equal
deleted
inserted
replaced
1 # HG changeset patch |
1 # HG changeset patch |
2 # Parent e7eb7e9e99204275532b04de030879c9548b88a3 |
2 # Parent f5fd2bbd77ef4b6554a7180c9c4768e64aca3b2a |
3 |
3 |
4 diff --git a/gfx/skia/skia/include/codec/SkEncodedOrigin.h b/gfx/skia/skia/include/codec/SkEncodedOrigin.h |
4 diff --git a/gfx/skia/skia/include/codec/SkEncodedOrigin.h b/gfx/skia/skia/include/codec/SkEncodedOrigin.h |
5 --- a/gfx/skia/skia/include/codec/SkEncodedOrigin.h |
5 --- a/gfx/skia/skia/include/codec/SkEncodedOrigin.h |
6 +++ b/gfx/skia/skia/include/codec/SkEncodedOrigin.h |
6 +++ b/gfx/skia/skia/include/codec/SkEncodedOrigin.h |
7 @@ -36,16 +36,17 @@ static inline SkMatrix SkEncodedOriginTo |
7 @@ -36,16 +36,17 @@ static inline SkMatrix SkEncodedOriginTo |
524 return kVideoCodecVP9; |
524 return kVideoCodecVP9; |
525 if (absl::EqualsIgnoreCase(name, kPayloadNameAv1) || |
525 if (absl::EqualsIgnoreCase(name, kPayloadNameAv1) || |
526 diff --git a/third_party/libwebrtc/api/video_codecs/video_encoder_software_fallback_wrapper.cc b/third_party/libwebrtc/api/video_codecs/video_encoder_software_fallback_wrapper.cc |
526 diff --git a/third_party/libwebrtc/api/video_codecs/video_encoder_software_fallback_wrapper.cc b/third_party/libwebrtc/api/video_codecs/video_encoder_software_fallback_wrapper.cc |
527 --- a/third_party/libwebrtc/api/video_codecs/video_encoder_software_fallback_wrapper.cc |
527 --- a/third_party/libwebrtc/api/video_codecs/video_encoder_software_fallback_wrapper.cc |
528 +++ b/third_party/libwebrtc/api/video_codecs/video_encoder_software_fallback_wrapper.cc |
528 +++ b/third_party/libwebrtc/api/video_codecs/video_encoder_software_fallback_wrapper.cc |
529 @@ -158,16 +158,17 @@ class VideoEncoderSoftwareFallbackWrappe |
529 @@ -183,16 +183,17 @@ class VideoEncoderSoftwareFallbackWrappe |
530 [[fallthrough]]; |
530 [[fallthrough]]; |
531 case EncoderState::kMainEncoderUsed: |
531 case EncoderState::kMainEncoderUsed: |
532 return encoder_.get(); |
532 return encoder_.get(); |
533 case EncoderState::kFallbackDueToFailure: |
533 case EncoderState::kFallbackDueToFailure: |
534 case EncoderState::kForcedFallback: |
534 case EncoderState::kForcedFallback: |
542 // etc. |
542 // etc. |
543 void PrimeEncoder(VideoEncoder* encoder) const; |
543 void PrimeEncoder(VideoEncoder* encoder) const; |
544 |
544 |
545 // Settings used in the last InitEncode call and used if a dynamic fallback to |
545 // Settings used in the last InitEncode call and used if a dynamic fallback to |
546 // software is required. |
546 // software is required. |
547 @@ -338,16 +339,17 @@ int32_t VideoEncoderSoftwareFallbackWrap |
547 @@ -363,16 +364,17 @@ int32_t VideoEncoderSoftwareFallbackWrap |
548 case EncoderState::kMainEncoderUsed: { |
548 case EncoderState::kMainEncoderUsed: { |
549 return EncodeWithMainEncoder(frame, frame_types); |
549 return EncodeWithMainEncoder(frame, frame_types); |
550 } |
550 } |
551 case EncoderState::kFallbackDueToFailure: |
551 case EncoderState::kFallbackDueToFailure: |
552 case EncoderState::kForcedFallback: |
552 case EncoderState::kForcedFallback: |
682 |
682 |
683 std::string VideoSendStream::StreamStats::ToString() const { |
683 std::string VideoSendStream::StreamStats::ToString() const { |
684 diff --git a/third_party/libwebrtc/media/base/codec.cc b/third_party/libwebrtc/media/base/codec.cc |
684 diff --git a/third_party/libwebrtc/media/base/codec.cc b/third_party/libwebrtc/media/base/codec.cc |
685 --- a/third_party/libwebrtc/media/base/codec.cc |
685 --- a/third_party/libwebrtc/media/base/codec.cc |
686 +++ b/third_party/libwebrtc/media/base/codec.cc |
686 +++ b/third_party/libwebrtc/media/base/codec.cc |
687 @@ -195,16 +195,17 @@ bool Codec::Matches(const Codec& codec, |
687 @@ -201,16 +201,17 @@ bool Codec::Matches(const Codec& codec, |
688 (codec.bitrate == 0 || bitrate <= 0 || |
688 (codec.bitrate == 0 || bitrate <= 0 || |
689 bitrate == codec.bitrate) && |
689 bitrate == codec.bitrate) && |
690 ((codec.channels < 2 && channels < 2) || |
690 ((codec.channels < 2 && channels < 2) || |
691 channels == codec.channels); |
691 channels == codec.channels); |
692 |
692 |
697 }; |
697 }; |
698 |
698 |
699 return matches_id && matches_type_specific(); |
699 return matches_id && matches_type_specific(); |
700 } |
700 } |
701 |
701 |
702 bool Codec::MatchesCapability( |
702 bool Codec::MatchesRtpCodec(const webrtc::RtpCodec& codec_capability) const { |
703 const webrtc::RtpCodecCapability& codec_capability) const { |
|
704 webrtc::RtpCodecParameters codec_parameters = ToCodecParameters(); |
703 webrtc::RtpCodecParameters codec_parameters = ToCodecParameters(); |
|
704 |
705 diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/clipping_predictor.cc b/third_party/libwebrtc/modules/audio_processing/agc2/clipping_predictor.cc |
705 diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/clipping_predictor.cc b/third_party/libwebrtc/modules/audio_processing/agc2/clipping_predictor.cc |
706 --- a/third_party/libwebrtc/modules/audio_processing/agc2/clipping_predictor.cc |
706 --- a/third_party/libwebrtc/modules/audio_processing/agc2/clipping_predictor.cc |
707 +++ b/third_party/libwebrtc/modules/audio_processing/agc2/clipping_predictor.cc |
707 +++ b/third_party/libwebrtc/modules/audio_processing/agc2/clipping_predictor.cc |
708 @@ -373,12 +373,12 @@ std::unique_ptr<ClippingPredictor> Creat |
708 @@ -373,12 +373,12 @@ std::unique_ptr<ClippingPredictor> Creat |
709 config.reference_window_delay, config.clipping_threshold, |
709 config.reference_window_delay, config.clipping_threshold, |
955 extensions_map.IsRegistered(kRtpExtensionTransmissionTimeOffset); |
955 extensions_map.IsRegistered(kRtpExtensionTransmissionTimeOffset); |
956 } |
956 } |
957 diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc |
957 diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc |
958 --- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc |
958 --- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc |
959 +++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc |
959 +++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc |
960 @@ -40,16 +40,17 @@ namespace { |
960 @@ -41,16 +41,17 @@ namespace { |
961 case AudioFrameType::kEmptyFrame: |
961 case AudioFrameType::kEmptyFrame: |
962 return "empty"; |
962 return "empty"; |
963 case AudioFrameType::kAudioFrameSpeech: |
963 case AudioFrameType::kAudioFrameSpeech: |
964 return "audio_speech"; |
964 return "audio_speech"; |
965 case AudioFrameType::kAudioFrameCN: |
965 case AudioFrameType::kAudioFrameCN: |