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92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
131 lines
4.8 KiB
C++
131 lines
4.8 KiB
C++
/*
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* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef AUDIO_AUDIO_SEND_STREAM_H_
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#define AUDIO_AUDIO_SEND_STREAM_H_
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#include <memory>
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#include <vector>
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#include "audio/time_interval.h"
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#include "call/audio_send_stream.h"
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#include "call/audio_state.h"
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#include "call/bitrate_allocator.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp.h"
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#include "rtc_base/constructormagic.h"
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#include "rtc_base/thread_checker.h"
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#include "voice_engine/transport_feedback_packet_loss_tracker.h"
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namespace webrtc {
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class VoiceEngine;
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class RtcEventLog;
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class RtcpBandwidthObserver;
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class RtcpRttStats;
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class RtpTransportControllerSendInterface;
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namespace voe {
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class ChannelProxy;
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} // namespace voe
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namespace internal {
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class AudioSendStream final : public webrtc::AudioSendStream,
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public webrtc::BitrateAllocatorObserver,
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public webrtc::PacketFeedbackObserver {
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public:
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AudioSendStream(const webrtc::AudioSendStream::Config& config,
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const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
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rtc::TaskQueue* worker_queue,
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RtpTransportControllerSendInterface* transport,
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BitrateAllocator* bitrate_allocator,
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RtcEventLog* event_log,
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RtcpRttStats* rtcp_rtt_stats,
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const rtc::Optional<RtpState>& suspended_rtp_state);
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~AudioSendStream() override;
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// webrtc::AudioSendStream implementation.
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const webrtc::AudioSendStream::Config& GetConfig() const override;
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void Reconfigure(const webrtc::AudioSendStream::Config& config) override;
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void Start() override;
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void Stop() override;
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bool SendTelephoneEvent(int payload_type, int payload_frequency, int event,
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int duration_ms) override;
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void SetMuted(bool muted) override;
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webrtc::AudioSendStream::Stats GetStats() const override;
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void SignalNetworkState(NetworkState state);
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bool DeliverRtcp(const uint8_t* packet, size_t length);
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// Implements BitrateAllocatorObserver.
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uint32_t OnBitrateUpdated(uint32_t bitrate_bps,
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uint8_t fraction_loss,
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int64_t rtt,
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int64_t bwe_period_ms) override;
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// From PacketFeedbackObserver.
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void OnPacketAdded(uint32_t ssrc, uint16_t seq_num) override;
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void OnPacketFeedbackVector(
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const std::vector<PacketFeedback>& packet_feedback_vector) override;
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void SetTransportOverhead(int transport_overhead_per_packet);
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RtpState GetRtpState() const;
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const TimeInterval& GetActiveLifetime() const;
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private:
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class TimedTransport;
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VoiceEngine* voice_engine() const;
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// These are all static to make it less likely that (the old) config_ is
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// accessed unintentionally.
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static void ConfigureStream(AudioSendStream* stream,
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const Config& new_config,
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bool first_time);
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static bool SetupSendCodec(AudioSendStream* stream, const Config& new_config);
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static bool ReconfigureSendCodec(AudioSendStream* stream,
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const Config& new_config);
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static void ReconfigureANA(AudioSendStream* stream, const Config& new_config);
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static void ReconfigureCNG(AudioSendStream* stream, const Config& new_config);
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static void ReconfigureBitrateObserver(AudioSendStream* stream,
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const Config& new_config);
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void ConfigureBitrateObserver(int min_bitrate_bps, int max_bitrate_bps);
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void RemoveBitrateObserver();
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void RegisterCngPayloadType(int payload_type, int clockrate_hz);
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rtc::ThreadChecker worker_thread_checker_;
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rtc::ThreadChecker pacer_thread_checker_;
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rtc::TaskQueue* worker_queue_;
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webrtc::AudioSendStream::Config config_;
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rtc::scoped_refptr<webrtc::AudioState> audio_state_;
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std::unique_ptr<voe::ChannelProxy> channel_proxy_;
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RtcEventLog* const event_log_;
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BitrateAllocator* const bitrate_allocator_;
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RtpTransportControllerSendInterface* const transport_;
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std::unique_ptr<RtcpBandwidthObserver> bandwidth_observer_;
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rtc::CriticalSection packet_loss_tracker_cs_;
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TransportFeedbackPacketLossTracker packet_loss_tracker_
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RTC_GUARDED_BY(&packet_loss_tracker_cs_);
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RtpRtcp* rtp_rtcp_module_;
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rtc::Optional<RtpState> const suspended_rtp_state_;
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std::unique_ptr<TimedTransport> timed_send_transport_adapter_;
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TimeInterval active_lifetime_;
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RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream);
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};
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} // namespace internal
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} // namespace webrtc
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#endif // AUDIO_AUDIO_SEND_STREAM_H_
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