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Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away

Browse Source 
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
This commit is contained in:
kwiberg@webrtc.org
2015-02-26 14:34:55 +00:00
parent ac2d27d9ae
commit 00b8f6b364
391 changed files with 1427 additions and 1402 deletions
Download Patch File Download Diff File Expand all files Collapse all files

10
webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h
View File

@ -13,6 +13,7 @@
#include <map>
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
@ -21,7 +22,6 @@
#include "webrtc/modules/audio_coding/neteq/interface/neteq.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/system_wrappers/interface/trace.h"
#define MAX_FRAME_SIZE_10MSEC 6
@ -72,7 +72,7 @@ class AudioDecoderProxy final : public AudioDecoder {
CNG_dec_inst* CngDecoderInstance() override;
private:
scoped_ptr<CriticalSectionWrapper> decoder_lock_;
rtc::scoped_ptr<CriticalSectionWrapper> decoder_lock_;
AudioDecoder* decoder_ GUARDED_BY(decoder_lock_);
};
@ -472,9 +472,9 @@ class ACMGenericCodec {
OpusApplicationMode GetOpusApplication(int num_channels) const
EXCLUSIVE_LOCKS_REQUIRED(codec_wrapper_lock_);
scoped_ptr<AudioEncoder> audio_encoder_ GUARDED_BY(codec_wrapper_lock_);
scoped_ptr<AudioEncoder> cng_encoder_ GUARDED_BY(codec_wrapper_lock_);
scoped_ptr<AudioEncoder> red_encoder_ GUARDED_BY(codec_wrapper_lock_);
rtc::scoped_ptr<AudioEncoder> audio_encoder_ GUARDED_BY(codec_wrapper_lock_);
rtc::scoped_ptr<AudioEncoder> cng_encoder_ GUARDED_BY(codec_wrapper_lock_);
rtc::scoped_ptr<AudioEncoder> red_encoder_ GUARDED_BY(codec_wrapper_lock_);
AudioEncoder* encoder_ GUARDED_BY(codec_wrapper_lock_);
AudioDecoderProxy decoder_proxy_ GUARDED_BY(codec_wrapper_lock_);
std::vector<int16_t> input_ GUARDED_BY(codec_wrapper_lock_);

2
webrtc/modules/audio_coding/main/acm2/acm_generic_codec_opus_test.cc
View File

@ -43,7 +43,7 @@ class AcmGenericCodecOpusTest : public ::testing::Test {
return ptr;
}
WebRtcACMCodecParams acm_codec_params_;
scoped_ptr<ACMGenericCodec> codec_wrapper_;
rtc::scoped_ptr<ACMGenericCodec> codec_wrapper_;
};
TEST_F(AcmGenericCodecOpusTest, DefaultApplicationModeMono) {

2
webrtc/modules/audio_coding/main/acm2/acm_generic_codec_test.cc
View File

@ -73,7 +73,7 @@ class AcmGenericCodecTest : public ::testing::Test {
}
WebRtcACMCodecParams acm_codec_params_;
scoped_ptr<ACMGenericCodec> codec_;
rtc::scoped_ptr<ACMGenericCodec> codec_;
uint32_t timestamp_;
};

2
webrtc/modules/audio_coding/main/acm2/acm_receive_test.cc
View File

@ -86,7 +86,7 @@ void AcmReceiveTest::RegisterNetEqTestCodecs() {
}
void AcmReceiveTest::Run() {
for (scoped_ptr<Packet> packet(packet_source_->NextPacket()); packet;
for (rtc::scoped_ptr<Packet> packet(packet_source_->NextPacket()); packet;
packet.reset(packet_source_->NextPacket())) {
// Pull audio until time to insert packet.
while (clock_.TimeInMilliseconds() < packet->time_ms()) {

4
webrtc/modules/audio_coding/main/acm2/acm_receive_test.h
View File

@ -12,8 +12,8 @@
#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVE_TEST_H_
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/system_wrappers/interface/clock.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
namespace webrtc {
class AudioCoding;
@ -50,7 +50,7 @@ class AcmReceiveTest {
private:
SimulatedClock clock_;
scoped_ptr<AudioCoding> acm_;
rtc::scoped_ptr<AudioCoding> acm_;
PacketSource* packet_source_;
AudioSink* audio_sink_;
const int output_freq_hz_;

2
webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.cc
View File

@ -144,7 +144,7 @@ void AcmReceiveTestOldApi::RegisterNetEqTestCodecs() {
}
void AcmReceiveTestOldApi::Run() {
for (scoped_ptr<Packet> packet(packet_source_->NextPacket()); packet;
for (rtc::scoped_ptr<Packet> packet(packet_source_->NextPacket()); packet;
packet.reset(packet_source_->NextPacket())) {
// Pull audio until time to insert packet.
while (clock_.TimeInMilliseconds() < packet->time_ms()) {

4
webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.h
View File

@ -12,8 +12,8 @@
#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVE_TEST_H_
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/system_wrappers/interface/clock.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
namespace webrtc {
class AudioCodingModule;
@ -52,7 +52,7 @@ class AcmReceiveTestOldApi {
virtual void AfterGetAudio() {}
SimulatedClock clock_;
scoped_ptr<AudioCodingModule> acm_;
rtc::scoped_ptr<AudioCodingModule> acm_;
PacketSource* packet_source_;
AudioSink* audio_sink_;
int output_freq_hz_;

16
webrtc/modules/audio_coding/main/acm2/acm_receiver.h
View File

@ -13,6 +13,7 @@
#include <vector>
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/common_audio/vad/include/webrtc_vad.h"
#include "webrtc/engine_configurations.h"
@ -23,7 +24,6 @@
#include "webrtc/modules/audio_coding/main/acm2/initial_delay_manager.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"
#include "webrtc/typedefs.h"
namespace webrtc {
@ -320,7 +320,7 @@ class AcmReceiver {
void InsertStreamOfSyncPackets(InitialDelayManager::SyncStream* sync_stream);
scoped_ptr<CriticalSectionWrapper> crit_sect_;
rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
int id_; // TODO(henrik.lundin) Make const.
int last_audio_decoder_ GUARDED_BY(crit_sect_);
AudioFrame::VADActivity previous_audio_activity_ GUARDED_BY(crit_sect_);
@ -328,9 +328,9 @@ class AcmReceiver {
ACMResampler resampler_ GUARDED_BY(crit_sect_);
// Used in GetAudio, declared as member to avoid allocating every 10ms.
// TODO(henrik.lundin) Stack-allocate in GetAudio instead?
scoped_ptr<int16_t[]> audio_buffer_ GUARDED_BY(crit_sect_);
scoped_ptr<int16_t[]> last_audio_buffer_ GUARDED_BY(crit_sect_);
scoped_ptr<Nack> nack_ GUARDED_BY(crit_sect_);
rtc::scoped_ptr<int16_t[]> audio_buffer_ GUARDED_BY(crit_sect_);
rtc::scoped_ptr<int16_t[]> last_audio_buffer_ GUARDED_BY(crit_sect_);
rtc::scoped_ptr<Nack> nack_ GUARDED_BY(crit_sect_);
bool nack_enabled_ GUARDED_BY(crit_sect_);
CallStatistics call_stats_ GUARDED_BY(crit_sect_);
NetEq* neteq_;
@ -342,15 +342,15 @@ class AcmReceiver {
// Indicates if a non-zero initial delay is set, and the receiver is in
// AV-sync mode.
bool av_sync_;
scoped_ptr<InitialDelayManager> initial_delay_manager_;
rtc::scoped_ptr<InitialDelayManager> initial_delay_manager_;
// The following are defined as members to avoid creating them in every
// iteration. |missing_packets_sync_stream_| is *ONLY* used in InsertPacket().
// |late_packets_sync_stream_| is only used in GetAudio(). Both of these
// member variables are allocated only when we AV-sync is enabled, i.e.
// initial delay is set.
scoped_ptr<InitialDelayManager::SyncStream> missing_packets_sync_stream_;
scoped_ptr<InitialDelayManager::SyncStream> late_packets_sync_stream_;
rtc::scoped_ptr<InitialDelayManager::SyncStream> missing_packets_sync_stream_;
rtc::scoped_ptr<InitialDelayManager::SyncStream> late_packets_sync_stream_;
};
} // namespace acm2

6
webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc
View File

@ -13,12 +13,12 @@
#include <algorithm> // std::min
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
#include "webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
#include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"
#include "webrtc/system_wrappers/interface/clock.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/test/test_suite.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/test/testsupport/gtest_disable.h"
@ -145,9 +145,9 @@ class AcmReceiverTest : public AudioPacketizationCallback,
return 0;
}
scoped_ptr<AcmReceiver> receiver_;
rtc::scoped_ptr<AcmReceiver> receiver_;
CodecInst codecs_[ACMCodecDB::kMaxNumCodecs];
scoped_ptr<AudioCoding> acm_;
rtc::scoped_ptr<AudioCoding> acm_;
WebRtcRTPHeader rtp_header_;
uint32_t timestamp_;
bool packet_sent_; // Set when SendData is called reset when inserting audio.

6
webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc
View File

@ -13,12 +13,12 @@
#include <algorithm> // std::min
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
#include "webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
#include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"
#include "webrtc/system_wrappers/interface/clock.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/test/test_suite.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/test/testsupport/gtest_disable.h"
@ -149,9 +149,9 @@ class AcmReceiverTestOldApi : public AudioPacketizationCallback,
return 0;
}
scoped_ptr<AcmReceiver> receiver_;
rtc::scoped_ptr<AcmReceiver> receiver_;
CodecInst codecs_[ACMCodecDB::kMaxNumCodecs];
scoped_ptr<AudioCodingModule> acm_;
rtc::scoped_ptr<AudioCodingModule> acm_;
WebRtcRTPHeader rtp_header_;
uint32_t timestamp_;
bool packet_sent_; // Set when SendData is called reset when inserting audio.

4
webrtc/modules/audio_coding/main/acm2/acm_send_test.h
View File

@ -14,10 +14,10 @@
#include <vector>
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
#include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
#include "webrtc/system_wrappers/interface/clock.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
namespace webrtc {
@ -61,7 +61,7 @@ class AcmSendTest : public AudioPacketizationCallback, public PacketSource {
Packet* CreatePacket();
SimulatedClock clock_;
scoped_ptr<AudioCoding> acm_;
rtc::scoped_ptr<AudioCoding> acm_;
InputAudioFile* audio_source_;
int source_rate_hz_;
const int input_block_size_samples_;

4
webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.h
View File

@ -14,10 +14,10 @@
#include <vector>
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
#include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
#include "webrtc/system_wrappers/interface/clock.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
namespace webrtc {
@ -65,7 +65,7 @@ class AcmSendTestOldApi : public AudioPacketizationCallback,
Packet* CreatePacket();
SimulatedClock clock_;
scoped_ptr<AudioCodingModule> acm_;
rtc::scoped_ptr<AudioCodingModule> acm_;
InputAudioFile* audio_source_;
int source_rate_hz_;
const int input_block_size_samples_;

4
webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
View File

@ -13,13 +13,13 @@
#include <vector>
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/common_types.h"
#include "webrtc/engine_configurations.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_receiver.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
namespace webrtc {
@ -429,7 +429,7 @@ class AudioCodingImpl : public AudioCoding {
int playout_frequency_hz_;
// TODO(henrik.lundin): All members below this line are temporary and should
// be removed after refactoring is completed.
scoped_ptr<acm2::AudioCodingModuleImpl> acm_old_;
rtc::scoped_ptr<acm2::AudioCodingModuleImpl> acm_old_;
CodecInst current_send_codec_;
};

26
webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc
View File

@ -14,6 +14,7 @@
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/md5digest.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_receive_test.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_send_test.h"
@ -29,7 +30,6 @@
#include "webrtc/system_wrappers/interface/clock.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/event_wrapper.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/system_wrappers/interface/sleep.h"
#include "webrtc/system_wrappers/interface/thread_wrapper.h"
#include "webrtc/test/testsupport/fileutils.h"
@ -112,7 +112,7 @@ class PacketizationCallbackStub : public AudioPacketizationCallback {
private:
int num_calls_ GUARDED_BY(crit_sect_);
std::vector<uint8_t> last_payload_vec_ GUARDED_BY(crit_sect_);
const scoped_ptr<CriticalSectionWrapper> crit_sect_;
const rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
};
class AudioCodingModuleTest : public ::testing::Test {
@ -188,8 +188,8 @@ class AudioCodingModuleTest : public ::testing::Test {
}
AudioCoding::Config config_;
scoped_ptr<RtpUtility> rtp_utility_;
scoped_ptr<AudioCoding> acm_;
rtc::scoped_ptr<RtpUtility> rtp_utility_;
rtc::scoped_ptr<AudioCoding> acm_;
PacketizationCallbackStub packet_cb_;
WebRtcRTPHeader rtp_header_;
AudioFrame input_frame_;
@ -404,16 +404,16 @@ class AudioCodingModuleMtTest : public AudioCodingModuleTest {
return true;
}
scoped_ptr<ThreadWrapper> send_thread_;
scoped_ptr<ThreadWrapper> insert_packet_thread_;
scoped_ptr<ThreadWrapper> pull_audio_thread_;
const scoped_ptr<EventWrapper> test_complete_;
rtc::scoped_ptr<ThreadWrapper> send_thread_;
rtc::scoped_ptr<ThreadWrapper> insert_packet_thread_;
rtc::scoped_ptr<ThreadWrapper> pull_audio_thread_;
const rtc::scoped_ptr<EventWrapper> test_complete_;
int send_count_;
int insert_packet_count_;
int pull_audio_count_ GUARDED_BY(crit_sect_);
const scoped_ptr<CriticalSectionWrapper> crit_sect_;
const rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
int64_t next_insert_packet_time_ms_ GUARDED_BY(crit_sect_);
scoped_ptr<SimulatedClock> fake_clock_;
rtc::scoped_ptr<SimulatedClock> fake_clock_;
};
TEST_F(AudioCodingModuleMtTest, DoTest) {
@ -531,7 +531,7 @@ class AcmReceiverBitExactness : public ::testing::Test {
void Run(int output_freq_hz, const std::string& checksum_ref) {
const std::string input_file_name =
webrtc::test::ResourcePath("audio_coding/neteq_universal_new", "rtp");
scoped_ptr<test::RtpFileSource> packet_source(
rtc::scoped_ptr<test::RtpFileSource> packet_source(
test::RtpFileSource::Create(input_file_name));
#ifdef WEBRTC_ANDROID
// Filter out iLBC and iSAC-swb since they are not supported on Android.
@ -755,8 +755,8 @@ class AcmSenderBitExactness : public ::testing::Test,
codec_frame_size_rtp_timestamps));
}
scoped_ptr<test::AcmSendTest> send_test_;
scoped_ptr<test::InputAudioFile> audio_source_;
rtc::scoped_ptr<test::AcmSendTest> send_test_;
rtc::scoped_ptr<test::InputAudioFile> audio_source_;
uint32_t frame_size_rtp_timestamps_;
int packet_count_;
uint8_t payload_type_;

26
webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
View File

@ -13,6 +13,7 @@
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/md5digest.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.h"
@ -29,7 +30,6 @@
#include "webrtc/system_wrappers/interface/clock.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/event_wrapper.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/system_wrappers/interface/sleep.h"
#include "webrtc/system_wrappers/interface/thread_wrapper.h"
#include "webrtc/test/testsupport/fileutils.h"
@ -131,7 +131,7 @@ class PacketizationCallbackStubOldApi : public AudioPacketizationCallback {
FrameType last_frame_type_ GUARDED_BY(crit_sect_);
int last_payload_type_ GUARDED_BY(crit_sect_);
std::vector<uint8_t> last_payload_vec_ GUARDED_BY(crit_sect_);
const scoped_ptr<CriticalSectionWrapper> crit_sect_;
const rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
};
class AudioCodingModuleTestOldApi : public ::testing::Test {
@ -205,8 +205,8 @@ class AudioCodingModuleTestOldApi : public ::testing::Test {
}
const int id_;
scoped_ptr<RtpUtility> rtp_utility_;
scoped_ptr<AudioCodingModule> acm_;
rtc::scoped_ptr<RtpUtility> rtp_utility_;
rtc::scoped_ptr<AudioCodingModule> acm_;
PacketizationCallbackStubOldApi packet_cb_;
WebRtcRTPHeader rtp_header_;
AudioFrame input_frame_;
@ -541,16 +541,16 @@ class AudioCodingModuleMtTestOldApi : public AudioCodingModuleTestOldApi {
return true;
}
scoped_ptr<ThreadWrapper> send_thread_;
scoped_ptr<ThreadWrapper> insert_packet_thread_;
scoped_ptr<ThreadWrapper> pull_audio_thread_;
const scoped_ptr<EventWrapper> test_complete_;
rtc::scoped_ptr<ThreadWrapper> send_thread_;
rtc::scoped_ptr<ThreadWrapper> insert_packet_thread_;
rtc::scoped_ptr<ThreadWrapper> pull_audio_thread_;
const rtc::scoped_ptr<EventWrapper> test_complete_;
int send_count_;
int insert_packet_count_;
int pull_audio_count_ GUARDED_BY(crit_sect_);
const scoped_ptr<CriticalSectionWrapper> crit_sect_;
const rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
int64_t next_insert_packet_time_ms_ GUARDED_BY(crit_sect_);
scoped_ptr<SimulatedClock> fake_clock_;
rtc::scoped_ptr<SimulatedClock> fake_clock_;
};
TEST_F(AudioCodingModuleMtTestOldApi, DoTest) {
@ -675,7 +675,7 @@ class AcmReceiverBitExactnessOldApi : public ::testing::Test {
void Run(int output_freq_hz, const std::string& checksum_ref) {
const std::string input_file_name =
webrtc::test::ResourcePath("audio_coding/neteq_universal_new", "rtp");
scoped_ptr<test::RtpFileSource> packet_source(
rtc::scoped_ptr<test::RtpFileSource> packet_source(
test::RtpFileSource::Create(input_file_name));
#ifdef WEBRTC_ANDROID
// Filter out iLBC and iSAC-swb since they are not supported on Android.
@ -907,8 +907,8 @@ class AcmSenderBitExactnessOldApi : public ::testing::Test,
codec_frame_size_rtp_timestamps));
}
scoped_ptr<test::AcmSendTestOldApi> send_test_;
scoped_ptr<test::InputAudioFile> audio_source_;
rtc::scoped_ptr<test::AcmSendTestOldApi> send_test_;
rtc::scoped_ptr<test::InputAudioFile> audio_source_;
uint32_t frame_size_rtp_timestamps_;
int packet_count_;
uint8_t payload_type_;

2
webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h
View File

@ -11,8 +11,8 @@
#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_INITIAL_DELAY_MANAGER_H_
#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_INITIAL_DELAY_MANAGER_H_
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
namespace webrtc {

2
webrtc/modules/audio_coding/main/acm2/initial_delay_manager_unittest.cc
View File

@ -78,7 +78,7 @@ class InitialDelayManagerTest : public ::testing::Test {
NextRtpHeader(rtp_info, rtp_receive_timestamp);
}
scoped_ptr<InitialDelayManager> manager_;
rtc::scoped_ptr<InitialDelayManager> manager_;
WebRtcRTPHeader rtp_info_;
uint32_t rtp_receive_timestamp_;
};

2
webrtc/modules/audio_coding/main/acm2/nack.h
View File

@ -14,8 +14,8 @@
#include <vector>
#include <map>
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/test/testsupport/gtest_prod_util.h"
//

24
webrtc/modules/audio_coding/main/acm2/nack_unittest.cc
View File

@ -15,9 +15,9 @@
#include <algorithm>
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/typedefs.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
namespace webrtc {
@ -58,7 +58,7 @@ bool IsNackListCorrect(const std::vector<uint16_t>& nack_list,
} // namespace
TEST(NackTest, EmptyListWhenNoPacketLoss) {
scoped_ptr<Nack> nack(Nack::Create(kNackThreshold));
rtc::scoped_ptr<Nack> nack(Nack::Create(kNackThreshold));
nack->UpdateSampleRate(kSampleRateHz);
int seq_num = 1;
@ -76,7 +76,7 @@ TEST(NackTest, EmptyListWhenNoPacketLoss) {
}
TEST(NackTest, NoNackIfReorderWithinNackThreshold) {
scoped_ptr<Nack> nack(Nack::Create(kNackThreshold));
rtc::scoped_ptr<Nack> nack(Nack::Create(kNackThreshold));
nack->UpdateSampleRate(kSampleRateHz);
int seq_num = 1;
@ -104,7 +104,7 @@ TEST(NackTest, LatePacketsMovedToNackThenNackListDoesNotChange) {
sizeof(kSequenceNumberLostPackets[0]);
for (int k = 0; k < 2; k++) { // Two iteration with/without wrap around.
scoped_ptr<Nack> nack(Nack::Create(kNackThreshold));
rtc::scoped_ptr<Nack> nack(Nack::Create(kNackThreshold));
nack->UpdateSampleRate(kSampleRateHz);
uint16_t sequence_num_lost_packets[kNumAllLostPackets];
@ -152,7 +152,7 @@ TEST(NackTest, ArrivedPacketsAreRemovedFromNackList) {
sizeof(kSequenceNumberLostPackets[0]);
for (int k = 0; k < 2; ++k) { // Two iteration with/without wrap around.
scoped_ptr<Nack> nack(Nack::Create(kNackThreshold));
rtc::scoped_ptr<Nack> nack(Nack::Create(kNackThreshold));
nack->UpdateSampleRate(kSampleRateHz);
uint16_t sequence_num_lost_packets[kNumAllLostPackets];
@ -215,7 +215,7 @@ TEST(NackTest, EstimateTimestampAndTimeToPlay) {
for (int k = 0; k < 4; ++k) {
scoped_ptr<Nack> nack(Nack::Create(kNackThreshold));
rtc::scoped_ptr<Nack> nack(Nack::Create(kNackThreshold));
nack->UpdateSampleRate(kSampleRateHz);
// Sequence number wrap around if |k| is 2 or 3;
@ -286,7 +286,7 @@ TEST(NackTest, EstimateTimestampAndTimeToPlay) {
TEST(NackTest, MissingPacketsPriorToLastDecodedRtpShouldNotBeInNackList) {
for (int m = 0; m < 2; ++m) {
uint16_t seq_num_offset = (m == 0) ? 0 : 65531; // Wrap around if |m| is 1.
scoped_ptr<Nack> nack(Nack::Create(kNackThreshold));
rtc::scoped_ptr<Nack> nack(Nack::Create(kNackThreshold));
nack->UpdateSampleRate(kSampleRateHz);
// Two consecutive packets to have a correct estimate of timestamp increase.
@ -337,7 +337,7 @@ TEST(NackTest, MissingPacketsPriorToLastDecodedRtpShouldNotBeInNackList) {
}
TEST(NackTest, Reset) {
scoped_ptr<Nack> nack(Nack::Create(kNackThreshold));
rtc::scoped_ptr<Nack> nack(Nack::Create(kNackThreshold));
nack->UpdateSampleRate(kSampleRateHz);
// Two consecutive packets to have a correct estimate of timestamp increase.
@ -364,7 +364,7 @@ TEST(NackTest, ListSizeAppliedFromBeginning) {
const size_t kNackListSize = 10;
for (int m = 0; m < 2; ++m) {
uint16_t seq_num_offset = (m == 0) ? 0 : 65525; // Wrap around if |m| is 1.
scoped_ptr<Nack> nack(Nack::Create(kNackThreshold));
rtc::scoped_ptr<Nack> nack(Nack::Create(kNackThreshold));
nack->UpdateSampleRate(kSampleRateHz);
nack->SetMaxNackListSize(kNackListSize);
@ -388,7 +388,7 @@ TEST(NackTest, ChangeOfListSizeAppliedAndOldElementsRemoved) {
const size_t kNackListSize = 10;
for (int m = 0; m < 2; ++m) {
uint16_t seq_num_offset = (m == 0) ? 0 : 65525; // Wrap around if |m| is 1.
scoped_ptr<Nack> nack(Nack::Create(kNackThreshold));
rtc::scoped_ptr<Nack> nack(Nack::Create(kNackThreshold));
nack->UpdateSampleRate(kSampleRateHz);
uint16_t seq_num = seq_num_offset;
@ -398,7 +398,7 @@ TEST(NackTest, ChangeOfListSizeAppliedAndOldElementsRemoved) {
// Packet lost more than NACK-list size limit.
uint16_t num_lost_packets = kNackThreshold + kNackListSize + 5;
scoped_ptr<uint16_t[]> seq_num_lost(new uint16_t[num_lost_packets]);
rtc::scoped_ptr<uint16_t[]> seq_num_lost(new uint16_t[num_lost_packets]);
for (int n = 0; n < num_lost_packets; ++n) {
seq_num_lost[n] = ++seq_num;
}
@ -454,7 +454,7 @@ TEST(NackTest, ChangeOfListSizeAppliedAndOldElementsRemoved) {
TEST(NackTest, RoudTripTimeIsApplied) {
const int kNackListSize = 200;
scoped_ptr<Nack> nack(Nack::Create(kNackThreshold));
rtc::scoped_ptr<Nack> nack(Nack::Create(kNackThreshold));
nack->UpdateSampleRate(kSampleRateHz);
nack->SetMaxNackListSize(kNackListSize);