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through streams related to a call object. The Call object does not know directly when packets pass through it, only which AudioSendStreams are used. Each AudioSendStream has a pointer to the Transport object through which its packets are send. This CL: By registering an internal wrapper class, TimedTransport, the AudioSendStream can stay up-to-date on when packets have passed through its Transport. This lifetime (as an interval) is then queried by the Call when the AudioSendStream is destroyed. When Call is destroyed, all streams are guaranteed to have been destroyed and hence Call is up-to-date on packet activity. The class TimeInterval keeps the code in Call and AudioSendStream smaller, with fewer get methods in their APIs and less code for updating values. Also modifies the unit test for AudioSendStream: it previously enforced that the stream registers (with its channel proxy) the same transport that it was constructed with. BUG=webrtc:7882 Review-Url: https://codereview.webrtc.org/2979833002 Cr-Commit-Position: refs/heads/master@{#19087}
133 lines
4.9 KiB
C++
133 lines
4.9 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 WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_
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#define WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_
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#include <memory>
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#include <vector>
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#include "webrtc/audio/time_interval.h"
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#include "webrtc/call/audio_send_stream.h"
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#include "webrtc/call/audio_state.h"
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#include "webrtc/call/bitrate_allocator.h"
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#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
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#include "webrtc/rtc_base/constructormagic.h"
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#include "webrtc/rtc_base/thread_checker.h"
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#include "webrtc/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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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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const webrtc::AudioSendStream::Config& config() const;
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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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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 // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_
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