fbcc5cb386
Reason for revert:
Suspected of WebRtcApprtcBrowserTest.MANUAL_WorksOnApprtc breakage, see
https://bugs.chromium.org/p/webrtc/issues/detail?id=7465
Original issue's description:
> Added the GetSources() to the RtpReceiverInterface and implemented
> it for the AudioRtpReceiver.
>
> This method returns a vector of RtpSource(both CSRC source and SSRC
> source) which contains the ID of a source, the timestamp, the source
> type (SSRC or CSRC) and the audio level.
>
> The RtpSource objects are buffered and maintained by the
> RtpReceiver in webrtc/modules/rtp_rtcp/. When the method is called,
> the info of the contributing source will be pulled along the object
> chain:
> AudioRtpReceiver -> VoiceChannel -> WebRtcVoiceMediaChannel ->
> AudioReceiveStream -> voe::Channel -> RtpRtcp module
>
> Spec:https://w3c.github.io/webrtc-pc/archives/20151006/webrtc.html#widl-RTCRtpReceiver-getContributingSources-sequence-RTCRtpContributingSource
>
> BUG=chromium:703122
> TBR=stefan@webrtc.org, danilchap@webrtc.org
>
> Review-Url: https://codereview.webrtc.org/2770233003
> Cr-Commit-Position: refs/heads/master@{#17591}
> Committed: 292084c376
TBR=deadbeef@webrtc.org,solenberg@webrtc.org,hbos@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=chromium:703122
Review-Url: https://codereview.webrtc.org/2809613002
Cr-Commit-Position: refs/heads/master@{#17616}
90 lines
3.1 KiB
C++
90 lines
3.1 KiB
C++
/*
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* Copyright (c) 2013 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_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_
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#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_
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#include <memory>
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#include "webrtc/base/criticalsection.h"
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#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
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#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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class RtpReceiverImpl : public RtpReceiver {
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public:
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// Callbacks passed in here may not be NULL (use Null Object callbacks if you
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// want callbacks to do nothing). This class takes ownership of the media
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// receiver but nothing else.
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RtpReceiverImpl(Clock* clock,
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RtpFeedback* incoming_messages_callback,
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RTPPayloadRegistry* rtp_payload_registry,
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RTPReceiverStrategy* rtp_media_receiver);
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virtual ~RtpReceiverImpl();
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int32_t RegisterReceivePayload(const CodecInst& audio_codec) override;
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int32_t RegisterReceivePayload(const VideoCodec& video_codec) override;
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int32_t DeRegisterReceivePayload(const int8_t payload_type) override;
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bool IncomingRtpPacket(const RTPHeader& rtp_header,
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const uint8_t* payload,
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size_t payload_length,
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PayloadUnion payload_specific,
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bool in_order) override;
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// Returns the last received timestamp.
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bool Timestamp(uint32_t* timestamp) const override;
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bool LastReceivedTimeMs(int64_t* receive_time_ms) const override;
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uint32_t SSRC() const override;
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int32_t CSRCs(uint32_t array_of_csrc[kRtpCsrcSize]) const override;
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int32_t Energy(uint8_t array_of_energy[kRtpCsrcSize]) const override;
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TelephoneEventHandler* GetTelephoneEventHandler() override;
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private:
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bool HaveReceivedFrame() const;
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void CheckSSRCChanged(const RTPHeader& rtp_header);
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void CheckCSRC(const WebRtcRTPHeader& rtp_header);
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int32_t CheckPayloadChanged(const RTPHeader& rtp_header,
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const int8_t first_payload_byte,
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bool* is_red,
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PayloadUnion* payload);
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Clock* clock_;
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RTPPayloadRegistry* rtp_payload_registry_;
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std::unique_ptr<RTPReceiverStrategy> rtp_media_receiver_;
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RtpFeedback* cb_rtp_feedback_;
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rtc::CriticalSection critical_section_rtp_receiver_;
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int64_t last_receive_time_;
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size_t last_received_payload_length_;
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// SSRCs.
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uint32_t ssrc_;
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uint8_t num_csrcs_;
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uint32_t current_remote_csrc_[kRtpCsrcSize];
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uint32_t last_received_timestamp_;
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int64_t last_received_frame_time_ms_;
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uint16_t last_received_sequence_number_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_
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