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Adopt absl::string_view in function parameters under audio/

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This is part of a large-scale effort to adopt absl::string_view
throughout the WebRTC code base. As a first step, function parameters
of type 'const std::string&' are being converted to absl::string_view.

Bug: webrtc:13579
Change-Id: Ib4618fad3bff2902cd3a4730506aca300949d76c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252982
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36102}
This commit is contained in:
Ali Tofigh 2022-03-01 10:32:20 +01:00 committed by WebRTC LUCI CQ
parent 8290040848
commit 4d278e2caf
4 changed files with 13 additions and 6 deletions
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6
audio/BUILD.gn
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@ -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" ]
}

4
audio/audio_receive_stream.cc
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@ -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) {

3
audio/audio_receive_stream.h
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@ -16,6 +16,7 @@
#include <string>
#include <vector>
#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);

6
audio/test/pc_low_bandwidth_audio_test.cc
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@ -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<webrtc_pc_e2e::PeerConnectionE2EQualityTestFixture>
CreateTestFixture(const std::string& test_case_name,
CreateTestFixture(absl::string_view test_case_name,
TimeController& time_controller,
std::pair<EmulatedNetworkManagerInterface*,
EmulatedNetworkManagerInterface*> network_links,
rtc::FunctionView<void(PeerConfigurer*)> alice_configurer,
rtc::FunctionView<void(PeerConfigurer*)> 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);