diff options
Diffstat (limited to 'third_party/libwebrtc/moz-patch-stack/0033.patch')
| -rw-r--r-- | third_party/libwebrtc/moz-patch-stack/0033.patch | 16 |
1 files changed, 8 insertions, 8 deletions
diff --git a/third_party/libwebrtc/moz-patch-stack/0033.patch b/third_party/libwebrtc/moz-patch-stack/0033.patch index 2742e376b0..5c69ef0bce 100644 --- a/third_party/libwebrtc/moz-patch-stack/0033.patch +++ b/third_party/libwebrtc/moz-patch-stack/0033.patch @@ -15,7 +15,7 @@ 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 0caf59a20e..bffb910832 100644 +index c9dc42c04e..e7ebb2bf4e 100644 --- a/audio/audio_send_stream.cc +++ b/audio/audio_send_stream.cc @@ -431,6 +431,7 @@ webrtc::AudioSendStream::Stats AudioSendStream::GetStats( @@ -27,10 +27,10 @@ index 0caf59a20e..bffb910832 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 81d5c66652..ddc3323df9 100644 +index 310e0517cf..549e65a59c 100644 --- a/audio/channel_send.cc +++ b/audio/channel_send.cc -@@ -55,6 +55,31 @@ constexpr int64_t kMinRetransmissionWindowMs = 30; +@@ -56,6 +56,31 @@ constexpr int64_t kMinRetransmissionWindowMs = 30; class RtpPacketSenderProxy; class TransportSequenceNumberProxy; @@ -62,7 +62,7 @@ index 81d5c66652..ddc3323df9 100644 class ChannelSend : public ChannelSendInterface, public AudioPacketizationCallback, // receive encoded // packets from the ACM -@@ -207,6 +232,8 @@ class ChannelSend : public ChannelSendInterface, +@@ -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; @@ -71,7 +71,7 @@ index 81d5c66652..ddc3323df9 100644 PacketRouter* packet_router_ RTC_GUARDED_BY(&worker_thread_checker_) = nullptr; const std::unique_ptr<RtpPacketSenderProxy> rtp_packet_pacer_proxy_; -@@ -387,6 +414,7 @@ ChannelSend::ChannelSend( +@@ -398,6 +425,7 @@ ChannelSend::ChannelSend( const FieldTrialsView& field_trials) : ssrc_(ssrc), event_log_(rtc_event_log), @@ -79,7 +79,7 @@ index 81d5c66652..ddc3323df9 100644 rtp_packet_pacer_proxy_(new RtpPacketSenderProxy()), retransmission_rate_limiter_( new RateLimiter(clock, kMaxRetransmissionWindowMs)), -@@ -411,6 +439,8 @@ ChannelSend::ChannelSend( +@@ -423,6 +451,8 @@ ChannelSend::ChannelSend( configuration.event_log = event_log_; configuration.rtt_stats = rtcp_rtt_stats; @@ -88,7 +88,7 @@ index 81d5c66652..ddc3323df9 100644 if (field_trials.IsDisabled("WebRTC-DisableRtxRateLimiter")) { configuration.retransmission_rate_limiter = retransmission_rate_limiter_.get(); -@@ -673,6 +703,7 @@ CallSendStatistics ChannelSend::GetRTCPStatistics() const { +@@ -687,6 +717,7 @@ CallSendStatistics ChannelSend::GetRTCPStatistics() const { RTC_DCHECK_RUN_ON(&worker_thread_checker_); CallSendStatistics stats = {0}; stats.rttMs = GetRTT(); @@ -97,7 +97,7 @@ index 81d5c66652..ddc3323df9 100644 StreamDataCounters rtp_stats; StreamDataCounters rtx_stats; diff --git a/audio/channel_send.h b/audio/channel_send.h -index 00d954c952..f0c9232296 100644 +index b6a6a37bf5..f36085c1fa 100644 --- a/audio/channel_send.h +++ b/audio/channel_send.h @@ -43,6 +43,7 @@ struct CallSendStatistics { |