zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

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

4
webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
View File

@ -9,10 +9,10 @@
*/
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_audio/vad/mock/mock_vad.h"
#include "webrtc/modules/audio_coding/codecs/cng/include/audio_encoder_cng.h"
#include "webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
using ::testing::Return;
using ::testing::_;
@ -176,7 +176,7 @@ class AudioEncoderCngTest : public ::testing::Test {
}
AudioEncoderCng::Config config_;
scoped_ptr<AudioEncoderCng> cng_;
rtc::scoped_ptr<AudioEncoderCng> cng_;
MockAudioEncoder mock_encoder_;
MockVad* mock_vad_; // Ownership is transferred to |cng_|.
uint32_t timestamp_;

8
webrtc/modules/audio_coding/codecs/cng/include/audio_encoder_cng.h
View File

@ -13,10 +13,10 @@
#include <vector>
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_audio/vad/include/vad.h"
#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
#include "webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
namespace webrtc {
@ -63,7 +63,7 @@ class AudioEncoderCng final : public AudioEncoder {
private:
// Deleter for use with scoped_ptr. E.g., use as
// scoped_ptr<CNG_enc_inst, CngInstDeleter> cng_inst_;
// rtc::scoped_ptr<CNG_enc_inst, CngInstDeleter> cng_inst_;
struct CngInstDeleter {
inline void operator()(CNG_enc_inst* ptr) const { WebRtcCng_FreeEnc(ptr); }
};
@ -81,8 +81,8 @@ class AudioEncoderCng final : public AudioEncoder {
uint32_t first_timestamp_in_buffer_;
int frames_in_buffer_;
bool last_frame_active_;
scoped_ptr<Vad> vad_;
scoped_ptr<CNG_enc_inst, CngInstDeleter> cng_inst_;
rtc::scoped_ptr<Vad> vad_;
rtc::scoped_ptr<CNG_enc_inst, CngInstDeleter> cng_inst_;
};
} // namespace webrtc

10
webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h
View File

@ -11,9 +11,9 @@
#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_INCLUDE_AUDIO_ENCODER_G722_H_
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_INCLUDE_AUDIO_ENCODER_G722_H_
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
#include "webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
namespace webrtc {
@ -47,8 +47,8 @@ class AudioEncoderG722 : public AudioEncoder {
// The encoder state for one channel.
struct EncoderState {
G722EncInst* encoder;
scoped_ptr<int16_t[]> speech_buffer; // Queued up for encoding.
scoped_ptr<uint8_t[]> encoded_buffer; // Already encoded.
rtc::scoped_ptr<int16_t[]> speech_buffer; // Queued up for encoding.
rtc::scoped_ptr<uint8_t[]> encoded_buffer; // Already encoded.
EncoderState();
~EncoderState();
};
@ -58,8 +58,8 @@ class AudioEncoderG722 : public AudioEncoder {
const int num_10ms_frames_per_packet_;
int num_10ms_frames_buffered_;
uint32_t first_timestamp_in_buffer_;
const scoped_ptr<EncoderState[]> encoders_;
const scoped_ptr<uint8_t[]> interleave_buffer_;
const rtc::scoped_ptr<EncoderState[]> encoders_;
const rtc::scoped_ptr<uint8_t[]> interleave_buffer_;
};
} // namespace webrtc

2
webrtc/modules/audio_coding/codecs/ilbc/interface/audio_encoder_ilbc.h
View File

@ -11,9 +11,9 @@
#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_INTERFACE_AUDIO_ENCODER_ILBC_H_
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_INTERFACE_AUDIO_ENCODER_ILBC_H_
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
#include "webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
namespace webrtc {

6
webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
View File

@ -13,10 +13,10 @@
#include <vector>
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
namespace webrtc {
@ -112,14 +112,14 @@ class AudioEncoderDecoderIsacT : public AudioEncoder, public AudioDecoder {
// iSAC encoder/decoder state, guarded by a mutex to ensure that encode calls
// from one thread won't clash with decode calls from another thread.
// Note: PT_GUARDED_BY is disabled since it is not yet supported by clang.
const scoped_ptr<CriticalSectionWrapper> state_lock_;
const rtc::scoped_ptr<CriticalSectionWrapper> state_lock_;
typename T::instance_type* isac_state_
GUARDED_BY(state_lock_) /* PT_GUARDED_BY(lock_)*/;
int decoder_sample_rate_hz_ GUARDED_BY(state_lock_);
// Must be acquired before state_lock_.
const scoped_ptr<CriticalSectionWrapper> lock_;
const rtc::scoped_ptr<CriticalSectionWrapper> lock_;
// Have we accepted input but not yet emitted it in a packet?
bool packet_in_progress_ GUARDED_BY(lock_);

2
webrtc/modules/audio_coding/codecs/isac/main/source/audio_encoder_isac_red_unittest.cc
View File

@ -11,8 +11,8 @@
#include <stdlib.h>
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
namespace webrtc {

4
webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
View File

@ -9,8 +9,8 @@
*/
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
namespace webrtc {
@ -32,7 +32,7 @@ class AudioEncoderOpusTest : public ::testing::Test {
}
}
scoped_ptr<AudioEncoderOpus> opus_;
rtc::scoped_ptr<AudioEncoderOpus> opus_;
};
namespace {

8
webrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc
View File

@ -9,9 +9,9 @@
*/
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
using ::std::string;
using ::std::tr1::tuple;
@ -60,9 +60,9 @@ class OpusFecTest : public TestWithParam<coding_param> {
string in_filename_;
scoped_ptr<int16_t[]> in_data_;
scoped_ptr<int16_t[]> out_data_;
scoped_ptr<uint8_t[]> bit_stream_;
rtc::scoped_ptr<int16_t[]> in_data_;
rtc::scoped_ptr<int16_t[]> out_data_;
rtc::scoped_ptr<uint8_t[]> bit_stream_;
};
void OpusFecTest::SetUp() {

4
webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
View File

@ -13,8 +13,8 @@
#include <vector>
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
namespace webrtc {
@ -53,7 +53,7 @@ class AudioEncoderCopyRed : public AudioEncoder {
private:
AudioEncoder* speech_encoder_;
int red_payload_type_;
scoped_ptr<uint8_t[]> secondary_encoded_;
rtc::scoped_ptr<uint8_t[]> secondary_encoded_;
size_t secondary_allocated_;
EncodedInfoLeaf secondary_info_;
};

4
webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
View File

@ -10,9 +10,9 @@
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h"
#include "webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
using ::testing::Return;
using ::testing::_;
@ -63,7 +63,7 @@ class AudioEncoderCopyRedTest : public ::testing::Test {
}
MockAudioEncoder mock_encoder_;
scoped_ptr<AudioEncoderCopyRed> red_;
rtc::scoped_ptr<AudioEncoderCopyRed> red_;
uint32_t timestamp_;
int16_t audio_[kMaxNumSamples];
const int sample_rate_hz_;

8
webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h
View File

@ -13,7 +13,7 @@
#include <string>
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/typedefs.h"
namespace webrtc {
@ -60,11 +60,11 @@ class AudioCodecSpeedTest : public testing::TestWithParam<coding_param> {
// Expected output number of samples-per-channel in a frame.
int output_length_sample_;
scoped_ptr<int16_t[]> in_data_;
scoped_ptr<int16_t[]> out_data_;
rtc::scoped_ptr<int16_t[]> in_data_;
rtc::scoped_ptr<int16_t[]> out_data_;
size_t data_pointer_;
size_t loop_length_samples_;
scoped_ptr<uint8_t[]> bit_stream_;
rtc::scoped_ptr<uint8_t[]> bit_stream_;
// Maximum number of bytes in output bitstream for a frame of audio.
int max_bytes_;

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);

6
webrtc/modules/audio_coding/main/test/APITest.h
View File

@ -11,6 +11,7 @@
#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_APITEST_H_
#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_APITEST_H_
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
#include "webrtc/modules/audio_coding/main/test/Channel.h"
@ -18,7 +19,6 @@
#include "webrtc/modules/audio_coding/main/test/utility.h"
#include "webrtc/system_wrappers/interface/event_wrapper.h"
#include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
namespace webrtc {
@ -82,8 +82,8 @@ class APITest : public ACMTest {
bool APIRunB();
//--- ACMs
scoped_ptr<AudioCodingModule> _acmA;
scoped_ptr<AudioCodingModule> _acmB;
rtc::scoped_ptr<AudioCodingModule> _acmA;
rtc::scoped_ptr<AudioCodingModule> _acmB;
//--- Channels
Channel* _channel_A2B;

6
webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc
View File

@ -15,11 +15,11 @@
#include <stdlib.h>
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
#include "webrtc/modules/audio_coding/main/test/utility.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/system_wrappers/interface/trace.h"
#include "webrtc/test/testsupport/fileutils.h"
@ -276,7 +276,7 @@ void EncodeDecodeTest::Perform() {
codePars[1] = 0;
codePars[2] = 0;
scoped_ptr<AudioCodingModule> acm(AudioCodingModule::Create(0));
rtc::scoped_ptr<AudioCodingModule> acm(AudioCodingModule::Create(0));
struct CodecInst sendCodecTmp;
numCodecs = acm->NumberOfCodecs();
@ -332,7 +332,7 @@ std::string EncodeDecodeTest::EncodeToFile(int fileType,
int codeId,
int* codePars,
int testMode) {
scoped_ptr<AudioCodingModule> acm(AudioCodingModule::Create(1));
rtc::scoped_ptr<AudioCodingModule> acm(AudioCodingModule::Create(1));
RTPFile rtpFile;
std::string fileName = webrtc::test::TempFilename(webrtc::test::OutputPath(),
"encode_decode_rtp");

2
webrtc/modules/audio_coding/main/test/PacketLossTest.cc
View File

@ -126,7 +126,7 @@ void PacketLossTest::Perform() {
#ifndef WEBRTC_CODEC_OPUS
return;
#else
scoped_ptr<AudioCodingModule> acm(AudioCodingModule::Create(0));
rtc::scoped_ptr<AudioCodingModule> acm(AudioCodingModule::Create(0));
int codec_id = acm->Codec("opus", 48000, channels_);

6
webrtc/modules/audio_coding/main/test/PacketLossTest.h
View File

@ -12,8 +12,8 @@
#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_PACKETLOSSTEST_H_
#include <string>
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
namespace webrtc {
@ -54,8 +54,8 @@ class PacketLossTest : public ACMTest {
int channels_;
std::string in_file_name_;
int sample_rate_hz_;
scoped_ptr<SenderWithFEC> sender_;
scoped_ptr<ReceiverWithPacketLoss> receiver_;
rtc::scoped_ptr<SenderWithFEC> sender_;
rtc::scoped_ptr<ReceiverWithPacketLoss> receiver_;
int expected_loss_rate_;
int actual_loss_rate_;
int burst_length_;

8
webrtc/modules/audio_coding/main/test/SpatialAudio.h
View File

@ -11,12 +11,12 @@
#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_SPATIALAUDIO_H_
#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_SPATIALAUDIO_H_
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
#include "webrtc/modules/audio_coding/main/test/Channel.h"
#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
#include "webrtc/modules/audio_coding/main/test/utility.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#define MAX_FILE_NAME_LENGTH_BYTE 500
@ -33,9 +33,9 @@ class SpatialAudio : public ACMTest {
void EncodeDecode(double leftPanning, double rightPanning);
void EncodeDecode();
scoped_ptr<AudioCodingModule> _acmLeft;
scoped_ptr<AudioCodingModule> _acmRight;
scoped_ptr<AudioCodingModule> _acmReceiver;
rtc::scoped_ptr<AudioCodingModule> _acmLeft;
rtc::scoped_ptr<AudioCodingModule> _acmRight;
rtc::scoped_ptr<AudioCodingModule> _acmReceiver;
Channel* _channel;
PCMFile _inFile;
PCMFile _outFile;

6
webrtc/modules/audio_coding/main/test/TestAllCodecs.h
View File

@ -11,10 +11,10 @@
#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTALLCODECS_H_
#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTALLCODECS_H_
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
#include "webrtc/modules/audio_coding/main/test/Channel.h"
#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/typedefs.h"
namespace webrtc {
@ -70,8 +70,8 @@ class TestAllCodecs : public ACMTest {
void DisplaySendReceiveCodec();
int test_mode_;
scoped_ptr<AudioCodingModule> acm_a_;
scoped_ptr<AudioCodingModule> acm_b_;
rtc::scoped_ptr<AudioCodingModule> acm_a_;
rtc::scoped_ptr<AudioCodingModule> acm_b_;
TestPack* channel_a_to_b_;
PCMFile infile_a_;
PCMFile outfile_b_;

6
webrtc/modules/audio_coding/main/test/TestRedFec.h
View File

@ -12,10 +12,10 @@
#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TESTREDFEC_H_
#include <string>
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
#include "webrtc/modules/audio_coding/main/test/Channel.h"
#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
namespace webrtc {
@ -36,8 +36,8 @@ class TestRedFec : public ACMTest {
void Run();
void OpenOutFile(int16_t testNumber);
int32_t SetVAD(bool enableDTX, bool enableVAD, ACMVADMode vadMode);
scoped_ptr<AudioCodingModule> _acmA;
scoped_ptr<AudioCodingModule> _acmB;
rtc::scoped_ptr<AudioCodingModule> _acmA;
rtc::scoped_ptr<AudioCodingModule> _acmB;
Channel* _channelA2B;

6
webrtc/modules/audio_coding/main/test/TestStereo.h
View File

@ -13,7 +13,7 @@
#include <math.h>
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
#include "webrtc/modules/audio_coding/main/test/Channel.h"
#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
@ -82,8 +82,8 @@ class TestStereo : public ACMTest {
int test_mode_;
scoped_ptr<AudioCodingModule> acm_a_;
scoped_ptr<AudioCodingModule> acm_b_;
rtc::scoped_ptr<AudioCodingModule> acm_a_;
rtc::scoped_ptr<AudioCodingModule> acm_b_;
TestPackStereo* channel_a2b_;

6
webrtc/modules/audio_coding/main/test/TestVADDTX.h
View File

@ -11,10 +11,10 @@
#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTVADDTX_H_
#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTVADDTX_H_
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
#include "webrtc/modules/audio_coding/main/test/Channel.h"
#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
namespace webrtc {
@ -65,8 +65,8 @@ class TestVADDTX : public ACMTest {
void SetVAD(bool statusDTX, bool statusVAD, int16_t vadMode);
VADDTXstruct GetVAD();
int16_t VerifyTest();
scoped_ptr<AudioCodingModule> _acmA;
scoped_ptr<AudioCodingModule> _acmB;
rtc::scoped_ptr<AudioCodingModule> _acmA;
rtc::scoped_ptr<AudioCodingModule> _acmB;
Channel* _channelA2B;

2
webrtc/modules/audio_coding/main/test/TwoWayCommunication.cc
View File

@ -60,7 +60,7 @@ TwoWayCommunication::~TwoWayCommunication() {
void TwoWayCommunication::ChooseCodec(uint8_t* codecID_A,
uint8_t* codecID_B) {
scoped_ptr<AudioCodingModule> tmpACM(AudioCodingModule::Create(0));
rtc::scoped_ptr<AudioCodingModule> tmpACM(AudioCodingModule::Create(0));
uint8_t noCodec = tmpACM->NumberOfCodecs();
CodecInst codecInst;
printf("List of Supported Codecs\n");

10
webrtc/modules/audio_coding/main/test/TwoWayCommunication.h
View File

@ -11,12 +11,12 @@
#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TWOWAYCOMMUNICATION_H_
#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TWOWAYCOMMUNICATION_H_
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
#include "webrtc/modules/audio_coding/main/test/Channel.h"
#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
#include "webrtc/modules/audio_coding/main/test/utility.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
namespace webrtc {
@ -31,11 +31,11 @@ class TwoWayCommunication : public ACMTest {
void SetUp();
void SetUpAutotest();
scoped_ptr<AudioCodingModule> _acmA;
scoped_ptr<AudioCodingModule> _acmB;
rtc::scoped_ptr<AudioCodingModule> _acmA;
rtc::scoped_ptr<AudioCodingModule> _acmB;
scoped_ptr<AudioCodingModule> _acmRefA;
scoped_ptr<AudioCodingModule> _acmRefB;
rtc::scoped_ptr<AudioCodingModule> _acmRefA;
rtc::scoped_ptr<AudioCodingModule> _acmRefB;
Channel* _channel_A2B;
Channel* _channel_B2A;

6
webrtc/modules/audio_coding/main/test/delay_test.cc
View File

@ -15,6 +15,7 @@
#include "gflags/gflags.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common.h"
#include "webrtc/common_types.h"
#include "webrtc/engine_configurations.h"
@ -25,7 +26,6 @@
#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
#include "webrtc/modules/audio_coding/main/test/utility.h"
#include "webrtc/system_wrappers/interface/event_wrapper.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/test/testsupport/fileutils.h"
DEFINE_string(codec, "isac", "Codec Name");
@ -229,8 +229,8 @@ class DelayTest {
out_file_b_.Close();
}
scoped_ptr<AudioCodingModule> acm_a_;
scoped_ptr<AudioCodingModule> acm_b_;
rtc::scoped_ptr<AudioCodingModule> acm_a_;
rtc::scoped_ptr<AudioCodingModule> acm_b_;
Channel* channel_a2b_;

10
webrtc/modules/audio_coding/main/test/iSACTest.h
View File

@ -13,13 +13,13 @@
#include <string.h>
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
#include "webrtc/modules/audio_coding/main/test/Channel.h"
#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
#include "webrtc/modules/audio_coding/main/test/utility.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#define MAX_FILE_NAME_LENGTH_BYTE 500
#define NO_OF_CLIENTS 15
@ -53,11 +53,11 @@ class ISACTest : public ACMTest {
void SwitchingSamplingRate(int testNr, int maxSampRateChange);
scoped_ptr<AudioCodingModule> _acmA;
scoped_ptr<AudioCodingModule> _acmB;
rtc::scoped_ptr<AudioCodingModule> _acmA;
rtc::scoped_ptr<AudioCodingModule> _acmB;
scoped_ptr<Channel> _channel_A2B;
scoped_ptr<Channel> _channel_B2A;
rtc::scoped_ptr<Channel> _channel_A2B;
rtc::scoped_ptr<Channel> _channel_B2A;
PCMFile _inFileA;
PCMFile _inFileB;

6
webrtc/modules/audio_coding/main/test/initial_delay_unittest.cc
View File

@ -16,6 +16,7 @@
#include <iostream>
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_types.h"
#include "webrtc/engine_configurations.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
@ -23,7 +24,6 @@
#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
#include "webrtc/modules/audio_coding/main/test/utility.h"
#include "webrtc/system_wrappers/interface/event_wrapper.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/test/testsupport/gtest_disable.h"
@ -156,8 +156,8 @@ class InitialPlayoutDelayTest : public ::testing::Test {
ASSERT_LE(num_frames * 10, initial_delay_ms + 100);
}
scoped_ptr<AudioCodingModule> acm_a_;
scoped_ptr<AudioCodingModule> acm_b_;
rtc::scoped_ptr<AudioCodingModule> acm_a_;
rtc::scoped_ptr<AudioCodingModule> acm_b_;
Channel* channel_a2b_;
};

6
webrtc/modules/audio_coding/main/test/insert_packet_with_timing.cc
View File

@ -12,13 +12,13 @@
#include "gflags/gflags.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
#include "webrtc/modules/audio_coding/main/test/Channel.h"
#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/system_wrappers/interface/clock.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/test/testsupport/fileutils.h"
// Codec.
@ -249,8 +249,8 @@ class InsertPacketWithTiming {
SimulatedClock* sender_clock_;
SimulatedClock* receiver_clock_;
scoped_ptr<AudioCodingModule> send_acm_;
scoped_ptr<AudioCodingModule> receive_acm_;
rtc::scoped_ptr<AudioCodingModule> send_acm_;
rtc::scoped_ptr<AudioCodingModule> receive_acm_;
Channel* channel_;
FILE* seq_num_fid_; // Input (text), one sequence number per line.

4
webrtc/modules/audio_coding/main/test/opus_test.h
View File

@ -13,12 +13,12 @@
#include <math.h>
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h"
#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
#include "webrtc/modules/audio_coding/main/test/Channel.h"
#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
#include "webrtc/modules/audio_coding/main/test/TestStereo.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
namespace webrtc {
@ -35,7 +35,7 @@ class OpusTest : public ACMTest {
void OpenOutFile(int test_number);
scoped_ptr<AudioCodingModule> acm_receiver_;
rtc::scoped_ptr<AudioCodingModule> acm_receiver_;
TestPackStereo* channel_a2b_;
PCMFile in_file_stereo_;
PCMFile in_file_mono_;

4
webrtc/modules/audio_coding/main/test/target_delay_unittest.cc
View File

@ -9,12 +9,12 @@
*/
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
#include "webrtc/modules/audio_coding/main/test/utility.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/system_wrappers/interface/sleep.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/test/testsupport/gtest_disable.h"
@ -194,7 +194,7 @@ class TargetDelayTest : public ::testing::Test {
return acm_->LeastRequiredDelayMs();
}
scoped_ptr<AudioCodingModule> acm_;
rtc::scoped_ptr<AudioCodingModule> acm_;
WebRtcRTPHeader rtp_info_;
uint8_t payload_[kPayloadLenBytes];
};

2
webrtc/modules/audio_coding/neteq/audio_classifier.h
View File

@ -17,7 +17,7 @@ extern "C" {
#include "opus_private.h"
}
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/typedefs.h"
namespace webrtc {

2
webrtc/modules/audio_coding/neteq/audio_classifier_unittest.cc
View File

@ -39,7 +39,7 @@ void RunAnalysisTest(const std::string& audio_filename,
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");

20
webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
View File

@ -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 @@ class AudioDecoderTest : public ::testing::Test {
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 @@ class AudioDecoderTest : public ::testing::Test {
// 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 @@ class AudioDecoderTest : public ::testing::Test {
// 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 @@ class AudioDecoderTest : public ::testing::Test {
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 @@ class AudioDecoderIlbcTest : public AudioDecoderTest {
// 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);

2
webrtc/modules/audio_coding/neteq/audio_vector.cc
View File

@ -155,7 +155,7 @@ int16_t& AudioVector::operator[](size_t index) {
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;

4
webrtc/modules/audio_coding/neteq/audio_vector.h
View File

@ -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 @@ class AudioVector {
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.

4
webrtc/modules/audio_coding/neteq/background_noise.h
View File

@ -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 @@ class BackgroundNoise {
int32_t residual_energy);
size_t num_channels_;
scoped_ptr<ChannelParameters[]> channel_parameters_;
rtc::scoped_ptr<ChannelParameters[]> channel_parameters_;
bool initialized_;
NetEq::BackgroundNoiseMode mode_;

4
webrtc/modules/audio_coding/neteq/expand.h
View File

@ -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 @@ class Expand {
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);
};

5
webrtc/modules/audio_coding/neteq/merge.cc
View File

@ -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 @@ int16_t Merge::CorrelateAndPeakSearch(int16_t expanded_max, int16_t input_max,
// 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,

10
webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc
View File

@ -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 @@ class NetEqExternalDecoderUnitTest : public test::NetEqExternalDecoderTest {
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 @@ class NetEqExternalVsInternalDecoderTest : public NetEqExternalDecoderUnitTest,
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];
};

57
webrtc/modules/audio_coding/neteq/neteq_impl.h
View File

@ -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 @@ class NetEqImpl : public webrtc::NetEq {
// 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 @@ class NetEqImpl : public webrtc::NetEq {
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_);

4
webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
View File

@ -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 @@ struct NetEqNetworkStatsCheck {
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_;

4
webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc
View File

@ -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 @@ class NetEqStereoTest : public ::testing::TestWithParam<TestParameters> {
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 {

6
webrtc/modules/audio_coding/neteq/neteq_unittest.cc
View File

@ -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 @@ class NetEqDecodingTest : public ::testing::Test {
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_;

6
webrtc/modules/audio_coding/neteq/normal_unittest.cc
View File

@ -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 @@ TEST(Normal, AvoidDivideByZero) {
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 @@ TEST(Normal, InputLengthAndChannelsDoNotMatch) {
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;
}

22
webrtc/modules/audio_coding/neteq/payload_splitter_unittest.cc
View File

@ -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 @@ TEST(AudioPayloadSplitter, NonSplittable) {
// 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 @@ TEST_P(SplitBySamplesTest, PayloadSizes) {
// 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 @@ TEST_P(SplitIlbcTest, NumFrames) {
// 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 @@ TEST(IlbcPayloadSplitter, TooLargePayload) {
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 @@ TEST(IlbcPayloadSplitter, UnevenPayload) {
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()));

4
webrtc/modules/audio_coding/neteq/test/audio_classifier_test.cc
View File

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

4
webrtc/modules/audio_coding/neteq/time_stretch.cc
View File

@ -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 @@ TimeStretch::ReturnCodes TimeStretch::Process(
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;

4
webrtc/modules/audio_coding/neteq/tools/audio_loop.h
View File

@ -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);
};

4
webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h
View File

@ -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

18
webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h
View File

@ -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_;

20
webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
View File

@ -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.

4
webrtc/modules/audio_coding/neteq/tools/packet.cc
View File

@ -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);
}

4
webrtc/modules/audio_coding/neteq/tools/packet.h
View File

@ -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.

4
webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.cc
View File

@ -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(

6
webrtc/modules/audio_coding/neteq/tools/rtp_analyze.cc
View File

@ -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()) {

10
webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
View File

@ -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;

6
webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h
View File

@ -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);
};

4
webrtc/modules/audio_coding/neteq/tools/rtpcat.cc
View File

@ -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;