Index: webrtc/modules/audio_coding/acm2/rent_a_codec.cc |
diff --git a/webrtc/modules/audio_coding/acm2/rent_a_codec.cc b/webrtc/modules/audio_coding/acm2/rent_a_codec.cc |
index 44a38bb7065953ab0e1577aecca5007728112c21..91c5e4d2fc3e2ff84a0590e5579932f670dcc96c 100644 |
--- a/webrtc/modules/audio_coding/acm2/rent_a_codec.cc |
+++ b/webrtc/modules/audio_coding/acm2/rent_a_codec.cc |
@@ -10,6 +10,7 @@ |
#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h" |
+#include <memory> |
#include <utility> |
#include "webrtc/base/logging.h" |
@@ -144,52 +145,53 @@ namespace { |
// Returns a new speech encoder, or null on error. |
// TODO(kwiberg): Don't handle errors here (bug 5033) |
-rtc::scoped_ptr<AudioEncoder> CreateEncoder( |
- const CodecInst& speech_inst, |
- LockedIsacBandwidthInfo* bwinfo) { |
+std::unique_ptr<AudioEncoder> CreateEncoder(const CodecInst& speech_inst, |
+ LockedIsacBandwidthInfo* bwinfo) { |
#if defined(WEBRTC_CODEC_ISACFX) |
if (STR_CASE_CMP(speech_inst.plname, "isac") == 0) |
- return rtc_make_scoped_ptr(new AudioEncoderIsacFix(speech_inst, bwinfo)); |
+ return std::unique_ptr<AudioEncoder>( |
+ new AudioEncoderIsacFix(speech_inst, bwinfo)); |
#endif |
#if defined(WEBRTC_CODEC_ISAC) |
if (STR_CASE_CMP(speech_inst.plname, "isac") == 0) |
- return rtc_make_scoped_ptr(new AudioEncoderIsac(speech_inst, bwinfo)); |
+ return std::unique_ptr<AudioEncoder>( |
+ new AudioEncoderIsac(speech_inst, bwinfo)); |
#endif |
#ifdef WEBRTC_CODEC_OPUS |
if (STR_CASE_CMP(speech_inst.plname, "opus") == 0) |
- return rtc_make_scoped_ptr(new AudioEncoderOpus(speech_inst)); |
+ return std::unique_ptr<AudioEncoder>(new AudioEncoderOpus(speech_inst)); |
#endif |
if (STR_CASE_CMP(speech_inst.plname, "pcmu") == 0) |
- return rtc_make_scoped_ptr(new AudioEncoderPcmU(speech_inst)); |
+ return std::unique_ptr<AudioEncoder>(new AudioEncoderPcmU(speech_inst)); |
if (STR_CASE_CMP(speech_inst.plname, "pcma") == 0) |
- return rtc_make_scoped_ptr(new AudioEncoderPcmA(speech_inst)); |
+ return std::unique_ptr<AudioEncoder>(new AudioEncoderPcmA(speech_inst)); |
if (STR_CASE_CMP(speech_inst.plname, "l16") == 0) |
- return rtc_make_scoped_ptr(new AudioEncoderPcm16B(speech_inst)); |
+ return std::unique_ptr<AudioEncoder>(new AudioEncoderPcm16B(speech_inst)); |
#ifdef WEBRTC_CODEC_ILBC |
if (STR_CASE_CMP(speech_inst.plname, "ilbc") == 0) |
- return rtc_make_scoped_ptr(new AudioEncoderIlbc(speech_inst)); |
+ return std::unique_ptr<AudioEncoder>(new AudioEncoderIlbc(speech_inst)); |
#endif |
#ifdef WEBRTC_CODEC_G722 |
if (STR_CASE_CMP(speech_inst.plname, "g722") == 0) |
- return rtc_make_scoped_ptr(new AudioEncoderG722(speech_inst)); |
+ return std::unique_ptr<AudioEncoder>(new AudioEncoderG722(speech_inst)); |
#endif |
LOG_F(LS_ERROR) << "Could not create encoder of type " << speech_inst.plname; |
- return rtc::scoped_ptr<AudioEncoder>(); |
+ return std::unique_ptr<AudioEncoder>(); |
} |
-rtc::scoped_ptr<AudioEncoder> CreateRedEncoder(AudioEncoder* encoder, |
+std::unique_ptr<AudioEncoder> CreateRedEncoder(AudioEncoder* encoder, |
int red_payload_type) { |
#ifdef WEBRTC_CODEC_RED |
AudioEncoderCopyRed::Config config; |
config.payload_type = red_payload_type; |
config.speech_encoder = encoder; |
- return rtc::scoped_ptr<AudioEncoder>(new AudioEncoderCopyRed(config)); |
+ return std::unique_ptr<AudioEncoder>(new AudioEncoderCopyRed(config)); |
#else |
- return rtc::scoped_ptr<AudioEncoder>(); |
+ return std::unique_ptr<AudioEncoder>(); |
#endif |
} |
-rtc::scoped_ptr<AudioEncoder> CreateCngEncoder(AudioEncoder* encoder, |
+std::unique_ptr<AudioEncoder> CreateCngEncoder(AudioEncoder* encoder, |
int payload_type, |
ACMVADMode vad_mode) { |
AudioEncoderCng::Config config; |
@@ -212,18 +214,18 @@ rtc::scoped_ptr<AudioEncoder> CreateCngEncoder(AudioEncoder* encoder, |
default: |
FATAL(); |
} |
- return rtc::scoped_ptr<AudioEncoder>(new AudioEncoderCng(config)); |
+ return std::unique_ptr<AudioEncoder>(new AudioEncoderCng(config)); |
} |
-rtc::scoped_ptr<AudioDecoder> CreateIsacDecoder( |
+std::unique_ptr<AudioDecoder> CreateIsacDecoder( |
LockedIsacBandwidthInfo* bwinfo) { |
#if defined(WEBRTC_CODEC_ISACFX) |
- return rtc_make_scoped_ptr(new AudioDecoderIsacFix(bwinfo)); |
+ return std::unique_ptr<AudioDecoder>(new AudioDecoderIsacFix(bwinfo)); |
#elif defined(WEBRTC_CODEC_ISAC) |
- return rtc_make_scoped_ptr(new AudioDecoderIsac(bwinfo)); |
+ return std::unique_ptr<AudioDecoder>(new AudioDecoderIsac(bwinfo)); |
#else |
FATAL() << "iSAC is not supported."; |
- return rtc::scoped_ptr<AudioDecoder>(); |
+ return std::unique_ptr<AudioDecoder>(); |
#endif |
} |
@@ -233,7 +235,7 @@ RentACodec::RentACodec() = default; |
RentACodec::~RentACodec() = default; |
AudioEncoder* RentACodec::RentEncoder(const CodecInst& codec_inst) { |
- rtc::scoped_ptr<AudioEncoder> enc = |
+ std::unique_ptr<AudioEncoder> enc = |
CreateEncoder(codec_inst, &isac_bandwidth_info_); |
if (!enc) |
return nullptr; |