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

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 @@ class AudioLoop {
   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);
 };

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 @@ class NetEqExternalDecoderTest {
   AudioDecoder* decoder_;
   int sample_rate_hz_;
   int channels_;
-  scoped_ptr<NetEq> neteq_;
+  rtc::scoped_ptr<NetEq> neteq_;
 };
 
 }  // namespace test

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 @@ class GilbertElliotLoss : public LossModel {
   // 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 @@ class NetEqQualityTest : public ::testing::Test {
   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_;

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 @@ int CodecTimestampRate(uint8_t payload_type) {
 }
 
 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 @@ int main(int argc, char* argv[]) {
   }
 
   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 @@ int main(int argc, char* argv[]) {
 
   // 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 @@ int main(int argc, char* argv[]) {
   }
 
   // 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 @@ int main(int argc, char* argv[]) {
   // 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 @@ int main(int argc, char* argv[]) {
 
 
   // 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.

webrtc/modules/audio_coding/neteq/tools/packet.cc

@@ -55,7 +55,7 @@ Packet::Packet(uint8_t* packet_memory, size_t allocated_bytes, double time_ms)
       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 @@ Packet::Packet(uint8_t* packet_memory,
       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);
 }
 

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 @@ class Packet {
   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.

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 @@ bool ResampleInputAudioFile::Read(size_t samples,
   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(

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 @@ int main(int argc, char* argv[]) {
   }
 
   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 @@ int main(int argc, char* argv[]) {
   }
   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()) {

webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc

@@ -52,13 +52,11 @@ Packet* RtpFileSource::NextPacket() {
       // 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;

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 @@ class RtpFileSource : public PacketSource {
 
   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);
 };

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;