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adpcm streaming

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This commit is contained in:
David Marcec 2020-07-30 18:16:57 +10:00
parent 3dcbba38bf
commit 1b8fe7073b
4 changed files with 32 additions and 27 deletions
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45
src/audio_core/command_generator.cpp
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@ -470,6 +470,7 @@ s32 CommandGenerator::DecodePcm16(ServerVoiceInfo& voice_info, VoiceState& dsp_s
return samples_processed;
}
s32 CommandGenerator::DecodeAdpcm(ServerVoiceInfo& voice_info, VoiceState& dsp_state,
s32 sample_count, s32 channel, std::size_t mix_offset) {
auto& in_params = voice_info.GetInParams();
@ -486,33 +487,45 @@ s32 CommandGenerator::DecodeAdpcm(ServerVoiceInfo& voice_info, VoiceState& dsp_s
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 buffer_pos = wave_buffer.buffer_address + start_offset;
const auto end_offset = start_offset + samples_processed;
const auto samples_processed = std::min(sample_count, samples_remaining);
constexpr std::size_t FRAME_LEN = 8;
constexpr std::size_t SAMPLES_PER_FRAME = 14;
if (start_offset > dsp_state.adpcm_samples.size()) {
dsp_state.adpcm_samples.clear();
}
// 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;
// TODO(ogniK): Proper ADPCM streaming
if (dsp_state.adpcm_samples.empty()) {
Codec::ADPCM_Coeff coeffs;
memory.ReadBlock(in_params.additional_params_address, coeffs.data(),
sizeof(Codec::ADPCM_Coeff));
std::vector<u8> buffer(wave_buffer.buffer_size);
memory.ReadBlock(wave_buffer.buffer_address, buffer.data(), buffer.size());
dsp_state.adpcm_samples =
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 = i + start_offset;
sample_buffer[mix_offset + i] =
dsp_state.adpcm_samples[sample_offset * in_params.channel_count + channel];
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;
}
// 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];
return samples_processed;
}
@ -628,10 +641,6 @@ void CommandGenerator::DecodeFromWaveBuffers(ServerVoiceInfo& voice_info, s32* o
dsp_state.played_sample_count = 0;
}
} else {
if (in_params.sample_format == SampleFormat::Adpcm) {
// TODO(ogniK): Remove this when ADPCM streaming implemented
dsp_state.adpcm_samples.clear();
}
// Update our wave buffer states
dsp_state.is_wave_buffer_valid[dsp_state.wave_buffer_index] = false;

1
src/audio_core/command_generator.h
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@ -71,7 +71,6 @@ private:
s32 channel, std::size_t mix_offset);
void DecodeFromWaveBuffers(ServerVoiceInfo& voice_info, s32* output, VoiceState& dsp_state,
s32 channel, s32 target_sample_rate, s32 sample_count, s32 node_id);
void Resample(s32* output, s32* input, s32 pitch, s32& fraction, s32 sample_count);
AudioCommon::AudioRendererParameter& worker_params;
VoiceContext& voice_context;

3
src/audio_core/voice_context.cpp
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@ -520,8 +520,7 @@ void VoiceContext::SortInfo() {
}
void VoiceContext::UpdateStateByDspShared() {
std::memcpy(voice_states.data(), dsp_voice_states.data(),
sizeof(VoiceState) * dsp_voice_states.size());
voice_states = dsp_voice_states;
}
} // namespace AudioCore

2
src/audio_core/voice_context.h
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@ -101,8 +101,6 @@ struct VoiceState {
u32 external_context_size{};
bool is_external_context_used{};
bool voice_dropped{};
// TODO(ogniK): Hack until ADPCM streaming is implemented
std::vector<s16> adpcm_samples{};
};
class VoiceChannelResource {