Avoids hitting DCHECK in OpenSL ES player

TBR=glaznev
BUG=NONE

Review URL: https://codereview.webrtc.org/1467433002 .

Cr-Commit-Position: refs/heads/master@{#10727}
diff --git a/webrtc/modules/audio_device/android/opensles_player.cc b/webrtc/modules/audio_device/android/opensles_player.cc
index d1edef2..9dc001c 100644
--- a/webrtc/modules/audio_device/android/opensles_player.cc
+++ b/webrtc/modules/audio_device/android/opensles_player.cc
@@ -245,7 +245,7 @@
       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