diff --git a/include/SDL3/SDL_audio.h b/include/SDL3/SDL_audio.h
index 84894cebae..dcbb71154f 100644
--- a/include/SDL3/SDL_audio.h
+++ b/include/SDL3/SDL_audio.h
@@ -954,13 +954,22 @@ extern DECLSPEC int SDLCALL SDL_UnlockAudioStream(SDL_AudioStream *stream);
  * before your callback is called, so your callback does not need to
  * manage the lock explicitly.
  *
+ * Two values are offered here: one is the amount of additional data needed
+ * to satisfy the immediate request (which might be zero if the stream
+ * already has enough data queued) and the other is the total amount
+ * being requested. In a Get call triggering a Put callback, these
+ * values can be different. In a Put call triggering a Get callback,
+ * these values are always the same.
+ *
+ * Byte counts might be slightly overestimated due to buffering or
+ * resampling, and may change from call to call.
+ *
  * \param stream The SDL audio stream associated with this callback.
- * \param approx_amount The _approximate_ amount of data, in bytes, that is requested or available.
- *                       This might be slightly overestimated due to buffering or
- *                       resampling, and may change from call to call.
+ * \param additional The amount of data, in bytes, that is needed right now
+ * \param total_amount The total amount of data requested, in bytes, that is requested or available.
  * \param userdata An opaque pointer provided by the app for their personal use.
  */
-typedef void (SDLCALL *SDL_AudioStreamCallback)(void *userdata, SDL_AudioStream *stream, int approx_amount);
+typedef void (SDLCALL *SDL_AudioStreamCallback)(void *userdata, SDL_AudioStream *stream, int additional_amount, int total_amount);
 
 /**
  * Set a callback that runs when data is requested from an audio stream.
diff --git a/src/audio/SDL_audiocvt.c b/src/audio/SDL_audiocvt.c
index 70a405b15d..dc521f1349 100644
--- a/src/audio/SDL_audiocvt.c
+++ b/src/audio/SDL_audiocvt.c
@@ -1342,9 +1342,7 @@ int SDL_PutAudioStreamData(SDL_AudioStream *stream, const void *buf, int len)
 
     if ((retval == 0) && stream->put_callback) {
         const int newavail = SDL_GetAudioStreamAvailable(stream) - prev_available;
-        if (newavail > 0) {   // don't call the callback if we can't actually offer new data (still filling future buffer, only added 1 frame but downsampling needs more to produce new sound, etc).
-            stream->put_callback(stream->put_callback_userdata, stream, newavail);
-        }
+        stream->put_callback(stream->put_callback_userdata, stream, newavail, newavail);
     }
 
     SDL_UnlockMutex(stream->lock);
@@ -1647,7 +1645,8 @@ int SDL_GetAudioStreamData(SDL_AudioStream *stream, void *voidbuf, int len)
 
     // give the callback a chance to fill in more stream data if it wants.
     if (stream->get_callback) {
-        Sint64 approx_request = len / dst_frame_size;  // start with sample frames desired
+        Sint64 total_request = len / dst_frame_size;  // start with sample frames desired
+        Sint64 approx_request = total_request;
 
         const Sint64 available_frames = GetAudioStreamAvailableFrames(stream);
         approx_request -= SDL_min(available_frames, approx_request);
@@ -1655,14 +1654,13 @@ int SDL_GetAudioStreamData(SDL_AudioStream *stream, void *voidbuf, int len)
         const Sint64 resample_rate = GetStreamResampleRate(stream, stream->src_spec.freq);
 
         if (resample_rate) {
+            total_request = GetResamplerNeededInputFrames((int) total_request, resample_rate, 0);
             approx_request = GetResamplerNeededInputFrames((int) approx_request, resample_rate, 0);
         }
 
+        total_request *= SDL_AUDIO_FRAMESIZE(stream->src_spec);  // convert sample frames to bytes.
         approx_request *= SDL_AUDIO_FRAMESIZE(stream->src_spec);  // convert sample frames to bytes.
-
-        if (approx_request > 0) {  // don't call the callback if we can satisfy this request with existing data.
-            stream->get_callback(stream->get_callback_userdata, stream, (int) SDL_min(approx_request, SDL_INT_MAX));
-        }
+        stream->get_callback(stream->get_callback_userdata, stream, (int) SDL_min(approx_request, SDL_INT_MAX), (int) SDL_min(total_request, SDL_INT_MAX));
     }
 
     const int chunk_size = 4096;
diff --git a/test/testaudiostreamdynamicresample.c b/test/testaudiostreamdynamicresample.c
index 15d6dec272..7e9e09211b 100644
--- a/test/testaudiostreamdynamicresample.c
+++ b/test/testaudiostreamdynamicresample.c
@@ -356,7 +356,7 @@ static void loop(void)
     }
 }
 
-static void SDLCALL our_get_callback(void *userdata, SDL_AudioStream *strm, int approx_amount)
+static void SDLCALL our_get_callback(void *userdata, SDL_AudioStream *strm, int approx_amount, int total_amount)
 {
     last_get_callback = SDL_GetTicks();
     last_get_amount = approx_amount;
diff --git a/test/testsurround.c b/test/testsurround.c
index 001952ea1b..551f316b9d 100644
--- a/test/testsurround.c
+++ b/test/testsurround.c
@@ -96,7 +96,7 @@ static SDL_bool is_lfe_channel(int channel_index, int channel_count)
     return (channel_count == 3 && channel_index == 2) || (channel_count >= 6 && channel_index == 3);
 }
 
-static void SDLCALL fill_buffer(void *userdata, SDL_AudioStream *stream, int len)
+static void SDLCALL fill_buffer(void *userdata, SDL_AudioStream *stream, int len, int totallen)
 {
     const int samples = len / sizeof(Sint16);
     Sint16 *buffer = NULL;