ac2cda5fbd
The patch that introduced s32/float support included a fix/change in audio/format.c which is practically irrelevant and now reverted in upstream. Removing the part to avoid confusion in the future.
71 lines
2.5 KiB
Diff
71 lines
2.5 KiB
Diff
diff --git a/audio/out/ao_opensles.c b/audio/out/ao_opensles.c
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index ea48de892e..f3082a9aa6 100644
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--- a/audio/out/ao_opensles.c
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+++ b/audio/out/ao_opensles.c
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@@ -43,12 +43,6 @@ struct priv {
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int cfg_frames_per_buffer;
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};
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-static const int fmtmap[][2] = {
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- { AF_FORMAT_U8, SL_PCMSAMPLEFORMAT_FIXED_8 },
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- { AF_FORMAT_S16, SL_PCMSAMPLEFORMAT_FIXED_16 },
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- { 0 }
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-};
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-
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#define DESTROY(thing) \
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if (p->thing) { \
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(*p->thing)->Destroy(p->thing); \
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@@ -115,7 +109,7 @@ static int init(struct ao *ao)
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struct priv *p = ao->priv;
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SLDataLocator_BufferQueue locator_buffer_queue;
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SLDataLocator_OutputMix locator_output_mix;
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- SLDataFormat_PCM pcm;
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+ SLAndroidDataFormat_PCM_EX pcm;
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SLDataSource audio_source;
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SLDataSink audio_sink;
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@@ -131,29 +125,23 @@ static int init(struct ao *ao)
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locator_buffer_queue.locatorType = SL_DATALOCATOR_BUFFERQUEUE;
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locator_buffer_queue.numBuffers = 1;
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- pcm.formatType = SL_DATAFORMAT_PCM;
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- pcm.numChannels = 2;
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-
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- int compatible_formats[AF_FORMAT_COUNT + 1];
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- af_get_best_sample_formats(ao->format, compatible_formats);
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- pcm.bitsPerSample = 0;
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- for (int i = 0; compatible_formats[i] && !pcm.bitsPerSample; ++i)
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- for (int j = 0; fmtmap[j][0]; ++j)
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- if (compatible_formats[i] == fmtmap[j][0]) {
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- ao->format = fmtmap[j][0];
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- pcm.bitsPerSample = fmtmap[j][1];
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- break;
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- }
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- if (!pcm.bitsPerSample) {
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- MP_ERR(ao, "Cannot find compatible audio format\n");
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- goto error;
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+ if (af_fmt_is_int(ao->format)) {
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+ // Be future-proof
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+ if (af_fmt_to_bytes(ao->format) > 2)
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+ ao->format = AF_FORMAT_S32;
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+ else
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+ ao->format = af_fmt_from_planar(ao->format);
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+ pcm.formatType = SL_DATAFORMAT_PCM;
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+ } else {
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+ ao->format = AF_FORMAT_FLOAT;
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+ pcm.formatType = SL_ANDROID_DATAFORMAT_PCM_EX;
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+ pcm.representation = SL_ANDROID_PCM_REPRESENTATION_FLOAT;
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}
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- pcm.containerSize = 8 * af_fmt_to_bytes(ao->format);
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+ pcm.numChannels = ao->channels.num;
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+ pcm.containerSize = pcm.bitsPerSample = 8 * af_fmt_to_bytes(ao->format);
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pcm.channelMask = SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT;
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pcm.endianness = SL_BYTEORDER_LITTLEENDIAN;
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-
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- // samplesPerSec is misnamed, actually it's samples per ms
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- pcm.samplesPerSec = ao->samplerate * 1000;
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+ pcm.sampleRate = ao->samplerate * 1000;
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if (p->cfg_frames_per_buffer)
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ao->device_buffer = p->cfg_frames_per_buffer;
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