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Reworked ADPCM decoder to allow better streaming

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This commit is contained in:
David Marcec 2020-08-14 21:04:28 +10:00
parent 0947f613b1
commit 1b3d86c02f
2 changed files with 92 additions and 30 deletions
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113
src/audio_core/command_generator.cpp
View file

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

9
src/audio_core/voice_context.h
View file

@ -85,6 +85,13 @@ struct BehaviorFlags {
};
static_assert(sizeof(BehaviorFlags) == 0x4, "BehaviorFlags is an invalid size");
struct ADPCMContext {
u16 header{};
s16 yn1{};
s16 yn2{};
};
static_assert(sizeof(ADPCMContext) == 0x6, "ADPCMContext is an invalid size");
struct VoiceState {
s64 played_sample_count{};
s32 offset{};
@ -95,7 +102,7 @@ struct VoiceState {
s32 fraction{};
VAddr context_address{};
Codec::ADPCM_Coeff coeff{};
Codec::ADPCMState context{};
ADPCMContext context{};
std::array<s64, 2> biquad_filter_state{};
std::array<s32, AudioCommon::MAX_MIX_BUFFERS> previous_samples{};
u32 external_context_size{};