diff options
Diffstat (limited to 'third_party/libwebrtc/audio/audio_send_stream.h')
-rw-r--r-- | third_party/libwebrtc/audio/audio_send_stream.h | 28 |
1 files changed, 8 insertions, 20 deletions
diff --git a/third_party/libwebrtc/audio/audio_send_stream.h b/third_party/libwebrtc/audio/audio_send_stream.h index 09fd712d40..a37c8fd452 100644 --- a/third_party/libwebrtc/audio/audio_send_stream.h +++ b/third_party/libwebrtc/audio/audio_send_stream.h @@ -110,8 +110,7 @@ class AudioSendStream final : public webrtc::AudioSendStream, const voe::ChannelSendInterface* GetChannel() const; // Returns combined per-packet overhead. - size_t TestOnlyGetPerPacketOverheadBytes() const - RTC_LOCKS_EXCLUDED(overhead_per_packet_lock_); + size_t TestOnlyGetPerPacketOverheadBytes() const; private: class TimedTransport; @@ -152,19 +151,11 @@ class AudioSendStream final : public webrtc::AudioSendStream, // Sets per-packet overhead on encoded (for ANA) based on current known values // of transport and packetization overheads. - void UpdateOverheadForEncoder() - RTC_EXCLUSIVE_LOCKS_REQUIRED(overhead_per_packet_lock_); - - // Returns combined per-packet overhead. - size_t GetPerPacketOverheadBytes() const - RTC_EXCLUSIVE_LOCKS_REQUIRED(overhead_per_packet_lock_); + void UpdateOverheadPerPacket(); void RegisterCngPayloadType(int payload_type, int clockrate_hz) RTC_RUN_ON(worker_thread_checker_); - void UpdateCachedTargetAudioBitrateConstraints() - RTC_RUN_ON(worker_thread_checker_); - Clock* clock_; const FieldTrialsView& field_trials_; @@ -182,6 +173,7 @@ class AudioSendStream final : public webrtc::AudioSendStream, const std::unique_ptr<voe::ChannelSendInterface> channel_send_; RtcEventLog* const event_log_; const bool use_legacy_overhead_calculation_; + const bool enable_priority_bitrate_; int encoder_sample_rate_hz_ RTC_GUARDED_BY(worker_thread_checker_) = 0; size_t encoder_num_channels_ RTC_GUARDED_BY(worker_thread_checker_) = 0; @@ -193,9 +185,6 @@ class AudioSendStream final : public webrtc::AudioSendStream, BitrateAllocatorInterface* const bitrate_allocator_ RTC_GUARDED_BY(worker_thread_checker_); - absl::optional<AudioSendStream::TargetAudioBitrateConstraints> - cached_constraints_ RTC_GUARDED_BY(worker_thread_checker_) = - absl::nullopt; RtpTransportControllerSendInterface* const rtp_transport_; RtpRtcpInterface* const rtp_rtcp_module_; @@ -217,19 +206,18 @@ class AudioSendStream final : public webrtc::AudioSendStream, const std::vector<RtpExtension>& extensions); static int TransportSeqNumId(const Config& config); - mutable Mutex overhead_per_packet_lock_; - size_t overhead_per_packet_ RTC_GUARDED_BY(overhead_per_packet_lock_) = 0; - // Current transport overhead (ICE, TURN, etc.) size_t transport_overhead_per_packet_bytes_ - RTC_GUARDED_BY(overhead_per_packet_lock_) = 0; + RTC_GUARDED_BY(worker_thread_checker_) = 0; + // Total overhead, including transport and RTP headers. + size_t overhead_per_packet_ RTC_GUARDED_BY(worker_thread_checker_) = 0; bool registered_with_allocator_ RTC_GUARDED_BY(worker_thread_checker_) = false; - size_t total_packet_overhead_bytes_ RTC_GUARDED_BY(worker_thread_checker_) = - 0; absl::optional<std::pair<TimeDelta, TimeDelta>> frame_length_range_ RTC_GUARDED_BY(worker_thread_checker_); + absl::optional<std::pair<DataRate, DataRate>> bitrate_range_ + RTC_GUARDED_BY(worker_thread_checker_); }; } // namespace internal } // namespace webrtc |