Add passkey to TransformableFrameInterface to prevent external impls
This makes the downcasts currently used in eg
modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.cc
safer.
Bug: webrtc:339815768
Change-Id: Ie6c7ab84666d399dbca4c95846b850aac5327f1a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350361
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tony Herre <herre@google.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42325}
diff --git a/api/BUILD.gn b/api/BUILD.gn
index 464e1be..64028ad 100644
--- a/api/BUILD.gn
+++ b/api/BUILD.gn
@@ -393,7 +393,10 @@
rtc_source_set("frame_transformer_interface") {
visibility = [ "*" ]
- sources = [ "frame_transformer_interface.h" ]
+ sources = [
+ "frame_transformer_interface.cc",
+ "frame_transformer_interface.h",
+ ]
deps = [
":make_ref_counted",
":ref_count",
diff --git a/api/frame_transformer_interface.cc b/api/frame_transformer_interface.cc
new file mode 100644
index 0000000..88d4d19
--- /dev/null
+++ b/api/frame_transformer_interface.cc
@@ -0,0 +1,26 @@
+/*
+ * Copyright 2024 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/frame_transformer_interface.h"
+
+namespace webrtc {
+
+TransformableFrameInterface::TransformableFrameInterface(
+ TransformableFrameInterface::Passkey) {}
+
+TransformableVideoFrameInterface::TransformableVideoFrameInterface(
+ TransformableFrameInterface::Passkey passkey)
+ : TransformableFrameInterface(passkey) {}
+
+TransformableAudioFrameInterface::TransformableAudioFrameInterface(
+ TransformableFrameInterface::Passkey passkey)
+ : TransformableFrameInterface(passkey) {}
+
+} // namespace webrtc
diff --git a/api/frame_transformer_interface.h b/api/frame_transformer_interface.h
index 89356df..03f7b48 100644
--- a/api/frame_transformer_interface.h
+++ b/api/frame_transformer_interface.h
@@ -24,6 +24,18 @@
// Owns the frame payload data.
class TransformableFrameInterface {
public:
+ // Only a known list of internal implementations of transformable frames are
+ // permitted to allow internal downcasting. This is enforced via the
+ // internally-constructable Passkey.
+ // TODO: bugs.webrtc.org/339815768 - Remove this passkey once the
+ // downcasts are removed.
+ class Passkey;
+ RTC_EXPORT explicit TransformableFrameInterface(Passkey);
+
+ TransformableFrameInterface(TransformableFrameInterface&&) = default;
+ TransformableFrameInterface& operator=(TransformableFrameInterface&&) =
+ default;
+
virtual ~TransformableFrameInterface() = default;
// Returns the frame payload data. The data is valid until the next non-const
@@ -58,6 +70,7 @@
class TransformableVideoFrameInterface : public TransformableFrameInterface {
public:
+ RTC_EXPORT explicit TransformableVideoFrameInterface(Passkey passkey);
virtual ~TransformableVideoFrameInterface() = default;
virtual bool IsKeyFrame() const = 0;
@@ -69,6 +82,7 @@
// Extends the TransformableFrameInterface to expose audio-specific information.
class TransformableAudioFrameInterface : public TransformableFrameInterface {
public:
+ RTC_EXPORT explicit TransformableAudioFrameInterface(Passkey passkey);
virtual ~TransformableAudioFrameInterface() = default;
virtual rtc::ArrayView<const uint32_t> GetContributingSources() const = 0;
@@ -137,6 +151,27 @@
// virtual AddOutgoingMediaType(RtpCodec codec) = 0;
};
+//------------------------------------------------------------------------------
+// Implementation details follow
+//------------------------------------------------------------------------------
+class TransformableFrameInterface::Passkey {
+ public:
+ ~Passkey() = default;
+
+ private:
+ // Explicit list of allowed internal implmentations of
+ // TransformableFrameInterface.
+ friend class TransformableOutgoingAudioFrame;
+ friend class TransformableIncomingAudioFrame;
+ friend class TransformableVideoSenderFrame;
+ friend class TransformableVideoReceiverFrame;
+
+ friend class MockTransformableFrame;
+ friend class MockTransformableAudioFrame;
+ friend class MockTransformableVideoFrame;
+ Passkey() = default;
+};
+
} // namespace webrtc
#endif // API_FRAME_TRANSFORMER_INTERFACE_H_
diff --git a/api/test/mock_transformable_audio_frame.h b/api/test/mock_transformable_audio_frame.h
index f243e38..d1f4044 100644
--- a/api/test/mock_transformable_audio_frame.h
+++ b/api/test/mock_transformable_audio_frame.h
@@ -20,6 +20,8 @@
class MockTransformableAudioFrame : public TransformableAudioFrameInterface {
public:
+ MockTransformableAudioFrame() : TransformableAudioFrameInterface(Passkey()) {}
+
MOCK_METHOD(rtc::ArrayView<const uint8_t>, GetData, (), (const, override));
MOCK_METHOD(void, SetData, (rtc::ArrayView<const uint8_t>), (override));
MOCK_METHOD(void, SetRTPTimestamp, (uint32_t), (override));
diff --git a/api/test/mock_transformable_frame.h b/api/test/mock_transformable_frame.h
index df20b62..d6e0856 100644
--- a/api/test/mock_transformable_frame.h
+++ b/api/test/mock_transformable_frame.h
@@ -23,8 +23,10 @@
namespace webrtc {
-class MockTransformableFrame : public webrtc::TransformableFrameInterface {
+class MockTransformableFrame : public TransformableFrameInterface {
public:
+ MockTransformableFrame() : TransformableFrameInterface(Passkey()) {}
+
MOCK_METHOD(rtc::ArrayView<const uint8_t>, GetData, (), (const, override));
MOCK_METHOD(void, SetData, (rtc::ArrayView<const uint8_t>), (override));
MOCK_METHOD(uint8_t, GetPayloadType, (), (const, override));
diff --git a/api/test/mock_transformable_video_frame.h b/api/test/mock_transformable_video_frame.h
index b3825dd..2cf7cb2 100644
--- a/api/test/mock_transformable_video_frame.h
+++ b/api/test/mock_transformable_video_frame.h
@@ -22,6 +22,7 @@
class MockTransformableVideoFrame
: public webrtc::TransformableVideoFrameInterface {
public:
+ MockTransformableVideoFrame() : TransformableVideoFrameInterface(Passkey()) {}
MOCK_METHOD(rtc::ArrayView<const uint8_t>, GetData, (), (const, override));
MOCK_METHOD(void, SetData, (rtc::ArrayView<const uint8_t> data), (override));
MOCK_METHOD(uint32_t, GetTimestamp, (), (const, override));
diff --git a/audio/channel_receive_frame_transformer_delegate.cc b/audio/channel_receive_frame_transformer_delegate.cc
index 953e27a..814127b 100644
--- a/audio/channel_receive_frame_transformer_delegate.cc
+++ b/audio/channel_receive_frame_transformer_delegate.cc
@@ -16,7 +16,6 @@
#include "rtc_base/buffer.h"
namespace webrtc {
-namespace {
class TransformableIncomingAudioFrame
: public TransformableAudioFrameInterface {
@@ -25,7 +24,8 @@
const RTPHeader& header,
uint32_t ssrc,
const std::string& codec_mime_type)
- : payload_(payload.data(), payload.size()),
+ : TransformableAudioFrameInterface(Passkey()),
+ payload_(payload.data(), payload.size()),
header_(header),
ssrc_(ssrc),
codec_mime_type_(codec_mime_type) {}
@@ -83,7 +83,6 @@
uint32_t ssrc_;
std::string codec_mime_type_;
};
-} // namespace
ChannelReceiveFrameTransformerDelegate::ChannelReceiveFrameTransformerDelegate(
ReceiveFrameCallback receive_frame_callback,
diff --git a/audio/channel_send_frame_transformer_delegate.cc b/audio/channel_send_frame_transformer_delegate.cc
index 8bf1963..ef6ec26 100644
--- a/audio/channel_send_frame_transformer_delegate.cc
+++ b/audio/channel_send_frame_transformer_delegate.cc
@@ -45,6 +45,7 @@
RTC_DCHECK_NOTREACHED();
return AudioFrameType::kEmptyFrame;
}
+} // namespace
class TransformableOutgoingAudioFrame
: public TransformableAudioFrameInterface {
@@ -61,7 +62,8 @@
const std::string& codec_mime_type,
absl::optional<uint16_t> sequence_number,
absl::optional<uint8_t> audio_level_dbov)
- : frame_type_(frame_type),
+ : TransformableAudioFrameInterface(Passkey()),
+ frame_type_(frame_type),
payload_type_(payload_type),
rtp_timestamp_with_offset_(rtp_timestamp_with_offset),
payload_(payload_data, payload_size),
@@ -119,7 +121,6 @@
absl::optional<uint16_t> sequence_number_;
absl::optional<uint8_t> audio_level_dbov_;
};
-} // namespace
ChannelSendFrameTransformerDelegate::ChannelSendFrameTransformerDelegate(
SendFrameCallback send_frame_callback,
diff --git a/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.cc b/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.cc
index 2bb7194..ae80f76 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.cc
@@ -28,6 +28,7 @@
// estimate of the RTT of the link,so 10ms should be a reasonable estimate for
// frames being re-transmitted to a peer, probably on the same network.
const TimeDelta kDefaultRetransmissionsTime = TimeDelta::Millis(10);
+} // namespace
class TransformableVideoSenderFrame : public TransformableVideoFrameInterface {
public:
@@ -39,7 +40,8 @@
TimeDelta expected_retransmission_time,
uint32_t ssrc,
std::vector<uint32_t> csrcs)
- : encoded_data_(encoded_image.GetEncodedData()),
+ : TransformableVideoFrameInterface(Passkey()),
+ encoded_data_(encoded_image.GetEncodedData()),
pre_transform_payload_size_(encoded_image.size()),
header_(video_header),
frame_type_(encoded_image._frameType),
@@ -128,7 +130,6 @@
uint32_t ssrc_;
std::vector<uint32_t> csrcs_;
};
-} // namespace
RTPSenderVideoFrameTransformerDelegate::RTPSenderVideoFrameTransformerDelegate(
RTPVideoFrameSenderInterface* sender,
diff --git a/modules/rtp_rtcp/source/rtp_video_stream_receiver_frame_transformer_delegate.cc b/modules/rtp_rtcp/source/rtp_video_stream_receiver_frame_transformer_delegate.cc
index fbd10c4..5ef0a80 100644
--- a/modules/rtp_rtcp/source/rtp_video_stream_receiver_frame_transformer_delegate.cc
+++ b/modules/rtp_rtcp/source/rtp_video_stream_receiver_frame_transformer_delegate.cc
@@ -16,20 +16,21 @@
#include "absl/memory/memory.h"
#include "modules/rtp_rtcp/source/rtp_descriptor_authentication.h"
+#include "modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/thread.h"
namespace webrtc {
-namespace {
class TransformableVideoReceiverFrame
: public TransformableVideoFrameInterface {
public:
TransformableVideoReceiverFrame(std::unique_ptr<RtpFrameObject> frame,
uint32_t ssrc,
RtpVideoFrameReceiver* receiver)
- : frame_(std::move(frame)),
+ : TransformableVideoFrameInterface(Passkey()),
+ frame_(std::move(frame)),
metadata_(frame_->GetRtpVideoHeader().GetAsMetadata()),
receiver_(receiver) {
metadata_.SetSsrc(ssrc);
@@ -89,7 +90,6 @@
VideoFrameMetadata metadata_;
RtpVideoFrameReceiver* receiver_;
};
-} // namespace
RtpVideoStreamReceiverFrameTransformerDelegate::
RtpVideoStreamReceiverFrameTransformerDelegate(
