From 922246a353acff8445d4b104f0271befff2bfa68 Mon Sep 17 00:00:00 2001 From: deadbeef Date: Sun, 26 Feb 2017 04:18:12 -0800 Subject: [PATCH] Replace NULL with nullptr or null in webrtc/audio/ and common_audio/. BUG=webrtc:7147 Review-Url: https://codereview.webrtc.org/2719733002 Cr-Commit-Position: refs/heads/master@{#16843} --- webrtc/audio/audio_receive_stream.cc | 2 +- webrtc/audio/audio_receive_stream_unittest.cc | 2 +- webrtc/audio/audio_send_stream_unittest.cc | 2 +- webrtc/call/audio_send_stream.cc | 2 +- webrtc/common_audio/fir_filter.cc | 4 ++-- webrtc/common_audio/resampler/resampler.cc | 14 ++++++------ .../common_audio/resampler/sinc_resampler.cc | 2 +- webrtc/common_audio/ring_buffer.h | 4 ++-- webrtc/common_audio/ring_buffer_unittest.cc | 22 +++++++++---------- .../signal_processing/include/real_fft.h | 4 ++-- .../include/signal_processing_library.h | 4 ++-- .../signal_processing/real_fft_unittest.cc | 6 ++--- webrtc/common_audio/vad/include/webrtc_vad.h | 4 ++-- webrtc/common_audio/vad/vad_core.h | 2 +- webrtc/common_audio/vad/vad_core_unittest.cc | 10 ++++----- webrtc/common_audio/wav_file.cc | 4 ++-- 16 files changed, 44 insertions(+), 44 deletions(-) diff --git a/webrtc/audio/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc index 9bc9468336..8dc14e850f 100644 --- a/webrtc/audio/audio_receive_stream.cc +++ b/webrtc/audio/audio_receive_stream.cc @@ -51,7 +51,7 @@ std::string AudioReceiveStream::Config::ToString() const { std::stringstream ss; ss << "{rtp: " << rtp.ToString(); ss << ", rtcp_send_transport: " - << (rtcp_send_transport ? "(Transport)" : "nullptr"); + << (rtcp_send_transport ? "(Transport)" : "null"); ss << ", voe_channel_id: " << voe_channel_id; if (!sync_group.empty()) { ss << ", sync_group: " << sync_group; diff --git a/webrtc/audio/audio_receive_stream_unittest.cc b/webrtc/audio/audio_receive_stream_unittest.cc index b28c62a627..e6b81a8bab 100644 --- a/webrtc/audio/audio_receive_stream_unittest.cc +++ b/webrtc/audio/audio_receive_stream_unittest.cc @@ -225,7 +225,7 @@ TEST(AudioReceiveStreamTest, ConfigToString) { "{rtp: {remote_ssrc: 1234, local_ssrc: 5678, transport_cc: off, nack: " "{rtp_history_ms: 0}, extensions: [{uri: " "urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 3}]}, " - "rtcp_send_transport: nullptr, voe_channel_id: 2}", + "rtcp_send_transport: null, voe_channel_id: 2}", config.ToString()); } diff --git a/webrtc/audio/audio_send_stream_unittest.cc b/webrtc/audio/audio_send_stream_unittest.cc index 9ae0a87999..8e8b0e965e 100644 --- a/webrtc/audio/audio_send_stream_unittest.cc +++ b/webrtc/audio/audio_send_stream_unittest.cc @@ -276,7 +276,7 @@ TEST(AudioSendStreamTest, ConfigToString) { EXPECT_EQ( "{rtp: {ssrc: 1234, extensions: [{uri: " "urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 2}], nack: " - "{rtp_history_ms: 0}, c_name: foo_name}, send_transport: nullptr, " + "{rtp_history_ms: 0}, c_name: foo_name}, send_transport: null, " "voe_channel_id: 1, min_bitrate_bps: 12000, max_bitrate_bps: 34000, " "send_codec_spec: {nack_enabled: true, transport_cc_enabled: false, " "enable_codec_fec: true, enable_opus_dtx: false, opus_max_playback_rate: " diff --git a/webrtc/call/audio_send_stream.cc b/webrtc/call/audio_send_stream.cc index 8b6dd9e416..6091462470 100644 --- a/webrtc/call/audio_send_stream.cc +++ b/webrtc/call/audio_send_stream.cc @@ -40,7 +40,7 @@ AudioSendStream::Config::~Config() = default; std::string AudioSendStream::Config::ToString() const { std::stringstream ss; ss << "{rtp: " << rtp.ToString(); - ss << ", send_transport: " << (send_transport ? "(Transport)" : "nullptr"); + ss << ", send_transport: " << (send_transport ? "(Transport)" : "null"); ss << ", voe_channel_id: " << voe_channel_id; ss << ", min_bitrate_bps: " << min_bitrate_bps; ss << ", max_bitrate_bps: " << max_bitrate_bps; diff --git a/webrtc/common_audio/fir_filter.cc b/webrtc/common_audio/fir_filter.cc index d0a2d2f3ee..5f5e210ce1 100644 --- a/webrtc/common_audio/fir_filter.cc +++ b/webrtc/common_audio/fir_filter.cc @@ -40,10 +40,10 @@ FIRFilter* FIRFilter::Create(const float* coefficients, size_t max_input_length) { if (!coefficients || coefficients_length <= 0 || max_input_length <= 0) { RTC_NOTREACHED(); - return NULL; + return nullptr; } - FIRFilter* filter = NULL; + FIRFilter* filter = nullptr; // If we know the minimum architecture at compile time, avoid CPU detection. #if defined(WEBRTC_ARCH_X86_FAMILY) #if defined(__SSE2__) diff --git a/webrtc/common_audio/resampler/resampler.cc b/webrtc/common_audio/resampler/resampler.cc index 7c690fc47a..c91a5de9ce 100644 --- a/webrtc/common_audio/resampler/resampler.cc +++ b/webrtc/common_audio/resampler/resampler.cc @@ -101,37 +101,37 @@ int Resampler::Reset(int inFreq, int outFreq, size_t num_channels) if (state1_) { free(state1_); - state1_ = NULL; + state1_ = nullptr; } if (state2_) { free(state2_); - state2_ = NULL; + state2_ = nullptr; } if (state3_) { free(state3_); - state3_ = NULL; + state3_ = nullptr; } if (in_buffer_) { free(in_buffer_); - in_buffer_ = NULL; + in_buffer_ = nullptr; } if (out_buffer_) { free(out_buffer_); - out_buffer_ = NULL; + out_buffer_ = nullptr; } if (slave_left_) { delete slave_left_; - slave_left_ = NULL; + slave_left_ = nullptr; } if (slave_right_) { delete slave_right_; - slave_right_ = NULL; + slave_right_ = nullptr; } in_buffer_size_ = 0; diff --git a/webrtc/common_audio/resampler/sinc_resampler.cc b/webrtc/common_audio/resampler/sinc_resampler.cc index 651bbf6954..1203f771f4 100644 --- a/webrtc/common_audio/resampler/sinc_resampler.cc +++ b/webrtc/common_audio/resampler/sinc_resampler.cc @@ -161,7 +161,7 @@ SincResampler::SincResampler(double io_sample_rate_ratio, input_buffer_(static_cast( AlignedMalloc(sizeof(float) * input_buffer_size_, 16))), #if defined(WEBRTC_CPU_DETECTION) - convolve_proc_(NULL), + convolve_proc_(nullptr), #endif r1_(input_buffer_.get()), r2_(input_buffer_.get() + kKernelSize / 2) { diff --git a/webrtc/common_audio/ring_buffer.h b/webrtc/common_audio/ring_buffer.h index 74951a8b2d..baa1df4fe0 100644 --- a/webrtc/common_audio/ring_buffer.h +++ b/webrtc/common_audio/ring_buffer.h @@ -31,7 +31,7 @@ typedef struct RingBuffer { char* data; } RingBuffer; -// Creates and initializes the buffer. Returns NULL on failure. +// Creates and initializes the buffer. Returns null on failure. RingBuffer* WebRtc_CreateBuffer(size_t element_count, size_t element_size); void WebRtc_InitBuffer(RingBuffer* handle); void WebRtc_FreeBuffer(void* handle); @@ -43,7 +43,7 @@ void WebRtc_FreeBuffer(void* handle); // user) and |data_ptr| points to the address of |data|. |data_ptr| is only // guaranteed to be valid until the next call to WebRtc_WriteBuffer(). // -// To force a copying to |data|, pass a NULL |data_ptr|. +// To force a copying to |data|, pass a null |data_ptr|. // // Returns number of elements read. size_t WebRtc_ReadBuffer(RingBuffer* handle, diff --git a/webrtc/common_audio/ring_buffer_unittest.cc b/webrtc/common_audio/ring_buffer_unittest.cc index d62e92d9bf..a5b53b027a 100644 --- a/webrtc/common_audio/ring_buffer_unittest.cc +++ b/webrtc/common_audio/ring_buffer_unittest.cc @@ -54,7 +54,7 @@ static void RandomStressTest(int** data_ptr) { const int kNumOps = 1000; const int kMaxBufferSize = 1000; - unsigned int seed = time(NULL); + unsigned int seed = time(nullptr); printf("seed=%u\n", seed); srand(seed); for (int i = 0; i < kNumTests; i++) { @@ -62,7 +62,7 @@ static void RandomStressTest(int** data_ptr) { std::unique_ptr write_data(new int[buffer_size]); std::unique_ptr read_data(new int[buffer_size]); scoped_ring_buffer buffer(WebRtc_CreateBuffer(buffer_size, sizeof(int))); - ASSERT_TRUE(buffer.get() != NULL); + ASSERT_TRUE(buffer.get() != nullptr); WebRtc_InitBuffer(buffer.get()); int buffer_consumed = 0; int write_element = 0; @@ -105,12 +105,12 @@ static void RandomStressTest(int** data_ptr) { } TEST(RingBufferTest, RandomStressTest) { - int* data_ptr = NULL; + int* data_ptr = nullptr; RandomStressTest(&data_ptr); } TEST(RingBufferTest, RandomStressTestWithNullPtr) { - RandomStressTest(NULL); + RandomStressTest(nullptr); } TEST(RingBufferTest, PassingNulltoReadBufferForcesMemcpy) { @@ -120,7 +120,7 @@ TEST(RingBufferTest, PassingNulltoReadBufferForcesMemcpy) { int* data_ptr; scoped_ring_buffer buffer(WebRtc_CreateBuffer(kDataSize, sizeof(int))); - ASSERT_TRUE(buffer.get() != NULL); + ASSERT_TRUE(buffer.get() != nullptr); WebRtc_InitBuffer(buffer.get()); SetIncrementingData(write_data, kDataSize, 0); @@ -133,17 +133,17 @@ TEST(RingBufferTest, PassingNulltoReadBufferForcesMemcpy) { CheckIncrementingData(read_data, kDataSize, kDataSize); EXPECT_EQ(kDataSize, WebRtc_WriteBuffer(buffer.get(), write_data, kDataSize)); - EXPECT_EQ(kDataSize, WebRtc_ReadBuffer(buffer.get(), NULL, read_data, - kDataSize)); - // Passing NULL forces a memcpy, so |read_data| is now updated. + EXPECT_EQ(kDataSize, + WebRtc_ReadBuffer(buffer.get(), nullptr, read_data, kDataSize)); + // Passing null forces a memcpy, so |read_data| is now updated. CheckIncrementingData(read_data, kDataSize, 0); } TEST(RingBufferTest, CreateHandlesErrors) { - EXPECT_TRUE(WebRtc_CreateBuffer(0, 1) == NULL); - EXPECT_TRUE(WebRtc_CreateBuffer(1, 0) == NULL); + EXPECT_TRUE(WebRtc_CreateBuffer(0, 1) == nullptr); + EXPECT_TRUE(WebRtc_CreateBuffer(1, 0) == nullptr); RingBuffer* buffer = WebRtc_CreateBuffer(1, 1); - EXPECT_TRUE(buffer != NULL); + EXPECT_TRUE(buffer != nullptr); WebRtc_FreeBuffer(buffer); } diff --git a/webrtc/common_audio/signal_processing/include/real_fft.h b/webrtc/common_audio/signal_processing/include/real_fft.h index e7942f04c4..ed0db767aa 100644 --- a/webrtc/common_audio/signal_processing/include/real_fft.h +++ b/webrtc/common_audio/signal_processing/include/real_fft.h @@ -57,7 +57,7 @@ void WebRtcSpl_FreeRealFFT(struct RealFFT* self); // // Return Value: // 0 - FFT calculation is successful. -// -1 - Error with bad arguments (NULL pointers). +// -1 - Error with bad arguments (null pointers). int WebRtcSpl_RealForwardFFT(struct RealFFT* self, const int16_t* real_data_in, int16_t* complex_data_out); @@ -85,7 +85,7 @@ int WebRtcSpl_RealForwardFFT(struct RealFFT* self, // 0 or a positive number - a value that the elements in the |real_data_out| // should be shifted left with in order to get // correct physical values. -// -1 - Error with bad arguments (NULL pointers). +// -1 - Error with bad arguments (null pointers). int WebRtcSpl_RealInverseFFT(struct RealFFT* self, const int16_t* complex_data_in, int16_t* real_data_out); diff --git a/webrtc/common_audio/signal_processing/include/signal_processing_library.h b/webrtc/common_audio/signal_processing/include/signal_processing_library.h index 89a281f410..7fa68e0422 100644 --- a/webrtc/common_audio/signal_processing/include/signal_processing_library.h +++ b/webrtc/common_audio/signal_processing/include/signal_processing_library.h @@ -343,8 +343,8 @@ void WebRtcSpl_ScaleAndAddVectors(const int16_t* in_vector1, // // Output: // - out_vector : Output vector -// Return value : 0 if OK, -1 if (in_vector1 == NULL -// || in_vector2 == NULL || out_vector == NULL +// Return value : 0 if OK, -1 if (in_vector1 == null +// || in_vector2 == null || out_vector == null // || length <= 0 || right_shift < 0). typedef int (*ScaleAndAddVectorsWithRound)(const int16_t* in_vector1, int16_t in_vector1_scale, diff --git a/webrtc/common_audio/signal_processing/real_fft_unittest.cc b/webrtc/common_audio/signal_processing/real_fft_unittest.cc index 8b1c431078..282342230f 100644 --- a/webrtc/common_audio/signal_processing/real_fft_unittest.cc +++ b/webrtc/common_audio/signal_processing/real_fft_unittest.cc @@ -42,9 +42,9 @@ class RealFFTTest : public ::testing::Test { TEST_F(RealFFTTest, CreateFailsOnBadInput) { RealFFT* fft = WebRtcSpl_CreateRealFFT(11); - EXPECT_TRUE(fft == NULL); + EXPECT_TRUE(fft == nullptr); fft = WebRtcSpl_CreateRealFFT(-1); - EXPECT_TRUE(fft == NULL); + EXPECT_TRUE(fft == nullptr); } TEST_F(RealFFTTest, RealAndComplexMatch) { @@ -64,7 +64,7 @@ TEST_F(RealFFTTest, RealAndComplexMatch) { // Create and run real forward FFT. RealFFT* fft = WebRtcSpl_CreateRealFFT(kOrder); - EXPECT_TRUE(fft != NULL); + EXPECT_TRUE(fft != nullptr); EXPECT_EQ(0, WebRtcSpl_RealForwardFFT(fft, real_fft_time, real_fft_freq)); // Run complex forward FFT. diff --git a/webrtc/common_audio/vad/include/webrtc_vad.h b/webrtc/common_audio/vad/include/webrtc_vad.h index 91308eef12..7c11b12964 100644 --- a/webrtc/common_audio/vad/include/webrtc_vad.h +++ b/webrtc/common_audio/vad/include/webrtc_vad.h @@ -39,7 +39,7 @@ void WebRtcVad_Free(VadInst* handle); // - handle [i/o] : Instance that should be initialized. // // returns : 0 - (OK), -// -1 - (NULL pointer or Default mode could not be set). +// -1 - (null pointer or Default mode could not be set). int WebRtcVad_Init(VadInst* handle); // Sets the VAD operating mode. A more aggressive (higher mode) VAD is more @@ -51,7 +51,7 @@ int WebRtcVad_Init(VadInst* handle); // - mode [i] : Aggressiveness mode (0, 1, 2, or 3). // // returns : 0 - (OK), -// -1 - (NULL pointer, mode could not be set or the VAD instance +// -1 - (null pointer, mode could not be set or the VAD instance // has not been initialized). int WebRtcVad_set_mode(VadInst* handle, int mode); diff --git a/webrtc/common_audio/vad/vad_core.h b/webrtc/common_audio/vad/vad_core.h index b38c515ea1..0a76d967dd 100644 --- a/webrtc/common_audio/vad/vad_core.h +++ b/webrtc/common_audio/vad/vad_core.h @@ -60,7 +60,7 @@ typedef struct VadInstT_ // // - self [i/o] : Instance that should be initialized // -// returns : 0 (OK), -1 (NULL pointer in or if the default mode can't be +// returns : 0 (OK), -1 (null pointer in or if the default mode can't be // set) int WebRtcVad_InitCore(VadInstT* self); diff --git a/webrtc/common_audio/vad/vad_core_unittest.cc b/webrtc/common_audio/vad/vad_core_unittest.cc index 43673ee62a..ba53c560e2 100644 --- a/webrtc/common_audio/vad/vad_core_unittest.cc +++ b/webrtc/common_audio/vad/vad_core_unittest.cc @@ -24,10 +24,10 @@ TEST_F(VadTest, InitCore) { // Test WebRtcVad_InitCore(). VadInstT* self = reinterpret_cast(malloc(sizeof(VadInstT))); - // NULL pointer test. - EXPECT_EQ(-1, WebRtcVad_InitCore(NULL)); + // null pointer test. + EXPECT_EQ(-1, WebRtcVad_InitCore(nullptr)); - // Verify return = 0 for non-NULL pointer. + // Verify return = 0 for non-null pointer. EXPECT_EQ(0, WebRtcVad_InitCore(self)); // Verify init_flag is set. EXPECT_EQ(42, self->init_flag); @@ -38,7 +38,7 @@ TEST_F(VadTest, InitCore) { TEST_F(VadTest, set_mode_core) { VadInstT* self = reinterpret_cast(malloc(sizeof(VadInstT))); - // TODO(bjornv): Add NULL pointer check if we take care of it in + // TODO(bjornv): Add null pointer check if we take care of it in // vad_core.c ASSERT_EQ(0, WebRtcVad_InitCore(self)); @@ -58,7 +58,7 @@ TEST_F(VadTest, CalcVad) { VadInstT* self = reinterpret_cast(malloc(sizeof(VadInstT))); int16_t speech[kMaxFrameLength]; - // TODO(bjornv): Add NULL pointer check if we take care of it in + // TODO(bjornv): Add null pointer check if we take care of it in // vad_core.c // Test WebRtcVad_CalcVadXXkhz() diff --git a/webrtc/common_audio/wav_file.cc b/webrtc/common_audio/wav_file.cc index 2b9098a6cd..0f37f9cdea 100644 --- a/webrtc/common_audio/wav_file.cc +++ b/webrtc/common_audio/wav_file.cc @@ -107,7 +107,7 @@ size_t WavReader::ReadSamples(size_t num_samples, float* samples) { void WavReader::Close() { RTC_CHECK_EQ(0, fclose(file_handle_)); - file_handle_ = NULL; + file_handle_ = nullptr; } WavWriter::WavWriter(const std::string& filename, int sample_rate, @@ -170,7 +170,7 @@ void WavWriter::Close() { kBytesPerSample, num_samples_); RTC_CHECK_EQ(1, fwrite(header, kWavHeaderSize, 1, file_handle_)); RTC_CHECK_EQ(0, fclose(file_handle_)); - file_handle_ = NULL; + file_handle_ = nullptr; } } // namespace webrtc