Pass the correct abs_capture_timestamp while cloning audio frame
This change replaces type of absolute_capture_timestamp_ms_ in
TransformableOutgoingAudioFrame from int to optional uint and makes
the function AbsoluteCaptureTimestamp() inside
TransformableAudioFrameInterface pure virtual.
Bug: webrtc:14949
Change-Id: Id3bdbcba63a5f91105ab198208e4f2b11eb3c7db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319000
Commit-Queue: Palak Agarwal <agpalak@google.com>
Reviewed-by: Tony Herre <herre@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40814}
diff --git a/api/frame_transformer_interface.h b/api/frame_transformer_interface.h
index bd9ea2d..40d2f36 100644
--- a/api/frame_transformer_interface.h
+++ b/api/frame_transformer_interface.h
@@ -78,11 +78,7 @@
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;
- }
+ virtual absl::optional<uint64_t> AbsoluteCaptureTimestamp() const = 0;
enum class FrameType { kEmptyFrame, kAudioFrameSpeech, kAudioFrameCN };
diff --git a/audio/channel_send_frame_transformer_delegate.cc b/audio/channel_send_frame_transformer_delegate.cc
index 9b88d4d..0f85216 100644
--- a/audio/channel_send_frame_transformer_delegate.cc
+++ b/audio/channel_send_frame_transformer_delegate.cc
@@ -48,13 +48,14 @@
class TransformableOutgoingAudioFrame
: public TransformableAudioFrameInterface {
public:
- TransformableOutgoingAudioFrame(AudioFrameType frame_type,
- uint8_t payload_type,
- uint32_t rtp_timestamp_with_offset,
- const uint8_t* payload_data,
- size_t payload_size,
- int64_t absolute_capture_timestamp_ms,
- uint32_t ssrc)
+ TransformableOutgoingAudioFrame(
+ AudioFrameType frame_type,
+ uint8_t payload_type,
+ uint32_t rtp_timestamp_with_offset,
+ const uint8_t* payload_data,
+ size_t payload_size,
+ absl::optional<uint64_t> absolute_capture_timestamp_ms,
+ uint32_t ssrc)
: frame_type_(frame_type),
payload_type_(payload_type),
rtp_timestamp_with_offset_(rtp_timestamp_with_offset),
@@ -97,7 +98,7 @@
uint8_t payload_type_;
uint32_t rtp_timestamp_with_offset_;
rtc::Buffer payload_;
- int64_t absolute_capture_timestamp_ms_;
+ absl::optional<uint64_t> absolute_capture_timestamp_ms_;
uint32_t ssrc_;
};
} // namespace
@@ -168,12 +169,11 @@
std::unique_ptr<TransformableAudioFrameInterface> CloneSenderAudioFrame(
TransformableAudioFrameInterface* original) {
- // TODO(crbug.com/webrtc/14949): Ensure the correct timestamps are passed.
return std::make_unique<TransformableOutgoingAudioFrame>(
InterfaceFrameTypeToInternalFrameType(original->Type()),
original->GetPayloadType(), original->GetTimestamp(),
original->GetData().data(), original->GetData().size(),
- original->GetTimestamp(), original->GetSsrc());
+ original->AbsoluteCaptureTimestamp(), original->GetSsrc());
}
} // namespace webrtc
diff --git a/test/mock_transformable_frame.h b/test/mock_transformable_frame.h
index 35b8d92..17d7e83 100644
--- a/test/mock_transformable_frame.h
+++ b/test/mock_transformable_frame.h
@@ -23,6 +23,10 @@
GetContributingSources,
(),
(const, override));
+ MOCK_METHOD(absl::optional<uint64_t>,
+ AbsoluteCaptureTimestamp,
+ (),
+ (const, override));
MOCK_METHOD(void, SetData, (rtc::ArrayView<const uint8_t>), (override));
MOCK_METHOD(uint8_t, GetPayloadType, (), (const, override));
MOCK_METHOD(uint32_t, GetSsrc, (), (const, override));