webrtc_m130/webrtc/media/engine/webrtcvideoencoderfactory.h

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/*
* Copyright (c) 2013 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.
*/
#ifndef WEBRTC_MEDIA_ENGINE_WEBRTCVIDEOENCODERFACTORY_H_
#define WEBRTC_MEDIA_ENGINE_WEBRTCVIDEOENCODERFACTORY_H_
#include "webrtc/base/refcount.h"
#include "webrtc/common_types.h"
Move talk/media to webrtc/media I removed the 'libjingle' target in talk/libjingle.gyp and replaced all users of it with base/base.gyp:rtc_base. It seems the jsoncpp and expat dependencies were not used by it's previous references. The files in talk/media/testdata were uploaded to Google Storage and added .sha1 files in resources/media instead of simply moving them. The previously disabled warnings that were inherited from talk/build/common.gypi are now replaced by target-specific disabling of only the failing warnings. Additional disabling was needed since the stricter compilation warnings that applies to code in webrtc/. License headers will be updated in a follow-up CL in order to not break Git history. Other modifications: * Updated the header guards. * Sorted the includes using chromium/src/tools/sort-headers.py except for these files: talk/app/webrtc/peerconnectionendtoend_unittest.cc talk/app/webrtc/java/jni/androidmediadecoder_jni.cc talk/app/webrtc/java/jni/androidmediaencoder_jni.cc webrtc/media/devices/win32devicemanager.cc. * Unused GYP reference to libjingle_tests_additional_deps was removed. * Removed duplicated GYP entries of webrtc/base/testutils.cc webrtc/base/testutils.h The HAVE_WEBRTC_VIDEO and HAVE_WEBRTC_VOICE defines were used by only talk/media, so they were moved to the media.gyp. I also checked that none of EXPAT_RELATIVE_PATH, FEATURE_ENABLE_VOICEMAIL, GTEST_RELATIVE_PATH, JSONCPP_RELATIVE_PATH, LOGGING=1, SRTP_RELATIVE_PATH, FEATURE_ENABLE_SSL, FEATURE_ENABLE_VOICEMAIL, FEATURE_ENABLE_PSTN, HAVE_SCTP, HAVE_SRTP, are used by the talk/media code. For Chromium, the following changes will need to be applied to the roll CL that updates the DEPS for WebRTC and libjingle: https://codereview.chromium.org/1604303002/ BUG=webrtc:5420 NOPRESUBMIT=True TBR=tommi@webrtc.org Review URL: https://codereview.webrtc.org/1587193006 Cr-Commit-Position: refs/heads/master@{#11495}
2016-02-04 23:52:28 -08:00
#include "webrtc/media/base/codec.h"
namespace webrtc {
class VideoEncoder;
}
namespace cricket {
class WebRtcVideoEncoderFactory {
public:
struct VideoCodec {
webrtc::VideoCodecType type;
std::string name;
int max_width;
int max_height;
int max_fps;
VideoCodec(webrtc::VideoCodecType t, const std::string& nm, int w, int h,
int fr)
: type(t), name(nm), max_width(w), max_height(h), max_fps(fr) {
}
};
virtual ~WebRtcVideoEncoderFactory() {}
// Caller takes the ownership of the returned object and it should be released
// by calling DestroyVideoEncoder().
virtual webrtc::VideoEncoder* CreateVideoEncoder(
webrtc::VideoCodecType type) = 0;
// Returns a list of supported codecs in order of preference.
virtual const std::vector<VideoCodec>& codecs() const = 0;
// Returns true if encoders created by this factory of the given codec type
// will use internal camera sources, meaning that they don't require/expect
// frames to be delivered via webrtc::VideoEncoder::Encode. This flag is used
// as the internal_source parameter to
// webrtc::ViEExternalCodec::RegisterExternalSendCodec.
virtual bool EncoderTypeHasInternalSource(webrtc::VideoCodecType type) const {
return false;
}
virtual void DestroyVideoEncoder(webrtc::VideoEncoder* encoder) = 0;
};
} // namespace cricket
#endif // WEBRTC_MEDIA_ENGINE_WEBRTCVIDEOENCODERFACTORY_H_