Pad with silence according to the input device sampling rate.

Contrary to earlier assumed, even in duplex audio the sampling rate of the input device doesn't always match the rate of the output device.

1 files changed, 23 insertions, 10 deletions

diff --git a/src/cubeb_audiounit.cpp b/src/cubeb_audiounit.cpp
index b9f2709..eb2daff 100644
--- a/src/cubeb_audiounit.cpp
+++ b/src/cubeb_audiounit.cpp
@@ -548,6 +548,19 @@ audiounit_mix_output_buffer(cubeb_stream * stm,
   }
 }
 
+// Return how many input frames (sampled at input_hw_rate) are needed to provide
+// output_frames (sampled at output_stream_params.rate)
+static uint32_t
+minimum_resampling_input_frames(cubeb_stream * stm, uint32_t output_frames)
+{
+  if (stm->input_hw_rate == stm->output_stream_params.rate) {
+    // Fast path.
+    return output_frames;
+  }
+  return ceil(stm->input_hw_rate * output_frames /
+              stm->output_stream_params.rate);
+}
+
 static OSStatus
 audiounit_output_callback(void * user_ptr,
                           AudioUnitRenderActionFlags * /* flags */,
@@ -569,7 +582,7 @@ audiounit_output_callback(void * user_ptr,
         (unsigned int) output_frames,
         stm->available_input_frames);
 
-  long input_frames = 0, input_frames_before_fill = 0;
+  long input_frames = 0;
   void * output_buffer = NULL, * input_buffer = NULL;
 
   if (stm->shutdown) {
@@ -607,10 +620,15 @@ audiounit_output_callback(void * user_ptr,
 
   /* If Full duplex get also input buffer */
   if (stm->input_unit != NULL) {
-    // In duplex mode, input rate is always output rate.
-    assert(stm->input_hw_rate == stm->output_hw_rate);
-    if (stm->available_input_frames < output_frames) {
-      long missing_frames = output_frames - stm->available_input_frames;
+    /* If the output callback came first and this is a duplex stream, we need to
+     * fill in some additional silence in the resampler.
+     * Otherwise, if we had more than expected callbacks in a row, or we're
+     * currently switching, we add some silence as well to compensate for the
+     * fact that we're lacking some input data. */
+    uint32_t input_frames_needed =
+      minimum_resampling_input_frames(stm, output_frames);
+    if (stm->available_input_frames < input_frames_needed) {
+      long missing_frames = input_frames_needed - stm->available_input_frames;
       stm->input_linear_buffer->push_silence(missing_frames * stm->input_desc.mChannelsPerFrame);
       stm->available_input_frames += missing_frames;
 
@@ -621,15 +639,10 @@ audiounit_output_callback(void * user_ptr,
     // Number of input frames in the buffer. It will change to actually used frames
     // inside fill
     input_frames = stm->input_linear_buffer->length() / stm->input_desc.mChannelsPerFrame;
-    // Number of input frames pushed inside resampler.
-    input_frames_before_fill = input_frames;
     assert(input_frames == stm->available_input_frames);
   }
 
   /* Call user callback through resampler. */
-  assert(!input_buffer ||
-         (stm->input_stream_params.rate == stm->output_stream_params.rate &&
-          input_frames >= output_frames));
   long outframes = cubeb_resampler_fill(stm->resampler.get(),
                                         input_buffer,
                                         input_buffer ? &input_frames : NULL,