6488761f2e
Removing assertion that SSRC count should be the same as the number of streams in the codec. It makes sense that you don't always use the same number of streams under one call. Dropping resolution due to CPU overuse for instance can require less streams, but the SSRCs should stay allocated so that operations can resume when not overusing any more. This change also means we can get rid of the ugly SendStreamState whose content wasn't defined. Instead we use SetCodec to change resolution etc. on the fly. Should something else have to be replaced on the fly then that functionality simply has to be implemented. BUG= R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3499005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5123 4adac7df-926f-26a2-2b94-8c16560cd09d
95 lines
2.9 KiB
C++
95 lines
2.9 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_VIDEO_ENGINE_NEW_INCLUDE_CALL_H_
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#define WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_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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// 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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: send_transport(send_transport),
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overuse_detection(false),
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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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newapi::Transport* send_transport;
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bool overuse_detection;
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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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};
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static Call* Create(const Call::Config& config);
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virtual std::vector<VideoCodec> GetVideoCodecs() = 0;
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virtual VideoSendStream::Config GetDefaultSendConfig() = 0;
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virtual VideoSendStream* CreateSendStream(
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const VideoSendStream::Config& config) = 0;
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// Returns the internal state of the send stream, for resume sending with a
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// new stream with different settings.
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// Note: Only the last returned send-stream state is valid.
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virtual void DestroySendStream(VideoSendStream* send_stream) = 0;
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virtual VideoReceiveStream::Config GetDefaultReceiveConfig() = 0;
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virtual VideoReceiveStream* CreateReceiveStream(
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const VideoReceiveStream::Config& config) = 0;
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virtual void DestroyReceiveStream(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_VIDEO_ENGINE_NEW_INCLUDE_CALL_H_
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