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/*
* Copyright 2024 jacqueline <me@jacqueline.id.au>
*
* SPDX-License-Identifier: GPL-3.0-only
*/
#include "tts/player.hpp"
#include <mutex>
#include "audio/audio_events.hpp"
#include "audio/processor.hpp"
#include "audio/resample.hpp"
#include "codec.hpp"
#include "esp_log.h"
#include "events/event_queue.hpp"
#include "freertos/projdefs.h"
#include "portmacro.h"
#include "sample.hpp"
#include "types.hpp"
namespace tts {
[[maybe_unused]] static constexpr char kTag[] = "ttsplay";
Player::Player(tasks::WorkerPool& worker,
drivers::PcmBuffer& output,
audio::FatfsStreamFactory& factory)
: bg_(worker),
stream_factory_(factory),
output_(output),
stream_playing_(false),
stream_cancelled_(false) {}
auto Player::playFile(const std::string& text, const std::string& file)
-> void {
bg_.Dispatch<void>([=, this]() {
{
std::scoped_lock<std::mutex> lock{new_stream_mutex_};
if (stream_playing_) {
stream_cancelled_ = true;
stream_playing_.wait(true);
}
stream_cancelled_ = false;
stream_playing_ = true;
}
openAndDecode(text, file);
if (!stream_cancelled_) {
events::Audio().Dispatch(audio::TtsPlaybackChanged{.is_playing = false});
}
stream_playing_ = false;
stream_playing_.notify_all();
});
}
auto Player::openAndDecode(const std::string& text, const std::string& path)
-> void {
auto stream = stream_factory_.create(path);
if (!stream) {
ESP_LOGW(kTag, "missing '%s' for '%s'", path.c_str(), text.c_str());
return;
}
// FIXME: Rather than hardcoding WAV support only, we should work out a
// proper subset of 'low memory' decoders that can all be used for TTS
// playback.
if (stream->type() != codecs::StreamType::kWav) {
ESP_LOGE(kTag, "'%s' has unsupported encoding", path.c_str());
return;
}
auto decoder = codecs::CreateCodecForType(stream->type());
if (!decoder) {
ESP_LOGE(kTag, "creating decoder failed");
return;
}
std::unique_ptr<codecs::ICodec> codec{*decoder};
auto open_res = codec->OpenStream(stream, 0);
if (open_res.has_error()) {
ESP_LOGE(kTag, "opening stream failed");
return;
}
decodeToSink(*open_res, std::move(codec));
}
auto Player::decodeToSink(const codecs::ICodec::OutputFormat& format,
std::unique_ptr<codecs::ICodec> codec) -> void {
// Set up buffers to hold samples between the intermediary parts of
// processing. We can just use the stack for these, since this method is
// called only from background workers, which have enormous stacks.
sample::Sample decode_storage[4096];
audio::Buffer decode_buf(decode_storage);
sample::Sample resample_storage[4096];
audio::Buffer resample_buf(resample_storage);
sample::Sample stereo_storage[4096];
audio::Buffer stereo_buf(stereo_storage);
// Work out what processing the codec's output needs.
std::unique_ptr<audio::Resampler> resampler;
if (format.sample_rate_hz != 48000) {
resampler = std::make_unique<audio::Resampler>(format.sample_rate_hz, 48000,
format.num_channels);
}
bool double_samples = format.num_channels == 1;
// Start our playback (wait for previous to end?)
events::Audio().Dispatch(audio::TtsPlaybackChanged{.is_playing = true});
// FIXME: This decode-and-process loop is substantially the same as the audio
// processor's filter loop. Ideally we should refactor both of these loops to
// reuse code, however I'm holding off on doing this until we've implemented
// more advanced audio processing features in the audio processor (EQ, tempo
// shifting, etc.) as it's not clear to me yet how much the two codepaths will
// be diverging later anyway.
while ((codec || !decode_buf.isEmpty() || !resample_buf.isEmpty() ||
!stereo_buf.isEmpty()) &&
!stream_cancelled_) {
if (codec) {
auto decode_res = codec->DecodeTo(decode_buf.writeAcquire());
if (decode_res.has_error()) {
ESP_LOGE(kTag, "decoding error");
break;
}
decode_buf.writeCommit(decode_res->samples_written);
if (decode_res->is_stream_finished) {
codec.reset();
}
}
if (!decode_buf.isEmpty()) {
auto resample_input = decode_buf.readAcquire();
auto resample_output = resample_buf.writeAcquire();
size_t read, wrote;
if (resampler) {
std::tie(read, wrote) =
resampler->Process(resample_input, resample_output, false);
} else {
read = wrote = std::min(resample_input.size(), resample_output.size());
std::copy_n(resample_input.begin(), read, resample_output.begin());
}
decode_buf.readCommit(read);
resample_buf.writeCommit(wrote);
}
if (!resample_buf.isEmpty()) {
auto channels_input = resample_buf.readAcquire();
auto channels_output = stereo_buf.writeAcquire();
size_t read, wrote;
if (double_samples) {
wrote = channels_output.size();
read = wrote / 2;
if (read > channels_input.size()) {
read = channels_input.size();
wrote = read * 2;
}
for (size_t i = 0; i < read; i++) {
channels_output[i * 2] = channels_input[i];
channels_output[(i * 2) + 1] = channels_input[i];
}
} else {
read = wrote = std::min(channels_input.size(), channels_output.size());
std::copy_n(channels_input.begin(), read, channels_output.begin());
}
resample_buf.readCommit(read);
stereo_buf.writeCommit(wrote);
}
// The mixin PcmBuffer should almost always be draining, so we can force
// samples into it more aggressively than with the main music PcmBuffer.
while (!stereo_buf.isEmpty() && !stream_cancelled_) {
size_t sent = output_.send(stereo_buf.readAcquire());
stereo_buf.readCommit(sent);
}
}
while (!output_.isEmpty()) {
if (stream_cancelled_) {
output_.clear();
} else {
vTaskDelay(pdMS_TO_TICKS(100));
}
}
}
} // namespace tts
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