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| author | jacqueline <me@jacqueline.id.au> | 2024-05-07 14:19:19 +1000 |
|---|---|---|
| committer | jacqueline <me@jacqueline.id.au> | 2024-05-07 14:19:19 +1000 |
| commit | b242ba998699208c87dc066158964de0866b61e2 (patch) | |
| tree | b0598526a28d6224e2290c35de5923d4234b89eb /src/tangara/audio/audio_decoder.cpp | |
| parent | 344a46d0664eb75d232eacea91a4957a25e071f6 (diff) | |
| download | tangara-fw-b242ba998699208c87dc066158964de0866b61e2.tar.gz | |
Improve decoder's interface to accept streams
Diffstat (limited to 'src/tangara/audio/audio_decoder.cpp')
| -rw-r--r-- | src/tangara/audio/audio_decoder.cpp | 151 |
1 files changed, 101 insertions, 50 deletions
diff --git a/src/tangara/audio/audio_decoder.cpp b/src/tangara/audio/audio_decoder.cpp index ae54a11c..0e38bca8 100644 --- a/src/tangara/audio/audio_decoder.cpp +++ b/src/tangara/audio/audio_decoder.cpp @@ -6,7 +6,7 @@ #include "audio/audio_decoder.hpp" -#include <algorithm> +#include <cassert> #include <cmath> #include <cstddef> #include <cstdint> @@ -23,14 +23,12 @@ #include "freertos/portmacro.h" #include "freertos/projdefs.h" #include "freertos/queue.h" -#include "freertos/ringbuf.h" -#include "audio/audio_converter.hpp" #include "audio/audio_events.hpp" #include "audio/audio_fsm.hpp" #include "audio/audio_sink.hpp" #include "audio/audio_source.hpp" -#include "audio/fatfs_audio_input.hpp" +#include "audio/processor.hpp" #include "codec.hpp" #include "database/track.hpp" #include "drivers/i2s_dac.hpp" @@ -42,21 +40,33 @@ namespace audio { -[[maybe_unused]] static const char* kTag = "audio_dec"; +static const char* kTag = "decoder"; +/* + * The size of the buffer used for holding decoded samples. This buffer is + * allocated in internal memory for greater speed, so be careful when + * increasing its size. + */ static constexpr std::size_t kCodecBufferLength = drivers::kI2SBufferLengthFrames * sizeof(sample::Sample); -auto Decoder::Start(std::shared_ptr<IAudioSource> source, - std::shared_ptr<SampleConverter> sink) -> Decoder* { - Decoder* task = new Decoder(source, sink); +auto Decoder::Start(std::shared_ptr<SampleProcessor> sink) -> Decoder* { + Decoder* task = new Decoder(sink); tasks::StartPersistent<tasks::Type::kAudioDecoder>([=]() { task->Main(); }); return task; } -Decoder::Decoder(std::shared_ptr<IAudioSource> source, - std::shared_ptr<SampleConverter> mixer) - : source_(source), converter_(mixer), codec_(), current_format_() { +auto Decoder::open(std::shared_ptr<TaggedStream> stream) -> void { + NextStream* next = new NextStream(); + next->stream = stream; + // The decoder services its queue very quickly, so blocking on this write + // should be fine. If we discover contention here, then adding more space for + // items to next_stream_ should be fine too. + xQueueSend(next_stream_, &next, portMAX_DELAY); +} + +Decoder::Decoder(std::shared_ptr<SampleProcessor> processor) + : processor_(processor), next_stream_(xQueueCreate(1, sizeof(void*))) { ESP_LOGI(kTag, "allocating codec buffer, %u KiB", kCodecBufferLength / 1024); codec_buffer_ = { reinterpret_cast<sample::Sample*>(heap_caps_calloc( @@ -64,81 +74,122 @@ Decoder::Decoder(std::shared_ptr<IAudioSource> source, kCodecBufferLength}; } +/* + * Main decoding loop. Handles watching for new streams, or continuing to nudge + * along the current stream if we have one. + */ void Decoder::Main() { for (;;) { - if (source_->HasNewStream() || !stream_) { - std::shared_ptr<TaggedStream> new_stream = source_->NextStream(); - if (new_stream && BeginDecoding(new_stream)) { - stream_ = new_stream; - } else { + // Check whether there's a new stream to begin. If we're idle, then we + // simply park and wait forever for a stream to arrive. + TickType_t wait_time = stream_ ? 0 : portMAX_DELAY; + NextStream* next; + if (xQueueReceive(next_stream_, &next, wait_time)) { + // Copy the data out of the queue, then clean up the item. + std::shared_ptr<TaggedStream> new_stream = next->stream; + delete next; + + // If we were already decoding, then make sure we finish up the current + // file gracefully. + if (stream_) { + finishDecode(); + } + + // Ensure there's actually stream data; we might have been given nullptr + // as a signal to stop. + if (!new_stream) { continue; } + + // Start decoding the new stream. + prepareDecode(new_stream); } - if (ContinueDecoding()) { - stream_.reset(); + if (!continueDecode()) { + finishDecode(); } } } -auto Decoder::BeginDecoding(std::shared_ptr<TaggedStream> stream) -> bool { - // Ensure any previous codec is freed before creating a new one. - codec_.reset(); +auto Decoder::prepareDecode(std::shared_ptr<TaggedStream> stream) -> void { + auto stub_track = std::make_shared<TrackInfo>(TrackInfo{ + .tags = stream->tags(), + .uri = stream->Filepath(), + .duration = {}, + .start_offset = {}, + .bitrate_kbps = {}, + .encoding = stream->type(), + .format = {}, + }); + codec_.reset(codecs::CreateCodecForType(stream->type()).value_or(nullptr)); if (!codec_) { ESP_LOGE(kTag, "no codec found for stream"); - return false; + events::Audio().Dispatch( + internal::DecodingFailedToStart{.track = stub_track}); + return; } auto open_res = codec_->OpenStream(stream, stream->Offset()); if (open_res.has_error()) { ESP_LOGE(kTag, "codec failed to start: %s", codecs::ICodec::ErrorString(open_res.error()).c_str()); - return false; - } - stream->SetPreambleFinished(); - current_sink_format_ = IAudioOutput::Format{ - .sample_rate = open_res->sample_rate_hz, - .num_channels = open_res->num_channels, - .bits_per_sample = 16, - }; - - std::optional<uint32_t> duration; - if (open_res->total_samples) { - duration = open_res->total_samples.value() / open_res->num_channels / - open_res->sample_rate_hz; + events::Audio().Dispatch( + internal::DecodingFailedToStart{.track = stub_track}); + return; } - converter_->beginStream(std::make_shared<TrackInfo>(TrackInfo{ + // Decoding started okay! Fill out the rest of the track info for this + // stream. + stream_ = stream; + track_ = std::make_shared<TrackInfo>(TrackInfo{ .tags = stream->tags(), .uri = stream->Filepath(), - .duration = duration, + .duration = {}, .start_offset = stream->Offset(), - .bitrate_kbps = open_res->sample_rate_hz, + .bitrate_kbps = {}, .encoding = stream->type(), - .format = *current_sink_format_, - })); + .format = + { + .sample_rate = open_res->sample_rate_hz, + .num_channels = open_res->num_channels, + .bits_per_sample = 16, + }, + }); + + if (open_res->total_samples) { + track_->duration = open_res->total_samples.value() / + open_res->num_channels / open_res->sample_rate_hz; + } - return true; + events::Audio().Dispatch(internal::DecodingStarted{.track = track_}); + processor_->beginStream(track_); } -auto Decoder::ContinueDecoding() -> bool { +auto Decoder::continueDecode() -> bool { auto res = codec_->DecodeTo(codec_buffer_); if (res.has_error()) { - converter_->endStream(); - return true; + return false; } if (res->samples_written > 0) { - converter_->continueStream(codec_buffer_.first(res->samples_written)); + processor_->continueStream(codec_buffer_.first(res->samples_written)); } - if (res->is_stream_finished) { - converter_->endStream(); - codec_.reset(); - } + return !res->is_stream_finished; +} - return res->is_stream_finished; +auto Decoder::finishDecode() -> void { + assert(track_); + + // Tell everyone we're finished. + events::Audio().Dispatch(internal::DecodingFinished{.track = track_}); + processor_->endStream(); + + // Clean up after ourselves. + stream_.reset(); + codec_.reset(); + track_.reset(); } } // namespace audio |
