ZeldaHLE: Implement more infrastructure and one sample source.
Now renders some of the cutscene audio in Zelda TWW. Most of the work around sample sources is in a good enough state, even though it is still missing features like Dolby mixing, IIR, etc.
This commit is contained in:
Pierre Bourdon
parent
d9188f1245
commit
e35b83fabf
|
|
@ -53,7 +53,7 @@ void ZeldaUCode::DoState(PointerWrap &p)
|
||||||
p.Do(m_mail_expected_cmd_mails);
|
p.Do(m_mail_expected_cmd_mails);
|
||||||
|
|
||||||
p.Do(m_sync_max_voice_id);
|
p.Do(m_sync_max_voice_id);
|
||||||
p.Do(m_sync_flags);
|
p.Do(m_sync_voice_skip_flags);
|
||||||
|
|
||||||
p.Do(m_cmd_buffer);
|
p.Do(m_cmd_buffer);
|
||||||
p.Do(m_read_offset);
|
p.Do(m_read_offset);
|
||||||
|
|
@ -63,11 +63,11 @@ void ZeldaUCode::DoState(PointerWrap &p)
|
||||||
|
|
||||||
p.Do(m_rendering_requested_frames);
|
p.Do(m_rendering_requested_frames);
|
||||||
p.Do(m_rendering_voices_per_frame);
|
p.Do(m_rendering_voices_per_frame);
|
||||||
p.Do(m_rendering_mram_lbuf_addr);
|
|
||||||
p.Do(m_rendering_mram_rbuf_addr);
|
|
||||||
p.Do(m_rendering_curr_frame);
|
p.Do(m_rendering_curr_frame);
|
||||||
p.Do(m_rendering_curr_voice);
|
p.Do(m_rendering_curr_voice);
|
||||||
|
|
||||||
|
m_renderer.DoState(p);
|
||||||
|
|
||||||
DoStateShared(p);
|
DoStateShared(p);
|
||||||
}
|
}
|
||||||
|
|
||||||
|
|
@ -139,7 +139,7 @@ void ZeldaUCode::HandleMail(u32 mail)
|
||||||
|
|
||||||
case MailState::RENDERING:
|
case MailState::RENDERING:
|
||||||
m_sync_max_voice_id = (((mail >> 16) & 0xF) + 1) << 4;
|
m_sync_max_voice_id = (((mail >> 16) & 0xF) + 1) << 4;
|
||||||
m_sync_flags[(mail >> 16) & 0xFF] = mail & 0xFFFF;
|
m_sync_voice_skip_flags[(mail >> 16) & 0xFF] = mail & 0xFFFF;
|
||||||
|
|
||||||
RenderAudio();
|
RenderAudio();
|
||||||
SetMailState(MailState::WAITING);
|
SetMailState(MailState::WAITING);
|
||||||
|
|
@ -206,21 +206,40 @@ void ZeldaUCode::RunPendingCommands()
|
||||||
SetMailState(MailState::HALTED);
|
SetMailState(MailState::HALTED);
|
||||||
return;
|
return;
|
||||||
|
|
||||||
// Command 01: TODO: find a name and implement.
|
// Command 01: Setup/initialization command. Provides the address to
|
||||||
|
// voice parameter blocks (VPBs) as well as some array of coefficients
|
||||||
|
// used for mixing.
|
||||||
case 0x01:
|
case 0x01:
|
||||||
WARN_LOG(DSPHLE, "CMD01: %08x %08x %08x %08x",
|
{
|
||||||
Read32(), Read32(), Read32(), Read32());
|
|
||||||
m_rendering_voices_per_frame = extra_data;
|
m_rendering_voices_per_frame = extra_data;
|
||||||
|
|
||||||
|
m_renderer.SetVPBBaseAddress(Read32());
|
||||||
|
|
||||||
|
u16* data_ptr = (u16*)HLEMemory_Get_Pointer(Read32());
|
||||||
|
|
||||||
|
std::array<s16, 0x100> resampling_coeffs;
|
||||||
|
for (size_t i = 0; i < 0x100; ++i)
|
||||||
|
resampling_coeffs[i] = Common::swap16(data_ptr[i]);
|
||||||
|
m_renderer.SetResamplingCoeffs(std::move(resampling_coeffs));
|
||||||
|
|
||||||
|
std::array<s16, 0x80> sine_table;
|
||||||
|
for (size_t i = 0; i < 0x80; ++i)
|
||||||
|
sine_table[i] = Common::swap16(data_ptr[0x200 + i]);
|
||||||
|
m_renderer.SetSineTable(std::move(sine_table));
|
||||||
|
|
||||||
SendCommandAck(CommandAck::STANDARD, sync);
|
SendCommandAck(CommandAck::STANDARD, sync);
|
||||||
|
Read32(); Read32();
|
||||||
break;
|
break;
|
||||||
|
}
|
||||||
|
|
||||||
// Command 02: starts audio processing. NOTE: this handler uses return,
|
// Command 02: starts audio processing. NOTE: this handler uses return,
|
||||||
// not break. This is because it hijacks the mail control flow and
|
// not break. This is because it hijacks the mail control flow and
|
||||||
// stops processing of further commands until audio processing is done.
|
// stops processing of further commands until audio processing is done.
|
||||||
case 0x02:
|
case 0x02:
|
||||||
m_rendering_requested_frames = (cmd_mail >> 16) & 0xFF;
|
m_rendering_requested_frames = (cmd_mail >> 16) & 0xFF;
|
||||||
m_rendering_mram_lbuf_addr = Read32();
|
m_renderer.SetOutputVolume(cmd_mail & 0xFFFF);
|
||||||
m_rendering_mram_rbuf_addr = Read32();
|
m_renderer.SetOutputLeftBufferAddr(Read32());
|
||||||
|
m_renderer.SetOutputRightBufferAddr(Read32());
|
||||||
|
|
||||||
m_rendering_curr_frame = 0;
|
m_rendering_curr_frame = 0;
|
||||||
m_rendering_curr_voice = 0;
|
m_rendering_curr_voice = 0;
|
||||||
|
|
@ -259,10 +278,12 @@ void ZeldaUCode::SendCommandAck(CommandAck ack_type, u16 sync_value)
|
||||||
|
|
||||||
void ZeldaUCode::RenderAudio()
|
void ZeldaUCode::RenderAudio()
|
||||||
{
|
{
|
||||||
|
#if 0
|
||||||
WARN_LOG(DSPHLE, "RenderAudio() frame %d/%d voice %d/%d (sync to %d)",
|
WARN_LOG(DSPHLE, "RenderAudio() frame %d/%d voice %d/%d (sync to %d)",
|
||||||
m_rendering_curr_frame, m_rendering_requested_frames,
|
m_rendering_curr_frame, m_rendering_requested_frames,
|
||||||
m_rendering_curr_voice, m_rendering_voices_per_frame,
|
m_rendering_curr_voice, m_rendering_voices_per_frame,
|
||||||
m_sync_max_voice_id);
|
m_sync_max_voice_id);
|
||||||
|
#endif
|
||||||
|
|
||||||
if (!RenderingInProgress())
|
if (!RenderingInProgress())
|
||||||
{
|
{
|
||||||
|
|
@ -272,6 +293,9 @@ void ZeldaUCode::RenderAudio()
|
||||||
|
|
||||||
while (m_rendering_curr_frame < m_rendering_requested_frames)
|
while (m_rendering_curr_frame < m_rendering_requested_frames)
|
||||||
{
|
{
|
||||||
|
if (m_rendering_curr_voice == 0)
|
||||||
|
m_renderer.PrepareFrame();
|
||||||
|
|
||||||
while (m_rendering_curr_voice < m_rendering_voices_per_frame)
|
while (m_rendering_curr_voice < m_rendering_voices_per_frame)
|
||||||
{
|
{
|
||||||
// If we are not meant to render this voice yet, go back to message
|
// If we are not meant to render this voice yet, go back to message
|
||||||
|
|
@ -279,13 +303,19 @@ void ZeldaUCode::RenderAudio()
|
||||||
if (m_rendering_curr_voice >= m_sync_max_voice_id)
|
if (m_rendering_curr_voice >= m_sync_max_voice_id)
|
||||||
return;
|
return;
|
||||||
|
|
||||||
// TODO(delroth): render.
|
// Test the sync flag for this voice, skip it if not set.
|
||||||
|
u16 flags = m_sync_voice_skip_flags[m_rendering_curr_voice >> 4];
|
||||||
|
u8 bit = 0xF - (m_rendering_curr_voice & 0xF);
|
||||||
|
if (flags & (1 << bit))
|
||||||
|
m_renderer.AddVoice(m_rendering_curr_voice);
|
||||||
|
|
||||||
m_rendering_curr_voice++;
|
m_rendering_curr_voice++;
|
||||||
}
|
}
|
||||||
|
|
||||||
SendCommandAck(CommandAck::STANDARD, 0xFF00 | m_rendering_curr_frame);
|
SendCommandAck(CommandAck::STANDARD, 0xFF00 | m_rendering_curr_frame);
|
||||||
|
|
||||||
|
m_renderer.FinalizeFrame();
|
||||||
|
|
||||||
m_rendering_curr_voice = 0;
|
m_rendering_curr_voice = 0;
|
||||||
m_sync_max_voice_id = 0;
|
m_sync_max_voice_id = 0;
|
||||||
m_rendering_curr_frame++;
|
m_rendering_curr_frame++;
|
||||||
|
|
@ -294,3 +324,511 @@ void ZeldaUCode::RenderAudio()
|
||||||
SendCommandAck(CommandAck::DONE_RENDERING, 0);
|
SendCommandAck(CommandAck::DONE_RENDERING, 0);
|
||||||
m_cmd_can_execute = false; // Block command execution until ACK is received.
|
m_cmd_can_execute = false; // Block command execution until ACK is received.
|
||||||
}
|
}
|
||||||
|
|
||||||
|
// Utility to define 32 bit accessors/modifiers methods based on two 16 bit
|
||||||
|
// fields named _l and _h.
|
||||||
|
#define DEFINE_32BIT_ACCESSOR(field_name, name) \
|
||||||
|
u32 Get##name() const { return (field_name##_h << 16) | field_name##_l; } \
|
||||||
|
void Set##name(u32 v) \
|
||||||
|
{ \
|
||||||
|
field_name##_h = v >> 16; \
|
||||||
|
field_name##_l = v & 0xFFFF; \
|
||||||
|
}
|
||||||
|
|
||||||
|
#pragma pack(push, 1)
|
||||||
|
struct ZeldaAudioRenderer::VPB
|
||||||
|
{
|
||||||
|
static constexpr u16 SIZE_IN_WORDS = 0xC0;
|
||||||
|
static constexpr u16 RW_SIZE_IN_WORDS = 0x80;
|
||||||
|
|
||||||
|
// If zero, skip processing this voice.
|
||||||
|
u16 enabled;
|
||||||
|
|
||||||
|
// If non zero, skip processing this voice.
|
||||||
|
u16 done;
|
||||||
|
|
||||||
|
// In 4.12 format. 1.0 (0x1000) means 0x50 raw samples from RAM/accelerator
|
||||||
|
// will be "resampled" to 0x50 input samples. 2.0 (0x2000) means 2 raw
|
||||||
|
// samples for one input samples. 0.5 (0x800) means one raw sample for 2
|
||||||
|
// input samples.
|
||||||
|
u16 resampling_ratio;
|
||||||
|
|
||||||
|
u16 unk_03;
|
||||||
|
|
||||||
|
// If non zero, reset some value in the VPB when processing it.
|
||||||
|
u16 reset_vpb;
|
||||||
|
|
||||||
|
u16 unk_05;
|
||||||
|
|
||||||
|
// If non zero, input samples to this VPB will be the fixed value from
|
||||||
|
// VPB[33] (constant_sample_value). This is used when a voice is being
|
||||||
|
// terminated in order to force silence.
|
||||||
|
u16 use_constant_sample;
|
||||||
|
|
||||||
|
// Number of samples that should be saved in the VPB for processing during
|
||||||
|
// future frames. Should be at most TODO.
|
||||||
|
u16 samples_to_keep_count;
|
||||||
|
|
||||||
|
// Channel mixing information. Each voice can be mixed to 6 different
|
||||||
|
// channels, with separate volume information.
|
||||||
|
//
|
||||||
|
// Used only if VPB[2C] (use_dolby_volume) is not set. Otherwise, the
|
||||||
|
// values from VPB[0x20:0x2C] are used to mix to all available channels.
|
||||||
|
struct Channel
|
||||||
|
{
|
||||||
|
// Can be treated as an ID, but in the real world this is actually the
|
||||||
|
// address in DRAM of a DSP buffer. The game passes that information to
|
||||||
|
// the DSP, which means the game must know the memory layout of the DSP
|
||||||
|
// UCode... that's terrible.
|
||||||
|
u16 id;
|
||||||
|
|
||||||
|
s16 target_volume;
|
||||||
|
s16 current_volume;
|
||||||
|
|
||||||
|
u16 unk;
|
||||||
|
};
|
||||||
|
Channel channels[6];
|
||||||
|
|
||||||
|
u16 unk_20_28[0x8];
|
||||||
|
|
||||||
|
// When using Dolby voice mixing (see VPB[2C] use_dolby_volume), the X
|
||||||
|
// (left/right) and Y (front/back) coordinates of the sound. 0x00 is all
|
||||||
|
// right/back, 0x7F is all left/front. Format is 0XXXXXXX0YYYYYYY.
|
||||||
|
u16 dolby_voice_position;
|
||||||
|
u8 GetDolbyVoiceX() const { return (dolby_voice_position >> 8) & 0x7F; }
|
||||||
|
u8 GetDolbyVoiceY() const { return dolby_voice_position & 0x7F; }
|
||||||
|
|
||||||
|
// How much reverbation to apply to the Dolby mixed voice. 0 is none,
|
||||||
|
// 0x7FFF is the maximum value.
|
||||||
|
s16 dolby_reverb_factor;
|
||||||
|
|
||||||
|
// The volume for the 0x50 samples being mixed will ramp between current
|
||||||
|
// and target. After the ramping is done, the current value is updated (to
|
||||||
|
// match target, usually).
|
||||||
|
s16 dolby_volume_current;
|
||||||
|
s16 dolby_volume_target;
|
||||||
|
|
||||||
|
// If non zero, use positional audio mixing. Instead of using the channels
|
||||||
|
// information, use the 4 Dolby related VPB fields defined above.
|
||||||
|
u16 use_dolby_volume;
|
||||||
|
|
||||||
|
u16 unk_2D;
|
||||||
|
u16 unk_2E;
|
||||||
|
u16 unk_2F;
|
||||||
|
|
||||||
|
// Fractional part of the current sample position, in 0.12 format (all
|
||||||
|
// decimal part, 0x0800 = 0.5). The 4 top bits are unused.
|
||||||
|
u16 current_pos_frac;
|
||||||
|
|
||||||
|
u16 unk_31;
|
||||||
|
u16 unk_32;
|
||||||
|
|
||||||
|
// Value used as the constant sample value if VPB[6] (use_constant_sample)
|
||||||
|
// is set. Reset to the last sample value after each round of resampling.
|
||||||
|
s16 constant_sample;
|
||||||
|
|
||||||
|
// Current position in the voice. Not needed for accelerator based voice
|
||||||
|
// types since the accelerator exposes a streaming based interface, but DMA
|
||||||
|
// based voice types (PCM16_FROM_MRAM for example) require it to know where
|
||||||
|
// to seek in the MRAM buffer.
|
||||||
|
u16 current_position_h;
|
||||||
|
u16 current_position_l;
|
||||||
|
DEFINE_32BIT_ACCESSOR(current_position, CurrentPosition)
|
||||||
|
|
||||||
|
// Number of samples that will be processed before the loop point of the
|
||||||
|
// voice is reached. Maintained by the UCode and used by the game to
|
||||||
|
// schedule some parameters updates.
|
||||||
|
u16 samples_before_loop;
|
||||||
|
|
||||||
|
u16 unk_37;
|
||||||
|
u16 unk_38;
|
||||||
|
u16 unk_39;
|
||||||
|
|
||||||
|
// Remaining number of samples to load before considering the voice
|
||||||
|
// rendering complete and setting the done flag. Note that this is an
|
||||||
|
// absolute value that does not take into account loops. If a loop of 100
|
||||||
|
// samples is played 4 times, remaining_length will have decreased by 400.
|
||||||
|
u16 remaining_length_h;
|
||||||
|
u16 remaining_length_l;
|
||||||
|
DEFINE_32BIT_ACCESSOR(remaining_length, RemainingLength)
|
||||||
|
|
||||||
|
// Stores the last 4 resampled input samples after each frame, so that they
|
||||||
|
// can be used for future linear interpolation.
|
||||||
|
u16 resample_buffer[4];
|
||||||
|
|
||||||
|
u16 unk[0x80 - 0x40];
|
||||||
|
|
||||||
|
enum SamplesSourceType
|
||||||
|
{
|
||||||
|
// Samples stored in MRAM at an arbitrary sample rate (resampling is
|
||||||
|
// applied, unlike PCM16_FROM_MRAM_RAW).
|
||||||
|
SRC_PCM16_FROM_MRAM = 33,
|
||||||
|
};
|
||||||
|
u16 samples_source_type;
|
||||||
|
|
||||||
|
u16 unk_81;
|
||||||
|
u16 unk_82;
|
||||||
|
u16 unk_83;
|
||||||
|
u16 unk_84;
|
||||||
|
|
||||||
|
// If true, ramp down quickly to a volume of zero, and end the voice (by
|
||||||
|
// setting VPB[1] done) when it reaches zero.
|
||||||
|
u16 end_requested;
|
||||||
|
|
||||||
|
u16 unk_86;
|
||||||
|
u16 unk_87;
|
||||||
|
|
||||||
|
// Base address used to download samples data after the loop point of the
|
||||||
|
// voice has been reached.
|
||||||
|
u16 dma_loop_address_h;
|
||||||
|
u16 dma_loop_address_l;
|
||||||
|
DEFINE_32BIT_ACCESSOR(dma_loop_address, DMALoopAddress)
|
||||||
|
|
||||||
|
// Offset (in number of raw samples) of the start of the loop area in the
|
||||||
|
// voice. Note: some sample sources only use the _h part of this.
|
||||||
|
u16 loop_start_position_h;
|
||||||
|
u16 loop_start_position_l;
|
||||||
|
DEFINE_32BIT_ACCESSOR(loop_start_position, LoopStartPosition)
|
||||||
|
|
||||||
|
// Base address used to download samples data before the loop point of the
|
||||||
|
// voice has been reached.
|
||||||
|
u16 dma_base_address_h;
|
||||||
|
u16 dma_base_address_l;
|
||||||
|
DEFINE_32BIT_ACCESSOR(dma_base_address, DMABaseAddress)
|
||||||
|
|
||||||
|
u16 padding[SIZE_IN_WORDS];
|
||||||
|
};
|
||||||
|
#pragma pack(pop)
|
||||||
|
|
||||||
|
void ZeldaAudioRenderer::PrepareFrame()
|
||||||
|
{
|
||||||
|
if (m_prepared)
|
||||||
|
return;
|
||||||
|
|
||||||
|
m_buf_front_left.fill(0);
|
||||||
|
m_buf_front_right.fill(0);
|
||||||
|
|
||||||
|
// TODO: Dolby/reverb mixing here.
|
||||||
|
|
||||||
|
m_prepared = true;
|
||||||
|
}
|
||||||
|
|
||||||
|
void ZeldaAudioRenderer::AddVoice(u16 voice_id)
|
||||||
|
{
|
||||||
|
VPB vpb;
|
||||||
|
FetchVPB(voice_id, &vpb);
|
||||||
|
|
||||||
|
if (!vpb.enabled || vpb.done)
|
||||||
|
return;
|
||||||
|
|
||||||
|
MixingBuffer input_samples;
|
||||||
|
LoadInputSamples(&input_samples, &vpb);
|
||||||
|
|
||||||
|
// TODO: In place effects.
|
||||||
|
|
||||||
|
// TODO: IIR.filter.
|
||||||
|
|
||||||
|
// TODO: Loop, etc.
|
||||||
|
|
||||||
|
if (vpb.use_dolby_volume)
|
||||||
|
{
|
||||||
|
if (vpb.end_requested)
|
||||||
|
{
|
||||||
|
vpb.dolby_volume_target = vpb.dolby_volume_current / 2;
|
||||||
|
if (vpb.dolby_volume_target == 0)
|
||||||
|
vpb.done = true;
|
||||||
|
}
|
||||||
|
|
||||||
|
// Each of these volumes is in 1.15 fixed format.
|
||||||
|
s16 right_volume = m_sine_table[vpb.GetDolbyVoiceX()];
|
||||||
|
s16 back_volume = m_sine_table[vpb.GetDolbyVoiceY()];
|
||||||
|
s16 left_volume = m_sine_table[vpb.GetDolbyVoiceX() ^ 0x7F];
|
||||||
|
s16 front_volume = m_sine_table[vpb.GetDolbyVoiceY() ^ 0x7F];
|
||||||
|
|
||||||
|
// Compute volume for each quadrant.
|
||||||
|
s16 quadrant_volumes[4] = {
|
||||||
|
(s16)((left_volume * front_volume) >> 16),
|
||||||
|
(s16)((left_volume * back_volume) >> 16),
|
||||||
|
(s16)((right_volume * front_volume) >> 16),
|
||||||
|
(s16)((right_volume * back_volume) >> 16),
|
||||||
|
};
|
||||||
|
|
||||||
|
// Compute the volume delta for each sample to match the difference
|
||||||
|
// between current and target volume.
|
||||||
|
s16 delta = vpb.dolby_volume_target - vpb.dolby_volume_current;
|
||||||
|
s16 volume_deltas[4];
|
||||||
|
for (size_t i = 0; i < 4; ++i)
|
||||||
|
volume_deltas[i] = ((u16)quadrant_volumes[i] * delta) >> 16;
|
||||||
|
|
||||||
|
// Apply master volume to each quadrant.
|
||||||
|
for (size_t i = 0; i < 4; ++i)
|
||||||
|
quadrant_volumes[i] = (quadrant_volumes[i] * vpb.dolby_volume_current) >> 16;
|
||||||
|
|
||||||
|
// Compute reverb volume and ramp deltas.
|
||||||
|
s16 reverb_volumes[4], reverb_volume_deltas[4];
|
||||||
|
s16 reverb_volume_factor = (vpb.dolby_volume_current * vpb.dolby_reverb_factor) >> 15;
|
||||||
|
for (size_t i = 0; i < 4; ++i)
|
||||||
|
{
|
||||||
|
reverb_volumes[i] = (quadrant_volumes[i] * reverb_volume_factor) >> 15;
|
||||||
|
reverb_volume_deltas[i] = (volume_deltas[i] * vpb.dolby_reverb_factor) >> 16;
|
||||||
|
}
|
||||||
|
|
||||||
|
struct {
|
||||||
|
MixingBuffer* buffer;
|
||||||
|
s16 volume;
|
||||||
|
s16 volume_delta;
|
||||||
|
} buffers[8] = {
|
||||||
|
{ &m_buf_front_left, quadrant_volumes[0], volume_deltas[0] },
|
||||||
|
{ &m_buf_back_left, quadrant_volumes[1], volume_deltas[1] },
|
||||||
|
{ &m_buf_front_right, quadrant_volumes[2], volume_deltas[2] },
|
||||||
|
{ &m_buf_back_right, quadrant_volumes[3], volume_deltas[3] },
|
||||||
|
|
||||||
|
{ &m_buf_front_left_reverb, reverb_volumes[0], reverb_volume_deltas[0] },
|
||||||
|
{ &m_buf_back_left_reverb, reverb_volumes[1], reverb_volume_deltas[1] },
|
||||||
|
{ &m_buf_front_right_reverb, reverb_volumes[2], reverb_volume_deltas[2] },
|
||||||
|
{ &m_buf_back_right_reverb, reverb_volumes[3], reverb_volume_deltas[3] },
|
||||||
|
};
|
||||||
|
for (const auto& buffer : buffers)
|
||||||
|
{
|
||||||
|
AddBuffersWithVolumeRamp(buffer.buffer, input_samples, buffer.volume << 16,
|
||||||
|
(buffer.volume_delta << 16) / (s32)buffer.buffer->size());
|
||||||
|
}
|
||||||
|
|
||||||
|
vpb.dolby_volume_current = vpb.dolby_volume_target;
|
||||||
|
}
|
||||||
|
else
|
||||||
|
{
|
||||||
|
// TODO: Store input samples if requested by the VPB.
|
||||||
|
|
||||||
|
if (vpb.end_requested)
|
||||||
|
{
|
||||||
|
bool all_mute = true;
|
||||||
|
for (auto& channel : vpb.channels)
|
||||||
|
{
|
||||||
|
channel.target_volume = channel.current_volume / 2;
|
||||||
|
all_mute &= (channel.target_volume == 0);
|
||||||
|
}
|
||||||
|
if (all_mute)
|
||||||
|
vpb.done = true;
|
||||||
|
}
|
||||||
|
|
||||||
|
// Map buffer "IDs"/addresses to our emulated buffers.
|
||||||
|
std::map<u16, MixingBuffer*> buffers = {
|
||||||
|
{ 0x0D00, &m_buf_front_left },
|
||||||
|
{ 0x0D60, &m_buf_front_right },
|
||||||
|
{ 0x0F40, &m_buf_back_left },
|
||||||
|
{ 0x0CA0, &m_buf_back_right },
|
||||||
|
{ 0x0E80, &m_buf_front_left_reverb },
|
||||||
|
{ 0x0EE0, &m_buf_front_right_reverb },
|
||||||
|
{ 0x0C00, &m_buf_back_left_reverb },
|
||||||
|
{ 0x0C50, &m_buf_back_right_reverb },
|
||||||
|
};
|
||||||
|
|
||||||
|
for (auto& channel : vpb.channels)
|
||||||
|
{
|
||||||
|
if (!channel.id)
|
||||||
|
continue;
|
||||||
|
|
||||||
|
s16 volume_delta = channel.target_volume - channel.current_volume;
|
||||||
|
s32 volume_step = (volume_delta << 16) / (s32)input_samples.size(); // In 1.31 format.
|
||||||
|
|
||||||
|
// TODO: The last value of each channel structure is used to
|
||||||
|
// determine whether a channel should be skipped or not. Not
|
||||||
|
// implemented yet.
|
||||||
|
|
||||||
|
if (!channel.current_volume && !volume_step)
|
||||||
|
continue;
|
||||||
|
|
||||||
|
MixingBuffer* dst_buffer = buffers[channel.id];
|
||||||
|
if (!dst_buffer)
|
||||||
|
{
|
||||||
|
ERROR_LOG(DSPHLE, "Mixing to an unmapped buffer: %04x", channel.id);
|
||||||
|
continue;
|
||||||
|
}
|
||||||
|
|
||||||
|
s32 new_volume = AddBuffersWithVolumeRamp(
|
||||||
|
dst_buffer, input_samples, channel.current_volume << 16,
|
||||||
|
volume_step);
|
||||||
|
channel.current_volume = new_volume >> 16;
|
||||||
|
}
|
||||||
|
}
|
||||||
|
|
||||||
|
StoreVPB(voice_id, vpb);
|
||||||
|
}
|
||||||
|
|
||||||
|
void ZeldaAudioRenderer::FinalizeFrame()
|
||||||
|
{
|
||||||
|
// TODO: Dolby mixing.
|
||||||
|
|
||||||
|
ApplyVolumeInPlace_4_12(&m_buf_front_left, m_output_volume);
|
||||||
|
ApplyVolumeInPlace_4_12(&m_buf_front_right, m_output_volume);
|
||||||
|
|
||||||
|
u16* ram_left_buffer = (u16*)HLEMemory_Get_Pointer(m_output_lbuf_addr);
|
||||||
|
u16* ram_right_buffer = (u16*)HLEMemory_Get_Pointer(m_output_rbuf_addr);
|
||||||
|
for (size_t i = 0; i < m_buf_front_left.size(); ++i)
|
||||||
|
{
|
||||||
|
ram_left_buffer[i] = Common::swap16(m_buf_front_left[i]);
|
||||||
|
ram_right_buffer[i] = Common::swap16(m_buf_front_right[i]);
|
||||||
|
}
|
||||||
|
m_output_lbuf_addr += sizeof (u16) * m_buf_front_left.size();
|
||||||
|
m_output_rbuf_addr += sizeof (u16) * m_buf_front_right.size();
|
||||||
|
|
||||||
|
// TODO: Some more Dolby mixing.
|
||||||
|
|
||||||
|
m_prepared = false;
|
||||||
|
}
|
||||||
|
|
||||||
|
void ZeldaAudioRenderer::FetchVPB(u16 voice_id, VPB* vpb)
|
||||||
|
{
|
||||||
|
u16* vpb_words = (u16*)vpb;
|
||||||
|
u16* ram_vpbs = (u16*)HLEMemory_Get_Pointer(m_vpb_base_addr);
|
||||||
|
|
||||||
|
size_t base_idx = voice_id * VPB::SIZE_IN_WORDS;
|
||||||
|
for (size_t i = 0; i < VPB::SIZE_IN_WORDS; ++i)
|
||||||
|
vpb_words[i] = Common::swap16(ram_vpbs[base_idx + i]);
|
||||||
|
}
|
||||||
|
|
||||||
|
void ZeldaAudioRenderer::StoreVPB(u16 voice_id, const VPB& vpb)
|
||||||
|
{
|
||||||
|
const u16* vpb_words = (const u16*)&vpb;
|
||||||
|
u16* ram_vpbs = (u16*)HLEMemory_Get_Pointer(m_vpb_base_addr);
|
||||||
|
|
||||||
|
size_t base_idx = voice_id * VPB::SIZE_IN_WORDS;
|
||||||
|
|
||||||
|
// Only the first 0x80 words are transferred back - the rest is read-only.
|
||||||
|
for (size_t i = 0; i < VPB::RW_SIZE_IN_WORDS; ++i)
|
||||||
|
ram_vpbs[base_idx + i] = Common::swap16(vpb_words[i]);
|
||||||
|
}
|
||||||
|
|
||||||
|
void ZeldaAudioRenderer::LoadInputSamples(MixingBuffer* buffer, VPB* vpb)
|
||||||
|
{
|
||||||
|
// Input data pre-resampling. Resampled into the mixing buffer parameter at
|
||||||
|
// the end of processing, if needed.
|
||||||
|
//
|
||||||
|
// Maximum of 0x500 samples here - see NeededRawSamplesCount to understand
|
||||||
|
// this practical limit (resampling_ratio = 0xFFFF -> 0x500 samples). Add a
|
||||||
|
// margin of 4 that is needed for samples source that do resampling.
|
||||||
|
std::array<s16, 0x500 + 4> raw_input_samples;
|
||||||
|
for (size_t i = 0; i < 4; ++i)
|
||||||
|
raw_input_samples[i] = vpb->resample_buffer[i];
|
||||||
|
|
||||||
|
if (vpb->use_constant_sample)
|
||||||
|
{
|
||||||
|
buffer->fill(vpb->constant_sample);
|
||||||
|
return;
|
||||||
|
}
|
||||||
|
|
||||||
|
switch (vpb->samples_source_type)
|
||||||
|
{
|
||||||
|
case VPB::SRC_PCM16_FROM_MRAM:
|
||||||
|
{
|
||||||
|
DownloadRawSamplesFromMRAM(raw_input_samples.data() + 4, vpb,
|
||||||
|
NeededRawSamplesCount(*vpb));
|
||||||
|
Resample(vpb, raw_input_samples.data(), buffer);
|
||||||
|
break;
|
||||||
|
}
|
||||||
|
|
||||||
|
default:
|
||||||
|
ERROR_LOG(DSPHLE, "Using an unknown/unimplemented sample source: %04x", vpb->samples_source_type);
|
||||||
|
buffer->fill(0);
|
||||||
|
return;
|
||||||
|
}
|
||||||
|
}
|
||||||
|
|
||||||
|
u16 ZeldaAudioRenderer::NeededRawSamplesCount(const VPB& vpb)
|
||||||
|
{
|
||||||
|
// Both of these are 4.12 fixed point, so shift by 12 to get the int part.
|
||||||
|
return (vpb.current_pos_frac + 0x50 * vpb.resampling_ratio) >> 12;
|
||||||
|
}
|
||||||
|
|
||||||
|
void ZeldaAudioRenderer::Resample(VPB* vpb, const s16* src, MixingBuffer* dst)
|
||||||
|
{
|
||||||
|
// Both in 20.12 format.
|
||||||
|
u32 ratio = vpb->resampling_ratio;
|
||||||
|
u32 pos = vpb->current_pos_frac;
|
||||||
|
|
||||||
|
// Check if we need to do some interpolation. If the resampling ratio is
|
||||||
|
// more than 4:1, it's not worth it.
|
||||||
|
if ((ratio >> 12) >= 4)
|
||||||
|
{
|
||||||
|
for (s16& dst_sample : *dst)
|
||||||
|
{
|
||||||
|
pos += ratio;
|
||||||
|
dst_sample = src[pos >> 12];
|
||||||
|
}
|
||||||
|
}
|
||||||
|
else
|
||||||
|
{
|
||||||
|
for (auto& dst_sample : *dst)
|
||||||
|
{
|
||||||
|
// We have 0x40 * 4 coeffs that need to be selected based on the
|
||||||
|
// most significant bits of the fractional part of the position. 12
|
||||||
|
// bits >> 6 = 6 bits = 0x40. Multiply by 4 since there are 4
|
||||||
|
// consecutive coeffs.
|
||||||
|
u32 coeffs_idx = ((pos & 0xFFF) >> 6) * 4;
|
||||||
|
const s16* coeffs = &m_resampling_coeffs[coeffs_idx];
|
||||||
|
const s16* input = &src[pos >> 12];
|
||||||
|
|
||||||
|
s64 dst_sample_unclamped = 0;
|
||||||
|
for (size_t i = 0; i < 4; ++i)
|
||||||
|
dst_sample_unclamped += (s64)2 * coeffs[i] * input[i];
|
||||||
|
dst_sample_unclamped >>= 16;
|
||||||
|
MathUtil::Clamp(&dst_sample_unclamped, (s64)-0x8000, (s64)0x7fff);
|
||||||
|
dst_sample = dst_sample_unclamped;
|
||||||
|
|
||||||
|
pos += ratio;
|
||||||
|
}
|
||||||
|
}
|
||||||
|
|
||||||
|
for (u32 i = 0; i < 4; ++i)
|
||||||
|
vpb->resample_buffer[i] = src[(pos >> 12) + i];
|
||||||
|
vpb->constant_sample = (*dst)[dst->size() - 1];
|
||||||
|
vpb->current_pos_frac = pos & 0xFFF;
|
||||||
|
}
|
||||||
|
|
||||||
|
void ZeldaAudioRenderer::DownloadRawSamplesFromMRAM(
|
||||||
|
s16* dst, VPB* vpb, u16 requested_samples_count)
|
||||||
|
{
|
||||||
|
u32 addr = vpb->GetDMABaseAddress() + vpb->current_position_h * sizeof (u16);
|
||||||
|
s16* src_ptr = (s16*)HLEMemory_Get_Pointer(addr);
|
||||||
|
|
||||||
|
if (requested_samples_count > vpb->GetRemainingLength())
|
||||||
|
{
|
||||||
|
s16 last_sample = 0;
|
||||||
|
for (u16 i = 0; i < vpb->GetRemainingLength(); ++i)
|
||||||
|
*dst++ = last_sample = Common::swap16(*src_ptr++);
|
||||||
|
for (u16 i = vpb->GetRemainingLength(); i < requested_samples_count; ++i)
|
||||||
|
*dst++ = last_sample;
|
||||||
|
|
||||||
|
vpb->current_position_h += vpb->GetRemainingLength();
|
||||||
|
vpb->SetRemainingLength(0);
|
||||||
|
vpb->done = true;
|
||||||
|
}
|
||||||
|
else
|
||||||
|
{
|
||||||
|
vpb->SetRemainingLength(vpb->GetRemainingLength() - requested_samples_count);
|
||||||
|
vpb->samples_before_loop = vpb->loop_start_position_h - vpb->current_position_h;
|
||||||
|
if (requested_samples_count <= vpb->samples_before_loop)
|
||||||
|
{
|
||||||
|
for (u16 i = 0; i < requested_samples_count; ++i)
|
||||||
|
*dst++ = Common::swap16(*src_ptr++);
|
||||||
|
vpb->current_position_h += requested_samples_count;
|
||||||
|
}
|
||||||
|
else
|
||||||
|
{
|
||||||
|
for (u16 i = 0; i < vpb->samples_before_loop; ++i)
|
||||||
|
*dst++ = Common::swap16(*src_ptr++);
|
||||||
|
vpb->SetDMABaseAddress(vpb->GetDMALoopAddress());
|
||||||
|
src_ptr = (s16*)HLEMemory_Get_Pointer(vpb->GetDMALoopAddress());
|
||||||
|
for (u16 i = vpb->samples_before_loop; i < requested_samples_count; ++i)
|
||||||
|
*dst++ = Common::swap16(*src_ptr++);
|
||||||
|
vpb->current_position_h = requested_samples_count - vpb->samples_before_loop;
|
||||||
|
}
|
||||||
|
}
|
||||||
|
}
|
||||||
|
|
||||||
|
void ZeldaAudioRenderer::DoState(PointerWrap& p)
|
||||||
|
{
|
||||||
|
p.Do(m_output_lbuf_addr);
|
||||||
|
p.Do(m_output_rbuf_addr);
|
||||||
|
}
|
||||||
|
|
|
||||||
|
|
@ -5,8 +5,142 @@
|
||||||
#pragma once
|
#pragma once
|
||||||
|
|
||||||
#include "Common/CommonTypes.h"
|
#include "Common/CommonTypes.h"
|
||||||
|
#include "Common/MathUtil.h"
|
||||||
#include "Core/HW/DSPHLE/UCodes/UCodes.h"
|
#include "Core/HW/DSPHLE/UCodes/UCodes.h"
|
||||||
|
|
||||||
|
class ZeldaAudioRenderer
|
||||||
|
{
|
||||||
|
public:
|
||||||
|
void PrepareFrame();
|
||||||
|
void AddVoice(u16 voice_id);
|
||||||
|
void FinalizeFrame();
|
||||||
|
|
||||||
|
void SetSineTable(std::array<s16, 0x80>&& sine_table) { m_sine_table = sine_table; }
|
||||||
|
void SetResamplingCoeffs(std::array<s16, 0x100>&& coeffs) { m_resampling_coeffs = coeffs; }
|
||||||
|
void SetVPBBaseAddress(u32 addr) { m_vpb_base_addr = addr; }
|
||||||
|
void SetOutputVolume(u16 volume) { m_output_volume = volume; }
|
||||||
|
void SetOutputLeftBufferAddr(u32 addr) { m_output_lbuf_addr = addr; }
|
||||||
|
void SetOutputRightBufferAddr(u32 addr) { m_output_rbuf_addr = addr; }
|
||||||
|
|
||||||
|
void DoState(PointerWrap& p);
|
||||||
|
|
||||||
|
private:
|
||||||
|
struct VPB;
|
||||||
|
|
||||||
|
// Utility functions for audio operations.
|
||||||
|
|
||||||
|
// Apply volume to a buffer. The volume is a fixed point integer, usually
|
||||||
|
// 1.15 or 4.12 in the DAC UCode.
|
||||||
|
template <size_t N, size_t B>
|
||||||
|
void ApplyVolumeInPlace(std::array<s16, N>* buf, u16 vol)
|
||||||
|
{
|
||||||
|
for (size_t i = 0; i < N; ++i)
|
||||||
|
{
|
||||||
|
s32 tmp = (u32)(*buf)[i] * (u32)vol;
|
||||||
|
tmp >>= 16 - B;
|
||||||
|
MathUtil::Clamp(&tmp, -0x8000, 0x7fff);
|
||||||
|
(*buf)[i] = (s16)tmp;
|
||||||
|
}
|
||||||
|
}
|
||||||
|
template <size_t N>
|
||||||
|
void ApplyVolumeInPlace_1_15(std::array<s16, N>* buf, u16 vol)
|
||||||
|
{
|
||||||
|
ApplyVolumeInPlace<N, 1>(buf, vol);
|
||||||
|
}
|
||||||
|
template <size_t N>
|
||||||
|
void ApplyVolumeInPlace_4_12(std::array<s16, N>* buf, u16 vol)
|
||||||
|
{
|
||||||
|
ApplyVolumeInPlace<N, 4>(buf, vol);
|
||||||
|
}
|
||||||
|
|
||||||
|
// Mixes two buffers together while applying a volume to one of them. The
|
||||||
|
// volume ramps up/down in N steps using the provided step delta value.
|
||||||
|
//
|
||||||
|
// Note: On a real GC, the stepping happens in 32 steps instead. But hey,
|
||||||
|
// we can do better here with very low risk. Why not? :)
|
||||||
|
template <size_t N>
|
||||||
|
s32 AddBuffersWithVolumeRamp(std::array<s16, N>* dst,
|
||||||
|
const std::array<s16, N>& src,
|
||||||
|
s32 vol, s32 step)
|
||||||
|
{
|
||||||
|
if (!vol && !step)
|
||||||
|
return vol;
|
||||||
|
|
||||||
|
for (size_t i = 0; i < N; ++i)
|
||||||
|
{
|
||||||
|
(*dst)[i] += ((vol >> 16) * src[i]) >> 16;
|
||||||
|
vol += step;
|
||||||
|
}
|
||||||
|
|
||||||
|
return vol;
|
||||||
|
}
|
||||||
|
|
||||||
|
// Whether the frame needs to be prepared or not.
|
||||||
|
bool m_prepared = false;
|
||||||
|
|
||||||
|
// MRAM addresses where output samples should be copied.
|
||||||
|
u32 m_output_lbuf_addr = 0;
|
||||||
|
u32 m_output_rbuf_addr = 0;
|
||||||
|
|
||||||
|
// Output volume applied to buffers before being uploaded to RAM.
|
||||||
|
u16 m_output_volume = 0;
|
||||||
|
|
||||||
|
// Mixing buffers.
|
||||||
|
typedef std::array<s16, 0x50> MixingBuffer;
|
||||||
|
MixingBuffer m_buf_front_left;
|
||||||
|
MixingBuffer m_buf_front_right;
|
||||||
|
MixingBuffer m_buf_back_left;
|
||||||
|
MixingBuffer m_buf_back_right;
|
||||||
|
MixingBuffer m_buf_front_left_reverb;
|
||||||
|
MixingBuffer m_buf_front_right_reverb;
|
||||||
|
MixingBuffer m_buf_back_left_reverb;
|
||||||
|
MixingBuffer m_buf_back_right_reverb;
|
||||||
|
|
||||||
|
// Base address where VPBs are stored linearly in RAM.
|
||||||
|
u32 m_vpb_base_addr;
|
||||||
|
void FetchVPB(u16 voice_id, VPB* vpb);
|
||||||
|
void StoreVPB(u16 voice_id, const VPB& vpb);
|
||||||
|
|
||||||
|
// Sine table transferred from MRAM. Contains sin(x) values for x in
|
||||||
|
// [0.0;pi/4] (sin(x) in [1.0;0.0]), in 1.15 fixed format.
|
||||||
|
std::array<s16, 0x80> m_sine_table;
|
||||||
|
|
||||||
|
// Fills up a buffer with the input samples for a voice, represented by its
|
||||||
|
// VPB.
|
||||||
|
void LoadInputSamples(MixingBuffer* buffer, VPB* vpb);
|
||||||
|
|
||||||
|
// Raw samples (pre-resampling) that need to be generated to result in 0x50
|
||||||
|
// post-resampling input samples.
|
||||||
|
u16 NeededRawSamplesCount(const VPB& vpb);
|
||||||
|
|
||||||
|
// Resamples raw samples to 0x50 input samples, using the resampling ratio
|
||||||
|
// and current position information from the VPB.
|
||||||
|
void Resample(VPB* vpb, const s16* src, MixingBuffer* dst);
|
||||||
|
|
||||||
|
// Coefficients used for resampling.
|
||||||
|
std::array<s16, 0x100> m_resampling_coeffs{};
|
||||||
|
|
||||||
|
// If non zero, base MRAM address for sound data transfers from ARAM. On
|
||||||
|
// the Wii, this points to some MRAM location since there is no ARAM to be
|
||||||
|
// used. If zero, use the top of ARAM.
|
||||||
|
u32 m_aram_base_addr = 0;
|
||||||
|
void* GetARAMPtr() const;
|
||||||
|
|
||||||
|
// Downloads PCM encoded samples from ARAM. Handles looping and other
|
||||||
|
// parameters appropriately.
|
||||||
|
template <typename T> void DownloadPCMSamplesFromARAM(s16* dst, VPB* vpb, u16 requested_samples_count);
|
||||||
|
|
||||||
|
// Downloads AFC encoded samples from ARAM and decode them. Handles looping
|
||||||
|
// and other parameters appropriately.
|
||||||
|
void DownloadAFCSamplesFromARAM(s16* dst, VPB* vpb, u16 requested_samples_count);
|
||||||
|
void DecodeAFC(VPB* vpb, s16* dst, size_t block_count);
|
||||||
|
std::array<s16, 0x20> m_afc_coeffs{};
|
||||||
|
|
||||||
|
// Downloads samples from MRAM while handling appropriate length / looping
|
||||||
|
// behavior.
|
||||||
|
void DownloadRawSamplesFromMRAM(s16* dst, VPB* vpb, u16 requested_samples_count);
|
||||||
|
};
|
||||||
|
|
||||||
class ZeldaUCode : public UCodeInterface
|
class ZeldaUCode : public UCodeInterface
|
||||||
{
|
{
|
||||||
public:
|
public:
|
||||||
|
|
@ -48,9 +182,10 @@ private:
|
||||||
// Voice synchronization / audio rendering flow control. When rendering an
|
// Voice synchronization / audio rendering flow control. When rendering an
|
||||||
// audio frame, only voices up to max_voice_id will be rendered until a
|
// audio frame, only voices up to max_voice_id will be rendered until a
|
||||||
// sync mail arrives, increasing the value of max_voice_id. Additionally,
|
// sync mail arrives, increasing the value of max_voice_id. Additionally,
|
||||||
// these sync mails contain 16 bit values that are used for TODO.
|
// these sync mails contain 16 bit values that are used as bitfields to
|
||||||
|
// control voice skipping on a voice per voice level.
|
||||||
u32 m_sync_max_voice_id = 0;
|
u32 m_sync_max_voice_id = 0;
|
||||||
std::array<u32, 256> m_sync_flags;
|
std::array<u32, 256> m_sync_voice_skip_flags;
|
||||||
|
|
||||||
// Command buffer (circular queue with r/w indices). Filled by HandleMail
|
// Command buffer (circular queue with r/w indices). Filled by HandleMail
|
||||||
// when the state machine is in WRITING_CMD state. Commands get executed
|
// when the state machine is in WRITING_CMD state. Commands get executed
|
||||||
|
|
@ -97,14 +232,15 @@ private:
|
||||||
};
|
};
|
||||||
void SendCommandAck(CommandAck ack_type, u16 sync_value);
|
void SendCommandAck(CommandAck ack_type, u16 sync_value);
|
||||||
|
|
||||||
// Audio rendering state.
|
// Audio rendering flow control state.
|
||||||
u32 m_rendering_requested_frames = 0;
|
u32 m_rendering_requested_frames = 0;
|
||||||
u16 m_rendering_voices_per_frame = 0;
|
u16 m_rendering_voices_per_frame = 0;
|
||||||
u32 m_rendering_mram_lbuf_addr = 0;
|
|
||||||
u32 m_rendering_mram_rbuf_addr = 0;
|
|
||||||
u32 m_rendering_curr_frame = 0;
|
u32 m_rendering_curr_frame = 0;
|
||||||
u32 m_rendering_curr_voice = 0;
|
u32 m_rendering_curr_voice = 0;
|
||||||
|
|
||||||
bool RenderingInProgress() const { return m_rendering_curr_frame != m_rendering_requested_frames; }
|
bool RenderingInProgress() const { return m_rendering_curr_frame != m_rendering_requested_frames; }
|
||||||
void RenderAudio();
|
void RenderAudio();
|
||||||
|
|
||||||
|
// Main object handling audio rendering logic and state.
|
||||||
|
ZeldaAudioRenderer m_renderer;
|
||||||
};
|
};
|
||||||
|
|
|
||||||
Loading…
Reference in New Issue