Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away
BUG=
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36229004
Cr-Commit-Position: refs/heads/master@{#8517}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8517 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/neteq/audio_classifier.h b/webrtc/modules/audio_coding/neteq/audio_classifier.h
index 7bf8513..2812ea2 100644
--- a/webrtc/modules/audio_coding/neteq/audio_classifier.h
+++ b/webrtc/modules/audio_coding/neteq/audio_classifier.h
@@ -17,7 +17,7 @@
#include "opus_private.h"
}
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/base/scoped_ptr.h"
#include "webrtc/typedefs.h"
namespace webrtc {
diff --git a/webrtc/modules/audio_coding/neteq/audio_classifier_unittest.cc b/webrtc/modules/audio_coding/neteq/audio_classifier_unittest.cc
index 530044e..e4db3a3 100644
--- a/webrtc/modules/audio_coding/neteq/audio_classifier_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/audio_classifier_unittest.cc
@@ -39,7 +39,7 @@
const std::string& data_filename,
size_t channels) {
AudioClassifier classifier;
- scoped_ptr<int16_t[]> in(new int16_t[channels * kFrameSize]);
+ rtc::scoped_ptr<int16_t[]> in(new int16_t[channels * kFrameSize]);
bool is_music_ref;
FILE* audio_file = fopen(audio_filename.c_str(), "rb");
diff --git a/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc b/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
index 1f0e881..4c326cc 100644
--- a/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
@@ -17,6 +17,7 @@
#include <vector>
#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h"
#include "webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h"
#include "webrtc/modules/audio_coding/codecs/ilbc/interface/audio_encoder_ilbc.h"
@@ -26,7 +27,6 @@
#include "webrtc/modules/audio_coding/codecs/pcm16b/include/audio_encoder_pcm16b.h"
#include "webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h"
#include "webrtc/system_wrappers/interface/data_log.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/test/testsupport/fileutils.h"
namespace webrtc {
@@ -139,7 +139,7 @@
const size_t samples_per_10ms = audio_encoder_->SampleRateHz() / 100;
CHECK_EQ(samples_per_10ms * audio_encoder_->Num10MsFramesInNextPacket(),
input_len_samples);
- scoped_ptr<int16_t[]> interleaved_input(
+ rtc::scoped_ptr<int16_t[]> interleaved_input(
new int16_t[channels_ * samples_per_10ms]);
for (int i = 0; i < audio_encoder_->Num10MsFramesInNextPacket(); ++i) {
EXPECT_EQ(0u, encoded_info_.encoded_bytes);
@@ -213,21 +213,21 @@
// decode. Verifies that the decoded result is the same.
void ReInitTest() {
InitEncoder();
- scoped_ptr<int16_t[]> input(new int16_t[frame_size_]);
+ rtc::scoped_ptr<int16_t[]> input(new int16_t[frame_size_]);
ASSERT_TRUE(
input_audio_.Read(frame_size_, codec_input_rate_hz_, input.get()));
size_t enc_len = EncodeFrame(input.get(), frame_size_, encoded_);
size_t dec_len;
AudioDecoder::SpeechType speech_type1, speech_type2;
EXPECT_EQ(0, decoder_->Init());
- scoped_ptr<int16_t[]> output1(new int16_t[frame_size_ * channels_]);
+ rtc::scoped_ptr<int16_t[]> output1(new int16_t[frame_size_ * channels_]);
dec_len = decoder_->Decode(encoded_, enc_len, codec_input_rate_hz_,
output1.get(), &speech_type1);
ASSERT_LE(dec_len, frame_size_ * channels_);
EXPECT_EQ(frame_size_ * channels_, dec_len);
// Re-init decoder and decode again.
EXPECT_EQ(0, decoder_->Init());
- scoped_ptr<int16_t[]> output2(new int16_t[frame_size_ * channels_]);
+ rtc::scoped_ptr<int16_t[]> output2(new int16_t[frame_size_ * channels_]);
dec_len = decoder_->Decode(encoded_, enc_len, codec_input_rate_hz_,
output2.get(), &speech_type2);
ASSERT_LE(dec_len, frame_size_ * channels_);
@@ -241,13 +241,13 @@
// Call DecodePlc and verify that the correct number of samples is produced.
void DecodePlcTest() {
InitEncoder();
- scoped_ptr<int16_t[]> input(new int16_t[frame_size_]);
+ rtc::scoped_ptr<int16_t[]> input(new int16_t[frame_size_]);
ASSERT_TRUE(
input_audio_.Read(frame_size_, codec_input_rate_hz_, input.get()));
size_t enc_len = EncodeFrame(input.get(), frame_size_, encoded_);
AudioDecoder::SpeechType speech_type;
EXPECT_EQ(0, decoder_->Init());
- scoped_ptr<int16_t[]> output(new int16_t[frame_size_ * channels_]);
+ rtc::scoped_ptr<int16_t[]> output(new int16_t[frame_size_ * channels_]);
size_t dec_len = decoder_->Decode(encoded_, enc_len, codec_input_rate_hz_,
output.get(), &speech_type);
EXPECT_EQ(frame_size_ * channels_, dec_len);
@@ -268,7 +268,7 @@
const int payload_type_;
AudioEncoder::EncodedInfo encoded_info_;
AudioDecoder* decoder_;
- scoped_ptr<AudioEncoder> audio_encoder_;
+ rtc::scoped_ptr<AudioEncoder> audio_encoder_;
};
class AudioDecoderPcmUTest : public AudioDecoderTest {
@@ -332,13 +332,13 @@
// not return any data. It simply resets a few states and returns 0.
void DecodePlcTest() {
InitEncoder();
- scoped_ptr<int16_t[]> input(new int16_t[frame_size_]);
+ rtc::scoped_ptr<int16_t[]> input(new int16_t[frame_size_]);
ASSERT_TRUE(
input_audio_.Read(frame_size_, codec_input_rate_hz_, input.get()));
size_t enc_len = EncodeFrame(input.get(), frame_size_, encoded_);
AudioDecoder::SpeechType speech_type;
EXPECT_EQ(0, decoder_->Init());
- scoped_ptr<int16_t[]> output(new int16_t[frame_size_ * channels_]);
+ rtc::scoped_ptr<int16_t[]> output(new int16_t[frame_size_ * channels_]);
size_t dec_len = decoder_->Decode(encoded_, enc_len, codec_input_rate_hz_,
output.get(), &speech_type);
EXPECT_EQ(frame_size_, dec_len);
diff --git a/webrtc/modules/audio_coding/neteq/audio_vector.cc b/webrtc/modules/audio_coding/neteq/audio_vector.cc
index ef24ea2..d0f1aca 100644
--- a/webrtc/modules/audio_coding/neteq/audio_vector.cc
+++ b/webrtc/modules/audio_coding/neteq/audio_vector.cc
@@ -155,7 +155,7 @@
void AudioVector::Reserve(size_t n) {
if (capacity_ < n) {
- scoped_ptr<int16_t[]> temp_array(new int16_t[n]);
+ rtc::scoped_ptr<int16_t[]> temp_array(new int16_t[n]);
memcpy(temp_array.get(), array_.get(), Size() * sizeof(int16_t));
array_.swap(temp_array);
capacity_ = n;
diff --git a/webrtc/modules/audio_coding/neteq/audio_vector.h b/webrtc/modules/audio_coding/neteq/audio_vector.h
index 3e025a4..28e53ee 100644
--- a/webrtc/modules/audio_coding/neteq/audio_vector.h
+++ b/webrtc/modules/audio_coding/neteq/audio_vector.h
@@ -14,7 +14,7 @@
#include <string.h> // Access to size_t.
#include "webrtc/base/constructormagic.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/base/scoped_ptr.h"
#include "webrtc/typedefs.h"
namespace webrtc {
@@ -108,7 +108,7 @@
void Reserve(size_t n);
- scoped_ptr<int16_t[]> array_;
+ rtc::scoped_ptr<int16_t[]> array_;
size_t first_free_ix_; // The first index after the last sample in array_.
// Note that this index may point outside of array_.
size_t capacity_; // Allocated number of samples in the array.
diff --git a/webrtc/modules/audio_coding/neteq/background_noise.h b/webrtc/modules/audio_coding/neteq/background_noise.h
index 5c9f39b..fd4e6a5 100644
--- a/webrtc/modules/audio_coding/neteq/background_noise.h
+++ b/webrtc/modules/audio_coding/neteq/background_noise.h
@@ -14,9 +14,9 @@
#include <string.h> // size_t
#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
#include "webrtc/modules/audio_coding/neteq/interface/neteq.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/typedefs.h"
namespace webrtc {
@@ -126,7 +126,7 @@
int32_t residual_energy);
size_t num_channels_;
- scoped_ptr<ChannelParameters[]> channel_parameters_;
+ rtc::scoped_ptr<ChannelParameters[]> channel_parameters_;
bool initialized_;
NetEq::BackgroundNoiseMode mode_;
diff --git a/webrtc/modules/audio_coding/neteq/expand.h b/webrtc/modules/audio_coding/neteq/expand.h
index 1acf951..7b41114 100644
--- a/webrtc/modules/audio_coding/neteq/expand.h
+++ b/webrtc/modules/audio_coding/neteq/expand.h
@@ -14,8 +14,8 @@
#include <assert.h>
#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/typedefs.h"
namespace webrtc {
@@ -167,7 +167,7 @@
int lag_index_direction_;
int current_lag_index_;
bool stop_muting_;
- scoped_ptr<ChannelParameters[]> channel_parameters_;
+ rtc::scoped_ptr<ChannelParameters[]> channel_parameters_;
DISALLOW_COPY_AND_ASSIGN(Expand);
};
diff --git a/webrtc/modules/audio_coding/neteq/merge.cc b/webrtc/modules/audio_coding/neteq/merge.cc
index d3d8077..bc22000 100644
--- a/webrtc/modules/audio_coding/neteq/merge.cc
+++ b/webrtc/modules/audio_coding/neteq/merge.cc
@@ -15,12 +15,12 @@
#include <algorithm> // min, max
+#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
#include "webrtc/modules/audio_coding/neteq/dsp_helper.h"
#include "webrtc/modules/audio_coding/neteq/expand.h"
#include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
namespace webrtc {
@@ -310,7 +310,8 @@
// Normalize correlation to 14 bits and copy to a 16-bit array.
const int pad_length = static_cast<int>(expand_->overlap_length() - 1);
const int correlation_buffer_size = 2 * pad_length + kMaxCorrelationLength;
- scoped_ptr<int16_t[]> correlation16(new int16_t[correlation_buffer_size]);
+ rtc::scoped_ptr<int16_t[]> correlation16(
+ new int16_t[correlation_buffer_size]);
memset(correlation16.get(), 0, correlation_buffer_size * sizeof(int16_t));
int16_t* correlation_ptr = &correlation16[pad_length];
int32_t max_correlation = WebRtcSpl_MaxAbsValueW32(correlation,
diff --git a/webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc
index 0449044..8a382e9 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc
@@ -11,11 +11,11 @@
// Test to verify correct operation for externally created decoders.
#include "testing/gmock/include/gmock/gmock.h"
+#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h"
#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
#include "webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h"
#include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/test/testsupport/fileutils.h"
namespace webrtc {
@@ -148,16 +148,16 @@
int samples_per_ms() const { return samples_per_ms_; }
private:
- scoped_ptr<MockExternalPcm16B> external_decoder_;
+ rtc::scoped_ptr<MockExternalPcm16B> external_decoder_;
int samples_per_ms_;
size_t frame_size_samples_;
- scoped_ptr<test::RtpGenerator> rtp_generator_;
+ rtc::scoped_ptr<test::RtpGenerator> rtp_generator_;
int16_t* input_;
uint8_t* encoded_;
size_t payload_size_bytes_;
uint32_t last_send_time_;
uint32_t last_arrival_time_;
- scoped_ptr<test::InputAudioFile> input_file_;
+ rtc::scoped_ptr<test::InputAudioFile> input_file_;
WebRtcRTPHeader rtp_header_;
};
@@ -228,7 +228,7 @@
private:
int sample_rate_hz_;
- scoped_ptr<NetEq> neteq_internal_;
+ rtc::scoped_ptr<NetEq> neteq_internal_;
int16_t output_internal_[kMaxBlockSize];
int16_t output_[kMaxBlockSize];
};
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.h b/webrtc/modules/audio_coding/neteq/neteq_impl.h
index fa96512..b82b43e 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.h
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.h
@@ -14,6 +14,7 @@
#include <vector>
#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
#include "webrtc/modules/audio_coding/neteq/defines.h"
@@ -22,7 +23,6 @@
#include "webrtc/modules/audio_coding/neteq/random_vector.h"
#include "webrtc/modules/audio_coding/neteq/rtcp.h"
#include "webrtc/modules/audio_coding/neteq/statistics_calculator.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/typedefs.h"
namespace webrtc {
@@ -334,37 +334,40 @@
// Creates DecisionLogic object with the mode given by |playout_mode_|.
virtual void CreateDecisionLogic() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
- const scoped_ptr<CriticalSectionWrapper> crit_sect_;
- const scoped_ptr<BufferLevelFilter> buffer_level_filter_
+ const rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
+ const rtc::scoped_ptr<BufferLevelFilter> buffer_level_filter_
GUARDED_BY(crit_sect_);
- const scoped_ptr<DecoderDatabase> decoder_database_ GUARDED_BY(crit_sect_);
- const scoped_ptr<DelayManager> delay_manager_ GUARDED_BY(crit_sect_);
- const scoped_ptr<DelayPeakDetector> delay_peak_detector_
+ const rtc::scoped_ptr<DecoderDatabase> decoder_database_
GUARDED_BY(crit_sect_);
- const scoped_ptr<DtmfBuffer> dtmf_buffer_ GUARDED_BY(crit_sect_);
- const scoped_ptr<DtmfToneGenerator> dtmf_tone_generator_
+ const rtc::scoped_ptr<DelayManager> delay_manager_ GUARDED_BY(crit_sect_);
+ const rtc::scoped_ptr<DelayPeakDetector> delay_peak_detector_
GUARDED_BY(crit_sect_);
- const scoped_ptr<PacketBuffer> packet_buffer_ GUARDED_BY(crit_sect_);
- const scoped_ptr<PayloadSplitter> payload_splitter_ GUARDED_BY(crit_sect_);
- const scoped_ptr<TimestampScaler> timestamp_scaler_ GUARDED_BY(crit_sect_);
- const scoped_ptr<PostDecodeVad> vad_ GUARDED_BY(crit_sect_);
- const scoped_ptr<ExpandFactory> expand_factory_ GUARDED_BY(crit_sect_);
- const scoped_ptr<AccelerateFactory> accelerate_factory_
+ const rtc::scoped_ptr<DtmfBuffer> dtmf_buffer_ GUARDED_BY(crit_sect_);
+ const rtc::scoped_ptr<DtmfToneGenerator> dtmf_tone_generator_
GUARDED_BY(crit_sect_);
- const scoped_ptr<PreemptiveExpandFactory> preemptive_expand_factory_
+ const rtc::scoped_ptr<PacketBuffer> packet_buffer_ GUARDED_BY(crit_sect_);
+ const rtc::scoped_ptr<PayloadSplitter> payload_splitter_
+ GUARDED_BY(crit_sect_);
+ const rtc::scoped_ptr<TimestampScaler> timestamp_scaler_
+ GUARDED_BY(crit_sect_);
+ const rtc::scoped_ptr<PostDecodeVad> vad_ GUARDED_BY(crit_sect_);
+ const rtc::scoped_ptr<ExpandFactory> expand_factory_ GUARDED_BY(crit_sect_);
+ const rtc::scoped_ptr<AccelerateFactory> accelerate_factory_
+ GUARDED_BY(crit_sect_);
+ const rtc::scoped_ptr<PreemptiveExpandFactory> preemptive_expand_factory_
GUARDED_BY(crit_sect_);
- scoped_ptr<BackgroundNoise> background_noise_ GUARDED_BY(crit_sect_);
- scoped_ptr<DecisionLogic> decision_logic_ GUARDED_BY(crit_sect_);
- scoped_ptr<AudioMultiVector> algorithm_buffer_ GUARDED_BY(crit_sect_);
- scoped_ptr<SyncBuffer> sync_buffer_ GUARDED_BY(crit_sect_);
- scoped_ptr<Expand> expand_ GUARDED_BY(crit_sect_);
- scoped_ptr<Normal> normal_ GUARDED_BY(crit_sect_);
- scoped_ptr<Merge> merge_ GUARDED_BY(crit_sect_);
- scoped_ptr<Accelerate> accelerate_ GUARDED_BY(crit_sect_);
- scoped_ptr<PreemptiveExpand> preemptive_expand_ GUARDED_BY(crit_sect_);
+ rtc::scoped_ptr<BackgroundNoise> background_noise_ GUARDED_BY(crit_sect_);
+ rtc::scoped_ptr<DecisionLogic> decision_logic_ GUARDED_BY(crit_sect_);
+ rtc::scoped_ptr<AudioMultiVector> algorithm_buffer_ GUARDED_BY(crit_sect_);
+ rtc::scoped_ptr<SyncBuffer> sync_buffer_ GUARDED_BY(crit_sect_);
+ rtc::scoped_ptr<Expand> expand_ GUARDED_BY(crit_sect_);
+ rtc::scoped_ptr<Normal> normal_ GUARDED_BY(crit_sect_);
+ rtc::scoped_ptr<Merge> merge_ GUARDED_BY(crit_sect_);
+ rtc::scoped_ptr<Accelerate> accelerate_ GUARDED_BY(crit_sect_);
+ rtc::scoped_ptr<PreemptiveExpand> preemptive_expand_ GUARDED_BY(crit_sect_);
RandomVector random_vector_ GUARDED_BY(crit_sect_);
- scoped_ptr<ComfortNoise> comfort_noise_ GUARDED_BY(crit_sect_);
+ rtc::scoped_ptr<ComfortNoise> comfort_noise_ GUARDED_BY(crit_sect_);
Rtcp rtcp_ GUARDED_BY(crit_sect_);
StatisticsCalculator stats_ GUARDED_BY(crit_sect_);
int fs_hz_ GUARDED_BY(crit_sect_);
@@ -372,9 +375,9 @@
int output_size_samples_ GUARDED_BY(crit_sect_);
int decoder_frame_length_ GUARDED_BY(crit_sect_);
Modes last_mode_ GUARDED_BY(crit_sect_);
- scoped_ptr<int16_t[]> mute_factor_array_ GUARDED_BY(crit_sect_);
+ rtc::scoped_ptr<int16_t[]> mute_factor_array_ GUARDED_BY(crit_sect_);
size_t decoded_buffer_length_ GUARDED_BY(crit_sect_);
- scoped_ptr<int16_t[]> decoded_buffer_ GUARDED_BY(crit_sect_);
+ rtc::scoped_ptr<int16_t[]> decoded_buffer_ GUARDED_BY(crit_sect_);
uint32_t playout_timestamp_ GUARDED_BY(crit_sect_);
bool new_codec_ GUARDED_BY(crit_sect_);
uint32_t timestamp_ GUARDED_BY(crit_sect_);
diff --git a/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
index cdcf0b3..c6195d0 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
@@ -9,10 +9,10 @@
*/
#include "testing/gmock/include/gmock/gmock.h"
+#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h"
#include "webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h"
#include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
namespace webrtc {
namespace test {
@@ -259,7 +259,7 @@
MockAudioDecoderOpus* external_decoder_;
const int samples_per_ms_;
const size_t frame_size_samples_;
- scoped_ptr<test::RtpGenerator> rtp_generator_;
+ rtc::scoped_ptr<test::RtpGenerator> rtp_generator_;
WebRtcRTPHeader rtp_header_;
uint32_t last_lost_time_;
uint32_t packet_loss_interval_;
diff --git a/webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc
index c9a10df..ea88f24 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc
@@ -15,11 +15,11 @@
#include <list>
#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h"
#include "webrtc/modules/audio_coding/neteq/interface/neteq.h"
#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
#include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/test/testsupport/gtest_disable.h"
@@ -260,7 +260,7 @@
int multi_payload_size_bytes_;
int last_send_time_;
int last_arrival_time_;
- scoped_ptr<test::InputAudioFile> input_file_;
+ rtc::scoped_ptr<test::InputAudioFile> input_file_;
};
class NetEqStereoTestNoJitter : public NetEqStereoTest {
diff --git a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
index 0d8f1a1..b3d6f25 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
@@ -25,10 +25,10 @@
#include "gflags/gflags.h"
#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h"
#include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"
#include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/test/testsupport/gtest_disable.h"
#include "webrtc/typedefs.h"
@@ -262,8 +262,8 @@
NetEq* neteq_;
NetEq::Config config_;
- scoped_ptr<test::RtpFileSource> rtp_source_;
- scoped_ptr<test::Packet> packet_;
+ rtc::scoped_ptr<test::RtpFileSource> rtp_source_;
+ rtc::scoped_ptr<test::Packet> packet_;
unsigned int sim_clock_;
int16_t out_data_[kMaxBlockSize];
int output_sample_rate_;
diff --git a/webrtc/modules/audio_coding/neteq/normal_unittest.cc b/webrtc/modules/audio_coding/neteq/normal_unittest.cc
index e96359a..796409b 100644
--- a/webrtc/modules/audio_coding/neteq/normal_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/normal_unittest.cc
@@ -15,6 +15,7 @@
#include <vector>
#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
#include "webrtc/modules/audio_coding/neteq/background_noise.h"
@@ -23,7 +24,6 @@
#include "webrtc/modules/audio_coding/neteq/mock/mock_expand.h"
#include "webrtc/modules/audio_coding/neteq/random_vector.h"
#include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
using ::testing::_;
@@ -53,7 +53,7 @@
Normal normal(fs, &db, bgn, &expand);
int16_t input[1000] = {0};
- scoped_ptr<int16_t[]> mute_factor_array(new int16_t[channels]);
+ rtc::scoped_ptr<int16_t[]> mute_factor_array(new int16_t[channels]);
for (size_t i = 0; i < channels; ++i) {
mute_factor_array[i] = 16384;
}
@@ -97,7 +97,7 @@
Normal normal(fs, &db, bgn, &expand);
int16_t input[1000] = {0};
- scoped_ptr<int16_t[]> mute_factor_array(new int16_t[channels]);
+ rtc::scoped_ptr<int16_t[]> mute_factor_array(new int16_t[channels]);
for (size_t i = 0; i < channels; ++i) {
mute_factor_array[i] = 16384;
}
diff --git a/webrtc/modules/audio_coding/neteq/payload_splitter_unittest.cc b/webrtc/modules/audio_coding/neteq/payload_splitter_unittest.cc
index 085e76f..305e526 100644
--- a/webrtc/modules/audio_coding/neteq/payload_splitter_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/payload_splitter_unittest.cc
@@ -17,9 +17,9 @@
#include <utility> // pair
#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h"
#include "webrtc/modules/audio_coding/neteq/packet.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
using ::testing::Return;
using ::testing::ReturnNull;
@@ -371,27 +371,27 @@
// Tell the mock decoder database to return DecoderInfo structs with different
// codec types.
// Use scoped pointers to avoid having to delete them later.
- scoped_ptr<DecoderDatabase::DecoderInfo> info0(
+ rtc::scoped_ptr<DecoderDatabase::DecoderInfo> info0(
new DecoderDatabase::DecoderInfo(kDecoderISAC, 16000, NULL, false));
EXPECT_CALL(decoder_database, GetDecoderInfo(0))
.WillRepeatedly(Return(info0.get()));
- scoped_ptr<DecoderDatabase::DecoderInfo> info1(
+ rtc::scoped_ptr<DecoderDatabase::DecoderInfo> info1(
new DecoderDatabase::DecoderInfo(kDecoderISACswb, 32000, NULL, false));
EXPECT_CALL(decoder_database, GetDecoderInfo(1))
.WillRepeatedly(Return(info1.get()));
- scoped_ptr<DecoderDatabase::DecoderInfo> info2(
+ rtc::scoped_ptr<DecoderDatabase::DecoderInfo> info2(
new DecoderDatabase::DecoderInfo(kDecoderRED, 8000, NULL, false));
EXPECT_CALL(decoder_database, GetDecoderInfo(2))
.WillRepeatedly(Return(info2.get()));
- scoped_ptr<DecoderDatabase::DecoderInfo> info3(
+ rtc::scoped_ptr<DecoderDatabase::DecoderInfo> info3(
new DecoderDatabase::DecoderInfo(kDecoderAVT, 8000, NULL, false));
EXPECT_CALL(decoder_database, GetDecoderInfo(3))
.WillRepeatedly(Return(info3.get()));
- scoped_ptr<DecoderDatabase::DecoderInfo> info4(
+ rtc::scoped_ptr<DecoderDatabase::DecoderInfo> info4(
new DecoderDatabase::DecoderInfo(kDecoderCNGnb, 8000, NULL, false));
EXPECT_CALL(decoder_database, GetDecoderInfo(4))
.WillRepeatedly(Return(info4.get()));
- scoped_ptr<DecoderDatabase::DecoderInfo> info5(
+ rtc::scoped_ptr<DecoderDatabase::DecoderInfo> info5(
new DecoderDatabase::DecoderInfo(kDecoderArbitrary, 8000, NULL, false));
EXPECT_CALL(decoder_database, GetDecoderInfo(5))
.WillRepeatedly(Return(info5.get()));
@@ -529,7 +529,7 @@
// codec types.
// Use scoped pointers to avoid having to delete them later.
// (Sample rate is set to 8000 Hz, but does not matter.)
- scoped_ptr<DecoderDatabase::DecoderInfo> info(
+ rtc::scoped_ptr<DecoderDatabase::DecoderInfo> info(
new DecoderDatabase::DecoderInfo(decoder_type_, 8000, NULL, false));
EXPECT_CALL(decoder_database, GetDecoderInfo(kPayloadType))
.WillRepeatedly(Return(info.get()));
@@ -608,7 +608,7 @@
// Tell the mock decoder database to return DecoderInfo structs with different
// codec types.
// Use scoped pointers to avoid having to delete them later.
- scoped_ptr<DecoderDatabase::DecoderInfo> info(
+ rtc::scoped_ptr<DecoderDatabase::DecoderInfo> info(
new DecoderDatabase::DecoderInfo(kDecoderILBC, 8000, NULL, false));
EXPECT_CALL(decoder_database, GetDecoderInfo(kPayloadType))
.WillRepeatedly(Return(info.get()));
@@ -671,7 +671,7 @@
packet_list.push_back(packet);
MockDecoderDatabase decoder_database;
- scoped_ptr<DecoderDatabase::DecoderInfo> info(
+ rtc::scoped_ptr<DecoderDatabase::DecoderInfo> info(
new DecoderDatabase::DecoderInfo(kDecoderILBC, 8000, NULL, false));
EXPECT_CALL(decoder_database, GetDecoderInfo(kPayloadType))
.WillRepeatedly(Return(info.get()));
@@ -702,7 +702,7 @@
packet_list.push_back(packet);
MockDecoderDatabase decoder_database;
- scoped_ptr<DecoderDatabase::DecoderInfo> info(
+ rtc::scoped_ptr<DecoderDatabase::DecoderInfo> info(
new DecoderDatabase::DecoderInfo(kDecoderILBC, 8000, NULL, false));
EXPECT_CALL(decoder_database, GetDecoderInfo(kPayloadType))
.WillRepeatedly(Return(info.get()));
diff --git a/webrtc/modules/audio_coding/neteq/test/audio_classifier_test.cc b/webrtc/modules/audio_coding/neteq/test/audio_classifier_test.cc
index aa2b61d..a14238c 100644
--- a/webrtc/modules/audio_coding/neteq/test/audio_classifier_test.cc
+++ b/webrtc/modules/audio_coding/neteq/test/audio_classifier_test.cc
@@ -18,7 +18,7 @@
#include <string>
#include <iostream>
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/base/scoped_ptr.h"
int main(int argc, char* argv[]) {
if (argc != 5) {
@@ -48,7 +48,7 @@
}
const int data_size = channels * kFrameSizeSamples;
- webrtc::scoped_ptr<int16_t[]> in(new int16_t[data_size]);
+ rtc::scoped_ptr<int16_t[]> in(new int16_t[data_size]);
std::string input_filename = argv[3];
std::string output_filename = argv[4];
diff --git a/webrtc/modules/audio_coding/neteq/time_stretch.cc b/webrtc/modules/audio_coding/neteq/time_stretch.cc
index a9228d4..02305c8 100644
--- a/webrtc/modules/audio_coding/neteq/time_stretch.cc
+++ b/webrtc/modules/audio_coding/neteq/time_stretch.cc
@@ -12,10 +12,10 @@
#include <algorithm> // min, max
+#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/modules/audio_coding/neteq/background_noise.h"
#include "webrtc/modules/audio_coding/neteq/dsp_helper.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
namespace webrtc {
@@ -29,7 +29,7 @@
int fs_mult_120 = fs_mult_ * 120; // Corresponds to 15 ms.
const int16_t* signal;
- scoped_ptr<int16_t[]> signal_array;
+ rtc::scoped_ptr<int16_t[]> signal_array;
size_t signal_len;
if (num_channels_ == 1) {
signal = input;
diff --git a/webrtc/modules/audio_coding/neteq/tools/audio_loop.h b/webrtc/modules/audio_coding/neteq/tools/audio_loop.h
index 9647d82..87ff688 100644
--- a/webrtc/modules/audio_coding/neteq/tools/audio_loop.h
+++ b/webrtc/modules/audio_coding/neteq/tools/audio_loop.h
@@ -14,7 +14,7 @@
#include <string>
#include "webrtc/base/constructormagic.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/base/scoped_ptr.h"
#include "webrtc/typedefs.h"
namespace webrtc {
@@ -49,7 +49,7 @@
size_t next_index_;
size_t loop_length_samples_;
size_t block_length_samples_;
- scoped_ptr<int16_t[]> audio_array_;
+ rtc::scoped_ptr<int16_t[]> audio_array_;
DISALLOW_COPY_AND_ASSIGN(AudioLoop);
};
diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h b/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h
index 8cf6ef8..0d4d2f9 100644
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h
@@ -11,10 +11,10 @@
#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_EXTERNAL_DECODER_TEST_H_
#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_EXTERNAL_DECODER_TEST_H_
+#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
#include "webrtc/modules/audio_coding/neteq/interface/neteq.h"
#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
namespace webrtc {
namespace test {
@@ -52,7 +52,7 @@
AudioDecoder* decoder_;
int sample_rate_hz_;
int channels_;
- scoped_ptr<NetEq> neteq_;
+ rtc::scoped_ptr<NetEq> neteq_;
};
} // namespace test
diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h b/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h
index 00a2499..6207fde 100644
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h
@@ -14,10 +14,10 @@
#include <gflags/gflags.h>
#include <string>
#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/neteq/interface/neteq.h"
#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
#include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/typedefs.h"
using google::RegisterFlagValidator;
@@ -57,7 +57,7 @@
// Prob. of losing current packet, when previous packet is not lost.
double prob_trans_01_;
bool lost_last_;
- scoped_ptr<UniformLoss> uniform_loss_model_;
+ rtc::scoped_ptr<UniformLoss> uniform_loss_model_;
};
class NetEqQualityTest : public ::testing::Test {
@@ -121,17 +121,17 @@
size_t payload_size_bytes_;
int max_payload_bytes_;
- scoped_ptr<InputAudioFile> in_file_;
+ rtc::scoped_ptr<InputAudioFile> in_file_;
FILE* out_file_;
FILE* log_file_;
- scoped_ptr<RtpGenerator> rtp_generator_;
- scoped_ptr<NetEq> neteq_;
- scoped_ptr<LossModel> loss_model_;
+ rtc::scoped_ptr<RtpGenerator> rtp_generator_;
+ rtc::scoped_ptr<NetEq> neteq_;
+ rtc::scoped_ptr<LossModel> loss_model_;
- scoped_ptr<int16_t[]> in_data_;
- scoped_ptr<uint8_t[]> payload_;
- scoped_ptr<int16_t[]> out_data_;
+ rtc::scoped_ptr<int16_t[]> in_data_;
+ rtc::scoped_ptr<uint8_t[]> payload_;
+ rtc::scoped_ptr<int16_t[]> out_data_;
WebRtcRTPHeader rtp_header_;
size_t total_payload_size_bytes_;
diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc b/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
index efa86d8..11dd20a 100644
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
@@ -23,6 +23,7 @@
#include "google/gflags.h"
#include "webrtc/base/checks.h"
+#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h"
#include "webrtc/modules/audio_coding/neteq/interface/neteq.h"
#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
@@ -31,7 +32,6 @@
#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
#include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"
#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/system_wrappers/interface/trace.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/typedefs.h"
@@ -270,8 +270,8 @@
}
size_t ReplacePayload(webrtc::test::InputAudioFile* replacement_audio_file,
- webrtc::scoped_ptr<int16_t[]>* replacement_audio,
- webrtc::scoped_ptr<uint8_t[]>* payload,
+ rtc::scoped_ptr<int16_t[]>* replacement_audio,
+ rtc::scoped_ptr<uint8_t[]>* payload,
size_t* payload_mem_size_bytes,
size_t* frame_size_samples,
WebRtcRTPHeader* rtp_header,
@@ -384,7 +384,7 @@
}
printf("Input file: %s\n", argv[1]);
- webrtc::scoped_ptr<webrtc::test::RtpFileSource> file_source(
+ rtc::scoped_ptr<webrtc::test::RtpFileSource> file_source(
webrtc::test::RtpFileSource::Create(argv[1]));
assert(file_source.get());
@@ -397,7 +397,7 @@
// Check if a replacement audio file was provided, and if so, open it.
bool replace_payload = false;
- webrtc::scoped_ptr<webrtc::test::InputAudioFile> replacement_audio_file;
+ rtc::scoped_ptr<webrtc::test::InputAudioFile> replacement_audio_file;
if (!FLAGS_replacement_audio_file.empty()) {
replacement_audio_file.reset(
new webrtc::test::InputAudioFile(FLAGS_replacement_audio_file));
@@ -405,7 +405,7 @@
}
// Read first packet.
- webrtc::scoped_ptr<webrtc::test::Packet> packet(file_source->NextPacket());
+ rtc::scoped_ptr<webrtc::test::Packet> packet(file_source->NextPacket());
if (!packet) {
printf(
"Warning: input file is empty, or the filters did not match any "
@@ -427,7 +427,7 @@
// for wav files.)
// Check output file type.
std::string output_file_name = argv[2];
- webrtc::scoped_ptr<webrtc::test::AudioSink> output;
+ rtc::scoped_ptr<webrtc::test::AudioSink> output;
if (output_file_name.size() >= 4 &&
output_file_name.substr(output_file_name.size() - 4) == ".wav") {
// Open a wav file.
@@ -454,11 +454,11 @@
// Set up variables for audio replacement if needed.
- webrtc::scoped_ptr<webrtc::test::Packet> next_packet;
+ rtc::scoped_ptr<webrtc::test::Packet> next_packet;
bool next_packet_available = false;
size_t input_frame_size_timestamps = 0;
- webrtc::scoped_ptr<int16_t[]> replacement_audio;
- webrtc::scoped_ptr<uint8_t[]> payload;
+ rtc::scoped_ptr<int16_t[]> replacement_audio;
+ rtc::scoped_ptr<uint8_t[]> payload;
size_t payload_mem_size_bytes = 0;
if (replace_payload) {
// Initially assume that the frame size is 30 ms at the initial sample rate.
diff --git a/webrtc/modules/audio_coding/neteq/tools/packet.cc b/webrtc/modules/audio_coding/neteq/tools/packet.cc
index 794c308..b8b27af 100644
--- a/webrtc/modules/audio_coding/neteq/tools/packet.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/packet.cc
@@ -55,7 +55,7 @@
virtual_packet_length_bytes_(allocated_bytes),
virtual_payload_length_bytes_(0),
time_ms_(time_ms) {
- scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
+ rtc::scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
valid_header_ = ParseHeader(*parser);
}
@@ -70,7 +70,7 @@
virtual_packet_length_bytes_(virtual_packet_length_bytes),
virtual_payload_length_bytes_(0),
time_ms_(time_ms) {
- scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
+ rtc::scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
valid_header_ = ParseHeader(*parser);
}
diff --git a/webrtc/modules/audio_coding/neteq/tools/packet.h b/webrtc/modules/audio_coding/neteq/tools/packet.h
index df7aeb7..a4e48d8 100644
--- a/webrtc/modules/audio_coding/neteq/tools/packet.h
+++ b/webrtc/modules/audio_coding/neteq/tools/packet.h
@@ -14,8 +14,8 @@
#include <list>
#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_types.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/typedefs.h"
namespace webrtc {
@@ -103,7 +103,7 @@
void CopyToHeader(RTPHeader* destination) const;
RTPHeader header_;
- scoped_ptr<uint8_t[]> payload_memory_;
+ rtc::scoped_ptr<uint8_t[]> payload_memory_;
const uint8_t* payload_; // First byte after header.
const size_t packet_length_bytes_; // Total length of packet.
size_t payload_length_bytes_; // Length of the payload, after RTP header.
diff --git a/webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.cc b/webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.cc
index f391466..ea88a3f 100644
--- a/webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.cc
@@ -11,7 +11,7 @@
#include "webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h"
#include "webrtc/base/checks.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/base/scoped_ptr.h"
namespace webrtc {
namespace test {
@@ -22,7 +22,7 @@
const size_t samples_to_read = samples * file_rate_hz_ / output_rate_hz;
CHECK_EQ(samples_to_read * output_rate_hz, samples * file_rate_hz_)
<< "Frame size and sample rates don't add up to an integer.";
- scoped_ptr<int16_t[]> temp_destination(new int16_t[samples_to_read]);
+ rtc::scoped_ptr<int16_t[]> temp_destination(new int16_t[samples_to_read]);
if (!InputAudioFile::Read(samples_to_read, temp_destination.get()))
return false;
resampler_.ResetIfNeeded(
diff --git a/webrtc/modules/audio_coding/neteq/tools/rtp_analyze.cc b/webrtc/modules/audio_coding/neteq/tools/rtp_analyze.cc
index ec604d2..d062b38 100644
--- a/webrtc/modules/audio_coding/neteq/tools/rtp_analyze.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/rtp_analyze.cc
@@ -13,9 +13,9 @@
#include <vector>
#include "google/gflags.h"
+#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
#include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
// Flag validator.
static bool ValidatePayloadType(const char* flagname, int32_t value) {
@@ -60,7 +60,7 @@
}
printf("Input file: %s\n", argv[1]);
- webrtc::scoped_ptr<webrtc::test::RtpFileSource> file_source(
+ rtc::scoped_ptr<webrtc::test::RtpFileSource> file_source(
webrtc::test::RtpFileSource::Create(argv[1]));
assert(file_source.get());
// Set RTP extension ID.
@@ -90,7 +90,7 @@
}
fprintf(out_file, "\n");
- webrtc::scoped_ptr<webrtc::test::Packet> packet;
+ rtc::scoped_ptr<webrtc::test::Packet> packet;
while (true) {
packet.reset(file_source->NextPacket());
if (!packet.get()) {
diff --git a/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc b/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
index 794c983..f5d323e 100644
--- a/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
@@ -52,13 +52,11 @@
// Read the next one.
continue;
}
- scoped_ptr<uint8_t[]> packet_memory(new uint8_t[temp_packet.length]);
+ rtc::scoped_ptr<uint8_t[]> packet_memory(new uint8_t[temp_packet.length]);
memcpy(packet_memory.get(), temp_packet.data, temp_packet.length);
- scoped_ptr<Packet> packet(new Packet(packet_memory.release(),
- temp_packet.length,
- temp_packet.original_length,
- temp_packet.time_ms,
- *parser_.get()));
+ rtc::scoped_ptr<Packet> packet(new Packet(
+ packet_memory.release(), temp_packet.length,
+ temp_packet.original_length, temp_packet.time_ms, *parser_.get()));
if (!packet->valid_header()) {
assert(false);
return NULL;
diff --git a/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h b/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h
index d309280..70b5216 100644
--- a/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h
+++ b/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h
@@ -15,10 +15,10 @@
#include <string>
#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
namespace webrtc {
@@ -52,8 +52,8 @@
bool OpenFile(const std::string& file_name);
- scoped_ptr<RtpFileReader> rtp_reader_;
- scoped_ptr<RtpHeaderParser> parser_;
+ rtc::scoped_ptr<RtpFileReader> rtp_reader_;
+ rtc::scoped_ptr<RtpHeaderParser> parser_;
DISALLOW_COPY_AND_ASSIGN(RtpFileSource);
};
diff --git a/webrtc/modules/audio_coding/neteq/tools/rtpcat.cc b/webrtc/modules/audio_coding/neteq/tools/rtpcat.cc
index 089d4ca..f7490de 100644
--- a/webrtc/modules/audio_coding/neteq/tools/rtpcat.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/rtpcat.cc
@@ -11,11 +11,11 @@
#include <stdio.h>
#include "webrtc/base/checks.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/base/scoped_ptr.h"
#include "webrtc/test/rtp_file_reader.h"
#include "webrtc/test/rtp_file_writer.h"
-using webrtc::scoped_ptr;
+using rtc::scoped_ptr;
using webrtc::test::RtpFileReader;
using webrtc::test::RtpFileWriter;