ff761fba82
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
104 lines
3.8 KiB
C++
104 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 WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_
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#define 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/typedefs.h"
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namespace webrtc {
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class RTPPayloadRegistry;
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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 forwarding of outband telephone events turned on/off?
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virtual bool TelephoneEventForwardToDecoder() const = 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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RtpFeedback* incoming_messages_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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RtpAudioFeedback* incoming_audio_feedback,
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RtpData* incoming_payload_callback,
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RtpFeedback* incoming_messages_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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const char payload_name[RTP_PAYLOAD_NAME_SIZE],
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const int8_t payload_type,
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const uint32_t frequency,
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const uint8_t channels,
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const uint32_t rate) = 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,
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bool in_order) = 0;
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// Returns the currently configured NACK method.
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virtual NACKMethod NACK() const = 0;
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// Turn negative acknowledgement (NACK) requests on/off.
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virtual void SetNACKStatus(const NACKMethod method) = 0;
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// Gets the last received timestamp. Returns true if a packet has been
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// received, false otherwise.
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virtual bool Timestamp(uint32_t* timestamp) const = 0;
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// Gets the time in milliseconds when the last timestamp was received.
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// Returns true if a packet has been received, false otherwise.
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virtual bool LastReceivedTimeMs(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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// Returns the current energy of the RTP stream received.
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virtual int32_t Energy(uint8_t array_of_energy[kRtpCsrcSize]) const = 0;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_
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