diff --git a/audio/BUILD.gn b/audio/BUILD.gn index 23976ab45f..b31ab0451e 100644 --- a/audio/BUILD.gn +++ b/audio/BUILD.gn @@ -104,6 +104,7 @@ rtc_library("audio") { ] absl_deps = [ "//third_party/abseil-cpp/absl/memory", + "//third_party/abseil-cpp/absl/strings", "//third_party/abseil-cpp/absl/types:optional", ] } @@ -225,7 +226,10 @@ if (rtc_include_tests) { "../test/pc/e2e:network_quality_metrics_reporter", "//testing/gtest", ] - absl_deps = [ "//third_party/abseil-cpp/absl/flags:flag" ] + absl_deps = [ + "//third_party/abseil-cpp/absl/flags:flag", + "//third_party/abseil-cpp/absl/strings", + ] if (is_android) { deps += [ "//testing/android/native_test:native_test_native_code" ] } diff --git a/audio/audio_receive_stream.cc b/audio/audio_receive_stream.cc index 6f2444901b..6717878abb 100644 --- a/audio/audio_receive_stream.cc +++ b/audio/audio_receive_stream.cc @@ -453,9 +453,9 @@ void AudioReceiveStream::DeliverRtcp(const uint8_t* packet, size_t length) { channel_receive_->ReceivedRTCPPacket(packet, length); } -void AudioReceiveStream::SetSyncGroup(const std::string& sync_group) { +void AudioReceiveStream::SetSyncGroup(absl::string_view sync_group) { RTC_DCHECK_RUN_ON(&packet_sequence_checker_); - config_.sync_group = sync_group; + config_.sync_group = std::string(sync_group); } void AudioReceiveStream::SetLocalSsrc(uint32_t local_ssrc) { diff --git a/audio/audio_receive_stream.h b/audio/audio_receive_stream.h index ee518c645b..3f4675cd72 100644 --- a/audio/audio_receive_stream.h +++ b/audio/audio_receive_stream.h @@ -16,6 +16,7 @@ #include #include +#include "absl/strings/string_view.h" #include "api/audio/audio_mixer.h" #include "api/neteq/neteq_factory.h" #include "api/rtp_headers.h" @@ -121,7 +122,7 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream, void AssociateSendStream(AudioSendStream* send_stream); void DeliverRtcp(const uint8_t* packet, size_t length); - void SetSyncGroup(const std::string& sync_group); + void SetSyncGroup(absl::string_view sync_group); void SetLocalSsrc(uint32_t local_ssrc); diff --git a/audio/test/pc_low_bandwidth_audio_test.cc b/audio/test/pc_low_bandwidth_audio_test.cc index 9cecc8dd41..0364670b91 100644 --- a/audio/test/pc_low_bandwidth_audio_test.cc +++ b/audio/test/pc_low_bandwidth_audio_test.cc @@ -12,6 +12,7 @@ #include "absl/flags/declare.h" #include "absl/flags/flag.h" +#include "absl/strings/string_view.h" #include "api/test/create_network_emulation_manager.h" #include "api/test/create_peerconnection_quality_test_fixture.h" #include "api/test/network_emulation_manager.h" @@ -71,14 +72,15 @@ CreateTwoNetworkLinks(NetworkEmulationManager* emulation, } std::unique_ptr -CreateTestFixture(const std::string& test_case_name, +CreateTestFixture(absl::string_view test_case_name, TimeController& time_controller, std::pair network_links, rtc::FunctionView alice_configurer, rtc::FunctionView bob_configurer) { auto fixture = webrtc_pc_e2e::CreatePeerConnectionE2EQualityTestFixture( - test_case_name, time_controller, /*audio_quality_analyzer=*/nullptr, + std::string(test_case_name), time_controller, + /*audio_quality_analyzer=*/nullptr, /*video_quality_analyzer=*/nullptr); fixture->AddPeer(network_links.first->network_dependencies(), alice_configurer);