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Audio: Implement PCM8.

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This commit is contained in:
MerryMage 2016-01-24 10:38:17 +00:00 committed by Sean Maas
parent d3a48335b8
commit 16a5a8792a
2 changed files with 47 additions and 14 deletions
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59
src/core/audio/stream.cpp
View File

@ -11,6 +11,8 @@
#include <queue> #include <queue>
namespace Audio { namespace Audio {
std::vector<u16> DecodeADPCM(u8* data, size_t sample_count, u16 adpcm_ps, s16 adpcm_yn[2], std::array<u8, 16> adpcm_coeff);
static const int BASE_SAMPLE_RATE = 22050; static const int BASE_SAMPLE_RATE = 22050;
struct Buffer { struct Buffer {
@ -63,8 +65,9 @@ namespace Audio {
return 0; return 0;
} }
ALuint source, buffer; ALuint silencebuffer;
ALCint dev_rate; ALCint dev_rate;
std::array<u8, 10000> silence;
void Init() { void Init() {
InitAL(); InitAL();
@ -92,21 +95,51 @@ namespace Audio {
} }
void EnqueueBuffer(int chanid, u16 buffer_id, void EnqueueBuffer(int chanid, u16 buffer_id,
void* data, int sample_count, void* data, int sample_count,
bool has_adpcm, u16 adpcm_ps, s16 adpcm_yn[2], bool has_adpcm, u16 adpcm_ps, s16 adpcm_yn[2],
bool is_looping) { bool is_looping) {
if (chans[chanid].format != FORMAT_PCM16) { if (is_looping) {
LOG_ERROR(Audio, "Unimplemented format"); LOG_WARNING(Audio, "Looped buffers are unimplemented");
return;
} }
// TODO: ADPCM processing should happen here
ALuint b; ALuint b;
alGenBuffers(1, &b); alGenBuffers(1, &b);
alBufferData(b, AL_FORMAT_MONO16, data, sample_count*2, BASE_SAMPLE_RATE);
if (alGetError() != AL_NO_ERROR) LOG_CRITICAL(Audio, "Failed to init buffer"); if (chans[chanid].format == FORMAT_PCM16) {
switch (chans[chanid].mono_or_stereo) {
case 2:
alBufferData(b, AL_FORMAT_STEREO16, data, sample_count * 4, BASE_SAMPLE_RATE);
break;
case 1:
default:
alBufferData(b, AL_FORMAT_MONO16, data, sample_count * 2, BASE_SAMPLE_RATE);
break;
}
if (alGetError() != AL_NO_ERROR) LOG_CRITICAL(Audio, "Failed to init buffer");
} else if (chans[chanid].format == FORMAT_PCM8) {
switch (chans[chanid].mono_or_stereo) {
case 2:
alBufferData(b, AL_FORMAT_STEREO8, data, sample_count * 2, BASE_SAMPLE_RATE);
break;
case 1:
default:
alBufferData(b, AL_FORMAT_MONO8, data, sample_count * 1, BASE_SAMPLE_RATE);
break;
}
if (alGetError() != AL_NO_ERROR) LOG_CRITICAL(Audio, "Failed to init buffer");
} /*else if (chans[chanid].format == FORMAT_ADPCM) {
if (chans[chanid].mono_or_stereo != 1) {
LOG_ERROR(Audio, "Being fed non-mono ADPCM");
}
std::vector<u16> decoded = DecodeADPCM(data, sample_count, adpcm_ps, adpcm_yn, chans[chanid].adpcm_coeff);
alBufferData(b, AL_FORMAT_MONO16, decoded.data(), decoded.size() * 2, BASE_SAMPLE_RATE);
if (alGetError() != AL_NO_ERROR) LOG_CRITICAL(Audio, "Failed to init buffer");
}*/ else {
LOG_ERROR(Audio, "Unrecognised audio format in buffer 0x%04x (size: %i samples)", buffer_id, sample_count);
alBufferData(b, AL_FORMAT_MONO8, silence.data(), silence.size(), BASE_SAMPLE_RATE);
if (alGetError() != AL_NO_ERROR) LOG_CRITICAL(Audio, "Failed to init buffer");
}
chans[chanid].queue.emplace( Buffer { buffer_id, b, is_looping }); chans[chanid].queue.emplace( Buffer { buffer_id, b, is_looping });
} }
@ -119,7 +152,7 @@ namespace Audio {
alSourceQueueBuffers(c.source, 1, &c.queue.top().buffer); alSourceQueueBuffers(c.source, 1, &c.queue.top().buffer);
if (alGetError() != AL_NO_ERROR) LOG_CRITICAL(Audio, "Failed to enqueue buffer"); if (alGetError() != AL_NO_ERROR) LOG_CRITICAL(Audio, "Failed to enqueue buffer");
c.playing.emplace(c.queue.top()); c.playing.emplace(c.queue.top());
LOG_INFO(Audio, "Enqueued buffer id %i", c.queue.top().id); LOG_INFO(Audio, "Enqueued buffer id 0x%04x", c.queue.top().id);
c.queue.pop(); c.queue.pop();
} }
@ -141,7 +174,7 @@ namespace Audio {
alSourceUnqueueBuffers(c.source, 1, &buf); alSourceUnqueueBuffers(c.source, 1, &buf);
processed--; processed--;
LOG_INFO(Audio, "Finished buffer id %i", c.playing.front().id); LOG_INFO(Audio, "Finished buffer id 0x%04x", c.playing.front().id);
while (!c.playing.empty() && c.playing.front().buffer != buf) { while (!c.playing.empty() && c.playing.front().buffer != buf) {
c.playing.pop(); c.playing.pop();

2
src/core/hle/service/dsp_dsp.cpp
View File

@ -200,7 +200,7 @@ static void AudioTick(u64, int cycles_late) {
if (TestAndUnsetBit(ctx.dirty, 16)) { if (TestAndUnsetBit(ctx.dirty, 16)) {
// Is Active? // Is Active?
LOG_WARNING(Service_DSP, "Unimplemented dirty bit 16"); //LOG_WARNING(Service_DSP, "Unimplemented dirty bit 16");
} }
if (TestAndUnsetBit(ctx.dirty, 2)) { if (TestAndUnsetBit(ctx.dirty, 2)) {