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authorDavid Marcec <dmarcecguzman@gmail.com>2020-07-30 10:16:57 +0200
committerDavid Marcec <dmarcecguzman@gmail.com>2020-07-30 10:16:57 +0200
commit1b8fe7073b4877b697bd4101a48d77011100579e (patch)
treee4d369207b07b25dd88997317ac8ab5584809f5d
parentFix perf regression (diff)
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-rw-r--r--src/audio_core/command_generator.cpp53
-rw-r--r--src/audio_core/command_generator.h1
-rw-r--r--src/audio_core/voice_context.cpp3
-rw-r--r--src/audio_core/voice_context.h2
4 files changed, 32 insertions, 27 deletions
diff --git a/src/audio_core/command_generator.cpp b/src/audio_core/command_generator.cpp
index 440bfc140..0c3b278ea 100644
--- a/src/audio_core/command_generator.cpp
+++ b/src/audio_core/command_generator.cpp
@@ -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;
- // 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::move(Codec::DecodeADPCM(buffer.data(), buffer.size(), coeffs, dsp_state.context));
- }
+ 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;
+
+ 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 = 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;
diff --git a/src/audio_core/command_generator.h b/src/audio_core/command_generator.h
index 3ad8973b5..3f49c1303 100644
--- a/src/audio_core/command_generator.h
+++ b/src/audio_core/command_generator.h
@@ -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;
diff --git a/src/audio_core/voice_context.cpp b/src/audio_core/voice_context.cpp
index 16ae1afe8..1d8f69844 100644
--- a/src/audio_core/voice_context.cpp
+++ b/src/audio_core/voice_context.cpp
@@ -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
diff --git a/src/audio_core/voice_context.h b/src/audio_core/voice_context.h
index b1d554766..13b0a7f0f 100644
--- a/src/audio_core/voice_context.h
+++ b/src/audio_core/voice_context.h
@@ -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 {