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Adopt absl::string_view in call/

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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}
This commit is contained in:
Ali Tofigh 2022-05-17 11:48:46 +02:00 committed by WebRTC LUCI CQ
parent d01e692e01
commit 641a1b11b6
29 changed files with 117 additions and 78 deletions
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9
call/BUILD.gn
View file

@ -78,6 +78,7 @@ rtc_library("call_interfaces") {
]
absl_deps = [
"//third_party/abseil-cpp/absl/functional:bind_front",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
}
@ -127,6 +128,7 @@ rtc_library("rtp_interfaces") {
]
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
}
@ -339,6 +341,7 @@ rtc_library("call") {
]
absl_deps = [
"//third_party/abseil-cpp/absl/functional:bind_front",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
}
@ -591,7 +594,10 @@ if (rtc_include_tests) {
"../video",
"//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",
]
}
}
@ -616,6 +622,7 @@ if (rtc_include_tests) {
"../rtc_base/network:sent_packet",
"../test:test_support",
]
absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
}
rtc_source_set("mock_bitrate_allocator") {
testonly = true

6
call/adaptation/BUILD.gn
View file

@ -57,6 +57,7 @@ rtc_library("resource_adaptation") {
]
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
"//third_party/abseil-cpp/absl/types:variant",
]
@ -123,6 +124,9 @@ if (rtc_include_tests) {
"../../rtc_base/task_utils:to_queued_task",
"../../test:test_support",
]
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
absl_deps = [
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
}
}

3
call/adaptation/broadcast_resource_listener.cc
View file

@ -14,6 +14,7 @@
#include <string>
#include <utility>
#include "absl/strings/string_view.h"
#include "rtc_base/checks.h"
#include "rtc_base/ref_counted_object.h"
#include "rtc_base/synchronization/mutex.h"
@ -24,7 +25,7 @@ namespace webrtc {
// a single ResourceListener.
class BroadcastResourceListener::AdapterResource : public Resource {
public:
explicit AdapterResource(std::string name) : name_(std::move(name)) {}
explicit AdapterResource(absl::string_view name) : name_(std::move(name)) {}
~AdapterResource() override { RTC_DCHECK(!listener_); }
// The parent is letting us know we have a usage neasurement.

6
call/adaptation/resource_adaptation_processor.cc
View file

@ -15,6 +15,7 @@
#include <utility>
#include "absl/algorithm/container.h"
#include "absl/strings/string_view.h"
#include "api/sequence_checker.h"
#include "api/video/video_adaptation_counters.h"
#include "call/adaptation/video_stream_adapter.h"
@ -59,8 +60,9 @@ ResourceAdaptationProcessor::MitigationResultAndLogMessage::
: result(MitigationResult::kAdaptationApplied), message() {}
ResourceAdaptationProcessor::MitigationResultAndLogMessage::
MitigationResultAndLogMessage(MitigationResult result, std::string message)
: result(result), message(std::move(message)) {}
MitigationResultAndLogMessage(MitigationResult result,
absl::string_view message)
: result(result), message(message) {}
ResourceAdaptationProcessor::ResourceAdaptationProcessor(
VideoStreamAdapter* stream_adapter)

4
call/adaptation/resource_adaptation_processor.h
View file

@ -17,6 +17,7 @@
#include <utility>
#include <vector>
#include "absl/strings/string_view.h"
#include "absl/types/optional.h"
#include "api/adaptation/resource.h"
#include "api/rtp_parameters.h"
@ -108,7 +109,8 @@ class ResourceAdaptationProcessor : public ResourceAdaptationProcessorInterface,
struct MitigationResultAndLogMessage {
MitigationResultAndLogMessage();
MitigationResultAndLogMessage(MitigationResult result, std::string message);
MitigationResultAndLogMessage(MitigationResult result,
absl::string_view message);
MitigationResult result;
std::string message;
};

6
call/adaptation/test/fake_adaptation_constraint.cc
View file

@ -12,10 +12,12 @@
#include <utility>
#include "absl/strings/string_view.h"
namespace webrtc {
FakeAdaptationConstraint::FakeAdaptationConstraint(std::string name)
: name_(std::move(name)), is_adaptation_up_allowed_(true) {}
FakeAdaptationConstraint::FakeAdaptationConstraint(absl::string_view name)
: name_(name), is_adaptation_up_allowed_(true) {}
FakeAdaptationConstraint::~FakeAdaptationConstraint() = default;

3
call/adaptation/test/fake_adaptation_constraint.h
View file

@ -13,13 +13,14 @@
#include <string>
#include "absl/strings/string_view.h"
#include "call/adaptation/adaptation_constraint.h"
namespace webrtc {
class FakeAdaptationConstraint : public AdaptationConstraint {
public:
explicit FakeAdaptationConstraint(std::string name);
explicit FakeAdaptationConstraint(absl::string_view name);
~FakeAdaptationConstraint() override;
void set_is_adaptation_up_allowed(bool is_adaptation_up_allowed);

7
call/adaptation/test/fake_resource.cc
View file

@ -13,17 +13,18 @@
#include <algorithm>
#include <utility>
#include "absl/strings/string_view.h"
#include "rtc_base/ref_counted_object.h"
namespace webrtc {
// static
rtc::scoped_refptr<FakeResource> FakeResource::Create(std::string name) {
rtc::scoped_refptr<FakeResource> FakeResource::Create(absl::string_view name) {
return rtc::make_ref_counted<FakeResource>(name);
}
FakeResource::FakeResource(std::string name)
: Resource(), name_(std::move(name)), listener_(nullptr) {}
FakeResource::FakeResource(absl::string_view name)
: Resource(), name_(name), listener_(nullptr) {}
FakeResource::~FakeResource() {}

5
call/adaptation/test/fake_resource.h
View file

@ -14,6 +14,7 @@
#include <string>
#include <vector>
#include "absl/strings/string_view.h"
#include "absl/types/optional.h"
#include "api/adaptation/resource.h"
#include "api/scoped_refptr.h"
@ -23,9 +24,9 @@ namespace webrtc {
// Fake resource used for testing.
class FakeResource : public Resource {
public:
static rtc::scoped_refptr<FakeResource> Create(std::string name);
static rtc::scoped_refptr<FakeResource> Create(absl::string_view name);
explicit FakeResource(std::string name);
explicit FakeResource(absl::string_view name);
~FakeResource() override;
void SetUsageState(ResourceUsageState usage_state);

3
call/bitrate_allocator_unittest.cc
View file

@ -14,6 +14,7 @@
#include <memory>
#include <vector>
#include "absl/strings/string_view.h"
#include "system_wrappers/include/clock.h"
#include "test/gmock.h"
#include "test/gtest.h"
@ -306,7 +307,7 @@ class BitrateAllocatorTestNoEnforceMin : public ::testing::Test {
uint32_t max_bitrate_bps,
uint32_t pad_up_bitrate_bps,
bool enforce_min_bitrate,
std::string track_id,
absl::string_view track_id,
double bitrate_priority) {
allocator_->AddObserver(
observer, {min_bitrate_bps, max_bitrate_bps, pad_up_bitrate_bps, 0,

6
call/bitrate_estimator_tests.cc
View file

@ -41,7 +41,7 @@ class LogObserver {
~LogObserver() { rtc::LogMessage::RemoveLogToStream(&callback_); }
void PushExpectedLogLine(const std::string& expected_log_line) {
void PushExpectedLogLine(absl::string_view expected_log_line) {
callback_.PushExpectedLogLine(expected_log_line);
}
@ -83,9 +83,9 @@ class LogObserver {
bool Wait() { return done_.Wait(test::CallTest::kDefaultTimeoutMs); }
void PushExpectedLogLine(const std::string& expected_log_line) {
void PushExpectedLogLine(absl::string_view expected_log_line) {
MutexLock lock(&mutex_);
expected_log_lines_.push_back(expected_log_line);
expected_log_lines_.emplace_back(expected_log_line);
}
private:

13
call/call.cc
View file

@ -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);

3
call/call.h
View file

@ -15,6 +15,7 @@
#include <string>
#include <vector>
#include "absl/strings/string_view.h"
#include "api/adaptation/resource.h"
#include "api/media_types.h"
#include "api/task_queue/task_queue_base.h"
@ -171,7 +172,7 @@ class Call {
uint32_t local_ssrc) = 0;
virtual void OnUpdateSyncGroup(AudioReceiveStream& stream,
const std::string& sync_group) = 0;
absl::string_view sync_group) = 0;
virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0;

13
call/call_perf_tests.cc
View file

@ -13,6 +13,7 @@
#include <memory>
#include <string>
#include "absl/strings/string_view.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/task_queue/task_queue_base.h"
@ -76,7 +77,7 @@ class CallPerfTest : public test::CallTest {
float video_ntp_speed,
float video_rtp_speed,
float audio_rtp_speed,
const std::string& test_label);
absl::string_view test_label);
void TestMinTransmitBitrate(bool pad_to_min_bitrate);
@ -91,7 +92,7 @@ class CallPerfTest : public test::CallTest {
int start_bwe,
int max_bwe);
void TestEncodeFramerate(VideoEncoderFactory* encoder_factory,
const std::string& payload_name,
absl::string_view payload_name,
const std::vector<int>& max_framerates);
};
@ -104,7 +105,7 @@ class VideoRtcpAndSyncObserver : public test::RtpRtcpObserver,
public:
explicit VideoRtcpAndSyncObserver(TaskQueueBase* task_queue,
Clock* clock,
const std::string& test_label)
absl::string_view test_label)
: test::RtpRtcpObserver(CallPerfTest::kLongTimeoutMs),
clock_(clock),
test_label_(test_label),
@ -169,7 +170,7 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
float video_ntp_speed,
float video_rtp_speed,
float audio_rtp_speed,
const std::string& test_label) {
absl::string_view test_label) {
const char* kSyncGroup = "av_sync";
const uint32_t kAudioSendSsrc = 1234;
const uint32_t kAudioRecvSsrc = 5678;
@ -1055,7 +1056,7 @@ TEST_F(CallPerfTest, MAYBE_Min_Bitrate_VideoAndAudio) {
}
void CallPerfTest::TestEncodeFramerate(VideoEncoderFactory* encoder_factory,
const std::string& payload_name,
absl::string_view payload_name,
const std::vector<int>& max_framerates) {
static constexpr double kAllowedFpsDiff = 1.5;
static constexpr TimeDelta kMinGetStatsInterval = TimeDelta::Millis(400);
@ -1067,7 +1068,7 @@ void CallPerfTest::TestEncodeFramerate(VideoEncoderFactory* encoder_factory,
public test::FrameGeneratorCapturer::SinkWantsObserver {
public:
FramerateObserver(VideoEncoderFactory* encoder_factory,
const std::string& payload_name,
absl::string_view payload_name,
const std::vector<int>& max_framerates,
TaskQueueBase* task_queue)
: EndToEndTest(kDefaultTimeoutMs),

5
call/call_unittest.cc
View file

@ -16,6 +16,7 @@
#include <utility>
#include "absl/memory/memory.h"
#include "absl/strings/string_view.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/task_queue/default_task_queue_factory.h"
@ -82,9 +83,9 @@ namespace {
rtc::scoped_refptr<Resource> FindResourceWhoseNameContains(
const std::vector<rtc::scoped_refptr<Resource>>& resources,
const std::string& name_contains) {
absl::string_view name_contains) {
for (const auto& resource : resources) {
if (resource->Name().find(name_contains) != std::string::npos)
if (resource->Name().find(std::string(name_contains)) != std::string::npos)
return resource;
}
return nullptr;

3
call/degraded_call.cc
View file

@ -13,6 +13,7 @@
#include <memory>
#include <utility>
#include "absl/strings/string_view.h"
#include "rtc_base/location.h"
namespace webrtc {
@ -315,7 +316,7 @@ void DegradedCall::OnLocalSsrcUpdated(FlexfecReceiveStream& stream,
}
void DegradedCall::OnUpdateSyncGroup(AudioReceiveStream& stream,
const std::string& sync_group) {
absl::string_view sync_group) {
call_->OnUpdateSyncGroup(stream, sync_group);
}

3
call/degraded_call.h
View file

@ -19,6 +19,7 @@
#include <string>
#include <vector>
#include "absl/strings/string_view.h"
#include "absl/types/optional.h"
#include "api/call/transport.h"
#include "api/fec_controller.h"
@ -105,7 +106,7 @@ class DegradedCall : public Call, private PacketReceiver {
void OnLocalSsrcUpdated(FlexfecReceiveStream& stream,
uint32_t local_ssrc) override;
void OnUpdateSyncGroup(AudioReceiveStream& stream,
const std::string& sync_group) override;
absl::string_view sync_group) override;
void OnSentPacket(const rtc::SentPacket& sent_packet) override;
protected:

9
call/rampup_tests.cc
View file

@ -13,6 +13,7 @@
#include <memory>
#include "absl/flags/flag.h"
#include "absl/strings/string_view.h"
#include "api/rtc_event_log/rtc_event_log_factory.h"
#include "api/rtc_event_log_output_file.h"
#include "api/task_queue/default_task_queue_factory.h"
@ -58,7 +59,7 @@ RampUpTester::RampUpTester(size_t num_video_streams,
size_t num_flexfec_streams,
unsigned int start_bitrate_bps,
int64_t min_run_time_ms,
const std::string& extension_type,
absl::string_view extension_type,
bool rtx,
bool red,
bool report_perf_stats,
@ -329,9 +330,9 @@ void RampUpTester::PollStats() {
}
void RampUpTester::ReportResult(
const std::string& measurement,
absl::string_view measurement,
size_t value,
const std::string& units,
absl::string_view units,
test::ImproveDirection improve_direction) const {
webrtc::test::PrintResult(
measurement, "",
@ -420,7 +421,7 @@ RampUpDownUpTester::RampUpDownUpTester(size_t num_video_streams,
size_t num_audio_streams,
size_t num_flexfec_streams,
unsigned int start_bitrate_bps,
const std::string& extension_type,
absl::string_view extension_type,
bool rtx,
bool red,
const std::vector<int>& loss_rates,

9
call/rampup_tests.h
View file

@ -17,6 +17,7 @@
#include <utility>
#include <vector>
#include "absl/strings/string_view.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/task_queue/task_queue_base.h"
#include "api/test/simulated_network.h"
@ -43,7 +44,7 @@ class RampUpTester : public test::EndToEndTest {
size_t num_flexfec_streams,
unsigned int start_bitrate_bps,
int64_t min_run_time_ms,
const std::string& extension_type,
absl::string_view extension_type,
bool rtx,
bool red,
bool report_perf_stats,
@ -65,9 +66,9 @@ class RampUpTester : public test::EndToEndTest {
size_t* padding_sent,
size_t* media_sent) const;
void ReportResult(const std::string& measurement,
void ReportResult(absl::string_view measurement,
size_t value,
const std::string& units,
absl::string_view units,
test::ImproveDirection improve_direction) const;
void TriggerTestDone();
@ -128,7 +129,7 @@ class RampUpDownUpTester : public RampUpTester {
size_t num_audio_streams,
size_t num_flexfec_streams,
unsigned int start_bitrate_bps,
const std::string& extension_type,
absl::string_view extension_type,
bool rtx,
bool red,
const std::vector<int>& loss_rates,

18
call/rtp_demuxer.cc
View file

@ -10,6 +10,7 @@
#include "call/rtp_demuxer.h"
#include "absl/strings/string_view.h"
#include "call/rtp_packet_sink_interface.h"
#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
@ -238,8 +239,7 @@ bool RtpDemuxer::AddSink(uint32_t ssrc, RtpPacketSinkInterface* sink) {
return AddSink(criteria, sink);
}
void RtpDemuxer::AddSink(const std::string& rsid,
RtpPacketSinkInterface* sink) {
void RtpDemuxer::AddSink(absl::string_view rsid, RtpPacketSinkInterface* sink) {
RtpDemuxerCriteria criteria(absl::string_view() /* mid */, rsid);
AddSink(criteria, sink);
}
@ -363,7 +363,7 @@ RtpPacketSinkInterface* RtpDemuxer::ResolveSink(
return ResolveSinkByPayloadType(packet.PayloadType(), ssrc);
}
RtpPacketSinkInterface* RtpDemuxer::ResolveSinkByMid(const std::string& mid,
RtpPacketSinkInterface* RtpDemuxer::ResolveSinkByMid(absl::string_view mid,
uint32_t ssrc) {
const auto it = sink_by_mid_.find(mid);
if (it != sink_by_mid_.end()) {
@ -374,11 +374,11 @@ RtpPacketSinkInterface* RtpDemuxer::ResolveSinkByMid(const std::string& mid,
return nullptr;
}
RtpPacketSinkInterface* RtpDemuxer::ResolveSinkByMidRsid(
const std::string& mid,
const std::string& rsid,
uint32_t ssrc) {
const auto it = sink_by_mid_and_rsid_.find(std::make_pair(mid, rsid));
RtpPacketSinkInterface* RtpDemuxer::ResolveSinkByMidRsid(absl::string_view mid,
absl::string_view rsid,
uint32_t ssrc) {
const auto it = sink_by_mid_and_rsid_.find(
std::make_pair(std::string(mid), std::string(rsid)));
if (it != sink_by_mid_and_rsid_.end()) {
RtpPacketSinkInterface* sink = it->second;
AddSsrcSinkBinding(ssrc, sink);
@ -387,7 +387,7 @@ RtpPacketSinkInterface* RtpDemuxer::ResolveSinkByMidRsid(
return nullptr;
}
RtpPacketSinkInterface* RtpDemuxer::ResolveSinkByRsid(const std::string& rsid,
RtpPacketSinkInterface* RtpDemuxer::ResolveSinkByRsid(absl::string_view rsid,
uint32_t ssrc) {
const auto it = sink_by_rsid_.find(rsid);
if (it != sink_by_rsid_.end()) {

10
call/rtp_demuxer.h
View file

@ -145,7 +145,7 @@ class RtpDemuxer {
// Registers a sink's association to an RSID. Only one sink may be associated
// with a given RSID. Null pointer is not allowed.
void AddSink(const std::string& rsid, RtpPacketSinkInterface* sink);
void AddSink(absl::string_view rsid, RtpPacketSinkInterface* sink);
// Removes a sink. Return value reports if anything was actually removed.
// Null pointer is not allowed.
@ -167,12 +167,12 @@ class RtpDemuxer {
RtpPacketSinkInterface* ResolveSink(const RtpPacketReceived& packet);
// Used by the ResolveSink algorithm.
RtpPacketSinkInterface* ResolveSinkByMid(const std::string& mid,
RtpPacketSinkInterface* ResolveSinkByMid(absl::string_view mid,
uint32_t ssrc);
RtpPacketSinkInterface* ResolveSinkByMidRsid(const std::string& mid,
const std::string& rsid,
RtpPacketSinkInterface* ResolveSinkByMidRsid(absl::string_view mid,
absl::string_view rsid,
uint32_t ssrc);
RtpPacketSinkInterface* ResolveSinkByRsid(const std::string& rsid,
RtpPacketSinkInterface* ResolveSinkByRsid(absl::string_view rsid,
uint32_t ssrc);
RtpPacketSinkInterface* ResolveSinkByPayloadType(uint8_t payload_type,
uint32_t ssrc);

23
call/rtp_demuxer_unittest.cc
View file

@ -14,6 +14,7 @@
#include <set>
#include <string>
#include "absl/strings/string_view.h"
#include "call/test/mock_rtp_packet_sink_interface.h"
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
@ -60,18 +61,18 @@ class RtpDemuxerTest : public ::testing::Test {
return AddSink(criteria, sink);
}
bool AddSinkOnlyRsid(const std::string& rsid, RtpPacketSinkInterface* sink) {
bool AddSinkOnlyRsid(absl::string_view rsid, RtpPacketSinkInterface* sink) {
RtpDemuxerCriteria criteria(absl::string_view(), rsid);
return AddSink(criteria, sink);
}
bool AddSinkOnlyMid(const std::string& mid, RtpPacketSinkInterface* sink) {
bool AddSinkOnlyMid(absl::string_view mid, RtpPacketSinkInterface* sink) {
RtpDemuxerCriteria criteria(mid);
return AddSink(criteria, sink);
}
bool AddSinkBothMidRsid(const std::string& mid,
const std::string& rsid,
bool AddSinkBothMidRsid(absl::string_view mid,
absl::string_view rsid,
RtpPacketSinkInterface* sink) {
RtpDemuxerCriteria criteria(mid, rsid);
return AddSink(criteria, sink);
@ -110,7 +111,7 @@ class RtpDemuxerTest : public ::testing::Test {
std::unique_ptr<RtpPacketReceived> CreatePacketWithSsrcMid(
uint32_t ssrc,
const std::string& mid) {
absl::string_view mid) {
RtpPacketReceived::ExtensionManager extension_manager;
extension_manager.Register<RtpMid>(11);
@ -121,7 +122,7 @@ class RtpDemuxerTest : public ::testing::Test {
std::unique_ptr<RtpPacketReceived> CreatePacketWithSsrcRsid(
uint32_t ssrc,
const std::string& rsid) {
absl::string_view rsid) {
RtpPacketReceived::ExtensionManager extension_manager;
extension_manager.Register<RtpStreamId>(6);
@ -132,7 +133,7 @@ class RtpDemuxerTest : public ::testing::Test {
std::unique_ptr<RtpPacketReceived> CreatePacketWithSsrcRrid(
uint32_t ssrc,
const std::string& rrid) {
absl::string_view rrid) {
RtpPacketReceived::ExtensionManager extension_manager;
extension_manager.Register<RepairedRtpStreamId>(7);
@ -143,8 +144,8 @@ class RtpDemuxerTest : public ::testing::Test {
std::unique_ptr<RtpPacketReceived> CreatePacketWithSsrcMidRsid(
uint32_t ssrc,
const std::string& mid,
const std::string& rsid) {
absl::string_view mid,
absl::string_view rsid) {
RtpPacketReceived::ExtensionManager extension_manager;
extension_manager.Register<RtpMid>(11);
extension_manager.Register<RtpStreamId>(6);
@ -157,8 +158,8 @@ class RtpDemuxerTest : public ::testing::Test {
std::unique_ptr<RtpPacketReceived> CreatePacketWithSsrcRsidRrid(
uint32_t ssrc,
const std::string& rsid,
const std::string& rrid) {
absl::string_view rsid,
absl::string_view rrid) {
RtpPacketReceived::ExtensionManager extension_manager;
extension_manager.Register<RtpStreamId>(6);
extension_manager.Register<RepairedRtpStreamId>(7);

3
call/rtp_transport_controller_send.cc
View file

@ -14,6 +14,7 @@
#include <vector>
#include "absl/strings/match.h"
#include "absl/strings/string_view.h"
#include "absl/types/optional.h"
#include "api/transport/goog_cc_factory.h"
#include "api/transport/network_types.h"
@ -270,7 +271,7 @@ bool RtpTransportControllerSend::IsRelevantRouteChange(
}
void RtpTransportControllerSend::OnNetworkRouteChanged(
const std::string& transport_name,
absl::string_view transport_name,
const rtc::NetworkRoute& network_route) {
// Check if the network route is connected.

3
call/rtp_transport_controller_send.h
View file

@ -17,6 +17,7 @@
#include <string>
#include <vector>
#include "absl/strings/string_view.h"
#include "api/network_state_predictor.h"
#include "api/sequence_checker.h"
#include "api/transport/network_control.h"
@ -94,7 +95,7 @@ class RtpTransportControllerSend final
StreamFeedbackProvider* GetStreamFeedbackProvider() override;
void RegisterTargetTransferRateObserver(
TargetTransferRateObserver* observer) override;
void OnNetworkRouteChanged(const std::string& transport_name,
void OnNetworkRouteChanged(absl::string_view transport_name,
const rtc::NetworkRoute& network_route) override;
void OnNetworkAvailability(bool network_available) override;
RtcpBandwidthObserver* GetBandwidthObserver() override;

3
call/rtp_transport_controller_send_interface.h
View file

@ -18,6 +18,7 @@
#include <string>
#include <vector>
#include "absl/strings/string_view.h"
#include "absl/types/optional.h"
#include "api/crypto/crypto_options.h"
#include "api/fec_controller.h"
@ -127,7 +128,7 @@ class RtpTransportControllerSendInterface {
virtual void RegisterTargetTransferRateObserver(
TargetTransferRateObserver* observer) = 0;
virtual void OnNetworkRouteChanged(
const std::string& transport_name,
absl::string_view transport_name,
const rtc::NetworkRoute& network_route) = 0;
virtual void OnNetworkAvailability(bool network_available) = 0;
virtual RtcpBandwidthObserver* GetBandwidthObserver() = 0;

6
call/rtp_video_sender.cc
View file

@ -17,6 +17,7 @@
#include "absl/algorithm/container.h"
#include "absl/strings/match.h"
#include "absl/strings/string_view.h"
#include "api/array_view.h"
#include "api/transport/field_trial_based_config.h"
#include "api/video_codecs/video_codec.h"
@ -55,9 +56,10 @@ static const size_t kPathMTU = 1500;
using webrtc_internal_rtp_video_sender::RtpStreamSender;
bool PayloadTypeSupportsSkippingFecPackets(const std::string& payload_name,
bool PayloadTypeSupportsSkippingFecPackets(absl::string_view payload_name,
const FieldTrialsView& trials) {
const VideoCodecType codecType = PayloadStringToCodecType(payload_name);
const VideoCodecType codecType =
PayloadStringToCodecType(std::string(payload_name));
if (codecType == kVideoCodecVP8 || codecType == kVideoCodecVP9) {
return true;
}

3
call/test/mock_rtp_transport_controller_send.h
View file

@ -16,6 +16,7 @@
#include <string>
#include <vector>
#include "absl/strings/string_view.h"
#include "api/crypto/crypto_options.h"
#include "api/crypto/frame_encryptor_interface.h"
#include "api/frame_transformer_interface.h"
@ -76,7 +77,7 @@ class MockRtpTransportControllerSend
(override));
MOCK_METHOD(void,
OnNetworkRouteChanged,
(const std::string&, const rtc::NetworkRoute&),
(absl::string_view, const rtc::NetworkRoute&),
(override));
MOCK_METHOD(void, OnNetworkAvailability, (bool), (override));
MOCK_METHOD(RtcpBandwidthObserver*, GetBandwidthObserver, (), (override));

3
media/engine/fake_webrtc_call.cc
View file

@ -13,6 +13,7 @@
#include <utility>
#include "absl/algorithm/container.h"
#include "absl/strings/string_view.h"
#include "api/call/audio_sink.h"
#include "modules/rtp_rtcp/source/rtp_util.h"
#include "rtc_base/checks.h"
@ -728,7 +729,7 @@ void FakeCall::OnLocalSsrcUpdated(webrtc::FlexfecReceiveStream& stream,
}
void FakeCall::OnUpdateSyncGroup(webrtc::AudioReceiveStream& stream,
const std::string& sync_group) {
absl::string_view sync_group) {
auto& fake_stream = static_cast<FakeAudioReceiveStream&>(stream);
fake_stream.SetSyncGroup(sync_group);
}

7
media/engine/fake_webrtc_call.h
View file

@ -25,6 +25,7 @@
#include <string>
#include <vector>
#include "absl/strings/string_view.h"
#include "api/transport/field_trial_based_config.h"
#include "api/video/video_frame.h"
#include "call/audio_receive_stream.h"
@ -105,8 +106,8 @@ class FakeAudioReceiveStream final : public webrtc::AudioReceiveStream {
config_.rtp.local_ssrc = local_ssrc;
}
void SetSyncGroup(const std::string& sync_group) {
config_.sync_group = sync_group;
void SetSyncGroup(absl::string_view sync_group) {
config_.sync_group = std::string(sync_group);
}
private:
@ -424,7 +425,7 @@ class FakeCall final : public webrtc::Call, public webrtc::PacketReceiver {
void OnLocalSsrcUpdated(webrtc::FlexfecReceiveStream& stream,
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;
webrtc::TaskQueueBase* const network_thread_;