Reworked ADPCM decoder to allow better streaming
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@ -546,46 +546,101 @@ s32 CommandGenerator::DecodeAdpcm(ServerVoiceInfo& voice_info, VoiceState& dsp_s
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return 0;
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}
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const auto samples_remaining =
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(wave_buffer.end_sample_offset - wave_buffer.start_sample_offset) - dsp_state.offset;
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const auto samples_processed = std::min(sample_count, samples_remaining);
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const auto start_offset =
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((wave_buffer.start_sample_offset + dsp_state.offset) * in_params.channel_count);
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const auto end_offset = start_offset + samples_processed;
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constexpr std::array<int, 16> SIGNED_NIBBLES = {
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{0, 1, 2, 3, 4, 5, 6, 7, -8, -7, -6, -5, -4, -3, -2, -1}};
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constexpr std::size_t FRAME_LEN = 8;
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constexpr std::size_t NIBBLES_PER_SAMPLE = 16;
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constexpr std::size_t SAMPLES_PER_FRAME = 14;
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// Base buffer position
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const auto start_frame_index = start_offset / SAMPLES_PER_FRAME;
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const auto start_frame_buffer = start_frame_index * FRAME_LEN;
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const auto end_frame_index = end_offset / SAMPLES_PER_FRAME;
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const auto end_frame_buffer = end_frame_index * FRAME_LEN;
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const auto position_in_frame = start_offset % SAMPLES_PER_FRAME;
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const auto buffer_size = (1 + (end_frame_index - start_frame_index)) * FRAME_LEN;
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auto frame_header = dsp_state.context.header;
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s32 idx = (frame_header >> 4) & 0xf;
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s32 scale = frame_header & 0xf;
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s16 yn1 = dsp_state.context.yn1;
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s16 yn2 = dsp_state.context.yn2;
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Codec::ADPCM_Coeff coeffs;
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memory.ReadBlock(in_params.additional_params_address, coeffs.data(),
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sizeof(Codec::ADPCM_Coeff));
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std::vector<u8> buffer(buffer_size);
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memory.ReadBlock(wave_buffer.buffer_address + start_frame_buffer, buffer.data(), buffer.size());
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const auto adpcm_samples =
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std::move(Codec::DecodeADPCM(buffer.data(), buffer.size(), coeffs, dsp_state.context));
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for (std::size_t i = 0; i < samples_processed; i++) {
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const auto sample_offset = position_in_frame + i * in_params.channel_count + channel;
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const auto sample = adpcm_samples[sample_offset];
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sample_buffer[mix_offset + i] = sample;
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s32 coef1 = coeffs[idx * 2];
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s32 coef2 = coeffs[idx * 2 + 1];
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const auto samples_remaining =
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(wave_buffer.end_sample_offset - wave_buffer.start_sample_offset) - dsp_state.offset;
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const auto samples_processed = std::min(sample_count, samples_remaining);
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const auto sample_pos = wave_buffer.start_sample_offset + dsp_state.offset;
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const auto samples_remaining_in_frame = sample_pos % SAMPLES_PER_FRAME;
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auto position_in_frame = ((sample_pos / SAMPLES_PER_FRAME) * NIBBLES_PER_SAMPLE) +
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samples_remaining_in_frame + (samples_remaining_in_frame != 0 ? 2 : 0);
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const auto decode_sample = [&](const int nibble) -> s16 {
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const int xn = nibble * (1 << scale);
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// We first transform everything into 11 bit fixed point, perform the second order
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// digital filter, then transform back.
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// 0x400 == 0.5 in 11 bit fixed point.
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// Filter: y[n] = x[n] + 0.5 + c1 * y[n-1] + c2 * y[n-2]
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int val = ((xn << 11) + 0x400 + coef1 * yn1 + coef2 * yn2) >> 11;
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// Clamp to output range.
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val = std::clamp<s32>(val, -32768, 32767);
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// Advance output feedback.
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yn2 = yn1;
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yn1 = val;
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return static_cast<s16>(val);
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};
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std::size_t buffer_offset{};
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std::vector<u8> buffer(
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std::max((samples_processed / FRAME_LEN) * SAMPLES_PER_FRAME, FRAME_LEN));
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memory.ReadBlock(wave_buffer.buffer_address + (position_in_frame / 2), buffer.data(),
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buffer.size());
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std::size_t cur_mix_offset = mix_offset;
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auto remaining_samples = samples_processed;
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while (remaining_samples > 0) {
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if (position_in_frame % NIBBLES_PER_SAMPLE == 0) {
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// Read header
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frame_header = buffer[buffer_offset++];
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idx = (frame_header >> 4) & 0xf;
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scale = frame_header & 0xf;
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coef1 = coeffs[idx * 2];
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coef2 = coeffs[idx * 2 + 1];
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position_in_frame += 2;
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// Decode entire frame
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if (remaining_samples >= SAMPLES_PER_FRAME) {
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for (std::size_t i = 0; i < SAMPLES_PER_FRAME / 2; i++) {
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// Sample 1
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const s32 s0 = SIGNED_NIBBLES[buffer[buffer_offset] >> 4];
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const s32 s1 = SIGNED_NIBBLES[buffer[buffer_offset++] & 0xf];
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const s16 sample_1 = decode_sample(s0);
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const s16 sample_2 = decode_sample(s1);
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sample_buffer[cur_mix_offset++] = sample_1;
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sample_buffer[cur_mix_offset++] = sample_2;
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}
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remaining_samples -= SAMPLES_PER_FRAME;
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position_in_frame += SAMPLES_PER_FRAME;
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continue;
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}
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}
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// Decode mid frame
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s32 current_nibble = buffer[buffer_offset];
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if (position_in_frame++ & 0x1) {
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current_nibble &= 0xf;
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buffer_offset++;
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} else {
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current_nibble >>= 4;
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}
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const s16 sample = decode_sample(SIGNED_NIBBLES[current_nibble]);
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sample_buffer[cur_mix_offset++] = sample;
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remaining_samples--;
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}
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// Manually set our context
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const auto frame_before_final = (end_frame_index - start_frame_index) - 1;
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const auto frame_before_final_off = frame_before_final * SAMPLES_PER_FRAME;
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dsp_state.context.yn2 = adpcm_samples[frame_before_final_off + 12];
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dsp_state.context.yn1 = adpcm_samples[frame_before_final_off + 13];
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dsp_state.context.header = frame_header;
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dsp_state.context.yn1 = yn1;
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dsp_state.context.yn2 = yn2;
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return samples_processed;
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}
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@ -85,6 +85,13 @@ struct BehaviorFlags {
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};
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static_assert(sizeof(BehaviorFlags) == 0x4, "BehaviorFlags is an invalid size");
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struct ADPCMContext {
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u16 header{};
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s16 yn1{};
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s16 yn2{};
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};
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static_assert(sizeof(ADPCMContext) == 0x6, "ADPCMContext is an invalid size");
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struct VoiceState {
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s64 played_sample_count{};
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s32 offset{};
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@ -95,7 +102,7 @@ struct VoiceState {
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s32 fraction{};
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VAddr context_address{};
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Codec::ADPCM_Coeff coeff{};
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Codec::ADPCMState context{};
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ADPCMContext context{};
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std::array<s64, 2> biquad_filter_state{};
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std::array<s32, AudioCommon::MAX_MIX_BUFFERS> previous_samples{};
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u32 external_context_size{};
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