Stop using TransformableAudioFrameInterface::GetHeader() within webrtc Instead switch to specific getters, or methods only defined on specific implementations rather than part of the public API. Once uses are removed from Chromium, I'll mark GetHeader() deprecated and eventually remove it. Bug: chromium:1456628 Change-Id: I19b80489b3a0322c201e24994494cfbb742ee13e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/309780 Commit-Queue: Tony Herre <herre@google.com> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40344}
diff --git a/api/frame_transformer_interface.h b/api/frame_transformer_interface.h index c47bad4..47058d5 100644 --- a/api/frame_transformer_interface.h +++ b/api/frame_transformer_interface.h
@@ -83,6 +83,18 @@ virtual const absl::optional<uint16_t> SequenceNumber() const { return absl::nullopt; } + + // TODO(crbug.com/1456628): Change this to pure virtual after it + // is implemented everywhere. + virtual absl::optional<uint64_t> AbsoluteCaptureTimestamp() const { + return absl::nullopt; + } + + enum class FrameType { kEmptyFrame, kAudioFrameSpeech, kAudioFrameCN }; + + // TODO(crbug.com/1456628): Change this to pure virtual after it + // is implemented everywhere. + virtual FrameType Type() const { return FrameType::kEmptyFrame; } }; // Objects implement this interface to be notified with the transformed frame.
diff --git a/api/test/mock_transformable_audio_frame.h b/api/test/mock_transformable_audio_frame.h index 9b52e5c..680d7a0 100644 --- a/api/test/mock_transformable_audio_frame.h +++ b/api/test/mock_transformable_audio_frame.h
@@ -37,6 +37,14 @@ GetDirection, (), (const, override)); + MOCK_METHOD(absl::optional<uint64_t>, + AbsoluteCaptureTimestamp, + (), + (const, override)); + MOCK_METHOD(TransformableAudioFrameInterface::FrameType, + Type, + (), + (const, override)); }; } // namespace webrtc
diff --git a/audio/channel_receive_frame_transformer_delegate.cc b/audio/channel_receive_frame_transformer_delegate.cc index 8f817c6..3cc9a01 100644 --- a/audio/channel_receive_frame_transformer_delegate.cc +++ b/audio/channel_receive_frame_transformer_delegate.cc
@@ -50,6 +50,18 @@ return header_.sequenceNumber; } + absl::optional<uint64_t> AbsoluteCaptureTimestamp() const override { + // This could be extracted from received header extensions + extrapolation, + // if required in future, eg for being able to re-send received frames. + return absl::nullopt; + } + const RTPHeader& Header() const { return header_; } + + FrameType Type() const override { + return header_.extension.voiceActivity ? FrameType::kAudioFrameSpeech + : FrameType::kAudioFrameCN; + } + private: rtc::Buffer payload_; RTPHeader header_; @@ -106,6 +118,6 @@ auto* transformed_frame = static_cast<TransformableIncomingAudioFrame*>(frame.get()); receive_frame_callback_(transformed_frame->GetData(), - transformed_frame->GetHeader()); + transformed_frame->Header()); } } // namespace webrtc
diff --git a/audio/channel_send_frame_transformer_delegate.cc b/audio/channel_send_frame_transformer_delegate.cc index 93aaa43..8c97256 100644 --- a/audio/channel_send_frame_transformer_delegate.cc +++ b/audio/channel_send_frame_transformer_delegate.cc
@@ -15,6 +15,36 @@ namespace webrtc { namespace { +using IfaceFrameType = TransformableAudioFrameInterface::FrameType; + +IfaceFrameType InternalFrameTypeToInterfaceFrameType( + const AudioFrameType frame_type) { + switch (frame_type) { + case AudioFrameType::kEmptyFrame: + return IfaceFrameType::kEmptyFrame; + case AudioFrameType::kAudioFrameSpeech: + return IfaceFrameType::kAudioFrameSpeech; + case AudioFrameType::kAudioFrameCN: + return IfaceFrameType::kAudioFrameCN; + } + RTC_DCHECK_NOTREACHED(); + return IfaceFrameType::kEmptyFrame; +} + +AudioFrameType InterfaceFrameTypeToInternalFrameType( + const IfaceFrameType frame_type) { + switch (frame_type) { + case IfaceFrameType::kEmptyFrame: + return AudioFrameType::kEmptyFrame; + case IfaceFrameType::kAudioFrameSpeech: + return AudioFrameType::kAudioFrameSpeech; + case IfaceFrameType::kAudioFrameCN: + return AudioFrameType::kAudioFrameCN; + } + RTC_DCHECK_NOTREACHED(); + return AudioFrameType::kEmptyFrame; +} + class TransformableOutgoingAudioFrame : public TransformableAudioFrameInterface { public: @@ -44,11 +74,11 @@ uint32_t GetStartTimestamp() const { return rtp_start_timestamp_; } uint32_t GetSsrc() const override { return ssrc_; } - AudioFrameType GetFrameType() const { return frame_type_; } - uint8_t GetPayloadType() const override { return payload_type_; } - int64_t GetAbsoluteCaptureTimestampMs() const { - return absolute_capture_timestamp_ms_; + IfaceFrameType Type() const override { + return InternalFrameTypeToInterfaceFrameType(frame_type_); } + + uint8_t GetPayloadType() const override { return payload_type_; } Direction GetDirection() const override { return Direction::kSender; } // TODO(crbug.com/1453226): Remove once GetHeader() is removed from @@ -67,6 +97,10 @@ rtp_timestamp_ = timestamp - rtp_start_timestamp_; } + absl::optional<uint64_t> AbsoluteCaptureTimestamp() const override { + return absolute_capture_timestamp_ms_; + } + private: AudioFrameType frame_type_; uint8_t payload_type_; @@ -140,24 +174,21 @@ return; auto* transformed_frame = static_cast<TransformableOutgoingAudioFrame*>(frame.get()); - send_frame_callback_(transformed_frame->GetFrameType(), - transformed_frame->GetPayloadType(), - transformed_frame->GetTimestamp() - - transformed_frame->GetStartTimestamp(), - transformed_frame->GetData(), - transformed_frame->GetAbsoluteCaptureTimestampMs()); + send_frame_callback_( + InterfaceFrameTypeToInternalFrameType(transformed_frame->Type()), + transformed_frame->GetPayloadType(), + transformed_frame->GetTimestamp() - + transformed_frame->GetStartTimestamp(), + transformed_frame->GetData(), + *transformed_frame->AbsoluteCaptureTimestamp()); } std::unique_ptr<TransformableAudioFrameInterface> CloneSenderAudioFrame( TransformableAudioFrameInterface* original) { - AudioFrameType audio_frame_type = - original->GetHeader().extension.voiceActivity - ? AudioFrameType::kAudioFrameSpeech - : AudioFrameType::kAudioFrameCN; - // TODO(crbug.com/webrtc/14949): Ensure the correct timestamps are passed. return std::make_unique<TransformableOutgoingAudioFrame>( - audio_frame_type, original->GetPayloadType(), original->GetTimestamp(), + InterfaceFrameTypeToInternalFrameType(original->Type()), + original->GetPayloadType(), original->GetTimestamp(), /*rtp_start_timestamp=*/0u, original->GetData().data(), original->GetData().size(), original->GetTimestamp(), original->GetSsrc());