mirror of
https://github.com/ClassiCube/ClassiCube.git
synced 2025-09-13 09:35:23 -04:00
Refactor audio backends to allow explicitly setting volume
This commit is contained in:
UnknownShadow200
parent
e06f54e61b
commit
e22be93ab3
13
src/Audio.c
13
src/Audio.c
@ -329,7 +329,6 @@ static cc_result Music_Buffer(cc_int16* data, int maxSamples, struct VorbisState
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cur = &data[samples];
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samples += Vorbis_OutputFrame(ctx, cur);
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}
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if (Audio_MusicVolume < 100) { Audio_ApplyVolume(data, samples, Audio_MusicVolume); }
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res2 = Audio_QueueChunk(&music_ctx, data, samples * 2);
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if (res2) { music_stopping = true; return res2; }
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@ -339,7 +338,7 @@ static cc_result Music_Buffer(cc_int16* data, int maxSamples, struct VorbisState
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static cc_result Music_PlayOgg(struct Stream* source) {
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struct OggState ogg;
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struct VorbisState vorbis = { 0 };
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int channels, sampleRate;
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int channels, sampleRate, volume;
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int chunkSize, samplesPerSecond;
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void* chunks[AUDIO_MAX_BUFFERS] = { 0 };
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@ -366,7 +365,9 @@ static cc_result Music_PlayOgg(struct Stream* source) {
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res = ERR_OUT_OF_MEMORY; goto cleanup;
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}
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volume = Audio_MusicVolume;
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Audio_SetVolume(&music_ctx, volume);
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/* fill up with some samples before playing */
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for (i = 0; i < AUDIO_MAX_BUFFERS && !res; i++)
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@ -391,6 +392,10 @@ static cc_result Music_PlayOgg(struct Stream* source) {
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Audio_Play(&music_ctx);
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}
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#endif
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if (volume != Audio_MusicVolume) {
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volume = Audio_MusicVolume;
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Audio_SetVolume(&music_ctx, volume);
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}
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res = Audio_Poll(&music_ctx, &inUse);
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if (res) { music_stopping = true; break; }
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@ -550,4 +555,4 @@ static void OnFree(void) {
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struct IGameComponent Audio_Component = {
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OnInit, /* Init */
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OnFree /* Free */
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};
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};
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@ -52,6 +52,8 @@ void Audio_Close(struct AudioContext* ctx);
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/* Sets the format of the audio data to be played. */
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/* NOTE: Changing the format can be expensive, depending on the backend. */
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cc_result Audio_SetFormat(struct AudioContext* ctx, int channels, int sampleRate);
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/* Sets the volume audio data is played at */
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void Audio_SetVolume(struct AudioContext* ctx, int volume);
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/* Queues the given audio chunk for playing. */
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/* NOTE: You MUST ensure Audio_Poll indicates a buffer is free before calling this. */
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/* NOTE: Some backends directly read from the data - therefore you MUST NOT modify it */
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@ -77,7 +79,6 @@ void Audio_AllocChunks(cc_uint32 size, void** chunks, int numChunks);
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void Audio_FreeChunks(void** chunks, int numChunks);
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extern struct AudioContext music_ctx;
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void Audio_ApplyVolume(cc_int16* samples, int count, int volume);
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void Audio_Warn(cc_result res, const char* action);
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cc_result AudioPool_Play(struct AudioData* data);
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@ -6,27 +6,6 @@
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#include "Utils.h"
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#include "Platform.h"
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void Audio_ApplyVolume(cc_int16* samples, int count, int volume) {
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int i;
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for (i = 0; i < (count & ~0x07); i += 8, samples += 8) {
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samples[0] = (samples[0] * volume / 100);
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samples[1] = (samples[1] * volume / 100);
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samples[2] = (samples[2] * volume / 100);
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samples[3] = (samples[3] * volume / 100);
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samples[4] = (samples[4] * volume / 100);
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samples[5] = (samples[5] * volume / 100);
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samples[6] = (samples[6] * volume / 100);
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samples[7] = (samples[7] * volume / 100);
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}
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for (; i < count; i++, samples++) {
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samples[0] = (samples[0] * volume / 100);
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}
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}
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void Audio_Warn(cc_result res, const char* action) {
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Logger_Warn(res, action, Audio_DescribeError);
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}
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@ -239,6 +218,11 @@ cc_result Audio_SetFormat(struct AudioContext* ctx, int channels, int sampleRate
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return 0;
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}
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void Audio_SetVolume(struct AudioContext* ctx, int volume) {
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_alSourcef(ctx->source, AL_GAIN, volume / 100.0f);
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_alGetError(); /* Reset error state */
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}
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cc_result Audio_QueueChunk(struct AudioContext* ctx, void* chunk, cc_uint32 size) {
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ALuint buffer;
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ALenum err;
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@ -286,11 +270,8 @@ cc_bool Audio_FastPlay(struct AudioContext* ctx, struct AudioData* data) {
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}
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cc_result Audio_PlayData(struct AudioContext* ctx, struct AudioData* data) {
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cc_result res;
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cc_result res;
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data->sampleRate = Audio_AdjustSampleRate(data);
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_alSourcef(ctx->source, AL_GAIN, data->volume / 100.0f);
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_alGetError(); /* Reset error state */
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if ((res = Audio_SetFormat(ctx, data->channels, data->sampleRate))) return res;
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if ((res = Audio_QueueChunk(ctx, data->data, data->size))) return res;
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@ -384,7 +365,7 @@ WINMMAPI UINT WINAPI waveOutGetNumDevs(void);
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struct AudioContext {
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HWAVEOUT handle;
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WAVEHDR headers[AUDIO_MAX_BUFFERS];
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int count, channels, sampleRate;
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int count, channels, sampleRate, volume;
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cc_uint32 _tmpSize; void* _tmpData;
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};
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#define AUDIO_COMMON_VOLUME
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@ -399,7 +380,8 @@ cc_result Audio_Init(struct AudioContext* ctx, int buffers) {
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for (i = 0; i < buffers; i++) {
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ctx->headers[i].dwFlags = WHDR_DONE;
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}
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ctx->count = buffers;
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ctx->count = buffers;
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ctx->volume = 100;
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return 0;
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}
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@ -454,6 +436,8 @@ cc_result Audio_SetFormat(struct AudioContext* ctx, int channels, int sampleRate
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return res;
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}
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void Audio_SetVolume(struct AudioContext* ctx, int volume) { ctx->volume = volume; }
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cc_result Audio_QueueChunk(struct AudioContext* ctx, void* chunk, cc_uint32 dataSize) {
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cc_result res = 0;
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WAVEHDR* hdr;
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@ -536,19 +520,19 @@ void Audio_FreeChunks(void** chunks, int numChunks) {
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*#########################################################################################################################*/
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#include <SLES/OpenSLES.h>
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#include <SLES/OpenSLES_Android.h>
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#include "ExtMath.h"
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static SLObjectItf slEngineObject;
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static SLEngineItf slEngineEngine;
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static SLObjectItf slOutputObject;
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struct AudioContext {
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int count, channels, sampleRate;
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SLObjectItf playerObject;
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SLPlayItf playerPlayer;
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SLBufferQueueItf playerQueue;
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SLObjectItf playerObject;
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SLPlayItf playerPlayer;
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SLBufferQueueItf playerQueue;
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SLPlaybackRateItf playerRate;
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cc_uint32 _tmpSize; void* _tmpData;
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SLVolumeItf playerVolume;
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};
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#define AUDIO_COMMON_VOLUME
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#define AUDIO_COMMON_ALLOC
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static SLresult (SLAPIENTRY *_slCreateEngine)(SLObjectItf* engine, SLuint32 numOptions, const SLEngineOption* engineOptions,
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@ -558,13 +542,15 @@ static SLInterfaceID* _SL_IID_PLAY;
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static SLInterfaceID* _SL_IID_ENGINE;
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static SLInterfaceID* _SL_IID_BUFFERQUEUE;
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static SLInterfaceID* _SL_IID_PLAYBACKRATE;
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static SLInterfaceID* _SL_IID_VOLUME;
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static const cc_string slLib = String_FromConst("libOpenSLES.so");
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static cc_bool LoadSLFuncs(void) {
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static const struct DynamicLibSym funcs[] = {
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DynamicLib_Sym(slCreateEngine), DynamicLib_Sym(SL_IID_NULL),
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DynamicLib_Sym(SL_IID_PLAY), DynamicLib_Sym(SL_IID_ENGINE),
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DynamicLib_Sym(SL_IID_BUFFERQUEUE), DynamicLib_Sym(SL_IID_PLAYBACKRATE)
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DynamicLib_Sym(SL_IID_BUFFERQUEUE), DynamicLib_Sym(SL_IID_PLAYBACKRATE),
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DynamicLib_Sym(SL_IID_VOLUME)
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};
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void* lib;
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@ -581,8 +567,7 @@ cc_bool AudioBackend_Init(void) {
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if (!LoadSLFuncs()) { Logger_WarnFunc(&msg); return false; }
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/* mixer doesn't use any effects */
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ids[0] = *_SL_IID_NULL;
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req[0] = SL_BOOLEAN_FALSE;
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ids[0] = *_SL_IID_NULL; req[0] = SL_BOOLEAN_FALSE;
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res = _slCreateEngine(&slEngineObject, 0, NULL, 0, NULL, NULL);
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if (res) { Audio_Warn(res, "creating OpenSL ES engine"); return false; }
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@ -637,12 +622,16 @@ static void Audio_Reset(struct AudioContext* ctx) {
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ctx->playerPlayer = NULL;
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ctx->playerQueue = NULL;
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ctx->playerRate = NULL;
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ctx->playerVolume = NULL;
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}
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void Audio_Close(struct AudioContext* ctx) {
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Audio_Stop(ctx);
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Audio_Reset(ctx);
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AudioBase_Clear(ctx);
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ctx->count = 0;
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ctx->channels = 0;
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ctx->sampleRate = 0;
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}
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cc_result Audio_SetFormat(struct AudioContext* ctx, int channels, int sampleRate) {
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@ -650,8 +639,8 @@ cc_result Audio_SetFormat(struct AudioContext* ctx, int channels, int sampleRate
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SLDataLocator_OutputMix output;
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SLObjectItf playerObject;
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SLDataFormat_PCM fmt;
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SLInterfaceID ids[3];
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SLboolean req[3];
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SLInterfaceID ids[4];
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SLboolean req[4];
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SLDataSource src;
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SLDataSink dst;
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cc_result res;
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@ -682,8 +671,9 @@ cc_result Audio_SetFormat(struct AudioContext* ctx, int channels, int sampleRate
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ids[0] = *_SL_IID_BUFFERQUEUE; req[0] = SL_BOOLEAN_TRUE;
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ids[1] = *_SL_IID_PLAY; req[1] = SL_BOOLEAN_TRUE;
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ids[2] = *_SL_IID_PLAYBACKRATE; req[2] = SL_BOOLEAN_TRUE;
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ids[3] = *_SL_IID_VOLUME; req[3] = SL_BOOLEAN_TRUE;
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res = (*slEngineEngine)->CreateAudioPlayer(slEngineEngine, &playerObject, &src, &dst, 3, ids, req);
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res = (*slEngineEngine)->CreateAudioPlayer(slEngineEngine, &playerObject, &src, &dst, 4, ids, req);
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ctx->playerObject = playerObject;
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if (res) return res;
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@ -691,9 +681,19 @@ cc_result Audio_SetFormat(struct AudioContext* ctx, int channels, int sampleRate
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if ((res = (*playerObject)->GetInterface(playerObject, *_SL_IID_PLAY, &ctx->playerPlayer))) return res;
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if ((res = (*playerObject)->GetInterface(playerObject, *_SL_IID_BUFFERQUEUE, &ctx->playerQueue))) return res;
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if ((res = (*playerObject)->GetInterface(playerObject, *_SL_IID_PLAYBACKRATE, &ctx->playerRate))) return res;
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if ((res = (*playerObject)->GetInterface(playerObject, *_SL_IID_VOLUME, &ctx->playerVolume))) return res;
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return 0;
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}
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static float Log10(float volume) { return Math_Log(volume) / Math_Log(10); }
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void Audio_SetVolume(struct AudioContext* ctx, int volume) {
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// log of 0 is undefined
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SLmillibel attenuation = volume == 0 ? SL_MILLIBEL_MIN : (2000 * Log10(volume / 100.0f));
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(*ctx->playerVolume)->SetVolumeLevel(ctx->playerVolume, attenuation);
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}
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cc_result Audio_QueueChunk(struct AudioContext* ctx, void* chunk, cc_uint32 size) {
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return (*ctx->playerQueue)->Enqueue(ctx->playerQueue, chunk, size);
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}
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@ -722,9 +722,7 @@ cc_bool Audio_FastPlay(struct AudioContext* ctx, struct AudioData* data) {
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}
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cc_result Audio_PlayData(struct AudioContext* ctx, struct AudioData* data) {
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cc_bool ok = AudioBase_AdjustSound(ctx, data);
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cc_result res;
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if (!ok) return ERR_OUT_OF_MEMORY;
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if ((res = Audio_SetFormat(ctx, data->channels, data->sampleRate))) return res;
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/* rate is in milli, so 1000 = normal rate */
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@ -831,6 +829,14 @@ cc_result Audio_SetFormat(struct AudioContext* ctx, int channels, int sampleRate
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return 0;
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}
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void Audio_SetVolume(struct AudioContext* ctx, int volume) {
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float mix[12] = { 0 };
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mix[0] = volume / 100.0f;
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mix[1] = volume / 100.0f;
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ndspChnSetMix(ctx->chanID, mix);
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}
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cc_result Audio_QueueChunk(struct AudioContext* ctx, void* chunk, cc_uint32 dataSize) {
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ndspWaveBuf* buf;
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@ -885,11 +891,6 @@ cc_bool Audio_FastPlay(struct AudioContext* ctx, struct AudioData* data) {
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}
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cc_result Audio_PlayData(struct AudioContext* ctx, struct AudioData* data) {
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float mix[12] = { 0 };
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mix[0] = data->volume / 100.0f;
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mix[1] = data->volume / 100.0f;
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ndspChnSetMix(ctx->chanID, mix);
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data->sampleRate = Audio_AdjustSampleRate(data);
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cc_result res;
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@ -1031,6 +1032,10 @@ cc_result Audio_SetFormat(struct AudioContext* ctx, int channels, int sampleRate
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return 0;
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}
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void Audio_SetVolume(struct AudioContext* ctx, int volume) {
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audrvVoiceSetVolume(&drv, ctx->chanID, volume / 100.0f);
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}
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cc_result Audio_QueueChunk(struct AudioContext* ctx, void* chunk, cc_uint32 dataSize) {
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AudioDriverWaveBuf* buf;
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@ -1090,7 +1095,6 @@ cc_result Audio_Poll(struct AudioContext* ctx, int* inUse) {
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return 0;
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}
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cc_bool Audio_FastPlay(struct AudioContext* ctx, struct AudioData* data) {
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return true;
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}
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@ -1102,9 +1106,7 @@ cc_result Audio_PlayData(struct AudioContext* ctx, struct AudioData* data) {
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if ((res = Audio_SetFormat(ctx, data->channels, data->sampleRate))) return res;
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if ((res = Audio_QueueChunk(ctx, data->data, data->size))) return res;
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audrvVoiceSetVolume(&drv, ctx->chanID, data->volume/100.f);
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if ((res = Audio_Play(ctx))) return res;
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return 0;
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}
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@ -1233,6 +1235,10 @@ cc_result Audio_SetFormat(struct AudioContext* ctx, int channels, int sampleRate
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return 0;
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}
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void Audio_SetVolume(struct AudioContext* ctx, int volume) {
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snd_stream_volume(ctx->hnd, volume);
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}
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cc_result Audio_QueueChunk(struct AudioContext* ctx, void* chunk, cc_uint32 dataSize) {
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struct AudioBuffer* buf;
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@ -1274,7 +1280,6 @@ cc_bool Audio_FastPlay(struct AudioContext* ctx, struct AudioData* data) {
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}
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cc_result Audio_PlayData(struct AudioContext* ctx, struct AudioData* data) {
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snd_stream_volume(ctx->hnd, data->volume);
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data->sampleRate = Audio_AdjustSampleRate(data);
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cc_result res;
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@ -1311,8 +1316,9 @@ struct AudioContext { int contextID, count; };
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extern int interop_InitAudio(void);
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extern int interop_AudioCreate(void);
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extern void interop_AudioClose(int contextID);
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extern int interop_AudioPlay(int contextID, const void* name, int volume, int rate);
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extern int interop_AudioPoll(int contetID, int* inUse);
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extern int interop_AudioPlay(int contextID, const void* name, int rate);
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extern int interop_AudioPoll(int contextID, int* inUse);
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extern int interop_AudioVolume(int contextID, int volume);
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extern int interop_AudioDescribe(int res, char* buffer, int bufferLen);
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cc_bool AudioBackend_Init(void) {
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@ -1339,9 +1345,15 @@ void Audio_Close(struct AudioContext* ctx) {
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cc_result Audio_SetFormat(struct AudioContext* ctx, int channels, int sampleRate) {
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return ERR_NOT_SUPPORTED;
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}
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void Audio_SetVolume(struct AudioContext* ctx, int volume) {
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interop_AudioVolume(ctx->contextID, volume);
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}
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cc_result Audio_QueueChunk(struct AudioContext* ctx, void* chunk, cc_uint32 size) {
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return ERR_NOT_SUPPORTED;
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}
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cc_result Audio_Play(struct AudioContext* ctx) { return ERR_NOT_SUPPORTED; }
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cc_result Audio_Poll(struct AudioContext* ctx, int* inUse) {
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@ -1354,7 +1366,7 @@ cc_bool Audio_FastPlay(struct AudioContext* ctx, struct AudioData* data) {
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}
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cc_result Audio_PlayData(struct AudioContext* ctx, struct AudioData* data) {
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return interop_AudioPlay(ctx->contextID, data->data, data->volume, data->rate);
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return interop_AudioPlay(ctx->contextID, data->data, data->rate);
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}
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cc_bool Audio_DescribeError(cc_result res, cc_string* dst) {
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@ -1403,6 +1415,26 @@ cc_bool Audio_DescribeError(cc_result res, cc_string* dst) { return false; }
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*#########################################################################################################################*/
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#ifdef AUDIO_COMMON_VOLUME
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static void ApplyVolume(cc_int16* samples, int count, int volume) {
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int i;
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for (i = 0; i < (count & ~0x07); i += 8, samples += 8) {
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samples[0] = (samples[0] * volume / 100);
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samples[1] = (samples[1] * volume / 100);
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samples[2] = (samples[2] * volume / 100);
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samples[3] = (samples[3] * volume / 100);
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samples[4] = (samples[4] * volume / 100);
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samples[5] = (samples[5] * volume / 100);
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samples[6] = (samples[6] * volume / 100);
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samples[7] = (samples[7] * volume / 100);
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}
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for (; i < count; i++, samples++) {
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samples[0] = (samples[0] * volume / 100);
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}
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}
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static void AudioBase_Clear(struct AudioContext* ctx) {
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ctx->count = 0;
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ctx->channels = 0;
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@ -1414,7 +1446,7 @@ static void AudioBase_Clear(struct AudioContext* ctx) {
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static cc_bool AudioBase_AdjustSound(struct AudioContext* ctx, struct AudioData* data) {
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void* audio;
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if (data->volume >= 100) return true;
|
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if (ctx->volume >= 100) return true;
|
||||
|
||||
/* copy to temp buffer to apply volume */
|
||||
if (ctx->_tmpSize < data->size) {
|
||||
@ -1432,7 +1464,7 @@ static cc_bool AudioBase_AdjustSound(struct AudioContext* ctx, struct AudioData*
|
||||
|
||||
audio = ctx->_tmpData;
|
||||
Mem_Copy(audio, data->data, data->size);
|
||||
Audio_ApplyVolume((cc_int16*)audio, data->size / 2, data->volume);
|
||||
ApplyVolume((cc_int16*)audio, data->size / 2, ctx->volume);
|
||||
data->data = audio;
|
||||
return true;
|
||||
}
|
||||
@ -1463,6 +1495,11 @@ struct AudioContext music_ctx;
|
||||
static struct AudioContext context_pool[POOL_MAX_CONTEXTS];
|
||||
|
||||
#ifndef CC_BUILD_NOSOUNDS
|
||||
static cc_result PlayAudio(struct AudioContext* ctx, struct AudioData* data) {
|
||||
Audio_SetVolume(ctx, data->volume);
|
||||
return Audio_PlayData(ctx, data);
|
||||
}
|
||||
|
||||
cc_result AudioPool_Play(struct AudioData* data) {
|
||||
struct AudioContext* ctx;
|
||||
int inUse, i;
|
||||
@ -1477,7 +1514,7 @@ cc_result AudioPool_Play(struct AudioData* data) {
|
||||
if (inUse > 0) continue;
|
||||
|
||||
if (!Audio_FastPlay(ctx, data)) continue;
|
||||
return Audio_PlayData(ctx, data);
|
||||
return PlayAudio(ctx, data);
|
||||
}
|
||||
|
||||
/* Try again with all contexts, even if need to recreate one (expensive) */
|
||||
@ -1488,7 +1525,7 @@ cc_result AudioPool_Play(struct AudioData* data) {
|
||||
if (res) return res;
|
||||
if (inUse > 0) continue;
|
||||
|
||||
return Audio_PlayData(ctx, data);
|
||||
return PlayAudio(ctx, data);
|
||||
}
|
||||
return 0;
|
||||
}
|
||||
|
||||
@ -976,7 +976,7 @@ mergeInto(LibraryManager.library, {
|
||||
interop_AudioCreate: function() {
|
||||
var src = {
|
||||
source: null,
|
||||
gain: null,
|
||||
gain: AUDIO.context.createGain(),
|
||||
playing: false,
|
||||
};
|
||||
AUDIO.sources.push(src);
|
||||
@ -993,7 +993,11 @@ mergeInto(LibraryManager.library, {
|
||||
HEAP32[inUse >> 2] = src.playing; // only 1 buffer
|
||||
return 0;
|
||||
},
|
||||
interop_AudioPlay: function(ctxID, sndID, volume, rate) {
|
||||
interop_AudioVolume: function(ctxID, volume) {
|
||||
var src = AUDIO.sources[ctxID - 1|0];
|
||||
src.gain.gain.value = volume / 100;
|
||||
},
|
||||
interop_AudioPlay: function(ctxID, sndID, rate) {
|
||||
var src = AUDIO.sources[ctxID - 1|0];
|
||||
var name = UTF8ToString(sndID);
|
||||
|
||||
@ -1009,15 +1013,12 @@ mergeInto(LibraryManager.library, {
|
||||
if (!buffer) return 0;
|
||||
|
||||
try {
|
||||
if (!src.gain) src.gain = AUDIO.context.createGain();
|
||||
|
||||
// AudioBufferSourceNode only allows the buffer property
|
||||
// to be assigned *ONCE* (throws InvalidStateError next time)
|
||||
// MDN says that these nodes are very inexpensive to create though
|
||||
// https://developer.mozilla.org/en-US/docs/Web/API/AudioBufferSourceNode
|
||||
src.source = AUDIO.context.createBufferSource();
|
||||
src.source.buffer = buffer;
|
||||
src.gain.gain.value = volume / 100;
|
||||
src.source.playbackRate.value = rate / 100;
|
||||
|
||||
// source -> gain -> output
|
||||
@ -1444,4 +1445,4 @@ mergeInto(LibraryManager.library, {
|
||||
|
||||
CCFS.ensureErrnoError();
|
||||
},
|
||||
});
|
||||
});
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user