(OBSOLETE) Increasing audio buffer on Iphone 4S.

webrtc/modules/audio_device/ios/objc/RTCAudioSessionConfiguration.m

 /*
  *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #import "webrtc/modules/audio_device/ios/objc/RTCAudioSessionConfiguration.h"
 
+#import <sys/utsname.h>
+
 #import "WebRTC/RTCDispatcher.h"
 
 #import "webrtc/modules/audio_device/ios/objc/RTCAudioSession.h"
 
 // Try to use mono to save resources. Also avoids channel format conversion
 // in the I/O audio unit. Initial tests have shown that it is possible to use
 // mono natively for built-in microphones and for BT headsets but not for
 // wired headsets. Wired headsets only support stereo as native channel format
 // but it is a low cost operation to do a format conversion to mono in the
 // audio unit. Hence, we will not hit a RTC_CHECK in
@@ skipping 14 matching lines @@
 // Use a hardware I/O buffer size (unit is in seconds) that matches the 10ms
 // size used by WebRTC. The exact actual size will differ between devices.
 // Example: using 48kHz on iPhone 6 results in a native buffer size of
 // ~10.6667ms or 512 audio frames per buffer. The FineAudioBuffer instance will
 // take care of any buffering required to convert between native buffers and
 // buffers used by WebRTC. It is beneficial for the performance if the native
 // size is as close to 10ms as possible since it results in "clean" callback
 // sequence without bursts of callbacks back to back.
 const double kRTCAudioSessionHighPerformanceIOBufferDuration = 0.01;
 
+// Use a larger buffer size on iphone 4S to allow it running complex tasks
+// e.g., encoding long frames with Opus codec.

[henrika_webrtc, 2016/06/17 07:37:37]

nit, I would say large frames

[minyue-webrtc, 2016/06/23 11:35:07]

Done.

+NSString* const kIphone4sDeviceTyp = @"iPhone4,1";
+const double kRTCAudioSessionHighPerformanceIOBufferDurationIphone4s = 0.02;
+
 // Use a larger buffer size on devices with only one core (e.g. iPhone 4).
 // It will result in a lower CPU consumption at the cost of a larger latency.
 // The size of 60ms is based on instrumentation that shows a significant
 // reduction in CPU load compared with 10ms on low-end devices.
 // TODO(henrika): monitor this size and determine if it should be modified.
 const double kRTCAudioSessionLowComplexityIOBufferDuration = 0.06;
 
 static RTCAudioSessionConfiguration *gWebRTCConfiguration = nil;
 
 @implementation RTCAudioSessionConfiguration
@@ skipping 16 matching lines @@
     _categoryOptions = AVAudioSessionCategoryOptionAllowBluetooth;
 
     // Specify mode for two-way voice communication (e.g. VoIP).
     _mode = AVAudioSessionModeVoiceChat;
 
     // Set the session's sample rate or the hardware sample rate.
     // It is essential that we use the same sample rate as stream format
     // to ensure that the I/O unit does not have to do sample rate conversion.
     // Set the preferred audio I/O buffer duration, in seconds.
     NSUInteger processorCount = [NSProcessInfo processInfo].processorCount;
+
+    struct utsname systemInfo;

[henrika_webrtc, 2016/06/17 07:37:37]

Please compare with existing methods in
https://cs.chromium.org/chromium/src/third_party/webrtc/modules/utility/sourc....
They are designed for mixed C++/ObjC but should be useful here as well.

[minyue-webrtc, 2016/06/23 11:35:07]

Done.

+    uname(&systemInfo);
+    NSString *device_type = [NSString stringWithCString:systemInfo.machine
+                                               encoding:NSUTF8StringEncoding];
+
     // Use best sample rate and buffer duration if the CPU has more than one
     // core.
     if (processorCount > 1) {
       _sampleRate = kRTCAudioSessionHighPerformanceSampleRate;
-      _ioBufferDuration = kRTCAudioSessionHighPerformanceIOBufferDuration;
+      if ([device_type isEqualToString:kIphone4sDeviceTyp]) {
+        _ioBufferDuration =
+            kRTCAudioSessionHighPerformanceIOBufferDurationIphone4s;
+      } else {
+        _ioBufferDuration = kRTCAudioSessionHighPerformanceIOBufferDuration;
+      }
     } else {
       _sampleRate = kRTCAudioSessionLowComplexitySampleRate;
       _ioBufferDuration = kRTCAudioSessionLowComplexityIOBufferDuration;
     }
 
     // We try to use mono in both directions to save resources and format
     // conversions in the audio unit. Some devices does only support stereo;
     // e.g. wired headset on iPhone 6.
     // TODO(henrika): add support for stereo if needed.
     _inputNumberOfChannels = kRTCAudioSessionPreferredNumberOfChannels;
@@ skipping 26 matching lines @@
   }
 }
 
 + (void)setWebRTCConfiguration:(RTCAudioSessionConfiguration *)configuration {
   @synchronized(self) {
     gWebRTCConfiguration = configuration;
   }
 }
 
 @end