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| author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-05-15 03:34:50 +0000 |
|---|---|---|
| committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-05-15 03:34:50 +0000 |
| commit | def92d1b8e9d373e2f6f27c366d578d97d8960c6 (patch) | |
| tree | 2ef34b9ad8bb9a9220e05d60352558b15f513894 /third_party/libwebrtc/moz-patch-stack/0033.patch | |
| parent | Adding debian version 125.0.3-1. (diff) | |
| download | firefox-def92d1b8e9d373e2f6f27c366d578d97d8960c6.tar.xz firefox-def92d1b8e9d373e2f6f27c366d578d97d8960c6.zip | |
Merging upstream version 126.0.
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/moz-patch-stack/0033.patch')
| -rw-r--r-- | third_party/libwebrtc/moz-patch-stack/0033.patch | 12 |
1 files changed, 6 insertions, 6 deletions
diff --git a/third_party/libwebrtc/moz-patch-stack/0033.patch b/third_party/libwebrtc/moz-patch-stack/0033.patch index 5c69ef0bce..62695eeb23 100644 --- a/third_party/libwebrtc/moz-patch-stack/0033.patch +++ b/third_party/libwebrtc/moz-patch-stack/0033.patch @@ -15,10 +15,10 @@ Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/d380a43d59f4f7cbc 4 files changed, 35 insertions(+) diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc -index c9dc42c04e..e7ebb2bf4e 100644 +index 3701eafd6f..8dc78b18fa 100644 --- a/audio/audio_send_stream.cc +++ b/audio/audio_send_stream.cc -@@ -431,6 +431,7 @@ webrtc::AudioSendStream::Stats AudioSendStream::GetStats( +@@ -432,6 +432,7 @@ webrtc::AudioSendStream::Stats AudioSendStream::GetStats( stats.target_bitrate_bps = channel_send_->GetTargetBitrate(); webrtc::CallSendStatistics call_stats = channel_send_->GetRTCPStatistics(); @@ -27,7 +27,7 @@ index c9dc42c04e..e7ebb2bf4e 100644 stats.header_and_padding_bytes_sent = call_stats.header_and_padding_bytes_sent; diff --git a/audio/channel_send.cc b/audio/channel_send.cc -index 310e0517cf..549e65a59c 100644 +index ab1b72a2f3..db632b3aa8 100644 --- a/audio/channel_send.cc +++ b/audio/channel_send.cc @@ -56,6 +56,31 @@ constexpr int64_t kMinRetransmissionWindowMs = 30; @@ -64,14 +64,14 @@ index 310e0517cf..549e65a59c 100644 // packets from the ACM @@ -210,6 +235,8 @@ class ChannelSend : public ChannelSendInterface, bool input_mute_ RTC_GUARDED_BY(volume_settings_mutex_) = false; - bool previous_frame_muted_ RTC_GUARDED_BY(encoder_queue_) = false; + bool previous_frame_muted_ RTC_GUARDED_BY(encoder_queue_checker_) = false; + const std::unique_ptr<RtcpCounterObserver> rtcp_counter_observer_; + PacketRouter* packet_router_ RTC_GUARDED_BY(&worker_thread_checker_) = nullptr; const std::unique_ptr<RtpPacketSenderProxy> rtp_packet_pacer_proxy_; -@@ -398,6 +425,7 @@ ChannelSend::ChannelSend( +@@ -397,6 +424,7 @@ ChannelSend::ChannelSend( const FieldTrialsView& field_trials) : ssrc_(ssrc), event_log_(rtc_event_log), @@ -88,7 +88,7 @@ index 310e0517cf..549e65a59c 100644 if (field_trials.IsDisabled("WebRTC-DisableRtxRateLimiter")) { configuration.retransmission_rate_limiter = retransmission_rate_limiter_.get(); -@@ -687,6 +717,7 @@ CallSendStatistics ChannelSend::GetRTCPStatistics() const { +@@ -691,6 +721,7 @@ CallSendStatistics ChannelSend::GetRTCPStatistics() const { RTC_DCHECK_RUN_ON(&worker_thread_checker_); CallSendStatistics stats = {0}; stats.rttMs = GetRTT(); |