diff --git a/packages/mpv/mpv.conf b/packages/mpv/mpv.conf
index 265ced5c7..6cd46637c 100644
--- a/packages/mpv/mpv.conf
+++ b/packages/mpv/mpv.conf
@@ -12,5 +12,11 @@ ao=opensles
 # value in second instead), but it will only has effect on the
 # opensles ao if the above option is explicitly set to 0.
 
+# By default mpv allows s32 and float output. If they aren't
+# supported in your device (which should not happen with API
+# 21 and above), disable them with the following option:
+
+# audio-format=s16
+
 # Disable Video Decode and Output. Termux doesn't support video output (with the exception of "tct").
 vid=no
diff --git a/packages/mpv/sles_float.patch b/packages/mpv/sles_float.patch
new file mode 100644
index 000000000..a670a226e
--- /dev/null
+++ b/packages/mpv/sles_float.patch
@@ -0,0 +1,93 @@
+diff --git a/audio/format.c b/audio/format.c
+index 8a13698ff7..7004409d10 100644
+--- a/audio/format.c
++++ b/audio/format.c
+@@ -95,14 +95,16 @@ int af_fmt_to_planar(int format)
+ 
+ // Return the interleaved format corresponding to the given format.
+ // If the format is already interleaved, return it.
+-// Always succeeds if format is actually planar; otherwise return 0.
++// Return 0 if there's no equivalent.
+ int af_fmt_from_planar(int format)
+ {
+     for (int n = 0; n < MP_ARRAY_SIZE(planar_formats); n++) {
+         if (planar_formats[n][0] == format)
+             return planar_formats[n][1];
++        if (planar_formats[n][1] == format)
++            return format;
+     }
+-    return format;
++    return 0;
+ }
+ 
+ bool af_fmt_is_valid(int format)
+diff --git a/audio/out/ao_opensles.c b/audio/out/ao_opensles.c
+index ea48de892e..f3082a9aa6 100644
+--- a/audio/out/ao_opensles.c
++++ b/audio/out/ao_opensles.c
+@@ -43,12 +43,6 @@ struct priv {
+     int cfg_frames_per_buffer;
+ };
+ 
+-static const int fmtmap[][2] = {
+-    { AF_FORMAT_U8, SL_PCMSAMPLEFORMAT_FIXED_8 },
+-    { AF_FORMAT_S16, SL_PCMSAMPLEFORMAT_FIXED_16 },
+-    { 0 }
+-};
+-
+ #define DESTROY(thing) \
+     if (p->thing) { \
+         (*p->thing)->Destroy(p->thing); \
+@@ -115,7 +109,7 @@ static int init(struct ao *ao)
+     struct priv *p = ao->priv;
+     SLDataLocator_BufferQueue locator_buffer_queue;
+     SLDataLocator_OutputMix locator_output_mix;
+-    SLDataFormat_PCM pcm;
++    SLAndroidDataFormat_PCM_EX pcm;
+     SLDataSource audio_source;
+     SLDataSink audio_sink;
+ 
+@@ -131,29 +125,23 @@ static int init(struct ao *ao)
+     locator_buffer_queue.locatorType = SL_DATALOCATOR_BUFFERQUEUE;
+     locator_buffer_queue.numBuffers = 1;
+ 
+-    pcm.formatType = SL_DATAFORMAT_PCM;
+-    pcm.numChannels = 2;
+-
+-    int compatible_formats[AF_FORMAT_COUNT + 1];
+-    af_get_best_sample_formats(ao->format, compatible_formats);
+-    pcm.bitsPerSample = 0;
+-    for (int i = 0; compatible_formats[i] && !pcm.bitsPerSample; ++i)
+-        for (int j = 0; fmtmap[j][0]; ++j)
+-            if (compatible_formats[i] == fmtmap[j][0]) {
+-                ao->format = fmtmap[j][0];
+-                pcm.bitsPerSample = fmtmap[j][1];
+-                break;
+-            }
+-    if (!pcm.bitsPerSample) {
+-        MP_ERR(ao, "Cannot find compatible audio format\n");
+-        goto error;
++    if (af_fmt_is_int(ao->format)) {
++        // Be future-proof
++        if (af_fmt_to_bytes(ao->format) > 2)
++            ao->format = AF_FORMAT_S32;
++        else
++            ao->format = af_fmt_from_planar(ao->format);
++        pcm.formatType = SL_DATAFORMAT_PCM;
++    } else {
++        ao->format = AF_FORMAT_FLOAT;
++        pcm.formatType = SL_ANDROID_DATAFORMAT_PCM_EX;
++        pcm.representation = SL_ANDROID_PCM_REPRESENTATION_FLOAT;
+     }
+-    pcm.containerSize = 8 * af_fmt_to_bytes(ao->format);
++    pcm.numChannels = ao->channels.num;
++    pcm.containerSize = pcm.bitsPerSample = 8 * af_fmt_to_bytes(ao->format);
+     pcm.channelMask = SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT;
+     pcm.endianness = SL_BYTEORDER_LITTLEENDIAN;
+-
+-    // samplesPerSec is misnamed, actually it's samples per ms
+-    pcm.samplesPerSec = ao->samplerate * 1000;
++    pcm.sampleRate = ao->samplerate * 1000;
+ 
+     if (p->cfg_frames_per_buffer)
+         ao->device_buffer = p->cfg_frames_per_buffer;