Avoids hitting DCHECK in OpenSL ES player

webrtc/modules/audio_device/android/opensles_player.cc

 /*
  *  Copyright (c) 2015 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 227 matching lines @@
   RTC_CHECK(audio_device_buffer_);
   // Don't use the lowest possible size as native buffer size. Instead,
   // use 10ms to better match the frame size that WebRTC uses. It will result
   // in a reduced risk for audio glitches and also in a more "clean" sequence
   // of callbacks from the OpenSL ES thread in to WebRTC when asking for audio
   // to render.
   ALOGD("lowest possible buffer size: %" PRIuS,
       audio_parameters_.GetBytesPerBuffer());
   bytes_per_buffer_ = audio_parameters_.GetBytesPerFrame() *
       audio_parameters_.frames_per_10ms_buffer();
-  RTC_DCHECK_GT(bytes_per_buffer_, audio_parameters_.GetBytesPerBuffer());
+  RTC_DCHECK_GE(bytes_per_buffer_, audio_parameters_.GetBytesPerBuffer());
   ALOGD("native buffer size: %" PRIuS, bytes_per_buffer_);
   // Create a modified audio buffer class which allows us to ask for any number
   // of samples (and not only multiple of 10ms) to match the native OpenSL ES
   // buffer size.
   fine_buffer_.reset(new FineAudioBuffer(audio_device_buffer_,
                                          bytes_per_buffer_,
                                          audio_parameters_.sample_rate()));
   // Each buffer must be of this size to avoid unnecessary memcpy while caching
   // data between successive callbacks.
   const size_t required_buffer_size =
@@ skipping 199 matching lines @@
   RTC_DCHECK(player_);
   SLuint32 state;
   SLresult err = (*player_)->GetPlayState(player_, &state);
   if (SL_RESULT_SUCCESS != err) {
     ALOGE("GetPlayState failed: %d", err);
   }
   return state;
 }
 
 }  // namespace webrtc