From 76a31ca3d4ef10cd19cfac557b783e438907ef01 Mon Sep 17 00:00:00 2001 From: henrika Date: Fri, 20 Nov 2015 13:40:44 +0100 Subject: [PATCH] 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} --- webrtc/modules/audio_device/android/opensles_player.cc | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/webrtc/modules/audio_device/android/opensles_player.cc b/webrtc/modules/audio_device/android/opensles_player.cc index d1edef227e..9dc001c44d 100644 --- a/webrtc/modules/audio_device/android/opensles_player.cc +++ b/webrtc/modules/audio_device/android/opensles_player.cc @@ -245,7 +245,7 @@ void OpenSLESPlayer::AllocateDataBuffers() { 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