share all the needed linux code with OpenBSD in chrome and content

content/renderer/media/webrtc_audio_device_impl.cc

 // Copyright (c) 2011 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 
 #include "base/bind.h"
 #include "base/string_util.h"
 #include "content/common/view_messages.h"
 #include "content/renderer/render_thread_impl.h"
@@ skipping 348 matching lines @@
   // Use different buffer sizes depending on the current hardware sample rate.
   if (output_sample_rate == 48000) {
     input_buffer_size = 480;
     output_buffer_size = 480;
   } else {
     // We do run at 44.1kHz at the actual audio layer, but ask for frames
     // at 44.0kHz to ensure that we can feed them to the webrtc::VoiceEngine.
     input_buffer_size = 440;
     output_buffer_size = 440;
   }
-#elif defined(OS_LINUX)
+#elif defined(OS_LINUX) || defined(OS_OPENBSD)
   if (output_sample_rate != 48000) {
     DLOG(ERROR) << "Only 48kHz sample rate is supported on Linux.";
     return -1;
   }
   input_channels = 1;
   output_channels = 1;
 
   // Based on tests using the current ALSA implementation in Chrome, we have
   // found that the best combination is 20ms on the input side and 10ms on the
   // output side.
@@ skipping 564 matching lines @@
 }
 
 int32_t WebRtcAudioDeviceImpl::GetLoudspeakerStatus(bool* enabled) const {
   NOTIMPLEMENTED();
   return -1;
 }
 
 void WebRtcAudioDeviceImpl::SetSessionId(int session_id) {
   session_id_ = session_id;
 }