| 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;
|
|
|