink_(dest) {
input_stream_.reset(new RawStream(kInputBufferLength));
floating_point_stream_.reset(new RawStream(kReformatBufferLength, MALLOC_CAP_SPIRAM));
resampled_stream_.reset(new RawStream(kResampleBufferLength, MALLOC_CAP_SPIRAM));
quantisation_buffer_ = {
reinterpret_cast<std::byte*>(heap_caps_malloc(
kQuantisedBufferLength, MALLOC_CAP_INTERNAL | MALLOC_CAP_8BIT)),
kQuantisedBufferLength};
quantisation_buffer_as_ints_ = {
reinterpret_cast<int*>(quantisation_buffer_.data()),
quantisation_buffer_.size_bytes() / 4};
quantisation_buffer_as_shorts_ = {
reinterpret_cast<short*>(quantisation_buffer_.data()),
quantisation_buffer_.size_bytes() / 2};
input_stream_.reset(new RawStream(kSampleBufferLength));
resampled_stream_.reset(new RawStream(kSampleBufferLength));
tasks::StartPersistent<tasks::Type::kMixer>([&]() { Main(); });
}
@@ -56,10 +43,6 @@ SinkMixer::~SinkMixer() {
vQueueDelete(commands_);
vSemaphoreDelete(is_idle_);
vStreamBufferDelete(source_);
heap_caps_free(quantisation_buffer_.data());
if (resampler_ != nullptr) {
src_delete(resampler_);
}
}
auto SinkMixer::MixAndSend(InputStream& input, const StreamInfo::Pcm& target)
@@ -109,10 +92,12 @@ auto SinkMixer::Main() -> void {
case Command::kSetSourceFormat:
ESP_LOGI(kTag, "setting source format");
input_receiver.prepare(args.format, {});
resampler_.reset();
break;
case Command::kSetTargetFormat:
ESP_LOGI(kTag, "setting target format");
target_format_ = args.format;
resampler_.reset();
break;
case Command::kReadBytes:
xSemaphoreTake(is_idle_, 0);
@@ -150,152 +135,84 @@ auto SinkMixer::HandleBytes() -> void {
return;
}
// Work out the resampling ratio using floating point arithmetic, since
// relying on the FPU for this will be much faster, and the difference in
// accuracy is unlikely to be noticeable.
float src_ratio = static_cast<float>(target_format_.sample_rate) /
static_cast<float>(pcm->sample_rate);
// Loop until we don't have any complete frames left in the input stream,
// where a 'frame' is one complete sample per channel.
while (!input_stream_->empty()) {
// The first step of both resampling and requantising is to convert the
// fixed point pcm input data into 32 bit floating point samples.
OutputStream floating_writer{floating_point_stream_.get()};
if (pcm->bits_per_sample == 16) {
ConvertFixedToFloating<short>(input, floating_writer);
RawStream* output_source;
if (pcm->sample_rate != target_format_.sample_rate) {
OutputStream resampled_writer{resampled_stream_.get()};
if (Resample(input, resampled_writer)) {
// Zero samples used or written. We need more input.
break;
}
output_source = resampled_stream_.get();
} else {
// FIXME: We should consider treating 24 bit and 32 bit samples
// differently.
ConvertFixedToFloating<int>(input, floating_writer);
output_source = input_stream_.get();
}
InputStream floating_reader{floating_point_stream_.get()};
if (target_format_.bits_per_sample == 16) {
// This is slightly scary; we're basically reaching into the internals of
// the stream buffer to do in-place conversion of samples. Saving an
// extra buffer + copy into that buffer is certainly worth it however.
cpp::span<sample::Sample> src =
output_source->data_as<sample::Sample>().first(
output_source->info().bytes_in_stream() / sizeof(sample::Sample));
cpp::span<int16_t> dest = output_source->data_as<int16_t>().first(
output_source->info().bytes_in_stream() / sizeof(int16_t));
while (!floating_point_stream_->empty()) {
RawStream* quantisation_source;
if (pcm->sample_rate != target_format_.sample_rate) {
// The input data needs to be resampled before being sent to the sink.
OutputStream resample_writer{resampled_stream_.get()};
Resample(src_ratio, pcm->channels, floating_reader, resample_writer);
quantisation_source = resampled_stream_.get();
} else {
// The input data already has an acceptable sample rate. All we need to
// do is quantise it.
quantisation_source = floating_point_stream_.get();
}
ApplyDither(src, 16);
Downscale(src, dest);
InputStream quantise_reader{quantisation_source};
while (!quantisation_source->empty()) {
std::size_t samples_available;
if (target_format_.bits_per_sample == 16) {
samples_available = Quantise<short>(quantise_reader);
} else {
samples_available = Quantise<int>(quantise_reader);
}
assert(samples_available * target_format_.real_bytes_per_sample() <=
quantisation_buffer_.size_bytes());
std::size_t bytes_sent = xStreamBufferSend(
sink_, quantisation_buffer_.data(),
samples_available * target_format_.real_bytes_per_sample(),
portMAX_DELAY);
assert(bytes_sent ==
samples_available * target_format_.real_bytes_per_sample());
}
output_source->info().bytes_in_stream() /= 2;
}
InputStream output{output_source};
cpp::span<const std::byte> buf = output.data();
size_t bytes_sent = 0;
while (bytes_sent < buf.size_bytes()) {
auto cropped = buf.subspan(bytes_sent);
bytes_sent += xStreamBufferSend(sink_, cropped.data(),
cropped.size_bytes(), portMAX_DELAY);
}
output.consume(bytes_sent);
}
}
template <>
auto SinkMixer::ConvertFixedToFloating<short>(InputStream& in_str,
OutputStream& out_str) -> void {
auto in = in_str.data_as<short>();
auto out = out_str.data_as<float>();
std::size_t samples_converted = std::min(in.size(), out.size());
src_short_to_float_array(in.data(), out.data(), samples_converted);
in_str.consume(samples_converted * sizeof(short));
out_str.add(samples_converted * sizeof(float));
}
template <>
auto SinkMixer::ConvertFixedToFloating<int>(InputStream& in_str,
OutputStream& out_str) -> void {
auto in = in_str.data_as<int>();
auto out = out_str.data_as<float>();
std::size_t samples_converted = std::min(in.size(), out.size());
src_int_to_float_array(in.data(), out.data(), samples_converted);
in_str.consume(samples_converted * sizeof(int));
out_str.add(samples_converted * sizeof(float));
}
auto SinkMixer::Resample(float src_ratio,
int channels,
InputStream& in,
OutputStream& out) -> void {
if (resampler_ == nullptr || src_get_channels(resampler_) != channels) {
if (resampler_ != nullptr) {
src_delete(resampler_);
}
ESP_LOGI(kTag, "creating new resampler with %u channels", channels);
int err = 0;
resampler_ = src_new(SRC_LINEAR, channels, &err);
assert(resampler_ != NULL);
assert(err == 0);
auto SinkMixer::Resample(InputStream& in, OutputStream& out) -> bool {
if (resampler_ == nullptr) {
ESP_LOGI(kTag, "creating new resampler");
auto format = in.info().format_as<StreamInfo::Pcm>();
resampler_.reset(new Resampler(
format->sample_rate, target_format_.sample_rate, format->channels));
}
auto in_buf = in.data_as<float>();
auto out_buf = out.data_as<float>();
auto res = resampler_->Process(in.data_as<sample::Sample>(),
out.data_as<sample::Sample>(), false);
src_set_ratio(resampler_, src_ratio);
SRC_DATA args{
.data_in = in_buf.data(),
.data_out = out_buf.data(),
.input_frames = static_cast<long>(in_buf.size()),
.output_frames = static_cast<long>(out_buf.size()),
.input_frames_used = 0,
.output_frames_gen = 0,
.end_of_input = 0,
.src_ratio = src_ratio,
};
int err = src_process(resampler_, &args);
if (err != 0) {
ESP_LOGE(kTag, "resampler error: %s", src_strerror(err));
ESP_LOGI(kTag, "resampler sent %u samples, consumed %u, produced %u",
in.data().size(), res.first, res.second);
in.consume(res.first * sizeof(sample::Sample));
out.add(res.first * sizeof(sample::Sample));
return res.first == 0 && res.second == 0;
}
auto SinkMixer::Downscale(cpp::span<sample::Sample> samples,
cpp::span<int16_t> output) -> void {
for (size_t i = 0; i < samples.size(); i++) {
output[i] = sample::ToSigned16Bit(samples[i]);
}
in.consume(args.input_frames_used * sizeof(float));
out.add(args.output_frames_gen * sizeof(float));
}
template <>
auto SinkMixer::Quantise<short>(InputStream& in) -> std::size_t {
auto src = in.data_as<float>();
cpp::span<short> dest = quantisation_buffer_as_shorts_;
dest = dest.first(std::min(src.size(), dest.size()));
src_float_to_short_array(src.data(), dest.data(), dest.size());
in.consume(dest.size() * sizeof(float));
return dest.size();
}
template <>
auto SinkMixer::Quantise<int>(InputStream& in) -> std::size_t {
auto src = in.data_as<float>();
cpp::span<int> dest = quantisation_buffer_as_ints_;
dest = dest.first(std::min<int>(src.size(), dest.size()));
src_float_to_int_array(src.data(), dest.data(), dest.size());
in.consume(dest.size() * sizeof(float));
return dest.size();
auto SinkMixer::ApplyDither(cpp::span<sample::Sample> samples,
uint_fast8_t bits) -> void {
static uint32_t prnd;
for (auto& s : samples) {
prnd = (prnd * 0x19660dL + 0x3c6ef35fL) & 0xffffffffL;
s = sample::Clip(
static_cast<int64_t>(s) +
(static_cast<int>(prnd) >> (sizeof(sample::Sample) - bits)));
}
}
} // namespace audio
+3 -3
View File
@@ -12,6 +12,7 @@
#include "esp_log.h"
#include "foxen/flac.h"
#include "sample.hpp"
namespace codecs {
@@ -47,7 +48,6 @@ auto FoxenFlacDecoder::BeginStream(const cpp::span<const std::byte> input)
OutputFormat format{
.num_channels = static_cast<uint8_t>(channels),
.bits_per_sample = 32, // libfoxenflac output is fixed-size.
.sample_rate_hz = static_cast<uint32_t>(fs),
.duration_seconds = {},
.bits_per_second = {},
@@ -62,7 +62,7 @@ auto FoxenFlacDecoder::BeginStream(const cpp::span<const std::byte> input)
}
auto FoxenFlacDecoder::ContinueStream(cpp::span<const std::byte> input,
cpp::span<std::byte> output)
cpp::span<sample::Sample> output)
-> Result<OutputInfo> {
cpp::span<int32_t> output_as_samples{
reinterpret_cast<int32_t*>(output.data()), output.size_bytes() / 4};
@@ -78,7 +78,7 @@ auto FoxenFlacDecoder::ContinueStream(cpp::span<const std::byte> input,
if (samples_written > 0) {
return {bytes_read,
OutputInfo{.bytes_written = samples_written * 4,
OutputInfo{.samples_written = samples_written,
.is_finished_writing = state == FLAC_END_OF_FRAME}};
}
+3 -3
View File
@@ -16,6 +16,7 @@
#include <string>
#include <utility>
#include "sample.hpp"
#include "result.hpp"
#include "span.hpp"
#include "types.hpp"
@@ -61,7 +62,6 @@ class ICodec {
struct OutputFormat {
uint8_t num_channels;
uint8_t bits_per_sample;
uint32_t sample_rate_hz;
std::optional<uint32_t> duration_seconds;
@@ -76,7 +76,7 @@ class ICodec {
-> Result<OutputFormat> = 0;
struct OutputInfo {
std::size_t bytes_written;
std::size_t samples_written;
bool is_finished_writing;
};
@@ -84,7 +84,7 @@ class ICodec {
* Writes PCM samples to the given output buffer.
*/
virtual auto ContinueStream(cpp::span<const std::byte> input,
cpp::span<std::byte> output)
cpp::span<sample::Sample> output)
-> Result<OutputInfo> = 0;
virtual auto SeekStream(cpp::span<const std::byte> input,
+2 -1
View File
@@ -14,6 +14,7 @@
#include <utility>
#include "foxen/flac.h"
#include "sample.hpp"
#include "span.hpp"
#include "codec.hpp"
@@ -26,7 +27,7 @@ class FoxenFlacDecoder : public ICodec {
~FoxenFlacDecoder();
auto BeginStream(cpp::span<const std::byte>) -> Result<OutputFormat> override;
auto ContinueStream(cpp::span<const std::byte>, cpp::span<std::byte>)
auto ContinueStream(cpp::span<const std::byte>, cpp::span<sample::Sample>)
-> Result<OutputInfo> override;
auto SeekStream(cpp::span<const std::byte> input, std::size_t target_sample)
-> Result<void> override;
+2 -1
View File
@@ -13,6 +13,7 @@
#include <utility>
#include "mad.h"
#include "sample.hpp"
#include "span.hpp"
#include "codec.hpp"
@@ -35,7 +36,7 @@ class MadMp3Decoder : public ICodec {
* Writes samples for the current frame.
*/
auto ContinueStream(cpp::span<const std::byte> input,
cpp::span<std::byte> output)
cpp::span<sample::Sample> output)
-> Result<OutputInfo> override;
auto SeekStream(cpp::span<const std::byte> input, std::size_t target_sample)
+59
View File
@@ -0,0 +1,59 @@
#pragma once
#include <stdint.h>
#include <algorithm>
#include <mad.h>
namespace sample {
// A signed, 32-bit PCM sample.
typedef int32_t Sample;
constexpr auto Clip(int64_t v) -> Sample {
if (v > INT32_MAX)
return INT32_MAX;
if (v < INT32_MIN)
return INT32_MIN;
return v;
}
constexpr auto FromSigned(int32_t src, uint_fast8_t bits) -> Sample {
// Left-align samples, effectively scaling them up to 32 bits.
return src << (sizeof(Sample) * 8 - bits);
}
constexpr auto FromUnsigned(uint32_t src, uint_fast8_t bits) -> Sample {
// Left-align, then substract the max value / 2 to make the sample centred
// around zero.
return (src << (sizeof(uint32_t) * 8 - bits)) - (~0UL >> 1);
}
constexpr auto FromFloat(float src) -> Sample {
return std::clamp<float>(src, -1.0f, 1.0f) * static_cast<float>(INT32_MAX);
}
constexpr auto FromDouble(double src) -> Sample {
return std::clamp<double>(src, -1.0, 1.0) * static_cast<double>(INT32_MAX);
}
constexpr auto FromMad(mad_fixed_t src) -> Sample {
// Round the bottom bits.
src += (1L << (MAD_F_FRACBITS - 24));
// Clip the leftover bits to within range.
if (src >= MAD_F_ONE)
src = MAD_F_ONE - 1;
else if (src < -MAD_F_ONE)
src = -MAD_F_ONE;
// Quantize.
return FromSigned(src >> (MAD_F_FRACBITS + 1 - 24), 24);
}
constexpr auto ToSigned16Bit(Sample src) -> uint16_t {
return src >> 16;
}
} // namespace sample
+9 -31
View File
@@ -17,24 +17,11 @@
#include "codec.hpp"
#include "esp_log.h"
#include "result.hpp"
#include "sample.hpp"
#include "types.hpp"
namespace codecs {
static uint32_t mad_fixed_to_pcm(mad_fixed_t sample, uint8_t bits) {
// Round the bottom bits.
sample += (1L << (MAD_F_FRACBITS - bits));
// Clip the leftover bits to within range.
if (sample >= MAD_F_ONE)
sample = MAD_F_ONE - 1;
else if (sample < -MAD_F_ONE)
sample = -MAD_F_ONE;
// Quantize.
return sample >> (MAD_F_FRACBITS + 1 - bits);
}
MadMp3Decoder::MadMp3Decoder() {
mad_stream_init(&stream_);
mad_frame_init(&frame_);
@@ -83,7 +70,6 @@ auto MadMp3Decoder::BeginStream(const cpp::span<const std::byte> input)
uint8_t channels = MAD_NCHANNELS(&header);
OutputFormat output{
.num_channels = channels,
.bits_per_sample = 24, // We always scale to 24 bits
.sample_rate_hz = header.samplerate,
.duration_seconds = {},
.bits_per_second = {},
@@ -100,7 +86,7 @@ auto MadMp3Decoder::BeginStream(const cpp::span<const std::byte> input)
}
auto MadMp3Decoder::ContinueStream(cpp::span<const std::byte> input,
cpp::span<std::byte> output)
cpp::span<sample::Sample> output)
-> Result<OutputInfo> {
std::size_t bytes_read = 0;
if (current_sample_ < 0) {
@@ -133,32 +119,24 @@ auto MadMp3Decoder::ContinueStream(cpp::span<const std::byte> input,
bytes_read = GetBytesUsed(input.size_bytes());
}
size_t output_byte = 0;
size_t output_sample = 0;
while (current_sample_ < synth_.pcm.length) {
if (output_byte + (4 * synth_.pcm.channels) >= output.size()) {
// We can't fit the next sample into the buffer. Stop now, and also avoid
// writing the sample for only half the channels.
return {bytes_read, OutputInfo{.bytes_written = output_byte,
if (output_sample + synth_.pcm.channels >= output.size()) {
// We can't fit the next full frame into the buffer.
return {bytes_read, OutputInfo{.samples_written = output_sample,
.is_finished_writing = false}};
}
for (int channel = 0; channel < synth_.pcm.channels; channel++) {
uint32_t sample_24 =
mad_fixed_to_pcm(synth_.pcm.samples[channel][current_sample_], 24);
// 24 bit samples must still be aligned to 32 bits. The LSB is ignored.
output[output_byte++] = static_cast<std::byte>(0);
output[output_byte++] = static_cast<std::byte>((sample_24)&0xFF);
output[output_byte++] = static_cast<std::byte>((sample_24 >> 8) & 0xFF);
output[output_byte++] = static_cast<std::byte>((sample_24 >> 16) & 0xFF);
output[output_sample++] =
sample::FromMad(synth_.pcm.samples[channel][current_sample_]);
}
current_sample_++;
}
// We wrote everything! Reset, ready for the next frame.
current_sample_ = -1;
return {bytes_read, OutputInfo{.bytes_written = output_byte,
return {bytes_read, OutputInfo{.samples_written = output_sample,
.is_finished_writing = true}};
}
+275
View File
@@ -0,0 +1,275 @@
#include "sample.hpp"
namespace audio {
namespace sample {
void siconv(int* dst, uint8_t* src, int bits, int skip, int count) {
int i, v, s, b;
b = (bits + 7) / 8;
s = sizeof(int) * 8 - bits;
while (count--) {
v = 0;
i = b;
switch (b) {
case 4:
v = src[--i];
case 3:
v = (v << 8) | src[--i];
case 2:
v = (v << 8) | src[--i];
case 1:
v = (v << 8) | src[--i];
}
*dst++ = v << s;
src += skip;
}
}
void Siconv(int* dst, uint8_t* src, int bits, int skip, int count) {
int i, v, s, b;
b = (bits + 7) / 8;
s = sizeof(int) * 8 - bits;
while (count--) {
v = 0;
i = 0;
switch (b) {
case 4:
v = src[i++];
case 3:
v = (v << 8) | src[i++];
case 2:
v = (v << 8) | src[i++];
case 1:
v = (v << 8) | src[i];
}
*dst++ = v << s;
src += skip;
}
}
void uiconv(int* dst, uint8_t* src, int bits, int skip, int count) {
int i, s, b;
uint32_t v;
b = (bits + 7) / 8;
s = sizeof(uint32_t) * 8 - bits;
while (count--) {
v = 0;
i = b;
switch (b) {
case 4:
v = src[--i];
case 3:
v = (v << 8) | src[--i];
case 2:
v = (v << 8) | src[--i];
case 1:
v = (v << 8) | src[--i];
}
*dst++ = (v << s) - (~0UL >> 1);
src += skip;
}
}
void Uiconv(int* dst, uint8_t* src, int bits, int skip, int count) {
int i, s, b;
uint32_t v;
b = (bits + 7) / 8;
s = sizeof(uint32_t) * 8 - bits;
while (count--) {
v = 0;
i = 0;
switch (b) {
case 4:
v = src[i++];
case 3:
v = (v << 8) | src[i++];
case 2:
v = (v << 8) | src[i++];
case 1:
v = (v << 8) | src[i];
}
*dst++ = (v << s) - (~0UL >> 1);
src += skip;
}
}
void ficonv(int* dst, uint8_t* src, int bits, int skip, int count) {
if (bits == 32) {
while (count--) {
float f;
f = *((float*)src), src += skip;
if (f > 1.0)
*dst++ = INT32_MAX;
else if (f < -1.0)
*dst++ = INT32_MIN;
else
*dst++ = f * ((float)INT32_MAX);
}
} else {
while (count--) {
double d;
d = *((double*)src), src += skip;
if (d > 1.0)
*dst++ = INT32_MAX;
else if (d < -1.0)
*dst++ = INT32_MIN;
else
*dst++ = d * ((double)INT32_MAX);
}
}
}
void aiconv(int* dst, uint8_t* src, int, int skip, int count) {
int t, seg;
uint8_t a;
while (count--) {
a = *src, src += skip;
a ^= 0x55;
t = (a & 0xf) << 4;
seg = (a & 0x70) >> 4;
switch (seg) {
case 0:
t += 8;
break;
case 1:
t += 0x108;
break;
default:
t += 0x108;
t <<= seg - 1;
}
t = (a & 0x80) ? t : -t;
*dst++ = t << (sizeof(int) * 8 - 16);
}
}
void µiconv(int* dst, uint8_t* src, int, int skip, int count) {
int t;
uint8_t u;
while (count--) {
u = *src, src += skip;
u = ~u;
t = ((u & 0xf) << 3) + 0x84;
t <<= (u & 0x70) >> 4;
t = u & 0x80 ? 0x84 - t : t - 0x84;
*dst++ = t << (sizeof(int) * 8 - 16);
}
}
void soconv(int* src, uint8_t* dst, int bits, int skip, int count) {
int i, v, s, b;
b = (bits + 7) / 8;
s = sizeof(int) * 8 - bits;
while (count--) {
v = *src++ >> s;
i = 0;
switch (b) {
case 4:
dst[i++] = v, v >>= 8;
case 3:
dst[i++] = v, v >>= 8;
case 2:
dst[i++] = v, v >>= 8;
case 1:
dst[i] = v;
}
dst += skip;
}
}
void Soconv(int* src, uint8_t* dst, int bits, int skip, int count) {
int i, v, s, b;
b = (bits + 7) / 8;
s = sizeof(int) * 8 - bits;
while (count--) {
v = *src++ >> s;
i = b;
switch (b) {
case 4:
dst[--i] = v, v >>= 8;
case 3:
dst[--i] = v, v >>= 8;
case 2:
dst[--i] = v, v >>= 8;
case 1:
dst[--i] = v;
}
dst += skip;
}
}
void uoconv(int* src, uint8_t* dst, int bits, int skip, int count) {
int i, s, b;
uint32_t v;
b = (bits + 7) / 8;
s = sizeof(uint32_t) * 8 - bits;
while (count--) {
v = ((~0UL >> 1) + *src++) >> s;
i = 0;
switch (b) {
case 4:
dst[i++] = v, v >>= 8;
case 3:
dst[i++] = v, v >>= 8;
case 2:
dst[i++] = v, v >>= 8;
case 1:
dst[i] = v;
}
dst += skip;
}
}
void Uoconv(int* src, uint8_t* dst, int bits, int skip, int count) {
int i, s, b;
uint32_t v;
b = (bits + 7) / 8;
s = sizeof(uint32_t) * 8 - bits;
while (count--) {
v = ((~0UL >> 1) + *src++) >> s;
i = b;
switch (b) {
case 4:
dst[--i] = v, v >>= 8;
case 3:
dst[--i] = v, v >>= 8;
case 2:
dst[--i] = v, v >>= 8;
case 1:
dst[--i] = v;
}
dst += skip;
}
}
void foconv(int* src, uint8_t* dst, int bits, int skip, int count) {
if (bits == 32) {
while (count--) {
*((float*)dst) = *src++ / ((float)INT32_MAX);
dst += skip;
}
} else {
while (count--) {
*((double*)dst) = *src++ / ((double)INT32_MAX);
dst += skip;
}
}
}
}
}