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synced 2026-05-04 04:54:42 +00:00
Add SDL_ConvertAudioSamples() helper function
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@@ -54,29 +54,28 @@ SDL_PauseAudioDevice() is only used to pause audio playback. Use SDL_PlayAudioDe
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SDL_FreeWAV has been removed and calls can be replaced with SDL_free.
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SDL_AudioCVT interface is removed, SDL_AudioStream can be used instead.
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SDL_AudioCVT interface is removed, SDL_AudioStream interface or SDL_ConvertAudioSamples() helper function can be used.
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Code that used to look like this:
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```c
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SDL_AudioCVT cvt;
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SDL_BuildAudioCVT(&cvt, spec.format, spec.channels, spec.freq, spec.format, cvtchans, cvtfreq);
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cvt.len = len;
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cvt.buf = (Uint8 *) SDL_malloc(len * cvt.len_mult);
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SDL_memcpy(cvt.buf, data, len);
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SDL_BuildAudioCVT(&cvt, src_format, src_channels, src_rate, dst_format, dst_channels, dst_rate);
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cvt.len = src_len;
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cvt.buf = (Uint8 *) SDL_malloc(src_len * cvt.len_mult);
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SDL_memcpy(cvt.buf, src_data, src_len);
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SDL_ConvertAudio(&cvt);
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do_something(cvt.buf, cvt.len_cvt);
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```
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should be changed to:
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```c
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SDL_AudioStream *stream = SDL_CreateAudioStream(spec.format, spec.channels, spec.freq, spec.format, cvtchans, cvtfreq);
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int src_samplesize = (SDL_AUDIO_BITSIZE(spec.format) / 8) * spec.channels;
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int src_len = len & ~(src_samplesize - 1); // need to be rounded to samplesize
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SDL_PutAudioStreamData(stream, data, src_len);
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SDL_FlushAudioStream(stream);
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int dst_len = expected_dst_len & ~(dst_samplesize - 1); // need to be rounded to samplesize
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Uint8 *dst_buf = (Uint8 *)SDL_malloc(dst_len);
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int real_dst_len = SDL_GetAudioStreamData(stream, dst_buf, dst_len);
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do_something(dst_buf, real_dst_len);
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Uint8 *dst_data = NULL;
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int dst_len = 0;
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if (SDL_ConvertAudioSamples(src_format, src_channels, src_rate, src_len, src_data
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dst_format, dst_channels, dst_rate, &dst_len, &dst_data) < 0) {
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/* error */
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}
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do_something(dst_data, dst_len);
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SDL_free(dst_data);
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```
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