208 lines
6.9 KiB
C++
208 lines
6.9 KiB
C++
/*
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* Copyright (c) 2004 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_MEDIA_BASE_MEDIAENGINE_H_
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#define WEBRTC_MEDIA_BASE_MEDIAENGINE_H_
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#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
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#include <CoreAudio/CoreAudio.h>
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#endif
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#include <string>
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#include <vector>
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#include "webrtc/audio_state.h"
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#include "webrtc/api/rtpparameters.h"
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#include "webrtc/base/fileutils.h"
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#include "webrtc/base/sigslotrepeater.h"
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#include "webrtc/media/base/codec.h"
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#include "webrtc/media/base/mediachannel.h"
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#include "webrtc/media/base/mediacommon.h"
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#include "webrtc/media/base/videocapturer.h"
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#include "webrtc/media/base/videocommon.h"
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#include "webrtc/modules/audio_coding/codecs/audio_decoder_factory.h"
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#if defined(GOOGLE_CHROME_BUILD) || defined(CHROMIUM_BUILD)
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#define DISABLE_MEDIA_ENGINE_FACTORY
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#endif
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namespace webrtc {
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class AudioDeviceModule;
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class Call;
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}
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namespace cricket {
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class VideoCapturer;
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struct RtpCapabilities {
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std::vector<webrtc::RtpExtension> header_extensions;
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};
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// MediaEngineInterface is an abstraction of a media engine which can be
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// subclassed to support different media componentry backends.
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// It supports voice and video operations in the same class to facilitate
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// proper synchronization between both media types.
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class MediaEngineInterface {
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public:
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virtual ~MediaEngineInterface() {}
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// Initialization
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// Starts the engine.
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virtual bool Init() = 0;
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// TODO(solenberg): Remove once VoE API refactoring is done.
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virtual rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const = 0;
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// MediaChannel creation
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// Creates a voice media channel. Returns NULL on failure.
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virtual VoiceMediaChannel* CreateChannel(webrtc::Call* call,
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const MediaConfig& config,
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const AudioOptions& options) = 0;
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// Creates a video media channel, paired with the specified voice channel.
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// Returns NULL on failure.
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virtual VideoMediaChannel* CreateVideoChannel(
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webrtc::Call* call,
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const MediaConfig& config,
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const VideoOptions& options) = 0;
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// Gets the current microphone level, as a value between 0 and 10.
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virtual int GetInputLevel() = 0;
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virtual const std::vector<AudioCodec>& audio_send_codecs() = 0;
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virtual const std::vector<AudioCodec>& audio_recv_codecs() = 0;
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virtual RtpCapabilities GetAudioCapabilities() = 0;
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virtual const std::vector<VideoCodec>& video_codecs() = 0;
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virtual RtpCapabilities GetVideoCapabilities() = 0;
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// Starts AEC dump using existing file, a maximum file size in bytes can be
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// specified. Logging is stopped just before the size limit is exceeded.
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// If max_size_bytes is set to a value <= 0, no limit will be used.
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virtual bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) = 0;
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// Stops recording AEC dump.
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virtual void StopAecDump() = 0;
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// Starts RtcEventLog using existing file. A maximum file size in bytes can be
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// specified. Logging is stopped just before the size limit is exceeded.
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// If max_size_bytes is set to a value <= 0, no limit will be used.
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virtual bool StartRtcEventLog(rtc::PlatformFile file,
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int64_t max_size_bytes) = 0;
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// Stops recording an RtcEventLog.
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virtual void StopRtcEventLog() = 0;
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};
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#if !defined(DISABLE_MEDIA_ENGINE_FACTORY)
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class MediaEngineFactory {
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public:
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typedef cricket::MediaEngineInterface* (*MediaEngineCreateFunction)();
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// Creates a media engine, using either the compiled system default or the
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// creation function specified in SetCreateFunction, if specified.
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static MediaEngineInterface* Create();
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// Sets the function used when calling Create. If unset, the compiled system
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// default will be used. Returns the old create function, or NULL if one
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// wasn't set. Likewise, NULL can be used as the |function| parameter to
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// reset to the default behavior.
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static MediaEngineCreateFunction SetCreateFunction(
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MediaEngineCreateFunction function);
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private:
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static MediaEngineCreateFunction create_function_;
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};
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#endif
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// CompositeMediaEngine constructs a MediaEngine from separate
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// voice and video engine classes.
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template<class VOICE, class VIDEO>
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class CompositeMediaEngine : public MediaEngineInterface {
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public:
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CompositeMediaEngine(
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webrtc::AudioDeviceModule* adm,
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const rtc::scoped_refptr<webrtc::AudioDecoderFactory>&
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audio_decoder_factory)
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: voice_(adm, audio_decoder_factory) {}
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virtual ~CompositeMediaEngine() {}
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virtual bool Init() {
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video_.Init();
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return true;
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}
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virtual rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const {
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return voice_.GetAudioState();
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}
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virtual VoiceMediaChannel* CreateChannel(webrtc::Call* call,
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const MediaConfig& config,
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const AudioOptions& options) {
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return voice_.CreateChannel(call, config, options);
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}
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virtual VideoMediaChannel* CreateVideoChannel(webrtc::Call* call,
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const MediaConfig& config,
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const VideoOptions& options) {
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return video_.CreateChannel(call, config, options);
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}
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virtual int GetInputLevel() {
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return voice_.GetInputLevel();
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}
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virtual const std::vector<AudioCodec>& audio_send_codecs() {
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return voice_.send_codecs();
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}
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virtual const std::vector<AudioCodec>& audio_recv_codecs() {
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return voice_.recv_codecs();
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}
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virtual RtpCapabilities GetAudioCapabilities() {
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return voice_.GetCapabilities();
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}
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virtual const std::vector<VideoCodec>& video_codecs() {
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return video_.codecs();
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}
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virtual RtpCapabilities GetVideoCapabilities() {
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return video_.GetCapabilities();
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}
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virtual bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) {
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return voice_.StartAecDump(file, max_size_bytes);
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}
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virtual void StopAecDump() {
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voice_.StopAecDump();
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}
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virtual bool StartRtcEventLog(rtc::PlatformFile file,
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int64_t max_size_bytes) {
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return voice_.StartRtcEventLog(file, max_size_bytes);
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}
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virtual void StopRtcEventLog() { voice_.StopRtcEventLog(); }
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protected:
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VOICE voice_;
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VIDEO video_;
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};
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enum DataChannelType {
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DCT_NONE = 0,
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DCT_RTP = 1,
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DCT_SCTP = 2
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};
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class DataEngineInterface {
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public:
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virtual ~DataEngineInterface() {}
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virtual DataMediaChannel* CreateChannel(DataChannelType type) = 0;
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virtual const std::vector<DataCodec>& data_codecs() = 0;
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};
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webrtc::RtpParameters CreateRtpParametersWithOneEncoding();
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} // namespace cricket
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#endif // WEBRTC_MEDIA_BASE_MEDIAENGINE_H_
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