2015-01-20 21:36:13 +00:00
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/*
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2016-02-07 20:46:45 -08:00
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* Copyright (c) 2010 The WebRTC project authors. All Rights Reserved.
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2015-01-20 21:36:13 +00:00
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*
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2016-02-07 20:46:45 -08:00
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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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2015-01-20 21:36:13 +00:00
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*/
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2013-07-10 00:45:36 +00:00
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2016-04-05 15:23:49 +02:00
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#ifndef WEBRTC_MEDIA_BASE_VIDEOADAPTER_H_
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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
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#define WEBRTC_MEDIA_BASE_VIDEOADAPTER_H_
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2013-07-10 00:45:36 +00:00
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2014-07-29 17:36:52 +00:00
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#include "webrtc/base/criticalsection.h"
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2016-02-29 00:04:41 -08:00
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#include "webrtc/base/optional.h"
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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
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#include "webrtc/media/base/videocommon.h"
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2013-07-10 00:45:36 +00:00
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namespace cricket {
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// VideoAdapter adapts an input video frame to an output frame based on the
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// specified input and output formats. The adaptation includes dropping frames
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2016-04-05 15:23:49 +02:00
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// to reduce frame rate and scaling frames.
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// VideoAdapter is thread safe.
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2013-07-10 00:45:36 +00:00
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class VideoAdapter {
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public:
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VideoAdapter();
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virtual ~VideoAdapter();
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2016-04-05 15:23:49 +02:00
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// Sets the expected frame interval. This controls how often frames should
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// be dropped if |OnOutputFormatRequest| is called with a lower frame
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// interval.
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void SetExpectedInputFrameInterval(int64_t interval);
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2014-11-19 18:09:14 +00:00
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// Return the adapted resolution given the input resolution. The returned
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// resolution will be 0x0 if the frame should be dropped.
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VideoFormat AdaptFrameResolution(int in_width, int in_height);
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// Requests the output frame size and frame interval from
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// |AdaptFrameResolution| to not be larger than |format|.
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void OnOutputFormatRequest(const VideoFormat& format);
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2013-08-30 21:24:16 +00:00
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// Requests the output frame size from |AdaptFrameResolution| to not have
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// more than |max_pixel_count| pixels and have "one step" up more pixels than
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// max_pixel_count_step_up.
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void OnResolutionRequest(rtc::Optional<int> max_pixel_count,
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rtc::Optional<int> max_pixel_count_step_up);
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2014-06-24 07:24:49 +00:00
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2016-04-05 15:23:49 +02:00
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const VideoFormat& input_format() const;
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private:
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void SetInputFormat(const VideoFormat& format);
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bool Adapt(int max_num_pixels, int max_pixel_count_step_up);
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2013-07-10 00:45:36 +00:00
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VideoFormat input_format_;
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VideoFormat output_format_;
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int output_num_pixels_;
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int frames_in_; // Number of input frames.
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int frames_out_; // Number of output frames.
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int frames_scaled_; // Number of frames scaled.
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int adaption_changes_; // Number of changes in scale factor.
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int previous_width_; // Previous adapter output width.
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int previous_height_; // Previous adapter output height.
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Use suffixed {uint,int}{8,16,32,64}_t types.
Removes the use of uint8, etc. in favor of uint8_t.
BUG=webrtc:5024
R=henrik.lundin@webrtc.org, henrikg@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org
Review URL: https://codereview.webrtc.org/1362503003 .
Cr-Commit-Position: refs/heads/master@{#10196}
2015-10-07 12:23:21 +02:00
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int64_t interval_next_frame_;
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// Max number of pixels requested via calls to OnOutputFormatRequest,
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// OnResolutionRequest respectively.
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// The adapted output format is the minimum of these.
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int format_request_max_pixel_count_;
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int resolution_request_max_pixel_count_;
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2013-07-10 00:45:36 +00:00
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// The critical section to protect the above variables.
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2014-07-29 17:36:52 +00:00
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rtc::CriticalSection critical_section_;
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2013-07-10 00:45:36 +00:00
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2015-09-16 05:37:44 -07:00
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RTC_DISALLOW_COPY_AND_ASSIGN(VideoAdapter);
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2013-07-10 00:45:36 +00:00
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};
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} // namespace cricket
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2016-04-05 15:23:49 +02:00
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#endif // WEBRTC_MEDIA_BASE_VIDEOADAPTER_H_
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