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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-06-12 05:43:14 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-06-12 05:43:14 +0000 |
commit | 8dd16259287f58f9273002717ec4d27e97127719 (patch) | |
tree | 3863e62a53829a84037444beab3abd4ed9dfc7d0 /third_party/libwebrtc/audio/channel_send_frame_transformer_delegate.cc | |
parent | Releasing progress-linux version 126.0.1-1~progress7.99u1. (diff) | |
download | firefox-8dd16259287f58f9273002717ec4d27e97127719.tar.xz firefox-8dd16259287f58f9273002717ec4d27e97127719.zip |
Merging upstream version 127.0.
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/audio/channel_send_frame_transformer_delegate.cc')
-rw-r--r-- | third_party/libwebrtc/audio/channel_send_frame_transformer_delegate.cc | 14 |
1 files changed, 9 insertions, 5 deletions
diff --git a/third_party/libwebrtc/audio/channel_send_frame_transformer_delegate.cc b/third_party/libwebrtc/audio/channel_send_frame_transformer_delegate.cc index ac32410aed..6d3c011862 100644 --- a/third_party/libwebrtc/audio/channel_send_frame_transformer_delegate.cc +++ b/third_party/libwebrtc/audio/channel_send_frame_transformer_delegate.cc @@ -58,7 +58,8 @@ class TransformableOutgoingAudioFrame absl::optional<uint64_t> absolute_capture_timestamp_ms, uint32_t ssrc, std::vector<uint32_t> csrcs, - const std::string& codec_mime_type) + const std::string& codec_mime_type, + absl::optional<uint16_t> sequence_number) : frame_type_(frame_type), payload_type_(payload_type), rtp_timestamp_with_offset_(rtp_timestamp_with_offset), @@ -66,7 +67,8 @@ class TransformableOutgoingAudioFrame absolute_capture_timestamp_ms_(absolute_capture_timestamp_ms), ssrc_(ssrc), csrcs_(std::move(csrcs)), - codec_mime_type_(codec_mime_type) {} + codec_mime_type_(codec_mime_type), + sequence_number_(sequence_number) {} ~TransformableOutgoingAudioFrame() override = default; rtc::ArrayView<const uint8_t> GetData() const override { return payload_; } void SetData(rtc::ArrayView<const uint8_t> data) override { @@ -88,7 +90,7 @@ class TransformableOutgoingAudioFrame } const absl::optional<uint16_t> SequenceNumber() const override { - return absl::nullopt; + return sequence_number_; } void SetRTPTimestamp(uint32_t rtp_timestamp_with_offset) override { @@ -108,6 +110,7 @@ class TransformableOutgoingAudioFrame uint32_t ssrc_; std::vector<uint32_t> csrcs_; std::string codec_mime_type_; + absl::optional<uint16_t> sequence_number_; }; } // namespace @@ -155,7 +158,8 @@ void ChannelSendFrameTransformerDelegate::Transform( std::make_unique<TransformableOutgoingAudioFrame>( frame_type, payload_type, rtp_timestamp, payload_data, payload_size, absolute_capture_timestamp_ms, ssrc, - /*csrcs=*/std::vector<uint32_t>(), codec_mimetype)); + /*csrcs=*/std::vector<uint32_t>(), codec_mimetype, + /*sequence_number=*/absl::nullopt)); } void ChannelSendFrameTransformerDelegate::OnTransformedFrame( @@ -203,7 +207,7 @@ std::unique_ptr<TransformableAudioFrameInterface> CloneSenderAudioFrame( original->GetPayloadType(), original->GetTimestamp(), original->GetData().data(), original->GetData().size(), original->AbsoluteCaptureTimestamp(), original->GetSsrc(), - std::move(csrcs), original->GetMimeType()); + std::move(csrcs), original->GetMimeType(), original->SequenceNumber()); } } // namespace webrtc |