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[Shinon] Enable Directsound2 player to use more than 1 channel audio. 

- DS2 Player has a similar structure to XAudio, so I don't see any reason why not to enable 1 channel+ audio.
 - Haven't tried with 5.1 channel sources but I believe it should be the same as 2 channel (As in, Directsound will downmix the audio to 2 channel)
 - Moved the volume setting to using the player directly and from some quick audio tests, -10000 is too soft. I tried with -5000 instead which seems to be alright.
This commit is contained in:
Ristellise 2022-08-11 12:28:35 +08:00
parent 994e50048a
commit 6906b6e5d7
1 changed files with 22 additions and 14 deletions
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36
src/audio_player_dsound2.cpp
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@ -48,7 +48,6 @@
#include <mmsystem.h>
#include <process.h>
#include <dsound.h>
#include <cguid.h>
namespace {
class DirectSoundPlayer2Thread;
@ -318,15 +317,14 @@ void DirectSoundPlayer2Thread::Run()
// Describe the wave format
WAVEFORMATEX waveFormat;
waveFormat.wFormatTag = WAVE_FORMAT_PCM;
waveFormat.nSamplesPerSec = provider->GetSampleRate();
//waveFormat.nChannels = provider->GetChannels();
//waveFormat.wBitsPerSample = provider->GetBytesPerSample() * 8;
waveFormat.nChannels = 1;
waveFormat.wBitsPerSample = sizeof(int16_t) * 8;
waveFormat.cbSize = 0;
waveFormat.wFormatTag = provider->AreSamplesFloat() ? 3 : WAVE_FORMAT_PCM; // Eh fuck it.
waveFormat.nChannels = provider->GetChannels();
waveFormat.wBitsPerSample = provider->GetBytesPerSample() * 8;
waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
waveFormat.cbSize = sizeof(waveFormat);
//waveFormat.cbSize = sizeof(waveFormat);
// And the buffer itself
int aim = waveFormat.nAvgBytesPerSec * (wanted_latency*buffer_length)/1000;
@ -335,7 +333,7 @@ void DirectSoundPlayer2Thread::Run()
DWORD bufSize = mid(min,aim,max); // size of entire playback buffer
DSBUFFERDESC desc;
desc.dwSize = sizeof(DSBUFFERDESC);
desc.dwFlags = DSBCAPS_GETCURRENTPOSITION2 | DSBCAPS_GLOBALFOCUS;
desc.dwFlags = DSBCAPS_CTRLVOLUME | DSBCAPS_GETCURRENTPOSITION2 | DSBCAPS_GLOBALFOCUS;
desc.dwBufferBytes = bufSize;
desc.dwReserved = 0;
desc.lpwfxFormat = &waveFormat;
@ -372,7 +370,7 @@ void DirectSoundPlayer2Thread::Run()
DWORD buffer_offset = 0;
bool playback_should_be_running = false;
int current_latency = wanted_latency;
const DWORD wanted_latency_bytes = wanted_latency*waveFormat.nSamplesPerSec*/*provider->GetBytesPerSample()*/sizeof(int16_t)/1000;
const DWORD wanted_latency_bytes = wanted_latency*waveFormat.nSamplesPerSec*provider->GetBytesPerSample()/1000;
while (running)
{
@ -425,7 +423,7 @@ void DirectSoundPlayer2Thread::Run()
if (bytes_filled < wanted_latency_bytes)
{
// Very short playback length, do without streaming playback
current_latency = (bytes_filled*1000) / (waveFormat.nSamplesPerSec*/*provider->GetBytesPerSample()*/sizeof(int16_t));
current_latency = (bytes_filled*1000) / (waveFormat.nSamplesPerSec*provider->GetBytesPerSample());
if (FAILED(bfr->Play(0, 0, 0)))
REPORT_ERROR("Could not start single-buffer playback.")
}
@ -464,6 +462,16 @@ stop_playback:
goto do_fill_buffer;
case WAIT_OBJECT_0+3:
{
LONG invert_volume = (LONG)((this->volume - 1.0) * 5000.0); // Hrmm weirdly it's half?
// Look, I would have used a min max but it just errored out for me lol.
if (invert_volume > DSBVOLUME_MAX)
invert_volume = DSBVOLUME_MAX;
else if (invert_volume < DSBVOLUME_MIN / 2)
invert_volume = DSBVOLUME_MIN / 2;
LOG_I("DS2") << "Earrape vlume: " <<invert_volume;
bfr->SetVolume(invert_volume);
}
// Change volume
// We aren't thread safe right now, filling the buffers grabs volume directly
// from the field set by the controlling thread, but it shouldn't be a major
@ -556,7 +564,7 @@ do_fill_buffer:
else if (bytes_filled < wanted_latency_bytes)
{
// Didn't fill as much as we wanted to, let's get back to filling sooner than normal
current_latency = (bytes_filled*1000) / (waveFormat.nSamplesPerSec*/*provider->GetBytesPerSample()*/sizeof(int16_t));
current_latency = (bytes_filled*1000) / (waveFormat.nSamplesPerSec*provider->GetBytesPerSample());
}
else
{
@ -580,7 +588,7 @@ DWORD DirectSoundPlayer2Thread::FillAndUnlockBuffers(void *buf1, DWORD buf1sz, v
{
// Assume buffers have been locked and are ready to be filled
DWORD bytes_per_frame = /*provider->GetChannels() * provider->GetBytesPerSample()*/sizeof(int16_t);
DWORD bytes_per_frame = provider->GetChannels() * provider->GetBytesPerSample();
DWORD buf1szf = buf1sz / bytes_per_frame;
DWORD buf2szf = buf2sz / bytes_per_frame;
@ -611,7 +619,7 @@ DWORD DirectSoundPlayer2Thread::FillAndUnlockBuffers(void *buf1, DWORD buf1sz, v
buf2sz = 0;
}
provider->GetInt16MonoAudioWithVolume(reinterpret_cast<int16_t*>(buf1), input_frame, buf1szf, volume);
provider->GetAudio(buf1, input_frame, buf1szf);
input_frame += buf1szf;
}
@ -624,7 +632,7 @@ DWORD DirectSoundPlayer2Thread::FillAndUnlockBuffers(void *buf1, DWORD buf1sz, v
buf2sz = buf2szf * bytes_per_frame;
}
provider->GetInt16MonoAudioWithVolume(reinterpret_cast<int16_t*>(buf2), input_frame, buf2szf, volume);
provider->GetAudio(buf2, input_frame, buf2szf);
input_frame += buf2szf;
}