Improved audio buffer handling for iOS

webrtc/modules/audio_device/ios/audio_device_ios.mm

Index: webrtc/modules/audio_device/ios/audio_device_ios.mm
diff --git a/webrtc/modules/audio_device/ios/audio_device_ios.mm b/webrtc/modules/audio_device/ios/audio_device_ios.mm
index 8cb1cd2b99ec23c6381fc76498dd46e325fd1514..ed948eadb94a4743c6849a2083ac434efb027761 100644
--- a/webrtc/modules/audio_device/ios/audio_device_ios.mm
+++ b/webrtc/modules/audio_device/ios/audio_device_ios.mm
@@ -15,6 +15,7 @@
 
 #include <cmath>
 
+#include "webrtc/base/array_view.h"
 #include "webrtc/base/atomicops.h"
 #include "webrtc/base/bind.h"
 #include "webrtc/base/checks.h"
@@ -69,6 +70,11 @@ enum AudioDeviceMessageType : uint32_t {
 using ios::CheckAndLogError;
 
 #if !defined(NDEBUG)
+// Returns true when the code runs on a device simulator.
+static bool DeviceIsSimulator() {
+  return ios::GetDeviceName() == "x86_64";
+}
+
 // Helper method that logs essential device information strings.
 static void LogDeviceInfo() {
   LOG(LS_INFO) << "LogDeviceInfo";
@@ -86,6 +92,10 @@ static void LogDeviceInfo() {
     && __IPHONE_OS_VERSION_MAX_ALLOWED >= __IPHONE_9_0
     LOG(LS_INFO) << " low power mode: " << ios::GetLowPowerModeEnabled();
 #endif
+#if TARGET_IPHONE_SIMULATOR
+    LOG(LS_INFO) << " TARGET_IPHONE_SIMULATOR is defined";
+#endif
+    LOG(LS_INFO) << " DeviceIsSimulator: " << DeviceIsSimulator();
   }
 }
 #endif  // !defined(NDEBUG)
@@ -423,26 +433,11 @@ OSStatus AudioDeviceIOS::OnGetPlayoutData(AudioUnitRenderActionFlags* flags,
     memset(destination, 0, size_in_bytes);
     return noErr;
   }
-  // Produce silence and log a warning message for the case when Core Audio is
-  // asking for an invalid number of audio frames. I don't expect this to happen
-  // but it is done as a safety measure to avoid bad audio if such as case would
-  // ever be triggered e.g. in combination with BT devices.
-  const size_t frames_per_buffer = playout_parameters_.frames_per_buffer();
-  if (num_frames != frames_per_buffer) {
-    RTCLogWarning(@"Expected %u frames but got %u",
-                  static_cast<unsigned int>(frames_per_buffer),
-                  static_cast<unsigned int>(num_frames));
-    *flags |= kAudioUnitRenderAction_OutputIsSilence;
-    memset(destination, 0, size_in_bytes);
-    return noErr;
-  }
 
   // Read decoded 16-bit PCM samples from WebRTC (using a size that matches
-  // the native I/O audio unit) to a preallocated intermediate buffer and
-  // copy the result to the audio buffer in the |io_data| destination.
-  int8_t* source = playout_audio_buffer_.get();
-  fine_audio_buffer_->GetPlayoutData(source);
-  memcpy(destination, source, size_in_bytes);
+  // the native I/O audio unit) and copy the result to the audio buffer in the
+  // |io_data| destination.
+  fine_audio_buffer_->GetPlayoutData(rtc::ArrayView<int8_t>(destination, size_in_bytes));
   return noErr;
 }
 
@@ -632,13 +627,12 @@ void AudioDeviceIOS::SetupAudioBuffersForActiveAudioSession() {
 
   // Create a modified audio buffer class which allows us to ask for,
   // or deliver, any number of samples (and not only multiple of 10ms) to match
-  // the native audio unit buffer size.
+  // the native audio unit buffer size. Use a reasonable capacity to avoid
+  // reallocations while audio is played to reduce risk of glitches.
   RTC_DCHECK(audio_device_buffer_);
-  const size_t buffer_size_in_bytes = playout_parameters_.GetBytesPerBuffer();
+  const size_t capacity_in_bytes = 2 * playout_parameters_.GetBytesPerBuffer();
   fine_audio_buffer_.reset(new FineAudioBuffer(
-      audio_device_buffer_, buffer_size_in_bytes,
-      playout_parameters_.sample_rate()));
-  playout_audio_buffer_.reset(new SInt8[buffer_size_in_bytes]);
+      audio_device_buffer_, playout_parameters_.sample_rate(), capacity_in_bytes));
 
   // Allocate AudioBuffers to be used as storage for the received audio.
   // The AudioBufferList structure works as a placeholder for the