Implement using libmad to decode
This commit is contained in:
@@ -1,4 +1,6 @@
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idf_component_register(INCLUDE_DIRS "include")
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idf_component_register(
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SRCS "mad.cpp"
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INCLUDE_DIRS "include")
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#target_compile_options(${COMPONENT_LIB} PRIVATE ${EXTRA_WARNINGS})
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target_compile_options(${COMPONENT_LIB} PRIVATE ${EXTRA_WARNINGS})
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add_dependencies("${COMPONENT_LIB}" libmad)
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@@ -0,0 +1,9 @@
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#include "codec.hpp"
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namespace codecs {
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auto CreateCodecForExtension(std::string extension) -> cpp::result<std::unique_ptr<ICodec>, CreateCodecError> {
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return cpp::fail(UNKNOWN_EXTENSION);
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}
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} // namespace codecs
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@@ -1,5 +1,6 @@
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#pragma once
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#include <stdint.h>
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#include <cstddef>
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#include <cstdint>
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@@ -7,7 +8,9 @@
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namespace codecs {
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enum CreateCodecError {};
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enum CreateCodecError {
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UNKNOWN_EXTENSION
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};
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auto CreateCodecForExtension(std::string extension) -> cpp::result<std::unique_ptr<ICodec>, CreateCodecError>;
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@@ -17,13 +20,39 @@ namespace codecs {
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virtual auto CanHandleExtension(std::string extension) -> bool = 0;
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struct OutputFormat {
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uint8_t num_channels;
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uint8_t bits_per_sample;
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int sample_rate_hz;
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};
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virtual auto GetOutputFormat() -> OutputFormat = 0;
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enum Error {};
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struct Result {
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bool need_more_input;
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/*
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* For need_more_input, this is how far we got in the input buffer
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* before we were unable to process more data. Any remaining data in the
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* buffer should be moved to the start before the next call.
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*/
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std::size_t input_processed;
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bool flush_output;
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/*
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* For flush_output, this is how far we got in the output buffer before
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* we ran out of space for samples. The caller should flush this many
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* bytes downstream.
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*/
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std::size_t output_written;
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};
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virtual auto Process(
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uint8_t *input,
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std::size_t input_len,
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uint8_t *output,
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std::size_t output_length) -> cpp::result<size_t, Error> = 0;
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std::size_t output_length) -> cpp::result<Result, Error> = 0;
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};
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} // namespace codecs
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@@ -0,0 +1,26 @@
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#pragma once
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#include "codec.hpp"
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namespace codecs {
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class MadMp3Decoder : public ICodec {
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public:
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MadMp3Decoder();
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~MadMp3Decoder();
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auto ProcessInput(Result *res, uint8_t *input, std::size_t input_len) -> void;
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auto WriteOutputSamples(Result *res, uint8_t *output, std::size_t output_length) -> void;
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private:
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mad_stream stream_;
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mad_frame frame_;
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mad_synth synth_;
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mad_header header_;
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bool has_decoded_header_;
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int current_sample_ = -1;
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};
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} // namespace codecs
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@@ -0,0 +1,154 @@
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#include "mad.hpp"
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#include <cstdint>
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#include "mad.h"
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namespace codecs {
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static int32_t scaleTo24Bits(mad_fixed_t sample) {
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// Round the bottom bits.
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sample += (1L << (MAD_F_FRACBITS - 24));
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// Clip the leftover bits to within range.
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if (sample >= MAD_F_ONE)
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sample = MAD_F_ONE - 1;
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else if (sample < -MAD_F_ONE)
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sample = -MAD_F_ONE;
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/* quantize */
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return sample >> (MAD_F_FRACBITS + 1 - 24);
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}
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MadMp3Decoder::MadMp3Decoder() {
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mad_stream_init(&stream_);
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mad_frame_init(&frame_);
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mad_synth_init(&synth_);
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mad_header_init(&header_);
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}
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MadMp3Decoder::~MadMp3Decoder() {
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mad_stream_finish(&stream_);
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mad_frame_finish(&frame_);
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mad_synth_finish(&synth_);
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mad_header_finish(&header_);
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}
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auto MadMp3Decoder::CanHandleExtension(std::string extension) -> bool {
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return extension == "mp3";
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}
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auto GetOutputFormat() -> OutputFormat {
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return OutputFormat {
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.num_channels = synth_.pcm.channels,
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.bits_per_sample = 24,
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.sample_rate_hz = synth_.pcm.samplerate,
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};
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}
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auto MadMp3Decoder::Process(
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uint8_t *input,
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std::size_t input_len,
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uint8_t *output,
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std::size_t output_length) -> cpp::result<Result, Error> {
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Result res {
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.need_more_input = false,
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.input_processed = 0,
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.flush_output = false,
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.output_written = 0,
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}
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while (true) {
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// Only process more of the input if we're done sending off the
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// samples for the previous frame.
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if (current_sample_ == -1) {
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ProcessInput(&res, input, input_len);
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}
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// Write PCM samples to the output buffer. This always needs to be
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// done, even if we ran out of input, so that we don't keep the last
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// few samples buffered if the input stream has actually finished.
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WriteOutputSamples(&res, output, output_length);
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if (res.need_more_input || res.flush_output) {
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return res;
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}
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}
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auto MadMp3Decoder::ProcessInput(
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Result *res, uint8_t *input, std::size_t input_len) -> void {
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if (input != stream_.buffer) {
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mad_stream_buffer(&stream_, input, input_len);
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}
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if (!has_decoded_header_) {
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// The header of any given frame should be representative of the
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// entire stream, so only need to read it once.
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mad_header_decode(&header_, &stream);
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has_decoded_header_ = true;
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// TODO: Use the info in the header for something. I think the
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// duration will help with seeking?
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}
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// Decode the next frame. To signal errors, this returns -1 and
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// stashes an error code in the stream structure.
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if (mad_frame_decode(&frame_, &stream_) < 0) {
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if (MAD_RECOVERABLE(stream_.error)) {
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// Recoverable errors are usually malformed parts of the stream.
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// We can recover from them by just retrying the decode.
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continue;
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}
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if (stream_.error = MAD_ERROR_BUFLEN) {
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// The decoder ran out of bytes before it completed a frame. We
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// need to return back to the caller to give us more data. Note
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// that there might still be some unused data in the input, so we
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// should calculate that amount and return it.
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size_t remaining_bytes = stream.bufend - stream_.next_frame;
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return remaining_bytes;
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}
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// The error is unrecoverable. Give up.
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return cpp::fail(MALFORMED_DATA);
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}
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// We've successfully decoded a frame!
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// Now we need to synthesize PCM samples based on the frame, and send
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// them downstream.
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mad_synth_frame(&synth_, &frame_);
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up_to_sample = 0;
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}
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auto MadMp3Decoder::WriteOutputSamples(
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Result *res,
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uint8_t *output,
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std::size_t output_length) -> void {
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size_t output_byte = 0;
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// First ensure that we actually have some samples to send off.
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if (current_sample_ < 0) {
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return;
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}
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res->flush_output = true;
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while (current_sample_ < synth_.pcm.length) {
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if (output_byte + (3 * synth_.pcm.channels) >= output_length) {
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res->output_written = output_byte;
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return;
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}
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for (int channel = 0; channel < synth_.pcm.channels; channel++) {
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uint32_t sample_24 = scaleTo24Bits(synth_.pcm.samples[channel][sample]);
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output[output_byte++] = (sample_24 >> 0) & 0xff;
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output[output_byte++] = (sample_24 >> 8) & 0xff;
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output[output_byte++] = (sample_24 >> 16) & 0xff;
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}
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current_sample_++;
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}
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// We wrote everything! Reset, ready for the next frame.
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current_sample_ = -1;
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res->output_written = output_byte;
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}
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} // namespace codecs
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