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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 01:13:27 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 01:13:27 +0000 |
commit | 40a355a42d4a9444dc753c04c6608dade2f06a23 (patch) | |
tree | 871fc667d2de662f171103ce5ec067014ef85e61 /third_party/libwebrtc/audio/channel_send_frame_transformer_delegate.cc | |
parent | Adding upstream version 124.0.1. (diff) | |
download | firefox-40a355a42d4a9444dc753c04c6608dade2f06a23.tar.xz firefox-40a355a42d4a9444dc753c04c6608dade2f06a23.zip |
Adding upstream version 125.0.1.upstream/125.0.1
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 | 44 |
1 files changed, 37 insertions, 7 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 0f85216e92..2eea0d2387 100644 --- a/third_party/libwebrtc/audio/channel_send_frame_transformer_delegate.cc +++ b/third_party/libwebrtc/audio/channel_send_frame_transformer_delegate.cc @@ -11,6 +11,7 @@ #include "audio/channel_send_frame_transformer_delegate.h" #include <utility> +#include <vector> namespace webrtc { namespace { @@ -55,13 +56,17 @@ class TransformableOutgoingAudioFrame const uint8_t* payload_data, size_t payload_size, absl::optional<uint64_t> absolute_capture_timestamp_ms, - uint32_t ssrc) + uint32_t ssrc, + std::vector<uint32_t> csrcs, + const std::string& codec_mime_type) : frame_type_(frame_type), payload_type_(payload_type), rtp_timestamp_with_offset_(rtp_timestamp_with_offset), payload_(payload_data, payload_size), absolute_capture_timestamp_ms_(absolute_capture_timestamp_ms), - ssrc_(ssrc) {} + ssrc_(ssrc), + csrcs_(std::move(csrcs)), + codec_mime_type_(codec_mime_type) {} ~TransformableOutgoingAudioFrame() override = default; rtc::ArrayView<const uint8_t> GetData() const override { return payload_; } void SetData(rtc::ArrayView<const uint8_t> data) override { @@ -76,9 +81,10 @@ class TransformableOutgoingAudioFrame uint8_t GetPayloadType() const override { return payload_type_; } Direction GetDirection() const override { return Direction::kSender; } + std::string GetMimeType() const override { return codec_mime_type_; } rtc::ArrayView<const uint32_t> GetContributingSources() const override { - return {}; + return csrcs_; } const absl::optional<uint16_t> SequenceNumber() const override { @@ -100,6 +106,8 @@ class TransformableOutgoingAudioFrame rtc::Buffer payload_; absl::optional<uint64_t> absolute_capture_timestamp_ms_; uint32_t ssrc_; + std::vector<uint32_t> csrcs_; + std::string codec_mime_type_; }; } // namespace @@ -131,11 +139,23 @@ void ChannelSendFrameTransformerDelegate::Transform( const uint8_t* payload_data, size_t payload_size, int64_t absolute_capture_timestamp_ms, - uint32_t ssrc) { + uint32_t ssrc, + const std::string& codec_mimetype) { + { + MutexLock lock(&send_lock_); + if (short_circuit_) { + send_frame_callback_( + frame_type, payload_type, rtp_timestamp, + rtc::ArrayView<const uint8_t>(payload_data, payload_size), + absolute_capture_timestamp_ms, /*csrcs=*/{}); + return; + } + } frame_transformer_->Transform( std::make_unique<TransformableOutgoingAudioFrame>( frame_type, payload_type, rtp_timestamp, payload_data, payload_size, - absolute_capture_timestamp_ms, ssrc)); + absolute_capture_timestamp_ms, ssrc, + /*csrcs=*/std::vector<uint32_t>(), codec_mimetype)); } void ChannelSendFrameTransformerDelegate::OnTransformedFrame( @@ -150,6 +170,11 @@ void ChannelSendFrameTransformerDelegate::OnTransformedFrame( }); } +void ChannelSendFrameTransformerDelegate::StartShortCircuiting() { + MutexLock lock(&send_lock_); + short_circuit_ = true; +} + void ChannelSendFrameTransformerDelegate::SendFrame( std::unique_ptr<TransformableFrameInterface> frame) const { MutexLock lock(&send_lock_); @@ -164,16 +189,21 @@ void ChannelSendFrameTransformerDelegate::SendFrame( transformed_frame->GetData(), transformed_frame->AbsoluteCaptureTimestamp() ? *transformed_frame->AbsoluteCaptureTimestamp() - : 0); + : 0, + transformed_frame->GetContributingSources()); } std::unique_ptr<TransformableAudioFrameInterface> CloneSenderAudioFrame( TransformableAudioFrameInterface* original) { + std::vector<uint32_t> csrcs; + csrcs.assign(original->GetContributingSources().begin(), + original->GetContributingSources().end()); return std::make_unique<TransformableOutgoingAudioFrame>( InterfaceFrameTypeToInternalFrameType(original->Type()), original->GetPayloadType(), original->GetTimestamp(), original->GetData().data(), original->GetData().size(), - original->AbsoluteCaptureTimestamp(), original->GetSsrc()); + original->AbsoluteCaptureTimestamp(), original->GetSsrc(), + std::move(csrcs), original->GetMimeType()); } } // namespace webrtc |