From 2d014be554f63128ceed8ecba007dff89595b4ae Mon Sep 17 00:00:00 2001
From: henrika <henrika@webrtc.org>
Date: Thu, 16 Jun 2016 14:26:55 +0200
Subject: [PATCH] Resolves issue with bad audio using BT headsets on iOS.

BUG=webrtc:6004
R=tkchin@webrtc.org

Review URL: https://codereview.webrtc.org/2063733002 .

Cr-Commit-Position: refs/heads/master@{#13165}
---
 .../audio_device/ios/audio_device_ios.mm      | 14 +++++++++++
 .../audio_device/ios/objc/RTCAudioSession.mm  | 25 +++++++++++++++++++
 2 files changed, 39 insertions(+)

diff --git a/webrtc/modules/audio_device/ios/audio_device_ios.mm b/webrtc/modules/audio_device/ios/audio_device_ios.mm
index e564540595..0bb997db53 100644
--- a/webrtc/modules/audio_device/ios/audio_device_ios.mm
+++ b/webrtc/modules/audio_device/ios/audio_device_ios.mm
@@ -431,6 +431,20 @@ 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.
diff --git a/webrtc/modules/audio_device/ios/objc/RTCAudioSession.mm b/webrtc/modules/audio_device/ios/objc/RTCAudioSession.mm
index 7ef5110f47..159d34e993 100644
--- a/webrtc/modules/audio_device/ios/objc/RTCAudioSession.mm
+++ b/webrtc/modules/audio_device/ios/objc/RTCAudioSession.mm
@@ -636,6 +636,31 @@ NSInteger const kRTCAudioSessionErrorConfiguration = -2;
     return NO;
   }
 
+  // It can happen (e.g. in combination with BT devices) that the attempt to set
+  // the preferred sample rate for WebRTC (48kHz) fails. If so, make a new
+  // configuration attempt using the sample rate that worked using the active
+  // audio session. A typical case is that only 8 or 16kHz can be set, e.g. in
+  // combination with BT headsets. Using this "trick" seems to avoid a state
+  // where Core Audio asks for a different number of audio frames than what the
+  // session's I/O buffer duration corresponds to.
+  // TODO(henrika): this fix resolves bugs.webrtc.org/6004 but it has only been
+  // tested on a limited set of iOS devices and BT devices.
+  double sessionSampleRate = self.sampleRate;
+  double preferredSampleRate = webRTCConfig.sampleRate;
+  if (sessionSampleRate != preferredSampleRate) {
+    RTCLogWarning(
+        @"Current sample rate (%.2f) is not the preferred rate (%.2f)",
+        sessionSampleRate, preferredSampleRate);
+    if (![self setPreferredSampleRate:sessionSampleRate
+                                error:&error]) {
+      RTCLogError(@"Failed to set preferred sample rate: %@",
+                  error.localizedDescription);
+      if (outError) {
+        *outError = error;
+      }
+    }
+  }
+
   return YES;
 }