1bd66642c3
All users call SetTelephoneEventForwardToDecoder(true). Setting the flag to true on construction, enables deletion of those calls, followed by deletion of the flag itself. The unused getter method TelephoneEventForwardToDecoder() is deleted right away. Bug: webrtc:7135 Change-Id: I8c52c957b3f074be7ffc425b3588402d1e42b844 Reviewed-on: https://webrtc-review.googlesource.com/90402 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24141}
105 lines
3.8 KiB
C++
105 lines
3.8 KiB
C++
/*
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* Copyright (c) 2012 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 MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_
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#define MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_
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#include <vector>
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#include "api/rtpreceiverinterface.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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namespace webrtc {
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class RTPPayloadRegistry;
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class VideoCodec;
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class TelephoneEventHandler {
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public:
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virtual ~TelephoneEventHandler() {}
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// The following three methods implement the TelephoneEventHandler interface.
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// Forward DTMFs to decoder for playout.
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virtual void SetTelephoneEventForwardToDecoder(bool forward_to_decoder) = 0;
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// Is TelephoneEvent configured with payload type payload_type
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virtual bool TelephoneEventPayloadType(const int8_t payload_type) const = 0;
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};
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class RtpReceiver {
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public:
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// Creates a video-enabled RTP receiver.
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static RtpReceiver* CreateVideoReceiver(
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Clock* clock,
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RtpData* incoming_payload_callback,
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RTPPayloadRegistry* rtp_payload_registry);
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// Creates an audio-enabled RTP receiver.
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static RtpReceiver* CreateAudioReceiver(
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Clock* clock,
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RtpData* incoming_payload_callback,
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RTPPayloadRegistry* rtp_payload_registry);
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virtual ~RtpReceiver() {}
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// Returns a TelephoneEventHandler if available.
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virtual TelephoneEventHandler* GetTelephoneEventHandler() = 0;
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// Registers a receive payload in the payload registry and notifies the media
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// receiver strategy.
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virtual int32_t RegisterReceivePayload(
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int payload_type,
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const SdpAudioFormat& audio_format) = 0;
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// Deprecated version of the above.
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int32_t RegisterReceivePayload(const CodecInst& audio_codec);
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// Registers a receive payload in the payload registry.
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virtual int32_t RegisterReceivePayload(const VideoCodec& video_codec) = 0;
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// De-registers |payload_type| from the payload registry.
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virtual int32_t DeRegisterReceivePayload(const int8_t payload_type) = 0;
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// Parses the media specific parts of an RTP packet and updates the receiver
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// state. This for instance means that any changes in SSRC and payload type is
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// detected and acted upon.
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virtual 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) = 0;
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// TODO(nisse): Deprecated version, delete as soon as downstream
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// applications are updated.
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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 /* Ignored */) {
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return IncomingRtpPacket(rtp_header, payload, payload_length,
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payload_specific);
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}
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// Gets the RTP timestamp and the corresponding monotonic system
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// time for the most recent in-order packet. Returns true on
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// success, false if no packet has been received.
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virtual bool GetLatestTimestamps(uint32_t* timestamp,
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int64_t* receive_time_ms) const = 0;
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// Returns the remote SSRC of the currently received RTP stream.
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virtual uint32_t SSRC() const = 0;
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// Returns the current remote CSRCs.
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virtual int32_t CSRCs(uint32_t array_of_csrc[kRtpCsrcSize]) const = 0;
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virtual std::vector<RtpSource> GetSources() const = 0;
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};
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} // namespace webrtc
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#endif // MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_
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