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| author | Ali Tofigh <alito@webrtc.org> | 2022-05-17 11:48:46 +0200 |
|---|---|---|
| committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | 2022-05-17 12:00:45 +0000 |
| commit | 641a1b11b68f3b8eadee2d138bdf61dce78c8c66 (patch) | |
| tree | 0277f4921c3907d394b4b7642a88dccaaccf9100 /call/call.cc | |
| parent | d01e692e01e6cc139d4e22dbbeb70a6bf9ee10a1 (diff) | |
| download | webrtc-641a1b11b68f3b8eadee2d138bdf61dce78c8c66.tar.gz | |
Adopt absl::string_view in call/
Bug: webrtc:13579
Change-Id: Ib616eb3372da341fafb55c23038182751b9da5a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262780
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36910}
Diffstat (limited to 'call/call.cc')
| -rw-r--r-- | call/call.cc | 13 |
1 files changed, 7 insertions, 6 deletions
diff --git a/call/call.cc b/call/call.cc index 1582bd2341..f96bda5702 100644 --- a/call/call.cc +++ b/call/call.cc @@ -21,6 +21,7 @@ #include <vector> #include "absl/functional/bind_front.h" +#include "absl/strings/string_view.h" #include "absl/types/optional.h" #include "api/rtc_event_log/rtc_event_log.h" #include "api/sequence_checker.h" @@ -272,7 +273,7 @@ class Call final : public webrtc::Call, uint32_t local_ssrc) override; void OnUpdateSyncGroup(webrtc::AudioReceiveStream& stream, - const std::string& sync_group) override; + absl::string_view sync_group) override; void OnSentPacket(const rtc::SentPacket& sent_packet) override; @@ -349,9 +350,9 @@ class Call final : public webrtc::Call, rtc::CopyOnWriteBuffer packet, int64_t packet_time_us) RTC_RUN_ON(worker_thread_); - AudioReceiveStream* FindAudioStreamForSyncGroup(const std::string& sync_group) + AudioReceiveStream* FindAudioStreamForSyncGroup(absl::string_view sync_group) RTC_RUN_ON(worker_thread_); - void ConfigureSync(const std::string& sync_group) RTC_RUN_ON(worker_thread_); + void ConfigureSync(absl::string_view sync_group) RTC_RUN_ON(worker_thread_); void NotifyBweOfReceivedPacket(const RtpPacketReceived& packet, MediaType media_type, @@ -1403,7 +1404,7 @@ void Call::OnLocalSsrcUpdated(FlexfecReceiveStream& stream, } void Call::OnUpdateSyncGroup(webrtc::AudioReceiveStream& stream, - const std::string& sync_group) { + absl::string_view sync_group) { RTC_DCHECK_RUN_ON(worker_thread_); webrtc::internal::AudioReceiveStream& receive_stream = static_cast<webrtc::internal::AudioReceiveStream&>(stream); @@ -1476,7 +1477,7 @@ void Call::OnAllocationLimitsChanged(BitrateAllocationLimits limits) { // RTC_RUN_ON(worker_thread_) AudioReceiveStream* Call::FindAudioStreamForSyncGroup( - const std::string& sync_group) { + absl::string_view sync_group) { RTC_DCHECK_RUN_ON(&receive_11993_checker_); if (!sync_group.empty()) { for (AudioReceiveStream* stream : audio_receive_streams_) { @@ -1490,7 +1491,7 @@ AudioReceiveStream* Call::FindAudioStreamForSyncGroup( // TODO(bugs.webrtc.org/11993): Expect to be called on the network thread. // RTC_RUN_ON(worker_thread_) -void Call::ConfigureSync(const std::string& sync_group) { +void Call::ConfigureSync(absl::string_view sync_group) { // `audio_stream` may be nullptr when clearing the audio stream for a group. AudioReceiveStream* audio_stream = FindAudioStreamForSyncGroup(sync_group); |