Replace scoped_ptr with unique_ptr in webrtc/modules/audio_coding/neteq/
BUG=webrtc:5520
Review URL: https://codereview.webrtc.org/1697823002
Cr-Commit-Position: refs/heads/master@{#11616}
diff --git a/webrtc/modules/audio_coding/neteq/tools/audio_loop.h b/webrtc/modules/audio_coding/neteq/tools/audio_loop.h
index 14e20f6..40b2c55 100644
--- a/webrtc/modules/audio_coding/neteq/tools/audio_loop.h
+++ b/webrtc/modules/audio_coding/neteq/tools/audio_loop.h
@@ -11,11 +11,11 @@
#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_LOOP_H_
#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_LOOP_H_
+#include <memory>
#include <string>
#include "webrtc/base/array_view.h"
#include "webrtc/base/constructormagic.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_;
- rtc::scoped_ptr<int16_t[]> audio_array_;
+ std::unique_ptr<int16_t[]> audio_array_;
RTC_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 d7b01fe..1b36d8b 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
@@ -11,9 +11,9 @@
#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 <memory>
#include <string>
-#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
#include "webrtc/modules/include/module_common_types.h"
@@ -55,7 +55,7 @@
AudioDecoder* decoder_;
int sample_rate_hz_;
size_t channels_;
- rtc::scoped_ptr<NetEq> neteq_;
+ std::unique_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 c2b2eff..8bae160 100644
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h
@@ -12,9 +12,9 @@
#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_QUALITY_TEST_H_
#include <fstream>
+#include <memory>
#include <gflags/gflags.h>
#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
#include "webrtc/modules/audio_coding/neteq/tools/audio_sink.h"
#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
@@ -58,7 +58,7 @@
// Prob. of losing current packet, when previous packet is not lost.
double prob_trans_01_;
bool lost_last_;
- rtc::scoped_ptr<UniformLoss> uniform_loss_model_;
+ std::unique_ptr<UniformLoss> uniform_loss_model_;
};
class NetEqQualityTest : public ::testing::Test {
@@ -119,17 +119,17 @@
size_t payload_size_bytes_;
size_t max_payload_bytes_;
- rtc::scoped_ptr<InputAudioFile> in_file_;
- rtc::scoped_ptr<AudioSink> output_;
+ std::unique_ptr<InputAudioFile> in_file_;
+ std::unique_ptr<AudioSink> output_;
std::ofstream log_file_;
- rtc::scoped_ptr<RtpGenerator> rtp_generator_;
- rtc::scoped_ptr<NetEq> neteq_;
- rtc::scoped_ptr<LossModel> loss_model_;
+ std::unique_ptr<RtpGenerator> rtp_generator_;
+ std::unique_ptr<NetEq> neteq_;
+ std::unique_ptr<LossModel> loss_model_;
- rtc::scoped_ptr<int16_t[]> in_data_;
- rtc::scoped_ptr<uint8_t[]> payload_;
- rtc::scoped_ptr<int16_t[]> out_data_;
+ std::unique_ptr<int16_t[]> in_data_;
+ std::unique_ptr<uint8_t[]> payload_;
+ std::unique_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 57005ae..1701c47 100644
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
@@ -19,13 +19,13 @@
#include <algorithm>
#include <iostream>
+#include <memory>
#include <limits>
#include <string>
#include "gflags/gflags.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/safe_conversions.h"
-#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
@@ -295,8 +295,8 @@
}
size_t ReplacePayload(webrtc::test::InputAudioFile* replacement_audio_file,
- rtc::scoped_ptr<int16_t[]>* replacement_audio,
- rtc::scoped_ptr<uint8_t[]>* payload,
+ std::unique_ptr<int16_t[]>* replacement_audio,
+ std::unique_ptr<uint8_t[]>* payload,
size_t* payload_mem_size_bytes,
size_t* frame_size_samples,
WebRtcRTPHeader* rtp_header,
@@ -411,7 +411,7 @@
printf("Input file: %s\n", argv[1]);
bool is_rtp_dump = false;
- rtc::scoped_ptr<webrtc::test::PacketSource> file_source;
+ std::unique_ptr<webrtc::test::PacketSource> file_source;
webrtc::test::RtcEventLogSource* event_log_source = nullptr;
if (webrtc::test::RtpFileSource::ValidRtpDump(argv[1]) ||
webrtc::test::RtpFileSource::ValidPcap(argv[1])) {
@@ -433,7 +433,7 @@
// Check if a replacement audio file was provided, and if so, open it.
bool replace_payload = false;
- rtc::scoped_ptr<webrtc::test::InputAudioFile> replacement_audio_file;
+ std::unique_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));
@@ -441,7 +441,7 @@
}
// Read first packet.
- rtc::scoped_ptr<webrtc::test::Packet> packet(file_source->NextPacket());
+ std::unique_ptr<webrtc::test::Packet> packet(file_source->NextPacket());
if (!packet) {
printf(
"Warning: input file is empty, or the filters did not match any "
@@ -468,7 +468,7 @@
// for wav files.)
// Check output file type.
std::string output_file_name = argv[2];
- rtc::scoped_ptr<webrtc::test::AudioSink> output;
+ std::unique_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.
@@ -495,11 +495,11 @@
// Set up variables for audio replacement if needed.
- rtc::scoped_ptr<webrtc::test::Packet> next_packet;
+ std::unique_ptr<webrtc::test::Packet> next_packet;
bool next_packet_available = false;
size_t input_frame_size_timestamps = 0;
- rtc::scoped_ptr<int16_t[]> replacement_audio;
- rtc::scoped_ptr<uint8_t[]> payload;
+ std::unique_ptr<int16_t[]> replacement_audio;
+ std::unique_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 2b2fcc2..46fd0cb 100644
--- a/webrtc/modules/audio_coding/neteq/tools/packet.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/packet.cc
@@ -12,6 +12,8 @@
#include <string.h>
+#include <memory>
+
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
@@ -55,7 +57,7 @@
virtual_packet_length_bytes_(allocated_bytes),
virtual_payload_length_bytes_(0),
time_ms_(time_ms) {
- rtc::scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
+ std::unique_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
valid_header_ = ParseHeader(*parser);
}
@@ -70,7 +72,7 @@
virtual_packet_length_bytes_(virtual_packet_length_bytes),
virtual_payload_length_bytes_(0),
time_ms_(time_ms) {
- rtc::scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
+ std::unique_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 8e43633..86eedc0 100644
--- a/webrtc/modules/audio_coding/neteq/tools/packet.h
+++ b/webrtc/modules/audio_coding/neteq/tools/packet.h
@@ -12,9 +12,9 @@
#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_H_
#include <list>
+#include <memory>
#include "webrtc/base/constructormagic.h"
-#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_types.h"
#include "webrtc/typedefs.h"
@@ -103,7 +103,7 @@
void CopyToHeader(RTPHeader* destination) const;
RTPHeader header_;
- rtc::scoped_ptr<uint8_t[]> payload_memory_;
+ std::unique_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 7a0bb1a..f5fe166 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
@@ -10,8 +10,9 @@
#include "webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h"
+#include <memory>
+
#include "webrtc/base/checks.h"
-#include "webrtc/base/scoped_ptr.h"
namespace webrtc {
namespace test {
@@ -22,7 +23,7 @@
const size_t samples_to_read = samples * file_rate_hz_ / output_rate_hz;
RTC_CHECK_EQ(samples_to_read * output_rate_hz, samples * file_rate_hz_)
<< "Frame size and sample rates don't add up to an integer.";
- rtc::scoped_ptr<int16_t[]> temp_destination(new int16_t[samples_to_read]);
+ std::unique_ptr<int16_t[]> temp_destination(new int16_t[samples_to_read]);
if (!InputAudioFile::Read(samples_to_read, temp_destination.get()))
return false;
resampler_.ResetIfNeeded(file_rate_hz_, output_rate_hz, 1);
diff --git a/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h b/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h
index 90d5931..312338e 100644
--- a/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h
+++ b/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h
@@ -11,10 +11,10 @@
#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTC_EVENT_LOG_SOURCE_H_
#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTC_EVENT_LOG_SOURCE_H_
+#include <memory>
#include <string>
#include "webrtc/base/constructormagic.h"
-#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
@@ -58,8 +58,8 @@
int rtp_packet_index_ = 0;
int audio_output_index_ = 0;
- rtc::scoped_ptr<rtclog::EventStream> event_log_;
- rtc::scoped_ptr<RtpHeaderParser> parser_;
+ std::unique_ptr<rtclog::EventStream> event_log_;
+ std::unique_ptr<RtpHeaderParser> parser_;
RTC_DISALLOW_COPY_AND_ASSIGN(RtcEventLogSource);
};
diff --git a/webrtc/modules/audio_coding/neteq/tools/rtp_analyze.cc b/webrtc/modules/audio_coding/neteq/tools/rtp_analyze.cc
index faabdc2..0735b4c 100644
--- a/webrtc/modules/audio_coding/neteq/tools/rtp_analyze.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/rtp_analyze.cc
@@ -10,10 +10,11 @@
#include <assert.h>
#include <stdio.h>
+
+#include <memory>
#include <vector>
#include "gflags/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"
@@ -63,7 +64,7 @@
}
printf("Input file: %s\n", argv[1]);
- rtc::scoped_ptr<webrtc::test::RtpFileSource> file_source(
+ std::unique_ptr<webrtc::test::RtpFileSource> file_source(
webrtc::test::RtpFileSource::Create(argv[1]));
assert(file_source.get());
// Set RTP extension IDs.
@@ -104,7 +105,7 @@
uint32_t max_abs_send_time = 0;
int cycles = -1;
- rtc::scoped_ptr<webrtc::test::Packet> packet;
+ std::unique_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 b7a3109..039e1fa 100644
--- a/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
@@ -18,6 +18,8 @@
#include <netinet/in.h>
#endif
+#include <memory>
+
#include "webrtc/base/checks.h"
#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
@@ -33,13 +35,13 @@
}
bool RtpFileSource::ValidRtpDump(const std::string& file_name) {
- rtc::scoped_ptr<RtpFileReader> temp_file(
+ std::unique_ptr<RtpFileReader> temp_file(
RtpFileReader::Create(RtpFileReader::kRtpDump, file_name));
return !!temp_file;
}
bool RtpFileSource::ValidPcap(const std::string& file_name) {
- rtc::scoped_ptr<RtpFileReader> temp_file(
+ std::unique_ptr<RtpFileReader> temp_file(
RtpFileReader::Create(RtpFileReader::kPcap, file_name));
return !!temp_file;
}
@@ -64,9 +66,9 @@
// Read the next one.
continue;
}
- rtc::scoped_ptr<uint8_t[]> packet_memory(new uint8_t[temp_packet.length]);
+ std::unique_ptr<uint8_t[]> packet_memory(new uint8_t[temp_packet.length]);
memcpy(packet_memory.get(), temp_packet.data, temp_packet.length);
- rtc::scoped_ptr<Packet> packet(new Packet(
+ std::unique_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()) {
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 2febf68..b02e16a 100644
--- a/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h
+++ b/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h
@@ -12,10 +12,11 @@
#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_
#include <stdio.h>
+
+#include <memory>
#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/include/rtp_rtcp_defines.h"
@@ -56,8 +57,8 @@
bool OpenFile(const std::string& file_name);
- rtc::scoped_ptr<RtpFileReader> rtp_reader_;
- rtc::scoped_ptr<RtpHeaderParser> parser_;
+ std::unique_ptr<RtpFileReader> rtp_reader_;
+ std::unique_ptr<RtpHeaderParser> parser_;
RTC_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 f2b87a5..e1f49f7 100644
--- a/webrtc/modules/audio_coding/neteq/tools/rtpcat.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/rtpcat.cc
@@ -10,12 +10,12 @@
#include <stdio.h>
+#include <memory>
+
#include "webrtc/base/checks.h"
-#include "webrtc/base/scoped_ptr.h"
#include "webrtc/test/rtp_file_reader.h"
#include "webrtc/test/rtp_file_writer.h"
-using rtc::scoped_ptr;
using webrtc::test::RtpFileReader;
using webrtc::test::RtpFileWriter;
@@ -26,13 +26,13 @@
exit(1);
}
- scoped_ptr<RtpFileWriter> output(
+ std::unique_ptr<RtpFileWriter> output(
RtpFileWriter::Create(RtpFileWriter::kRtpDump, argv[argc - 1]));
RTC_CHECK(output.get() != NULL) << "Cannot open output file.";
printf("Output RTP file: %s\n", argv[argc - 1]);
for (int i = 1; i < argc - 1; i++) {
- scoped_ptr<RtpFileReader> input(
+ std::unique_ptr<RtpFileReader> input(
RtpFileReader::Create(RtpFileReader::kRtpDump, argv[i]));
RTC_CHECK(input.get() != NULL) << "Cannot open input file " << argv[i];
printf("Input RTP file: %s\n", argv[i]);