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Try to simplify audio backend code a bit

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This commit is contained in:
UnknownShadow200 2024-03-25 20:33:25 +11:00
parent e22be93ab3
commit b2fdf80be1
3 changed files with 64 additions and 97 deletions
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2
src/Audio.c
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@ -351,7 +351,7 @@ static cc_result Music_PlayOgg(struct Stream* source) {
channels = vorbis.channels;
sampleRate = vorbis.sampleRate;
if ((res = Audio_SetFormat(&music_ctx, channels, sampleRate))) goto cleanup;
if ((res = Audio_SetFormat(&music_ctx, channels, sampleRate, 100))) goto cleanup;
/* largest possible vorbis frame decodes to blocksize1 * channels samples, */
/* so can end up decoding slightly over a second of audio */

4
src/Audio.h
View File

@ -51,7 +51,7 @@ cc_result Audio_Init(struct AudioContext* ctx, int buffers);
void Audio_Close(struct AudioContext* ctx);
/* Sets the format of the audio data to be played. */
/* NOTE: Changing the format can be expensive, depending on the backend. */
cc_result Audio_SetFormat(struct AudioContext* ctx, int channels, int sampleRate);
cc_result Audio_SetFormat(struct AudioContext* ctx, int channels, int sampleRate, int playbackRate);
/* Sets the volume audio data is played at */
void Audio_SetVolume(struct AudioContext* ctx, int volume);
/* Queues the given audio chunk for playing. */
@ -66,8 +66,6 @@ cc_result Audio_Play(struct AudioContext* ctx);
cc_result Audio_Poll(struct AudioContext* ctx, int* inUse);
cc_result Audio_Pause(struct AudioContext* ctx); /* Only implemented with OpenSL ES backend */
/* Plays the given audio data */
cc_result Audio_PlayData(struct AudioContext* ctx, struct AudioData* data);
/* Whether the given audio data can be played without recreating the underlying audio device */
cc_bool Audio_FastPlay(struct AudioContext* ctx, struct AudioData* data);
/* Outputs more detailed information about errors with audio. */

155
src/AudioBackend.c
View File

@ -12,12 +12,12 @@ void Audio_Warn(cc_result res, const char* action) {
/* Common/Base methods */
static void AudioBase_Clear(struct AudioContext* ctx);
static cc_bool AudioBase_AdjustSound(struct AudioContext* ctx, struct AudioData* data);
static cc_bool AudioBase_AdjustSound(struct AudioContext* ctx, void* data, cc_uint32 size);
static void AudioBase_AllocChunks(int size, void** chunks, int numChunks);
static void AudioBase_FreeChunks(void** chunks, int numChunks);
/* achieve higher speed by playing samples at higher sample rate */
#define Audio_AdjustSampleRate(data) ((data->sampleRate * data->rate) / 100)
#define Audio_AdjustSampleRate(sampleRate, playbackRate) ((sampleRate * playbackRate) / 100)
#if defined CC_BUILD_OPENAL
/*########################################################################################################################*
@ -205,8 +205,8 @@ void Audio_Close(struct AudioContext* ctx) {
ctx->count = 0;
}
cc_result Audio_SetFormat(struct AudioContext* ctx, int channels, int sampleRate) {
ctx->sampleRate = sampleRate;
cc_result Audio_SetFormat(struct AudioContext* ctx, int channels, int sampleRate, int playbackRate) {
ctx->sampleRate = Audio_AdjustSampleRate(sampleRate, playbackRate);
if (channels == 1) {
ctx->format = AL_FORMAT_MONO16;
@ -269,16 +269,6 @@ cc_bool Audio_FastPlay(struct AudioContext* ctx, struct AudioData* data) {
return true;
}
cc_result Audio_PlayData(struct AudioContext* ctx, struct AudioData* data) {
cc_result res;
data->sampleRate = Audio_AdjustSampleRate(data);
if ((res = Audio_SetFormat(ctx, data->channels, data->sampleRate))) return res;
if ((res = Audio_QueueChunk(ctx, data->data, data->size))) return res;
if ((res = Audio_Play(ctx))) return res;
return 0;
}
static const char* GetError(cc_result res) {
switch (res) {
case AL_ERR_INIT_CONTEXT: return "Failed to init OpenAL context";
@ -408,11 +398,12 @@ void Audio_Close(struct AudioContext* ctx) {
AudioBase_Clear(ctx);
}
cc_result Audio_SetFormat(struct AudioContext* ctx, int channels, int sampleRate) {
cc_result Audio_SetFormat(struct AudioContext* ctx, int channels, int sampleRate, int playbackRate) {
WAVEFORMATEX fmt;
cc_result res;
int sampleSize;
sampleRate = Audio_AdjustSampleRate(sampleRate, playbackRate);
if (ctx->channels == channels && ctx->sampleRate == sampleRate) return 0;
ctx->channels = channels;
ctx->sampleRate = sampleRate;
@ -443,13 +434,16 @@ cc_result Audio_QueueChunk(struct AudioContext* ctx, void* chunk, cc_uint32 data
WAVEHDR* hdr;
int i;
cc_bool ok = AudioBase_AdjustSound(ctx, chunk, dataSize);
if (!ok) return ERR_OUT_OF_MEMORY;
for (i = 0; i < ctx->count; i++) {
hdr = &ctx->headers[i];
if (!(hdr->dwFlags & WHDR_DONE)) continue;
Mem_Set(hdr, 0, sizeof(WAVEHDR));
hdr->lpData = (LPSTR)chunk;
hdr->dwBufferLength = dataSize;
hdr->lpData = (LPSTR)ctx->_tmpData;
hdr->dwBufferLength = ctx->_tmpSize;
hdr->dwLoops = 1;
if ((res = waveOutPrepareHeader(ctx->handle, hdr, sizeof(WAVEHDR)))) return res;
@ -483,21 +477,10 @@ cc_result Audio_Poll(struct AudioContext* ctx, int* inUse) {
cc_bool Audio_FastPlay(struct AudioContext* ctx, struct AudioData* data) {
int channels = data->channels;
int sampleRate = Audio_AdjustSampleRate(data);
int sampleRate = Audio_AdjustSampleRate(data->sampleRate, data->rate);
return !ctx->channels || (ctx->channels == channels && ctx->sampleRate == sampleRate);
}
cc_result Audio_PlayData(struct AudioContext* ctx, struct AudioData* data) {
cc_bool ok = AudioBase_AdjustSound(ctx, data);
cc_result res;
if (!ok) return ERR_OUT_OF_MEMORY;
data->sampleRate = Audio_AdjustSampleRate(data);
if ((res = Audio_SetFormat(ctx, data->channels, data->sampleRate))) return res;
if ((res = Audio_QueueChunk(ctx, data->data, data->size))) return res;
return 0;
}
cc_bool Audio_DescribeError(cc_result res, cc_string* dst) {
char buffer[NATIVE_STR_LEN] = { 0 };
waveOutGetErrorTextA(res, buffer, NATIVE_STR_LEN);
@ -634,7 +617,7 @@ void Audio_Close(struct AudioContext* ctx) {
ctx->sampleRate = 0;
}
cc_result Audio_SetFormat(struct AudioContext* ctx, int channels, int sampleRate) {
cc_result Audio_SetFormat(struct AudioContext* ctx, int channels, int sampleRate, int playbackRate) {
SLDataLocator_AndroidSimpleBufferQueue input;
SLDataLocator_OutputMix output;
SLObjectItf playerObject;
@ -645,6 +628,9 @@ cc_result Audio_SetFormat(struct AudioContext* ctx, int channels, int sampleRate
SLDataSink dst;
cc_result res;
/* rate is in milli, so 1000 = normal rate */
if ((res = (*ctx->playerRate)->SetRate(ctx->playerRate, playbackRate * 10))) return res;
if (ctx->channels == channels && ctx->sampleRate == sampleRate) return 0;
ctx->channels = channels;
ctx->sampleRate = sampleRate;
@ -721,18 +707,6 @@ cc_bool Audio_FastPlay(struct AudioContext* ctx, struct AudioData* data) {
return !ctx->channels || (ctx->channels == data->channels && ctx->sampleRate == data->sampleRate);
}
cc_result Audio_PlayData(struct AudioContext* ctx, struct AudioData* data) {
cc_result res;
if ((res = Audio_SetFormat(ctx, data->channels, data->sampleRate))) return res;
/* rate is in milli, so 1000 = normal rate */
if ((res = (*ctx->playerRate)->SetRate(ctx->playerRate, data->rate * 10))) return res;
if ((res = Audio_QueueChunk(ctx, data->data, data->size))) return res;
if ((res = Audio_Play(ctx))) return res;
return 0;
}
static const char* GetError(cc_result res) {
switch (res) {
case SL_RESULT_PRECONDITIONS_VIOLATED: return "Preconditions violated";
@ -820,10 +794,11 @@ void Audio_Close(struct AudioContext* ctx) {
ctx->count = 0;
}
cc_result Audio_SetFormat(struct AudioContext* ctx, int channels, int sampleRate) {
cc_result Audio_SetFormat(struct AudioContext* ctx, int channels, int sampleRate, int playbackRate) {
ctx->stereo = (channels == 2);
int fmt = ctx->stereo ? NDSP_FORMAT_STEREO_PCM16 : NDSP_FORMAT_MONO_PCM16;
sampleRate = Audio_AdjustSampleRate(sampleRate, playbackRate);
ndspChnSetFormat(ctx->chanID, fmt);
ndspChnSetRate(ctx->chanID, sampleRate);
return 0;
@ -890,15 +865,6 @@ cc_bool Audio_FastPlay(struct AudioContext* ctx, struct AudioData* data) {
return true;
}
cc_result Audio_PlayData(struct AudioContext* ctx, struct AudioData* data) {
data->sampleRate = Audio_AdjustSampleRate(data);
cc_result res;
if ((res = Audio_SetFormat(ctx, data->channels, data->sampleRate))) return res;
if ((res = Audio_QueueChunk(ctx, data->data, data->size))) return res;
return 0;
}
cc_bool Audio_DescribeError(cc_result res, cc_string* dst) {
return false;
}
@ -1008,7 +974,8 @@ void Audio_Close(struct AudioContext* ctx) {
ctx->count = 0;
}
cc_result Audio_SetFormat(struct AudioContext* ctx, int channels, int sampleRate) {
cc_result Audio_SetFormat(struct AudioContext* ctx, int channels, int sampleRate, int playbackRate) {
sampleRate = Audio_AdjustSampleRate(sampleRate, playbackRate);
ctx->channels = channels;
ctx->sampleRate = sampleRate;
@ -1020,8 +987,7 @@ cc_result Audio_SetFormat(struct AudioContext* ctx, int channels, int sampleRate
// mono
audrvVoiceSetMixFactor(&drv, ctx->chanID, 1.0f, 0, 0);
audrvVoiceSetMixFactor(&drv, ctx->chanID, 1.0f, 0, 1);
}
else {
} else {
// stereo
audrvVoiceSetMixFactor(&drv, ctx->chanID, 1.0f, 0, 0);
audrvVoiceSetMixFactor(&drv, ctx->chanID, 0.0f, 0, 1);
@ -1099,17 +1065,6 @@ cc_bool Audio_FastPlay(struct AudioContext* ctx, struct AudioData* data) {
return true;
}
cc_result Audio_PlayData(struct AudioContext* ctx, struct AudioData* data) {
data->sampleRate = Audio_AdjustSampleRate(data);
cc_result res;
if ((res = Audio_SetFormat(ctx, data->channels, data->sampleRate))) return res;
if ((res = Audio_QueueChunk(ctx, data->data, data->size))) return res;
if ((res = Audio_Play(ctx))) return res;
return 0;
}
cc_bool Audio_DescribeError(cc_result res, cc_string* dst) {
return false;
}
@ -1229,7 +1184,8 @@ void Audio_Close(struct AudioContext* ctx) {
ctx->count = 0;
}
cc_result Audio_SetFormat(struct AudioContext* ctx, int channels, int sampleRate) {
cc_result Audio_SetFormat(struct AudioContext* ctx, int channels, int sampleRate, int playbackRate) {
sampleRate = Audio_AdjustSampleRate(sampleRate, playbackRate);
ctx->channels = channels;
ctx->sampleRate = sampleRate;
return 0;
@ -1279,16 +1235,6 @@ cc_bool Audio_FastPlay(struct AudioContext* ctx, struct AudioData* data) {
return true;
}
cc_result Audio_PlayData(struct AudioContext* ctx, struct AudioData* data) {
data->sampleRate = Audio_AdjustSampleRate(data);
cc_result res;
if ((res = Audio_SetFormat(ctx, data->channels, data->sampleRate))) return res;
if ((res = Audio_QueueChunk(ctx, data->data, data->size))) return res;
if ((res = Audio_Play(ctx))) return res;
return 0;
}
cc_bool Audio_DescribeError(cc_result res, cc_string* dst) {
return false;
}
@ -1310,7 +1256,7 @@ void Audio_FreeChunks(void** chunks, int numChunks) {
/*########################################################################################################################*
*-----------------------------------------------------WebAudio backend----------------------------------------------------*
*#########################################################################################################################*/
struct AudioContext { int contextID, count; };
struct AudioContext { int contextID, count, rate; void* data; };
#define AUDIO_COMMON_ALLOC
extern int interop_InitAudio(void);
@ -1333,6 +1279,8 @@ void AudioBackend_Free(void) { }
cc_result Audio_Init(struct AudioContext* ctx, int buffers) {
ctx->count = buffers;
ctx->contextID = interop_AudioCreate();
ctx->data = NULL;
ctx->rate = 100;
return 0;
}
@ -1342,8 +1290,8 @@ void Audio_Close(struct AudioContext* ctx) {
ctx->count = 0;
}
cc_result Audio_SetFormat(struct AudioContext* ctx, int channels, int sampleRate) {
return ERR_NOT_SUPPORTED;
cc_result Audio_SetFormat(struct AudioContext* ctx, int channels, int sampleRate, int playbackRate) {
ctx->rate = playbackRate; return 0;
}
void Audio_SetVolume(struct AudioContext* ctx, int volume) {
@ -1351,10 +1299,12 @@ void Audio_SetVolume(struct AudioContext* ctx, int volume) {
}
cc_result Audio_QueueChunk(struct AudioContext* ctx, void* chunk, cc_uint32 size) {
return ERR_NOT_SUPPORTED;
ctx->data = chunk; return 0;
}
cc_result Audio_Play(struct AudioContext* ctx) { return ERR_NOT_SUPPORTED; }
cc_result Audio_Play(struct AudioContext* ctx) {
return interop_AudioPlay(ctx->contextID, ctx->data, ctx->rate);
}
cc_result Audio_Poll(struct AudioContext* ctx, int* inUse) {
return interop_AudioPoll(ctx->contextID, inUse);
@ -1365,10 +1315,6 @@ cc_bool Audio_FastPlay(struct AudioContext* ctx, struct AudioData* data) {
return true;
}
cc_result Audio_PlayData(struct AudioContext* ctx, struct AudioData* data) {
return interop_AudioPlay(ctx->contextID, data->data, data->rate);
}
cc_bool Audio_DescribeError(cc_result res, cc_string* dst) {
char buffer[NATIVE_STR_LEN];
int len = interop_AudioDescribe(res, buffer, NATIVE_STR_LEN);
@ -1400,12 +1346,26 @@ cc_result Audio_Init(struct AudioContext* ctx, int buffers) {
void Audio_Close(struct AudioContext* ctx) { }
cc_bool Audio_FastPlay(struct AudioContext* ctx, struct AudioData* data) { return true; }
cc_result Audio_PlayData(struct AudioContext* ctx, struct AudioData* data) {
cc_result Audio_SetFormat(struct AudioContext* ctx, int channels, int sampleRate, int playbackRate) {
return ERR_NOT_SUPPORTED;
}
void Audio_SetVolume(struct AudioContext* ctx, int volume) { }
cc_result Audio_QueueChunk(struct AudioContext* ctx, void* chunk, cc_uint32 size) {
return ERR_NOT_SUPPORTED;
}
cc_result Audio_Play(struct AudioContext* ctx) {
return ERR_NOT_SUPPORTED;
}
cc_result Audio_Poll(struct AudioContext* ctx, int* inUse) {
return ERR_NOT_SUPPORTED;
}
cc_bool Audio_FastPlay(struct AudioContext* ctx, struct AudioData* data) { return false; }
cc_bool Audio_DescribeError(cc_result res, cc_string* dst) { return false; }
#endif
@ -1444,9 +1404,13 @@ static void AudioBase_Clear(struct AudioContext* ctx) {
ctx->_tmpSize = 0;
}
static cc_bool AudioBase_AdjustSound(struct AudioContext* ctx, struct AudioData* data) {
static cc_bool AudioBase_AdjustSound(struct AudioContext* ctx, void* data, cc_uint32 size) {
void* audio;
if (ctx->volume >= 100) return true;
if (ctx->volume >= 100) {
ctx->_tmpData = data;
ctx->_tmpSize = size;
return true;
}
/* copy to temp buffer to apply volume */
if (ctx->_tmpSize < data->size) {
@ -1496,8 +1460,13 @@ static struct AudioContext context_pool[POOL_MAX_CONTEXTS];
#ifndef CC_BUILD_NOSOUNDS
static cc_result PlayAudio(struct AudioContext* ctx, struct AudioData* data) {
cc_result res;
Audio_SetVolume(ctx, data->volume);
return Audio_PlayData(ctx, data);
if ((res = Audio_SetFormat(ctx, data->channels, data->sampleRate, data->rate))) return res;
if ((res = Audio_QueueChunk(ctx, data->data, data->size))) return res;
if ((res = Audio_Play(ctx))) return res;
return 0;
}
cc_result AudioPool_Play(struct AudioData* data) {