Index: webrtc/modules/audio_coding/test/opus_test.cc |
diff --git a/webrtc/modules/audio_coding/test/opus_test.cc b/webrtc/modules/audio_coding/test/opus_test.cc |
index 892eb37c3f6423424b153c377e57e8b1992e0f6b..551d57714519442a637daa61f5ff92639d02ca63 100644 |
--- a/webrtc/modules/audio_coding/test/opus_test.cc |
+++ b/webrtc/modules/audio_coding/test/opus_test.cc |
@@ -29,31 +29,31 @@ namespace webrtc { |
OpusTest::OpusTest() |
: acm_receiver_(AudioCodingModule::Create(0)), |
- channel_a2b_(NULL), |
+ channel_a2b_(nullptr), |
counter_(0), |
payload_type_(255), |
rtp_timestamp_(0) {} |
OpusTest::~OpusTest() { |
- if (channel_a2b_ != NULL) { |
+ if (channel_a2b_ != nullptr) { |
delete channel_a2b_; |
- channel_a2b_ = NULL; |
+ channel_a2b_ = nullptr; |
} |
- if (opus_mono_encoder_ != NULL) { |
+ if (opus_mono_encoder_ != nullptr) { |
WebRtcOpus_EncoderFree(opus_mono_encoder_); |
- opus_mono_encoder_ = NULL; |
+ opus_mono_encoder_ = nullptr; |
} |
- if (opus_stereo_encoder_ != NULL) { |
+ if (opus_stereo_encoder_ != nullptr) { |
WebRtcOpus_EncoderFree(opus_stereo_encoder_); |
- opus_stereo_encoder_ = NULL; |
+ opus_stereo_encoder_ = nullptr; |
} |
- if (opus_mono_decoder_ != NULL) { |
+ if (opus_mono_decoder_ != nullptr) { |
WebRtcOpus_DecoderFree(opus_mono_decoder_); |
- opus_mono_decoder_ = NULL; |
+ opus_mono_decoder_ = nullptr; |
} |
- if (opus_stereo_decoder_ != NULL) { |
+ if (opus_stereo_decoder_ != nullptr) { |
WebRtcOpus_DecoderFree(opus_stereo_decoder_); |
- opus_stereo_decoder_ = NULL; |
+ opus_stereo_decoder_ = nullptr; |
} |
} |
@@ -87,7 +87,7 @@ void OpusTest::Perform() { |
WebRtcOpus_DecoderInit(opus_mono_decoder_); |
WebRtcOpus_DecoderInit(opus_stereo_decoder_); |
- ASSERT_TRUE(acm_receiver_.get() != NULL); |
+ ASSERT_TRUE(acm_receiver_.get() != nullptr); |
EXPECT_EQ(0, acm_receiver_->InitializeReceiver()); |
// Register Opus stereo as receiving codec. |
@@ -326,7 +326,7 @@ void OpusTest::Run(TestPackStereo* channel, size_t channels, int bitrate, |
// Send data to the channel. "channel" will handle the loss simulation. |
channel->SendData(kAudioFrameSpeech, payload_type_, rtp_timestamp_, |
- bitstream, bitstream_len_byte, NULL); |
+ bitstream, bitstream_len_byte, nullptr); |
if (first_packet) { |
first_packet = false; |
start_time_stamp = rtp_timestamp_; |