[cleanup] Fix redundant webrtc name specifier
This CL was uploaded by git cl split.
R=hta@webrtc.org
Bug: webrtc:42232595
No-IWYU: LSC
Change-Id: I5df65e5bcf25eac9c8a34b7d785c53ab9b115e47
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390441
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#44571}
diff --git a/audio/audio_receive_stream.cc b/audio/audio_receive_stream.cc
index 344c4b4..1c4d634 100644
--- a/audio/audio_receive_stream.cc
+++ b/audio/audio_receive_stream.cc
@@ -77,9 +77,9 @@
namespace {
std::unique_ptr<voe::ChannelReceiveInterface> CreateChannelReceive(
const Environment& env,
- webrtc::AudioState* audio_state,
+ AudioState* audio_state,
NetEqFactory* neteq_factory,
- const webrtc::AudioReceiveStreamInterface::Config& config) {
+ const AudioReceiveStreamInterface::Config& config) {
RTC_DCHECK(audio_state);
internal::AudioState* internal_audio_state =
static_cast<internal::AudioState*>(audio_state);
@@ -98,8 +98,8 @@
const Environment& env,
PacketRouter* packet_router,
NetEqFactory* neteq_factory,
- const webrtc::AudioReceiveStreamInterface::Config& config,
- const scoped_refptr<webrtc::AudioState>& audio_state)
+ const AudioReceiveStreamInterface::Config& config,
+ const scoped_refptr<AudioState>& audio_state)
: AudioReceiveStreamImpl(
env,
packet_router,
@@ -111,8 +111,8 @@
AudioReceiveStreamImpl::AudioReceiveStreamImpl(
const Environment& /* env */,
PacketRouter* packet_router,
- const webrtc::AudioReceiveStreamInterface::Config& config,
- const scoped_refptr<webrtc::AudioState>& audio_state,
+ const AudioReceiveStreamInterface::Config& config,
+ const scoped_refptr<AudioState>& audio_state,
std::unique_ptr<voe::ChannelReceiveInterface> channel_receive)
: config_(config),
audio_state_(audio_state),
@@ -159,7 +159,7 @@
}
void AudioReceiveStreamImpl::ReconfigureForTesting(
- const webrtc::AudioReceiveStreamInterface::Config& config) {
+ const AudioReceiveStreamInterface::Config& config) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
// SSRC can't be changed mid-stream.
@@ -212,7 +212,7 @@
}
void AudioReceiveStreamImpl::SetDepacketizerToDecoderFrameTransformer(
- scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) {
+ scoped_refptr<FrameTransformerInterface> frame_transformer) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
channel_receive_->SetDepacketizerToDecoderFrameTransformer(
std::move(frame_transformer));
@@ -238,7 +238,7 @@
channel_receive_->SetNACKStatus(history_ms != 0, history_ms / 20);
}
-void AudioReceiveStreamImpl::SetRtcpMode(webrtc::RtcpMode mode) {
+void AudioReceiveStreamImpl::SetRtcpMode(RtcpMode mode) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
if (config_.rtp.rtcp_mode == mode)
@@ -255,17 +255,17 @@
}
void AudioReceiveStreamImpl::SetFrameDecryptor(
- scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor) {
+ scoped_refptr<FrameDecryptorInterface> frame_decryptor) {
// TODO(bugs.webrtc.org/11993): This is called via WebRtcAudioReceiveStream,
// expect to be called on the network thread.
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
channel_receive_->SetFrameDecryptor(std::move(frame_decryptor));
}
-webrtc::AudioReceiveStreamInterface::Stats AudioReceiveStreamImpl::GetStats(
+AudioReceiveStreamInterface::Stats AudioReceiveStreamImpl::GetStats(
bool get_and_clear_legacy_stats) const {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
- webrtc::AudioReceiveStreamInterface::Stats stats;
+ AudioReceiveStreamInterface::Stats stats;
stats.remote_ssrc = remote_ssrc();
auto receive_codec = channel_receive_->GetReceiveCodec();
@@ -274,8 +274,7 @@
stats.codec_payload_type = receive_codec->first;
}
- webrtc::CallReceiveStatistics call_stats =
- channel_receive_->GetRTCPStatistics();
+ CallReceiveStatistics call_stats = channel_receive_->GetRTCPStatistics();
stats.payload_bytes_received = call_stats.payload_bytes_received;
stats.header_and_padding_bytes_received =
call_stats.header_and_padding_bytes_received;
diff --git a/audio/audio_state_unittest.cc b/audio/audio_state_unittest.cc
index 0c278d7..6a98bdd 100644
--- a/audio/audio_state_unittest.cc
+++ b/audio/audio_state_unittest.cc
@@ -57,7 +57,7 @@
std::unique_ptr<TaskQueueBase, TaskQueueDeleter> CreateTaskQueue(
absl::string_view /* name */,
Priority /* priority */) const override {
- return std::unique_ptr<webrtc::TaskQueueBase, webrtc::TaskQueueDeleter>(
+ return std::unique_ptr<TaskQueueBase, TaskQueueDeleter>(
new FakeTaskQueue());
}
};
diff --git a/audio/channel_receive.cc b/audio/channel_receive.cc
index cfbf843..832f86a 100644
--- a/audio/channel_receive.cc
+++ b/audio/channel_receive.cc
@@ -132,7 +132,7 @@
scoped_refptr<AudioDecoderFactory> decoder_factory,
std::optional<AudioCodecPairId> codec_pair_id,
scoped_refptr<FrameDecryptorInterface> frame_decryptor,
- const webrtc::CryptoOptions& crypto_options,
+ const CryptoOptions& crypto_options,
scoped_refptr<FrameTransformerInterface> frame_transformer);
~ChannelReceive() override;
@@ -190,7 +190,7 @@
CallReceiveStatistics GetRTCPStatistics() const override;
void SetNACKStatus(bool enable, int max_packets) override;
- void SetRtcpMode(webrtc::RtcpMode mode) override;
+ void SetRtcpMode(RtcpMode mode) override;
void SetNonSenderRttMeasurement(bool enabled) override;
AudioMixer::Source::AudioFrameInfo GetAudioFrameWithInfo(
@@ -204,11 +204,10 @@
// Sets a frame transformer between the depacketizer and the decoder, to
// transform the received frames before decoding them.
void SetDepacketizerToDecoderFrameTransformer(
- scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
- override;
+ scoped_refptr<FrameTransformerInterface> frame_transformer) override;
void SetFrameDecryptor(
- scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor) override;
+ scoped_refptr<FrameDecryptorInterface> frame_decryptor) override;
void OnLocalSsrcChange(uint32_t local_ssrc) override;
@@ -233,7 +232,7 @@
RTC_RUN_ON(worker_thread_checker_);
void InitFrameTransformerDelegate(
- scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
+ scoped_refptr<FrameTransformerInterface> frame_transformer)
RTC_RUN_ON(worker_thread_checker_);
// Thread checkers document and lock usage of some methods to specific threads
@@ -293,7 +292,7 @@
mutable Mutex ts_stats_lock_;
- webrtc::RtpTimestampUnwrapper rtp_ts_wraparound_handler_;
+ RtpTimestampUnwrapper rtp_ts_wraparound_handler_;
// The rtp timestamp of the first played out audio frame.
int64_t capture_start_rtp_time_stamp_;
// The capture ntp time (in local timebase) of the first played out audio
@@ -310,12 +309,12 @@
// E2EE Audio Frame Decryption
scoped_refptr<FrameDecryptorInterface> frame_decryptor_
RTC_GUARDED_BY(worker_thread_checker_);
- webrtc::CryptoOptions crypto_options_;
+ CryptoOptions crypto_options_;
- webrtc::AbsoluteCaptureTimeInterpolator absolute_capture_time_interpolator_
+ AbsoluteCaptureTimeInterpolator absolute_capture_time_interpolator_
RTC_GUARDED_BY(worker_thread_checker_);
- webrtc::CaptureClockOffsetUpdater capture_clock_offset_updater_
+ CaptureClockOffsetUpdater capture_clock_offset_updater_
RTC_GUARDED_BY(ts_stats_lock_);
scoped_refptr<ChannelReceiveFrameTransformerDelegate>
@@ -381,7 +380,7 @@
}
void ChannelReceive::InitFrameTransformerDelegate(
- scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) {
+ scoped_refptr<FrameTransformerInterface> frame_transformer) {
RTC_DCHECK(frame_transformer);
RTC_DCHECK(!frame_transformer_delegate_);
RTC_DCHECK(worker_thread_->IsCurrent());
@@ -565,7 +564,7 @@
scoped_refptr<AudioDecoderFactory> decoder_factory,
std::optional<AudioCodecPairId> codec_pair_id,
scoped_refptr<FrameDecryptorInterface> frame_decryptor,
- const webrtc::CryptoOptions& crypto_options,
+ const CryptoOptions& crypto_options,
scoped_refptr<FrameTransformerInterface> frame_transformer)
: env_(env),
worker_thread_(TaskQueueBase::Current()),
@@ -713,14 +712,14 @@
Buffer decrypted_audio_payload;
if (frame_decryptor_ != nullptr) {
const size_t max_plaintext_size = frame_decryptor_->GetMaxPlaintextByteSize(
- webrtc::MediaType::AUDIO, payload_length);
+ MediaType::AUDIO, payload_length);
decrypted_audio_payload.SetSize(max_plaintext_size);
const std::vector<uint32_t> csrcs(header.arrOfCSRCs,
header.arrOfCSRCs + header.numCSRCs);
const FrameDecryptorInterface::Result decrypt_result =
frame_decryptor_->Decrypt(
- webrtc::MediaType::AUDIO, csrcs,
+ MediaType::AUDIO, csrcs,
/*additional_data=*/
nullptr, ArrayView<const uint8_t>(payload, payload_data_length),
decrypted_audio_payload);
@@ -747,7 +746,7 @@
char buf[1024];
SimpleStringBuilder mime_type(buf);
auto it = payload_type_map_.find(header.payloadType);
- mime_type << webrtc::MediaTypeToString(webrtc::MediaType::AUDIO) << "/"
+ mime_type << MediaTypeToString(MediaType::AUDIO) << "/"
<< (it != payload_type_map_.end() ? it->second.name
: "x-unknown");
frame_transformer_delegate_->Transform(payload_data, header, remote_ssrc_,
@@ -906,7 +905,7 @@
}
}
-void ChannelReceive::SetRtcpMode(webrtc::RtcpMode mode) {
+void ChannelReceive::SetRtcpMode(RtcpMode mode) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
rtp_rtcp_->SetRTCPStatus(mode);
}
@@ -933,7 +932,7 @@
}
void ChannelReceive::SetDepacketizerToDecoderFrameTransformer(
- scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) {
+ scoped_refptr<FrameTransformerInterface> frame_transformer) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
if (!frame_transformer) {
RTC_DCHECK_NOTREACHED() << "Not setting the transformer?";
@@ -953,7 +952,7 @@
}
void ChannelReceive::SetFrameDecryptor(
- scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor) {
+ scoped_refptr<FrameDecryptorInterface> frame_decryptor) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
frame_decryptor_ = std::move(frame_decryptor);
}
@@ -1189,7 +1188,7 @@
scoped_refptr<AudioDecoderFactory> decoder_factory,
std::optional<AudioCodecPairId> codec_pair_id,
scoped_refptr<FrameDecryptorInterface> frame_decryptor,
- const webrtc::CryptoOptions& crypto_options,
+ const CryptoOptions& crypto_options,
scoped_refptr<FrameTransformerInterface> frame_transformer) {
return std::make_unique<ChannelReceive>(
env, neteq_factory, audio_device_module, rtcp_send_transport, local_ssrc,
diff --git a/audio/channel_receive_frame_transformer_delegate_unittest.cc b/audio/channel_receive_frame_transformer_delegate_unittest.cc
index 95bfd50..4b60ea2 100644
--- a/audio/channel_receive_frame_transformer_delegate_unittest.cc
+++ b/audio/channel_receive_frame_transformer_delegate_unittest.cc
@@ -42,7 +42,7 @@
public:
MOCK_METHOD(void,
ReceiveFrame,
- (webrtc::ArrayView<const uint8_t> packet,
+ (ArrayView<const uint8_t> packet,
const RTPHeader& header,
Timestamp receive_time));
diff --git a/audio/channel_send.cc b/audio/channel_send.cc
index 52817f3..e953549 100644
--- a/audio/channel_send.cc
+++ b/audio/channel_send.cc
@@ -132,7 +132,7 @@
Transport* rtp_transport,
RtcpRttStats* rtcp_rtt_stats,
FrameEncryptorInterface* frame_encryptor,
- const webrtc::CryptoOptions& crypto_options,
+ const CryptoOptions& crypto_options,
bool extmap_allow_mixed,
int rtcp_report_interval_ms,
uint32_t ssrc,
@@ -204,8 +204,7 @@
// Sets a frame transformer between encoder and packetizer, to transform
// encoded frames before sending them out the network.
void SetEncoderToPacketizerFrameTransformer(
- scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
- override;
+ scoped_refptr<FrameTransformerInterface> frame_transformer) override;
// RtcpPacketTypeCounterObserver.
void RtcpPacketTypesCounterUpdated(
@@ -249,7 +248,7 @@
void OnReceivedRtt(int64_t rtt_ms);
void InitFrameTransformerDelegate(
- scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer);
+ scoped_refptr<FrameTransformerInterface> frame_transformer);
// Calls the encoder on the encoder queue (instead of blocking).
void CallEncoderAsync(absl::AnyInvocable<void(AudioEncoder*)> modifier);
@@ -301,7 +300,7 @@
scoped_refptr<FrameEncryptorInterface> frame_encryptor_
RTC_GUARDED_BY(encoder_queue_checker_);
// E2EE Frame Encryption Options
- const webrtc::CryptoOptions crypto_options_;
+ const CryptoOptions crypto_options_;
// Delegates calls to a frame transformer to transform audio, and
// receives callbacks with the transformed frames; delegates calls to
@@ -389,7 +388,7 @@
// is transformed, the delegate will call SendRtpAudio to send it.
char buf[1024];
SimpleStringBuilder mime_type(buf);
- mime_type << webrtc::MediaTypeToString(webrtc::MediaType::AUDIO) << "/"
+ mime_type << MediaTypeToString(MediaType::AUDIO) << "/"
<< encoder_format_.name;
frame_transformer_delegate_->Transform(
frameType, payloadType, rtp_timestamp + rtp_rtcp_->StartTimestamp(),
@@ -421,13 +420,13 @@
// TODO(benwright@webrtc.org) - Allocate enough to always encrypt inline.
// Allocate a buffer to hold the maximum possible encrypted payload.
size_t max_ciphertext_size = frame_encryptor_->GetMaxCiphertextByteSize(
- webrtc::MediaType::AUDIO, payload.size());
+ MediaType::AUDIO, payload.size());
encrypted_audio_payload.SetSize(max_ciphertext_size);
// Encrypt the audio payload into the buffer.
size_t bytes_written = 0;
int encrypt_status =
- frame_encryptor_->Encrypt(webrtc::MediaType::AUDIO, rtp_rtcp_->SSRC(),
+ frame_encryptor_->Encrypt(MediaType::AUDIO, rtp_rtcp_->SSRC(),
/*additional_data=*/nullptr, payload,
encrypted_audio_payload, &bytes_written);
if (encrypt_status != 0) {
@@ -489,7 +488,7 @@
Transport* rtp_transport,
RtcpRttStats* rtcp_rtt_stats,
FrameEncryptorInterface* frame_encryptor,
- const webrtc::CryptoOptions& crypto_options,
+ const CryptoOptions& crypto_options,
bool extmap_allow_mixed,
int rtcp_report_interval_ms,
uint32_t ssrc,
@@ -919,7 +918,7 @@
}
void ChannelSend::SetEncoderToPacketizerFrameTransformer(
- scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) {
+ scoped_refptr<FrameTransformerInterface> frame_transformer) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
if (!frame_transformer)
return;
@@ -937,7 +936,7 @@
}
void ChannelSend::InitFrameTransformerDelegate(
- scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) {
+ scoped_refptr<FrameTransformerInterface> frame_transformer) {
RTC_DCHECK_RUN_ON(&encoder_queue_checker_);
RTC_DCHECK(frame_transformer);
RTC_DCHECK(!frame_transformer_delegate_);
@@ -971,7 +970,7 @@
Transport* rtp_transport,
RtcpRttStats* rtcp_rtt_stats,
FrameEncryptorInterface* frame_encryptor,
- const webrtc::CryptoOptions& crypto_options,
+ const CryptoOptions& crypto_options,
bool extmap_allow_mixed,
int rtcp_report_interval_ms,
uint32_t ssrc,
diff --git a/audio/channel_send_frame_transformer_delegate_unittest.cc b/audio/channel_send_frame_transformer_delegate_unittest.cc
index 2a526f0..e9575c1 100644
--- a/audio/channel_send_frame_transformer_delegate_unittest.cc
+++ b/audio/channel_send_frame_transformer_delegate_unittest.cc
@@ -51,9 +51,9 @@
(AudioFrameType frameType,
uint8_t payloadType,
uint32_t rtp_timestamp,
- webrtc::ArrayView<const uint8_t> payload,
+ ArrayView<const uint8_t> payload,
int64_t absolute_capture_timestamp_ms,
- webrtc::ArrayView<const uint32_t> csrcs,
+ ArrayView<const uint32_t> csrcs,
std::optional<uint8_t> audio_level_dbov));
ChannelSendFrameTransformerDelegate::SendFrameCallback callback() {
@@ -104,7 +104,7 @@
AudioFrameType::kEmptyFrame, 0, 0, mock_data, sizeof(mock_data), 0,
/*ssrc=*/0, /*mimeType=*/"audio/opus", /*audio_level_dbov=*/123);
return absl::WrapUnique(
- static_cast<webrtc::TransformableAudioFrameInterface*>(frame.release()));
+ static_cast<TransformableAudioFrameInterface*>(frame.release()));
}
// Test that the delegate registers itself with the frame transformer on Init().
diff --git a/audio/channel_send_unittest.cc b/audio/channel_send_unittest.cc
index 03ce5bc..7636ee3 100644
--- a/audio/channel_send_unittest.cc
+++ b/audio/channel_send_unittest.cc
@@ -123,7 +123,7 @@
void ProcessNextFrame() { ProcessNextFrame(CreateAudioFrame()); }
GlobalSimulatedTimeController time_controller_;
- webrtc::test::ScopedKeyValueConfig field_trials_;
+ test::ScopedKeyValueConfig field_trials_;
Environment env_;
NiceMock<MockTransport> transport_;
CryptoOptions crypto_options_;
diff --git a/audio/test/audio_end_to_end_test.cc b/audio/test/audio_end_to_end_test.cc
index 746ae3f..d9c4594 100644
--- a/audio/test/audio_end_to_end_test.cc
+++ b/audio/test/audio_end_to_end_test.cc
@@ -63,8 +63,7 @@
AudioSendStream::Config* send_config,
std::vector<AudioReceiveStreamInterface::Config>* /* receive_configs */) {
// Large bitrate by default.
- const webrtc::SdpAudioFormat kDefaultFormat("opus", 48000, 2,
- {{"stereo", "1"}});
+ const SdpAudioFormat kDefaultFormat("opus", 48000, 2, {{"stereo", "1"}});
send_config->send_codec_spec = AudioSendStream::Config::SendCodecSpec(
test::VideoTestConstants::kAudioSendPayloadType, kDefaultFormat);
send_config->min_bitrate_bps = 32000;
diff --git a/audio/test/non_sender_rtt_test.cc b/audio/test/non_sender_rtt_test.cc
index 17dd97e..49b6cae 100644
--- a/audio/test/non_sender_rtt_test.cc
+++ b/audio/test/non_sender_rtt_test.cc
@@ -60,7 +60,7 @@
// cases it can take more than 10 seconds.
EXPECT_THAT(
WaitUntil([&] { return HasRoundTripTimeMeasurement(); }, IsTrue(),
- {.timeout = webrtc::TimeDelta::Millis(kLongTimeoutMs)}),
+ {.timeout = TimeDelta::Millis(kLongTimeoutMs)}),
IsRtcOk());
}
