f577ae9eac
Replaces VideoCodec in VideoSendStream::Config with an EncoderSettings struct. The EncoderSettings struct uses an external encoder for all codecs. This means that external users, such as libjingle, will provide the encoders themselves, removing the previous distinction of internal and external codecs. For now VideoSendStream translates to VideoCodec internally. In the interrim (before the corresponding change is implemented in VideoReceiveStream) tests convert EncoderSettings to VideoCodecs. Removes Call::GetVideoCodecs(). Disables RampUpTest.WithPacingAndRtx as its further exposed with changes to bitrates used in tests. BUG=2854,2992 R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/7919004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5722 4adac7df-926f-26a2-2b94-8c16560cd09d
113 lines
3.3 KiB
C++
113 lines
3.3 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_CALL_H_
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#define WEBRTC_CALL_H_
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#include <string>
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#include <vector>
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#include "webrtc/common_types.h"
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#include "webrtc/video_receive_stream.h"
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#include "webrtc/video_send_stream.h"
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namespace webrtc {
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class VoiceEngine;
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const char* Version();
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class PacketReceiver {
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public:
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virtual bool DeliverPacket(const uint8_t* packet, size_t length) = 0;
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protected:
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virtual ~PacketReceiver() {}
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};
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// Callback interface for reporting when a system overuse is detected.
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// The detection is based on the jitter of incoming captured frames.
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class OveruseCallback {
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public:
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// Called as soon as an overuse is detected.
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virtual void OnOveruse() = 0;
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// Called periodically when the system is not overused any longer.
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virtual void OnNormalUse() = 0;
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protected:
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virtual ~OveruseCallback() {}
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};
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// A Call instance can contain several send and/or receive streams. All streams
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// are assumed to have the same remote endpoint and will share bitrate estimates
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// etc.
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class Call {
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public:
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struct Config {
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explicit Config(newapi::Transport* send_transport)
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: webrtc_config(NULL),
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send_transport(send_transport),
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voice_engine(NULL),
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trace_callback(NULL),
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trace_filter(kTraceDefault),
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overuse_callback(NULL) {}
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webrtc::Config* webrtc_config;
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newapi::Transport* send_transport;
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// VoiceEngine used for audio/video synchronization for this Call.
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VoiceEngine* voice_engine;
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TraceCallback* trace_callback;
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uint32_t trace_filter;
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// Callback for overuse and normal usage based on the jitter of incoming
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// captured frames. 'NULL' disables the callback.
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OveruseCallback* overuse_callback;
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};
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static Call* Create(const Call::Config& config);
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static Call* Create(const Call::Config& config,
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const webrtc::Config& webrtc_config);
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virtual VideoSendStream::Config GetDefaultSendConfig() = 0;
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virtual VideoSendStream* CreateVideoSendStream(
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const VideoSendStream::Config& config) = 0;
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virtual void DestroyVideoSendStream(VideoSendStream* send_stream) = 0;
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virtual VideoReceiveStream::Config GetDefaultReceiveConfig() = 0;
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virtual VideoReceiveStream* CreateVideoReceiveStream(
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const VideoReceiveStream::Config& config) = 0;
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virtual void DestroyVideoReceiveStream(
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VideoReceiveStream* receive_stream) = 0;
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// All received RTP and RTCP packets for the call should be inserted to this
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// PacketReceiver. The PacketReceiver pointer is valid as long as the
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// Call instance exists.
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virtual PacketReceiver* Receiver() = 0;
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// Returns the estimated total send bandwidth. Note: this can differ from the
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// actual encoded bitrate.
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virtual uint32_t SendBitrateEstimate() = 0;
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// Returns the total estimated receive bandwidth for the call. Note: this can
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// differ from the actual receive bitrate.
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virtual uint32_t ReceiveBitrateEstimate() = 0;
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virtual ~Call() {}
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};
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} // namespace webrtc
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#endif // WEBRTC_CALL_H_
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