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Pass the correct abs_capture_timestamp while cloning audio frame

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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}
This commit is contained in:
Palak Agarwal 2023-09-20 10:01:18 +02:00 committed by WebRTC LUCI CQ
parent 3218d743be
commit 0505115b5c
3 changed files with 15 additions and 15 deletions
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6
api/frame_transformer_interface.h
View File

@ -78,11 +78,7 @@ class TransformableAudioFrameInterface : public TransformableFrameInterface {
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 };

20
audio/channel_send_frame_transformer_delegate.cc
View File

@ -48,13 +48,14 @@ AudioFrameType InterfaceFrameTypeToInternalFrameType(
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 @@ class TransformableOutgoingAudioFrame
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 @@ void ChannelSendFrameTransformerDelegate::SendFrame(
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

4
test/mock_transformable_frame.h
View File

@ -23,6 +23,10 @@ class MockTransformableAudioFrame : public TransformableAudioFrameInterface {
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));