webrtc_m130/webrtc/media/base/videoadapter.h

Ignoring revisions in .git-blame-ignore-revs. Click here to bypass and see the normal blame view.

85 lines
3.5 KiB
C
Raw Normal View History

/*
* Copyright (c) 2010 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_BASE_VIDEOADAPTER_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
#define WEBRTC_MEDIA_BASE_VIDEOADAPTER_H_
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/optional.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/videocommon.h"
namespace cricket {
// VideoAdapter adapts an input video frame to an output frame based on the
// specified input and output formats. The adaptation includes dropping frames
// to reduce frame rate and scaling frames.
// VideoAdapter is thread safe.
class VideoAdapter {
public:
VideoAdapter();
virtual ~VideoAdapter();
// Return the adapted resolution and cropping parameters given the
// input resolution. The input frame should first be cropped, then
// scaled to the final output resolution. Returns true if the frame
// should be adapted, and false if it should be dropped.
bool AdaptFrameResolution(int in_width,
int in_height,
int64_t in_timestamp_ns,
int* cropped_width,
int* cropped_height,
int* out_width,
int* out_height);
// Requests the output frame size and frame interval from
// |AdaptFrameResolution| to not be larger than |format|. Also, the input
// frame size will be cropped to match the requested aspect ratio. The
// requested aspect ratio is orientation agnostic and will be adjusted to
// maintain the input orientation, so it doesn't matter if e.g. 1280x720 or
// 720x1280 is requested.
void OnOutputFormatRequest(const VideoFormat& format);
// Requests the output frame size from |AdaptFrameResolution| to not have
// more than |max_pixel_count| pixels and have "one step" up more pixels than
// max_pixel_count_step_up.
void OnResolutionRequest(rtc::Optional<int> max_pixel_count,
rtc::Optional<int> max_pixel_count_step_up);
private:
// Determine if frame should be dropped based on input fps and requested fps.
bool KeepFrame(int64_t in_timestamp_ns);
int frames_in_; // Number of input frames.
int frames_out_; // Number of output frames.
int frames_scaled_; // Number of frames scaled.
int adaption_changes_; // Number of changes in scale factor.
int previous_width_; // Previous adapter output width.
int previous_height_; // Previous adapter output height.
// The target timestamp for the next frame based on requested format.
rtc::Optional<int64_t> next_frame_timestamp_ns_ GUARDED_BY(critical_section_);
// Max number of pixels requested via calls to OnOutputFormatRequest,
// OnResolutionRequest respectively.
// The adapted output format is the minimum of these.
rtc::Optional<VideoFormat> requested_format_ GUARDED_BY(critical_section_);
int resolution_request_max_pixel_count_ GUARDED_BY(critical_section_);
int resolution_request_max_pixel_count_step_up_ GUARDED_BY(critical_section_);
// The critical section to protect the above variables.
rtc::CriticalSection critical_section_;
RTC_DISALLOW_COPY_AND_ASSIGN(VideoAdapter);
};
} // namespace cricket
#endif // WEBRTC_MEDIA_BASE_VIDEOADAPTER_H_