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Search and replace gendered terms according to style guide:
https://chromium.googlesource.com/chromium/src/+/master/styleguide/inclusive_code.md#tools Not changin the transcipt in resources/audio_processing/conversational_speech/README.md BUG=webrtc:11680 Change-Id: I36af34e4a4e0ec6161093c0045b7bbe1dbe4eb45 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177016 Reviewed-by: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31514}
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9 changed files with 15 additions and 15 deletions
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@ -231,7 +231,7 @@ Yes, you still have the right to redistribute and you still have a patent
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license for Google's patents that cover the code that Google released.
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### What if my competitor uses the code and brings patent litigation against me for something unrelated to the code. Does he or she still have a patent license?
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### What if my competitor uses the code and brings patent litigation against me for something unrelated to the code. Do they still have a patent license?
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Yes, he/she still has the right to redistribute and he/she still has a patent
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Yes, they still have the right to redistribute and they still have a patent
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license for Google's patents that cover the code that Google released.
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@ -185,8 +185,8 @@ public class AppRTCAudioManager {
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// Note that, the sensor will not be active until start() has been called.
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proximitySensor = AppRTCProximitySensor.create(context,
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// This method will be called each time a state change is detected.
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// Example: user holds his hand over the device (closer than ~5 cm),
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// or removes his hand from the device.
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// Example: user holds their hand over the device (closer than ~5 cm),
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// or removes their hand from the device.
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this ::onProximitySensorChangedState);
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Log.d(TAG, "defaultAudioDevice: " + defaultAudioDevice);
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@ -43,7 +43,7 @@ namespace webrtc {
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//
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// Note that the returned std::string might have been reserved for significantly
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// more memory than it ends up using. If the caller to EncodeBlobs() intends
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// to store the result long-term, he should consider shrink_to_fit()-ing it.
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// to store the result long-term, they should consider shrink_to_fit()-ing it.
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std::string EncodeBlobs(const std::vector<std::string>& blobs);
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std::vector<absl::string_view> DecodeBlobs(absl::string_view encoded_blobs,
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size_t num_of_blobs);
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@ -230,7 +230,7 @@ TEST(RtcpReceiverTest, InjectSrPacketFromUnknownSender) {
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rtcp::SenderReport sr;
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sr.SetSenderSsrc(kUnknownSenderSsrc);
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// The parser will handle report blocks in Sender Report from other than his
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// The parser will handle report blocks in Sender Report from other than their
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// expected peer.
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EXPECT_CALL(mocks.rtp_rtcp_impl, OnReceivedRtcpReportBlocks);
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EXPECT_CALL(mocks.bandwidth_observer,
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@ -28,8 +28,8 @@ class MediaReceiverRtcpObserver {
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public:
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virtual ~MediaReceiverRtcpObserver() = default;
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// All message handlers have default empty implementation. This way user needs
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// to implement only those she is interested in.
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// All message handlers have default empty implementation. This way users only
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// need to implement the ones they are interested in.
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virtual void OnSenderReport(uint32_t sender_ssrc,
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NtpTime ntp_time,
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uint32_t rtp_time) {}
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@ -1480,7 +1480,7 @@ TEST_F(P2PTransportChannelTest, PeerReflexiveCandidateBeforeSignaling) {
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PauseCandidates(1);
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// Wait until the callee becomes writable to make sure that a ping request is
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// received by the caller before his remote ICE credentials are set.
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// received by the caller before their remote ICE credentials are set.
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ASSERT_TRUE_WAIT(ep2_ch1()->selected_connection() != nullptr, kMediumTimeout);
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// Add two sets of remote ICE credentials, so that the ones used by the
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// candidate will be generation 1 instead of 0.
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@ -1588,7 +1588,7 @@ TEST_F(P2PTransportChannelTest, PeerReflexiveCandidateBeforeSignalingWithNAT) {
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PauseCandidates(1);
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// Wait until the callee becomes writable to make sure that a ping request is
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// received by the caller before his remote ICE credentials are set.
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// received by the caller before their remote ICE credentials are set.
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ASSERT_TRUE_WAIT(ep2_ch1()->selected_connection() != nullptr, kMediumTimeout);
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// Add two sets of remote ICE credentials, so that the ones used by the
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// candidate will be generation 1 instead of 0.
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@ -69,7 +69,7 @@ RTC_NORETURN void rtc_FatalMessage(const char* file, int line, const char* msg);
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// the reason that it's better to terminate might simply be that the error
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// handling code isn't in place yet; in production, the reason might be that
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// the author of the code truly believes that x will always be true, but that
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// she recognizes that if she is wrong, abrupt and unpleasant process
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// they recognizes that if they are wrong, abrupt and unpleasant process
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// termination is still better than carrying on with the assumption violated.
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//
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// RTC_CHECK always evaluates its argument, so it's OK for x to have side
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@ -46,8 +46,8 @@ namespace webrtc_pc_e2e {
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// callback, where video analyzer will be called again and then decoded frame
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// will be passed to origin callback, provided by user.
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//
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// Quality decoder registers its own callback in origin decoder at the same
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// time, when user registers his callback in quality decoder.
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// Quality decoder registers its own callback in origin decoder, at the same
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// time the user registers their callback in quality decoder.
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class QualityAnalyzingVideoDecoder : public VideoDecoder {
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public:
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// Creates analyzing decoder. |id| is unique coding entity id, that will
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@ -50,8 +50,8 @@ constexpr int kAnalyzeAnySpatialStream = -1;
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// injected into EncodedImage with passed EncodedImageDataInjector. Then new
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// EncodedImage will be passed to origin callback, provided by user.
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//
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// Quality encoder registers its own callback in origin encoder at the same
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// time, when user registers his callback in quality encoder.
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// Quality encoder registers its own callback in origin encoder, at the same
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// time the user registers their callback in quality encoder.
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class QualityAnalyzingVideoEncoder : public VideoEncoder,
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public EncodedImageCallback {
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public:
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