webrtc_m130/webrtc/api/java/jni/androidmediacodeccommon.h

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/*
* Copyright 2015 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_API_JAVA_JNI_ANDROIDMEDIACODECCOMMON_H_
#define WEBRTC_API_JAVA_JNI_ANDROIDMEDIACODECCOMMON_H_
#include <android/log.h>
#include <string>
#include "webrtc/base/thread.h"
#include "webrtc/api/java/jni/classreferenceholder.h"
#include "webrtc/api/java/jni/jni_helpers.h"
#include "webrtc/base/logging.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/base/thread.h"
#include "webrtc/system_wrappers/include/tick_util.h"
namespace webrtc_jni {
// Uncomment this define to enable verbose logging for every encoded/decoded
// video frame.
//#define TRACK_BUFFER_TIMING
#define TAG_COMMON "MediaCodecVideo"
// Color formats supported by encoder - should mirror supportedColorList
// from MediaCodecVideoEncoder.java
enum COLOR_FORMATTYPE {
COLOR_FormatYUV420Planar = 0x13,
COLOR_FormatYUV420SemiPlanar = 0x15,
COLOR_QCOM_FormatYUV420SemiPlanar = 0x7FA30C00,
// NV12 color format supported by QCOM codec, but not declared in MediaCodec -
// see /hardware/qcom/media/mm-core/inc/OMX_QCOMExtns.h
// This format is presumably similar to COLOR_FormatYUV420SemiPlanar,
// but requires some (16, 32?) byte alignment.
COLOR_QCOM_FORMATYUV420PackedSemiPlanar32m = 0x7FA30C04
};
// Arbitrary interval to poll the codec for new outputs.
enum { kMediaCodecPollMs = 10 };
// Media codec maximum output buffer ready timeout.
enum { kMediaCodecTimeoutMs = 1000 };
// Interval to print codec statistics (bitrate, fps, encoding/decoding time).
enum { kMediaCodecStatisticsIntervalMs = 3000 };
// Maximum amount of pending frames for VP8 decoder.
enum { kMaxPendingFramesVp8 = 1 };
// Maximum amount of pending frames for VP9 decoder.
enum { kMaxPendingFramesVp9 = 1 };
// Maximum amount of pending frames for H.264 decoder.
enum { kMaxPendingFramesH264 = 8 };
// Maximum amount of decoded frames for which per-frame logging is enabled.
enum { kMaxDecodedLogFrames = 10 };
// Maximum amount of encoded frames for which per-frame logging is enabled.
enum { kMaxEncodedLogFrames = 10 };
static inline int64_t GetCurrentTimeMs() {
return webrtc::TickTime::Now().Ticks() / 1000000LL;
}
static inline void AllowBlockingCalls() {
rtc::Thread* current_thread = rtc::Thread::Current();
if (current_thread != NULL)
current_thread->SetAllowBlockingCalls(true);
}
// Return the (singleton) Java Enum object corresponding to |index|;
// |state_class_fragment| is something like "MediaSource$State".
static inline jobject JavaEnumFromIndexAndClassName(
JNIEnv* jni, const std::string& state_class_fragment, int index) {
const std::string state_class = "org/webrtc/" + state_class_fragment;
return JavaEnumFromIndex(jni, FindClass(jni, state_class.c_str()),
state_class, index);
}
// Checks for any Java exception, prints stack backtrace and clears
// currently thrown exception.
static inline bool CheckException(JNIEnv* jni) {
if (jni->ExceptionCheck()) {
LOG_TAG(rtc::LS_ERROR, TAG_COMMON) << "Java JNI exception.";
jni->ExceptionDescribe();
jni->ExceptionClear();
return true;
}
return false;
}
} // namespace webrtc_jni
#endif // WEBRTC_API_JAVA_JNI_ANDROIDMEDIACODECCOMMON_H_