mirror of
https://github.com/libretro/RetroArch
synced 2025-01-31 15:32:59 +00:00
Cleanups in audio mixer code
This commit is contained in:
twinaphex
parent
6d50ac7cb7
commit
a833c5f4a6
@ -115,7 +115,7 @@ audio_chunk_t* audio_mix_load_wav_file(const char *path, int sample_rate)
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int sample_size;
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int64_t len = 0;
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void *buf = NULL;
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audio_chunk_t *chunk = (audio_chunk_t*)calloc(1, sizeof(*chunk));
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audio_chunk_t *chunk = (audio_chunk_t*)malloc(sizeof(*chunk));
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if (!chunk)
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return NULL;
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@ -126,10 +126,26 @@ audio_chunk_t* audio_mix_load_wav_file(const char *path, int sample_rate)
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goto error;
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}
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chunk->sample_rate = sample_rate;
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chunk->buf = buf;
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chunk->len = len;
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chunk->rwav = (rwav_t*)malloc(sizeof(rwav_t));
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chunk->buf = buf;
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chunk->upsample_buf = NULL;
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chunk->float_buf = NULL;
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chunk->float_resample_buf = NULL;
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chunk->resample_buf = NULL;
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chunk->len = len;
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chunk->resample_len = 0;
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chunk->rwav = (rwav_t*)malloc(sizeof(rwav_t));
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chunk->sample_rate = sample_rate;
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chunk->resample = false;
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chunk->resampler = NULL;
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chunk->resampler_data = NULL;
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chunk->ratio = 0.00f;
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chunk->rwav->bitspersample = 0;
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chunk->rwav->numchannels = 0;
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chunk->rwav->samplerate = 0;
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chunk->rwav->numsamples = 0;
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chunk->rwav->subchunk2size = 0;
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chunk->rwav->samples = NULL;
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if (rwav_load(chunk->rwav, chunk->buf, chunk->len) == RWAV_ITERATE_ERROR)
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{
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@ -139,28 +155,42 @@ audio_chunk_t* audio_mix_load_wav_file(const char *path, int sample_rate)
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/* numsamples does not know or care about
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* multiple channels, but we need space for 2 */
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chunk->upsample_buf = (int16_t*)memalign_alloc(128,
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chunk->upsample_buf = (int16_t*)memalign_alloc(128,
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chunk->rwav->numsamples * 2 * sizeof(int16_t));
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sample_size = chunk->rwav->bitspersample / 8;
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sample_size = chunk->rwav->bitspersample / 8;
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if (sample_size == 1)
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{
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unsigned i;
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for (i = 0; i < chunk->rwav->numsamples; i++)
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{
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uint8_t *sample = (
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(uint8_t*)chunk->rwav->samples) +
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(i * chunk->rwav->numchannels);
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if (chunk->rwav->numchannels == 1)
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{
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for (i = 0; i < chunk->rwav->numsamples; i++)
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{
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uint8_t *sample = (
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(uint8_t*)chunk->rwav->samples) + i;
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chunk->upsample_buf[i * 2] = (int16_t)((sample[0] - 128) << 8);
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chunk->upsample_buf[i * 2] =
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(int16_t)((sample[0] - 128) << 8);
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chunk->upsample_buf[(i * 2) + 1] =
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(int16_t)((sample[0] - 128) << 8);
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}
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}
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else if (chunk->rwav->numchannels == 2)
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{
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for (i = 0; i < chunk->rwav->numsamples; i++)
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{
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uint8_t *sample = (
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(uint8_t*)chunk->rwav->samples) +
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(i * 2);
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if (chunk->rwav->numchannels == 1)
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chunk->upsample_buf[(i * 2) + 1] = (int16_t)((sample[0] - 128) << 8);
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else if (chunk->rwav->numchannels == 2)
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chunk->upsample_buf[(i * 2) + 1] = (int16_t)((sample[1] - 128) << 8);
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}
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chunk->upsample_buf[i * 2] =
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(int16_t)((sample[0] - 128) << 8);
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chunk->upsample_buf[(i * 2) + 1] =
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(int16_t)((sample[1] - 128) << 8);
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}
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}
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}
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else if (sample_size == 2)
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{
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@ -170,14 +200,16 @@ audio_chunk_t* audio_mix_load_wav_file(const char *path, int sample_rate)
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for (i = 0; i < chunk->rwav->numsamples; i++)
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{
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int16_t sample = ((int16_t*)chunk->rwav->samples)[i];
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int16_t sample = ((int16_t*)
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chunk->rwav->samples)[i];
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chunk->upsample_buf[i * 2] = sample;
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chunk->upsample_buf[(i * 2) + 1] = sample;
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}
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}
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else if (chunk->rwav->numchannels == 2)
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memcpy(chunk->upsample_buf, chunk->rwav->samples, chunk->rwav->subchunk2size);
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memcpy(chunk->upsample_buf, chunk->rwav->samples,
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chunk->rwav->subchunk2size);
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}
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else if (sample_size != 2)
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{
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@ -189,7 +221,7 @@ audio_chunk_t* audio_mix_load_wav_file(const char *path, int sample_rate)
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if (sample_rate != (int)chunk->rwav->samplerate)
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{
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chunk->resample = true;
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chunk->ratio = (double)sample_rate / chunk->rwav->samplerate;
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chunk->ratio = (double)sample_rate / chunk->rwav->samplerate;
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retro_resampler_realloc(&chunk->resampler_data,
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&chunk->resampler,
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@ -201,12 +233,18 @@ audio_chunk_t* audio_mix_load_wav_file(const char *path, int sample_rate)
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{
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struct resampler_data info;
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chunk->float_buf = (float*)memalign_alloc(128, chunk->rwav->numsamples * 2 * chunk->ratio * sizeof(float));
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chunk->float_buf = (float*)memalign_alloc(128,
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chunk->rwav->numsamples * 2 *
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chunk->ratio * sizeof(float));
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/* why is *3 needed instead of just *2? does the sinc driver require more space than we know about? */
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chunk->float_resample_buf = (float*)memalign_alloc(128, chunk->rwav->numsamples * 3 * chunk->ratio * sizeof(float));
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/* why is *3 needed instead of just *2? Does the
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* sinc driver require more space than we know about? */
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chunk->float_resample_buf = (float*)memalign_alloc(128,
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chunk->rwav->numsamples * 3 *
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chunk->ratio * sizeof(float));
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convert_s16_to_float(chunk->float_buf, chunk->upsample_buf, chunk->rwav->numsamples * 2, 1.0);
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convert_s16_to_float(chunk->float_buf,
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chunk->upsample_buf, chunk->rwav->numsamples * 2, 1.0);
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info.data_in = (const float*)chunk->float_buf;
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info.data_out = chunk->float_resample_buf;
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@ -218,10 +256,13 @@ audio_chunk_t* audio_mix_load_wav_file(const char *path, int sample_rate)
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chunk->resampler->process(chunk->resampler_data, &info);
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/* number of output_frames does not increase with multiple channels, but assume we need space for 2 */
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chunk->resample_buf = (int16_t*)memalign_alloc(128, info.output_frames * 2 * sizeof(int16_t));
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/* number of output_frames does not increase with
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* multiple channels, but assume we need space for 2 */
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chunk->resample_buf = (int16_t*)memalign_alloc(128,
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info.output_frames * 2 * sizeof(int16_t));
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chunk->resample_len = info.output_frames;
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convert_float_to_s16(chunk->resample_buf, chunk->float_resample_buf, info.output_frames * 2);
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convert_float_to_s16(chunk->resample_buf,
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chunk->float_resample_buf, info.output_frames * 2);
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}
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}
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@ -258,7 +299,8 @@ size_t audio_mix_get_chunk_num_samples(audio_chunk_t *chunk)
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*
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* Returns: A signed 16-bit audio sample.
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**/
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int16_t audio_mix_get_chunk_sample(audio_chunk_t *chunk, unsigned channel, size_t index)
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int16_t audio_mix_get_chunk_sample(audio_chunk_t *chunk,
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unsigned channel, size_t index)
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{
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if (!chunk)
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return 0;
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@ -273,10 +315,12 @@ int16_t audio_mix_get_chunk_sample(audio_chunk_t *chunk, unsigned channel, size_
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if (chunk->resample)
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sample = (uint8_t*)chunk->resample_buf +
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(sample_size * index * chunk->rwav->numchannels) + (channel * sample_size);
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(sample_size * index * chunk->rwav->numchannels)
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+ (channel * sample_size);
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else
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sample = (uint8_t*)chunk->upsample_buf +
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(sample_size * index * chunk->rwav->numchannels) + (channel * sample_size);
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(sample_size * index * chunk->rwav->numchannels)
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+ (channel * sample_size);
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sample_out = (int16_t)*sample;
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@ -184,10 +184,11 @@ struct audio_mixer_voice
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} types;
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};
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/* TODO/FIXME - static globals */
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static struct audio_mixer_voice s_voices[AUDIO_MIXER_MAX_VOICES] = {{0}};
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static unsigned s_rate = 0;
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static bool wav2float(const rwav_t* wav, float** pcm, size_t samples_out)
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static bool wav_to_float(const rwav_t* wav, float** pcm, size_t samples_out)
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{
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size_t i;
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/* Allocate on a 16-byte boundary, and pad to a multiple of 16 bytes */
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@ -334,7 +335,7 @@ audio_mixer_sound_t* audio_mixer_load_wav(void *buffer, int32_t size)
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samples = wav.numsamples * 2;
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if (!wav2float(&wav, &pcm, samples))
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if (!wav_to_float(&wav, &pcm, samples))
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return NULL;
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if (wav.samplerate != s_rate)
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@ -839,8 +840,8 @@ void audio_mixer_stop(audio_mixer_voice_t* voice)
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if (voice)
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{
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stop_cb = voice->stop_cb;
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sound = voice->sound;
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stop_cb = voice->stop_cb;
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sound = voice->sound;
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voice->type = AUDIO_MIXER_TYPE_NONE;
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@ -899,7 +900,7 @@ static void audio_mixer_mix_ogg(float* buffer, size_t num_frames,
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float volume)
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{
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int i;
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struct resampler_data info = { 0 };
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struct resampler_data info;
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float temp_buffer[AUDIO_MIXER_TEMP_BUFFER] = { 0 };
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unsigned buf_free = (unsigned)(num_frames * 2);
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unsigned temp_samples = 0;
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@ -922,14 +923,12 @@ again:
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stb_vorbis_seek_start(voice->types.ogg.stream);
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goto again;
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}
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else
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{
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if (voice->stop_cb)
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voice->stop_cb(voice->sound, AUDIO_MIXER_SOUND_FINISHED);
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voice->type = AUDIO_MIXER_TYPE_NONE;
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return;
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}
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if (voice->stop_cb)
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voice->stop_cb(voice->sound, AUDIO_MIXER_SOUND_FINISHED);
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voice->type = AUDIO_MIXER_TYPE_NONE;
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return;
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}
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info.data_in = temp_buffer;
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@ -939,11 +938,12 @@ again:
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info.ratio = voice->types.ogg.ratio;
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if (voice->types.ogg.resampler)
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{
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voice->types.ogg.resampler->process(voice->types.ogg.resampler_data, &info);
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}
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voice->types.ogg.resampler->process(
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voice->types.ogg.resampler_data, &info);
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else
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memcpy(voice->types.ogg.buffer, temp_buffer, temp_samples * sizeof(float));
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memcpy(voice->types.ogg.buffer, temp_buffer,
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temp_samples * sizeof(float));
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voice->types.ogg.position = 0;
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voice->types.ogg.samples = voice->types.ogg.buf_samples;
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}
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@ -958,15 +958,12 @@ again:
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buf_free -= voice->types.ogg.samples;
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goto again;
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}
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else
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{
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int i;
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for (i = buf_free; i != 0; --i )
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*buffer++ += *pcm++ * volume;
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voice->types.ogg.position += buf_free;
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voice->types.ogg.samples -= buf_free;
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}
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for (i = buf_free; i != 0; --i )
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*buffer++ += *pcm++ * volume;
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voice->types.ogg.position += buf_free;
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voice->types.ogg.samples -= buf_free;
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}
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#endif
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@ -1000,14 +997,12 @@ again:
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replay_seek( voice->types.mod.stream, 0);
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goto again;
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}
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else
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{
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if (voice->stop_cb)
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voice->stop_cb(voice->sound, AUDIO_MIXER_SOUND_FINISHED);
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voice->type = AUDIO_MIXER_TYPE_NONE;
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return;
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}
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if (voice->stop_cb)
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voice->stop_cb(voice->sound, AUDIO_MIXER_SOUND_FINISHED);
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voice->type = AUDIO_MIXER_TYPE_NONE;
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return;
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}
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voice->types.mod.position = 0;
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@ -1028,20 +1023,17 @@ again:
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buf_free -= voice->types.mod.samples;
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goto again;
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}
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else
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{
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int i;
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for (i = buf_free; i != 0; --i )
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{
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samplei = *pcm++ * volume;
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samplef = (float)((int)samplei + 32768) / 65535.0f;
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samplef = samplef * 2.0f - 1.0f;
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*buffer++ += samplef;
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}
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voice->types.mod.position += buf_free;
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voice->types.mod.samples -= buf_free;
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for (i = buf_free; i != 0; --i )
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{
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samplei = *pcm++ * volume;
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samplef = (float)((int)samplei + 32768) / 65535.0f;
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samplef = samplef * 2.0f - 1.0f;
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*buffer++ += samplef;
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}
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voice->types.mod.position += buf_free;
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voice->types.mod.samples -= buf_free;
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}
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#endif
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@ -1051,11 +1043,11 @@ static void audio_mixer_mix_flac(float* buffer, size_t num_frames,
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float volume)
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{
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int i;
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struct resampler_data info = { 0 };
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struct resampler_data info;
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float temp_buffer[AUDIO_MIXER_TEMP_BUFFER] = { 0 };
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unsigned buf_free = (unsigned)(num_frames * 2);
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unsigned temp_samples = 0;
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float* pcm = NULL;
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float *pcm = NULL;
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if (voice->types.flac.position == voice->types.flac.samples)
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{
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@ -1071,14 +1063,12 @@ again:
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drflac_seek_to_sample(voice->types.flac.stream,0);
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goto again;
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}
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else
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{
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if (voice->stop_cb)
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voice->stop_cb(voice->sound, AUDIO_MIXER_SOUND_FINISHED);
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voice->type = AUDIO_MIXER_TYPE_NONE;
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return;
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}
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if (voice->stop_cb)
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voice->stop_cb(voice->sound, AUDIO_MIXER_SOUND_FINISHED);
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voice->type = AUDIO_MIXER_TYPE_NONE;
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return;
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}
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info.data_in = temp_buffer;
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@ -1088,11 +1078,10 @@ again:
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info.ratio = voice->types.flac.ratio;
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if (voice->types.flac.resampler)
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{
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voice->types.flac.resampler->process(voice->types.flac.resampler_data, &info);
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}
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voice->types.flac.resampler->process(
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voice->types.flac.resampler_data, &info);
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else
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memcpy(voice->types.flac.buffer, temp_buffer, temp_samples * sizeof(float));
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memcpy(voice->types.flac.buffer, temp_buffer, temp_samples * sizeof(float));
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voice->types.flac.position = 0;
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voice->types.flac.samples = voice->types.flac.buf_samples;
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}
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@ -1107,15 +1096,12 @@ again:
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buf_free -= voice->types.flac.samples;
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goto again;
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}
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else
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{
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int i;
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for (i = buf_free; i != 0; --i )
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*buffer++ += *pcm++ * volume;
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voice->types.flac.position += buf_free;
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voice->types.flac.samples -= buf_free;
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}
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for (i = buf_free; i != 0; --i )
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*buffer++ += *pcm++ * volume;
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voice->types.flac.position += buf_free;
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voice->types.flac.samples -= buf_free;
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}
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#endif
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@ -1125,7 +1111,7 @@ static void audio_mixer_mix_mp3(float* buffer, size_t num_frames,
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float volume)
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{
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int i;
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struct resampler_data info = { 0 };
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struct resampler_data info;
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float temp_buffer[AUDIO_MIXER_TEMP_BUFFER] = { 0 };
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unsigned buf_free = (unsigned)(num_frames * 2);
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unsigned temp_samples = 0;
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@ -1134,7 +1120,9 @@ static void audio_mixer_mix_mp3(float* buffer, size_t num_frames,
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if (voice->types.mp3.position == voice->types.mp3.samples)
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{
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again:
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temp_samples = (unsigned)drmp3_read_f32(&voice->types.mp3.stream, AUDIO_MIXER_TEMP_BUFFER/2, temp_buffer) * 2;
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temp_samples = (unsigned)drmp3_read_f32(
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&voice->types.mp3.stream,
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AUDIO_MIXER_TEMP_BUFFER / 2, temp_buffer) * 2;
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if (temp_samples == 0)
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{
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||||
@ -1146,14 +1134,12 @@ again:
|
||||
drmp3_seek_to_frame(&voice->types.mp3.stream,0);
|
||||
goto again;
|
||||
}
|
||||
else
|
||||
{
|
||||
if (voice->stop_cb)
|
||||
voice->stop_cb(voice->sound, AUDIO_MIXER_SOUND_FINISHED);
|
||||
|
||||
voice->type = AUDIO_MIXER_TYPE_NONE;
|
||||
return;
|
||||
}
|
||||
if (voice->stop_cb)
|
||||
voice->stop_cb(voice->sound, AUDIO_MIXER_SOUND_FINISHED);
|
||||
|
||||
voice->type = AUDIO_MIXER_TYPE_NONE;
|
||||
return;
|
||||
}
|
||||
|
||||
info.data_in = temp_buffer;
|
||||
@ -1163,9 +1149,11 @@ again:
|
||||
info.ratio = voice->types.mp3.ratio;
|
||||
|
||||
if (voice->types.mp3.resampler)
|
||||
voice->types.mp3.resampler->process(voice->types.mp3.resampler_data, &info);
|
||||
voice->types.mp3.resampler->process(
|
||||
voice->types.mp3.resampler_data, &info);
|
||||
else
|
||||
memcpy(voice->types.mp3.buffer, temp_buffer, temp_samples * sizeof(float));
|
||||
memcpy(voice->types.mp3.buffer, temp_buffer,
|
||||
temp_samples * sizeof(float));
|
||||
voice->types.mp3.position = 0;
|
||||
voice->types.mp3.samples = voice->types.mp3.buf_samples;
|
||||
}
|
||||
@ -1180,19 +1168,17 @@ again:
|
||||
buf_free -= voice->types.mp3.samples;
|
||||
goto again;
|
||||
}
|
||||
else
|
||||
{
|
||||
int i;
|
||||
for (i = buf_free; i != 0; --i )
|
||||
*buffer++ += *pcm++ * volume;
|
||||
|
||||
voice->types.mp3.position += buf_free;
|
||||
voice->types.mp3.samples -= buf_free;
|
||||
}
|
||||
for (i = buf_free; i != 0; --i )
|
||||
*buffer++ += *pcm++ * volume;
|
||||
|
||||
voice->types.mp3.position += buf_free;
|
||||
voice->types.mp3.samples -= buf_free;
|
||||
}
|
||||
#endif
|
||||
|
||||
void audio_mixer_mix(float* buffer, size_t num_frames, float volume_override, bool override)
|
||||
void audio_mixer_mix(float* buffer, size_t num_frames,
|
||||
float volume_override, bool override)
|
||||
{
|
||||
unsigned i;
|
||||
size_t j = 0;
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user