Resolves issue with bad audio using BT headsets on iOS

webrtc/modules/audio_device/ios/objc/RTCAudioSession.mm

 /*
  *  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.
  */
 
@@ skipping 618 matching lines @@
   // TODO(tkchin): Figure out if this is really necessary.
   if (!self.inputAvailable) {
     RTCLogError(@"No audio input path is available!");
     [self unconfigureWebRTCSession:nil];
     if (outError) {
       *outError = [self configurationErrorWithDescription:@"No input path."];
     }
     return NO;
   }
 
+  // It can happen (e.g. in combination with BT devices) that the attempt to set

[tkchin_webrtc, 2016/06/14 19:13:28]

It seems like we should do this on route changes, and not on generic activation.

[henrika_webrtc, 2016/06/15 13:46:58]

But the problem is that it does not work since we don't get any route changes at
the second call. We only get it the first call and then everything works as is.
I did not come up with any other method that helped. Note that we used to
deregister all observers when closing a call before but we don't now. If you are
OK with adding back the old style, it might resolve the issue. I tried but it
seems like we now depend on these notifiers existing from start to end of the
app.

+  // the preferred sample rate for WebRTC (48kHz) fails. If so, disable the
+  // current session and make a new configuration attempt using the sample rate
+  // that worked. 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 b/29185303 but it has only been tested on

[tkchin_webrtc, 2016/06/14 19:13:28]

Don't reference buganizer in webrtc. File a corresponding webrtc bug if needed.

[henrika_webrtc, 2016/06/15 13:46:58]

Done.

+  // a limited set of iOS devices and BT devices.
+  double session_sample_rate = self.sampleRate;
+  double preferred_sample_rate = webRTCConfig.sampleRate;
+  if (session_sample_rate != preferred_sample_rate) {
+    RTCLogWarning(
+        @"Current sample rate (%.2f) is not the preferred rate (%.2f)",
+        session_sample_rate, preferred_sample_rate);
+    // Perform a second configuration attempt but use a sample rate that is
+    // known to work this time instead (e.g. 8 or 16kHz in combination with BT
+    // headsets).
+    RTCLog(@"Configuring audio session for WebRTC using new sample rate.");
+    [self unconfigureWebRTCSession:nil];

[tkchin_webrtc, 2016/06/14 19:13:28]

this won't work if there are multiple activations ([self setActive:NO] won't
actually deactivate the session unless the activation count is 1). You might
want to directly deactivate avaudiosession.

[henrika_webrtc, 2016/06/15 13:46:58]

Missed that. I actually started out by making the change of sample rate without
trying to deactivate and that worked as well. However, Apple recommends doing it
while session is inactive. I might not have full control of the current
reference counting scheme but let me try to come up with a new version.

+    webRTCConfig.sampleRate = session_sample_rate;
+    [RTCAudioSessionConfiguration setWebRTCConfiguration:webRTCConfig];

[tkchin_webrtc, 2016/06/14 19:13:28]

I don't think we should set the default because we will always want to try the
default first and see if it is usable.

[henrika_webrtc, 2016/06/15 13:46:58]

Agree that it is not a clean solution. Will rethink.

+    if (![self setConfiguration:webRTCConfig active:YES error:&error]) {
+      RTCLogError(@"Failed to set WebRTC audio configuration: %@",
+                  error.localizedDescription);
+      [self unconfigureWebRTCSession:nil];
+      if (outError) {
+        *outError = error;
+      }
+        return NO;
+      }
+      // Restore original WebRTC configuration parameters.
+      webRTCConfig.sampleRate = preferred_sample_rate;
+      [RTCAudioSessionConfiguration setWebRTCConfiguration:webRTCConfig];
+   }
+
   return YES;
 }
 
 - (BOOL)unconfigureWebRTCSession:(NSError **)outError {
   if (outError) {
     *outError = nil;
   }
   if (![self checkLock:outError]) {
     return NO;
   }
@@ skipping 114 matching lines @@
 - (void)notifyDidStopPlayOrRecord {
   for (auto delegate : self.delegates) {
     SEL sel = @selector(audioSessionDidStopPlayOrRecord:);
     if ([delegate respondsToSelector:sel]) {
       [delegate audioSessionDidStopPlayOrRecord:self];
     }
   }
 }
 
 @end