blob: 866898cbfaf17cd7e41239b242b1d27f7d2c0f12 [file] [log] [blame]
* Copyright (c) 2004 The WebRTC project authors. All Rights Reserved.
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
#include <CoreAudio/CoreAudio.h>
#include <string>
#include <vector>
#include "webrtc/api/call/audio_state.h"
#include "webrtc/api/rtpparameters.h"
#include "webrtc/base/fileutils.h"
#include "webrtc/base/sigslotrepeater.h"
#include "webrtc/media/base/codec.h"
#include "webrtc/media/base/mediachannel.h"
#include "webrtc/media/base/mediacommon.h"
#include "webrtc/media/base/videocommon.h"
#include "webrtc/modules/audio_coding/codecs/audio_decoder_factory.h"
namespace webrtc {
class AudioDeviceModule;
class Call;
namespace cricket {
struct RtpCapabilities {
std::vector<webrtc::RtpExtension> header_extensions;
// MediaEngineInterface is an abstraction of a media engine which can be
// subclassed to support different media componentry backends.
// It supports voice and video operations in the same class to facilitate
// proper synchronization between both media types.
class MediaEngineInterface {
virtual ~MediaEngineInterface() {}
// Initialization
// Starts the engine.
virtual bool Init() = 0;
// TODO(solenberg): Remove once VoE API refactoring is done.
virtual rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const = 0;
// MediaChannel creation
// Creates a voice media channel. Returns NULL on failure.
virtual VoiceMediaChannel* CreateChannel(webrtc::Call* call,
const MediaConfig& config,
const AudioOptions& options) = 0;
// Creates a video media channel, paired with the specified voice channel.
// Returns NULL on failure.
virtual VideoMediaChannel* CreateVideoChannel(
webrtc::Call* call,
const MediaConfig& config,
const VideoOptions& options) = 0;
// Gets the current microphone level, as a value between 0 and 10.
virtual int GetInputLevel() = 0;
virtual const std::vector<AudioCodec>& audio_send_codecs() = 0;
virtual const std::vector<AudioCodec>& audio_recv_codecs() = 0;
virtual RtpCapabilities GetAudioCapabilities() = 0;
virtual const std::vector<VideoCodec>& video_codecs() = 0;
virtual RtpCapabilities GetVideoCapabilities() = 0;
// Starts AEC dump using existing file, a maximum file size in bytes can be
// specified. Logging is stopped just before the size limit is exceeded.
// If max_size_bytes is set to a value <= 0, no limit will be used.
virtual bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) = 0;
// Stops recording AEC dump.
virtual void StopAecDump() = 0;
class MediaEngineFactory {
typedef cricket::MediaEngineInterface* (*MediaEngineCreateFunction)();
// Creates a media engine, using either the compiled system default or the
// creation function specified in SetCreateFunction, if specified.
static MediaEngineInterface* Create();
// Sets the function used when calling Create. If unset, the compiled system
// default will be used. Returns the old create function, or NULL if one
// wasn't set. Likewise, NULL can be used as the |function| parameter to
// reset to the default behavior.
static MediaEngineCreateFunction SetCreateFunction(
MediaEngineCreateFunction function);
static MediaEngineCreateFunction create_function_;
// CompositeMediaEngine constructs a MediaEngine from separate
// voice and video engine classes.
template<class VOICE, class VIDEO>
class CompositeMediaEngine : public MediaEngineInterface {
webrtc::AudioDeviceModule* adm,
const rtc::scoped_refptr<webrtc::AudioDecoderFactory>&
: voice_(adm, audio_decoder_factory) {}
virtual ~CompositeMediaEngine() {}
virtual bool Init() {
return true;
virtual rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const {
return voice_.GetAudioState();
virtual VoiceMediaChannel* CreateChannel(webrtc::Call* call,
const MediaConfig& config,
const AudioOptions& options) {
return voice_.CreateChannel(call, config, options);
virtual VideoMediaChannel* CreateVideoChannel(webrtc::Call* call,
const MediaConfig& config,
const VideoOptions& options) {
return video_.CreateChannel(call, config, options);
virtual int GetInputLevel() {
return voice_.GetInputLevel();
virtual const std::vector<AudioCodec>& audio_send_codecs() {
return voice_.send_codecs();
virtual const std::vector<AudioCodec>& audio_recv_codecs() {
return voice_.recv_codecs();
virtual RtpCapabilities GetAudioCapabilities() {
return voice_.GetCapabilities();
virtual const std::vector<VideoCodec>& video_codecs() {
return video_.codecs();
virtual RtpCapabilities GetVideoCapabilities() {
return video_.GetCapabilities();
virtual bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) {
return voice_.StartAecDump(file, max_size_bytes);
virtual void StopAecDump() {
VOICE voice_;
VIDEO video_;
enum DataChannelType { DCT_NONE = 0, DCT_RTP = 1, DCT_SCTP = 2, DCT_QUIC = 3 };
class DataEngineInterface {
virtual ~DataEngineInterface() {}
virtual DataMediaChannel* CreateChannel(DataChannelType type) = 0;
virtual const std::vector<DataCodec>& data_codecs() = 0;
webrtc::RtpParameters CreateRtpParametersWithOneEncoding();
} // namespace cricket