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authorAllan Sandfeld Jensen <allan.jensen@theqtcompany.com>2015-06-18 14:10:49 +0200
committerOswald Buddenhagen <oswald.buddenhagen@theqtcompany.com>2015-06-18 13:53:24 +0000
commit813fbf95af77a531c57a8c497345ad2c61d475b3 (patch)
tree821b2c8de8365f21b6c9ba17a236fb3006a1d506 /chromium/third_party/webrtc/modules/video_coding/codecs/test
parentaf6588f8d723931a298c995fa97259bb7f7deb55 (diff)
downloadqtwebengine-chromium-813fbf95af77a531c57a8c497345ad2c61d475b3.tar.gz
BASELINE: Update chromium to 44.0.2403.47
Change-Id: Ie056fedba95cf5e5c76b30c4b2c80fca4764aa2f Reviewed-by: Oswald Buddenhagen <oswald.buddenhagen@theqtcompany.com>
Diffstat (limited to 'chromium/third_party/webrtc/modules/video_coding/codecs/test')
-rw-r--r--chromium/third_party/webrtc/modules/video_coding/codecs/test/OWNERS5
-rw-r--r--chromium/third_party/webrtc/modules/video_coding/codecs/test/mock/mock_packet_manipulator.h33
-rw-r--r--chromium/third_party/webrtc/modules/video_coding/codecs/test/packet_manipulator.cc112
-rw-r--r--chromium/third_party/webrtc/modules/video_coding/codecs/test/packet_manipulator.h112
-rw-r--r--chromium/third_party/webrtc/modules/video_coding/codecs/test/packet_manipulator_unittest.cc153
-rw-r--r--chromium/third_party/webrtc/modules/video_coding/codecs/test/predictive_packet_manipulator.cc48
-rw-r--r--chromium/third_party/webrtc/modules/video_coding/codecs/test/predictive_packet_manipulator.h45
-rw-r--r--chromium/third_party/webrtc/modules/video_coding/codecs/test/stats.cc190
-rw-r--r--chromium/third_party/webrtc/modules/video_coding/codecs/test/stats.h70
-rw-r--r--chromium/third_party/webrtc/modules/video_coding/codecs/test/stats_unittest.cc64
-rw-r--r--chromium/third_party/webrtc/modules/video_coding/codecs/test/videoprocessor.cc415
-rw-r--r--chromium/third_party/webrtc/modules/video_coding/codecs/test/videoprocessor.h254
-rw-r--r--chromium/third_party/webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest.cc936
-rw-r--r--chromium/third_party/webrtc/modules/video_coding/codecs/test/videoprocessor_unittest.cc99
14 files changed, 2536 insertions, 0 deletions
diff --git a/chromium/third_party/webrtc/modules/video_coding/codecs/test/OWNERS b/chromium/third_party/webrtc/modules/video_coding/codecs/test/OWNERS
new file mode 100644
index 00000000000..3ee6b4bf5f9
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/video_coding/codecs/test/OWNERS
@@ -0,0 +1,5 @@
+
+# These are for the common case of adding or renaming files. If you're doing
+# structural changes, please get a review from a reviewer in this file.
+per-file *.gyp=*
+per-file *.gypi=*
diff --git a/chromium/third_party/webrtc/modules/video_coding/codecs/test/mock/mock_packet_manipulator.h b/chromium/third_party/webrtc/modules/video_coding/codecs/test/mock/mock_packet_manipulator.h
new file mode 100644
index 00000000000..1e2c9b8fb64
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/video_coding/codecs/test/mock/mock_packet_manipulator.h
@@ -0,0 +1,33 @@
+/*
+ * Copyright (c) 2011 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_MODULES_VIDEO_CODING_CODECS_TEST_MOCK_MOCK_PACKET_MANIPULATOR_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_MOCK_MOCK_PACKET_MANIPULATOR_H_
+
+#include "webrtc/modules/video_coding/codecs/test/packet_manipulator.h"
+
+#include <string>
+
+#include "testing/gmock/include/gmock/gmock.h"
+#include "webrtc/typedefs.h"
+#include "webrtc/video_frame.h"
+
+namespace webrtc {
+namespace test {
+
+class MockPacketManipulator : public PacketManipulator {
+ public:
+ MOCK_METHOD1(ManipulatePackets, int(webrtc::EncodedImage* encoded_image));
+};
+
+} // namespace test
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_MOCK_MOCK_PACKET_MANIPULATOR_H_
diff --git a/chromium/third_party/webrtc/modules/video_coding/codecs/test/packet_manipulator.cc b/chromium/third_party/webrtc/modules/video_coding/codecs/test/packet_manipulator.cc
new file mode 100644
index 00000000000..36ba0e8272d
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/video_coding/codecs/test/packet_manipulator.cc
@@ -0,0 +1,112 @@
+/*
+ * Copyright (c) 2012 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.
+ */
+
+#include "webrtc/modules/video_coding/codecs/test/packet_manipulator.h"
+
+#include <assert.h>
+#include <stdio.h>
+
+#include "webrtc/base/format_macros.h"
+
+namespace webrtc {
+namespace test {
+
+PacketManipulatorImpl::PacketManipulatorImpl(PacketReader* packet_reader,
+ const NetworkingConfig& config,
+ bool verbose)
+ : packet_reader_(packet_reader),
+ config_(config),
+ active_burst_packets_(0),
+ critsect_(CriticalSectionWrapper::CreateCriticalSection()),
+ random_seed_(1),
+ verbose_(verbose) {
+ assert(packet_reader);
+}
+
+PacketManipulatorImpl::~PacketManipulatorImpl() {
+ delete critsect_;
+}
+
+int PacketManipulatorImpl::ManipulatePackets(
+ webrtc::EncodedImage* encoded_image) {
+ int nbr_packets_dropped = 0;
+ // There's no need to build a copy of the image data since viewing an
+ // EncodedImage object, setting the length to a new lower value represents
+ // that everything is dropped after that position in the byte array.
+ // EncodedImage._size is the allocated bytes.
+ // EncodedImage._length is how many that are filled with data.
+ int new_length = 0;
+ packet_reader_->InitializeReading(encoded_image->_buffer,
+ encoded_image->_length,
+ config_.packet_size_in_bytes);
+ uint8_t* packet = NULL;
+ int nbr_bytes_to_read;
+ // keep track of if we've lost any packets, since then we shall loose
+ // the remains of the current frame:
+ bool packet_loss_has_occurred = false;
+ while ((nbr_bytes_to_read = packet_reader_->NextPacket(&packet)) > 0) {
+ // Check if we're currently in a packet loss burst that is not completed:
+ if (active_burst_packets_ > 0) {
+ active_burst_packets_--;
+ nbr_packets_dropped++;
+ } else if (RandomUniform() < config_.packet_loss_probability ||
+ packet_loss_has_occurred) {
+ packet_loss_has_occurred = true;
+ nbr_packets_dropped++;
+ if (config_.packet_loss_mode == kBurst) {
+ // Initiate a new burst
+ active_burst_packets_ = config_.packet_loss_burst_length - 1;
+ }
+ } else {
+ new_length += nbr_bytes_to_read;
+ }
+ }
+ encoded_image->_length = new_length;
+ if (nbr_packets_dropped > 0) {
+ // Must set completeFrame to false to inform the decoder about this:
+ encoded_image->_completeFrame = false;
+ if (verbose_) {
+ printf("Dropped %d packets for frame %d (frame length: %" PRIuS ")\n",
+ nbr_packets_dropped, encoded_image->_timeStamp,
+ encoded_image->_length);
+ }
+ }
+ return nbr_packets_dropped;
+}
+
+void PacketManipulatorImpl::InitializeRandomSeed(unsigned int seed) {
+ random_seed_ = seed;
+}
+
+inline double PacketManipulatorImpl::RandomUniform() {
+ // Use the previous result as new seed before each rand() call. Doing this
+ // it doesn't matter if other threads are calling rand() since we'll always
+ // get the same behavior as long as we're using a fixed initial seed.
+ critsect_->Enter();
+ srand(random_seed_);
+ random_seed_ = rand();
+ critsect_->Leave();
+ return (random_seed_ + 1.0)/(RAND_MAX + 1.0);
+}
+
+const char* PacketLossModeToStr(PacketLossMode e) {
+ switch (e) {
+ case kUniform:
+ return "Uniform";
+ case kBurst:
+ return "Burst";
+ default:
+ assert(false);
+ return "Unknown";
+ }
+}
+
+} // namespace test
+} // namespace webrtcc
diff --git a/chromium/third_party/webrtc/modules/video_coding/codecs/test/packet_manipulator.h b/chromium/third_party/webrtc/modules/video_coding/codecs/test/packet_manipulator.h
new file mode 100644
index 00000000000..5a1654a2a7d
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/video_coding/codecs/test/packet_manipulator.h
@@ -0,0 +1,112 @@
+/*
+ * Copyright (c) 2012 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_MODULES_VIDEO_CODING_CODECS_TEST_PACKET_MANIPULATOR_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_PACKET_MANIPULATOR_H_
+
+#include <stdlib.h>
+
+#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
+#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/test/testsupport/packet_reader.h"
+
+namespace webrtc {
+namespace test {
+
+// Which mode the packet loss shall be performed according to.
+enum PacketLossMode {
+ // Drops packets with a configured probability independently for each packet
+ kUniform,
+ // Drops packets similar to uniform but when a packet is being dropped,
+ // the number of lost packets in a row is equal to the configured burst
+ // length.
+ kBurst
+};
+// Returns a string representation of the enum value.
+const char* PacketLossModeToStr(PacketLossMode e);
+
+// Contains configurations related to networking and simulation of
+// scenarios caused by network interference.
+struct NetworkingConfig {
+ NetworkingConfig()
+ : packet_size_in_bytes(1500), max_payload_size_in_bytes(1440),
+ packet_loss_mode(kUniform), packet_loss_probability(0.0),
+ packet_loss_burst_length(1) {
+ }
+
+ // Packet size in bytes. Default: 1500 bytes.
+ size_t packet_size_in_bytes;
+
+ // Encoder specific setting of maximum size in bytes of each payload.
+ // Default: 1440 bytes.
+ size_t max_payload_size_in_bytes;
+
+ // Packet loss mode. Two different packet loss models are supported:
+ // uniform or burst. This setting has no effect unless
+ // packet_loss_probability is >0.
+ // Default: uniform.
+ PacketLossMode packet_loss_mode;
+
+ // Packet loss probability. A value between 0.0 and 1.0 that defines the
+ // probability of a packet being lost. 0.1 means 10% and so on.
+ // Default: 0 (no loss).
+ double packet_loss_probability;
+
+ // Packet loss burst length. Defines how many packets will be lost in a burst
+ // when a packet has been decided to be lost. Must be >=1. Default: 1.
+ int packet_loss_burst_length;
+};
+
+// Class for simulating packet loss on the encoded frame data.
+// When a packet loss has occurred in a frame, the remaining data in that
+// frame is lost (even if burst length is only a single packet).
+// TODO(kjellander): Support discarding only individual packets in the frame
+// when CL 172001 has been submitted. This also requires a correct
+// fragmentation header to be passed to the decoder.
+//
+// To get a repeatable packet drop pattern, re-initialize the random seed
+// using InitializeRandomSeed before each test run.
+class PacketManipulator {
+ public:
+ virtual ~PacketManipulator() {}
+
+ // Manipulates the data of the encoded_image to simulate parts being lost
+ // during transport.
+ // If packets are dropped from frame data, the completedFrame field will be
+ // set to false.
+ // Returns the number of packets being dropped.
+ virtual int ManipulatePackets(webrtc::EncodedImage* encoded_image) = 0;
+};
+
+class PacketManipulatorImpl : public PacketManipulator {
+ public:
+ PacketManipulatorImpl(PacketReader* packet_reader,
+ const NetworkingConfig& config,
+ bool verbose);
+ virtual ~PacketManipulatorImpl();
+ int ManipulatePackets(webrtc::EncodedImage* encoded_image) override;
+ virtual void InitializeRandomSeed(unsigned int seed);
+ protected:
+ // Returns a uniformly distributed random value between 0.0 and 1.0
+ virtual double RandomUniform();
+ private:
+ PacketReader* packet_reader_;
+ const NetworkingConfig& config_;
+ // Used to simulate a burst over several frames.
+ int active_burst_packets_;
+ CriticalSectionWrapper* critsect_;
+ unsigned int random_seed_;
+ bool verbose_;
+};
+
+} // namespace test
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_PACKET_MANIPULATOR_H_
diff --git a/chromium/third_party/webrtc/modules/video_coding/codecs/test/packet_manipulator_unittest.cc b/chromium/third_party/webrtc/modules/video_coding/codecs/test/packet_manipulator_unittest.cc
new file mode 100644
index 00000000000..ace7bc05076
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/video_coding/codecs/test/packet_manipulator_unittest.cc
@@ -0,0 +1,153 @@
+/*
+ * Copyright (c) 2012 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.
+ */
+
+#include "webrtc/modules/video_coding/codecs/test/packet_manipulator.h"
+
+#include <queue>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
+#include "webrtc/modules/video_coding/codecs/test/predictive_packet_manipulator.h"
+#include "webrtc/test/testsupport/unittest_utils.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+namespace test {
+
+const double kNeverDropProbability = 0.0;
+const double kAlwaysDropProbability = 1.0;
+const int kBurstLength = 1;
+
+class PacketManipulatorTest: public PacketRelatedTest {
+ protected:
+ PacketReader packet_reader_;
+ EncodedImage image_;
+ NetworkingConfig drop_config_;
+ NetworkingConfig no_drop_config_;
+
+ PacketManipulatorTest() {
+ image_._buffer = packet_data_;
+ image_._length = kPacketDataLength;
+ image_._size = kPacketDataLength;
+
+ drop_config_.packet_size_in_bytes = kPacketSizeInBytes;
+ drop_config_.packet_loss_probability = kAlwaysDropProbability;
+ drop_config_.packet_loss_burst_length = kBurstLength;
+ drop_config_.packet_loss_mode = kUniform;
+
+ no_drop_config_.packet_size_in_bytes = kPacketSizeInBytes;
+ no_drop_config_.packet_loss_probability = kNeverDropProbability;
+ no_drop_config_.packet_loss_burst_length = kBurstLength;
+ no_drop_config_.packet_loss_mode = kUniform;
+ }
+
+ virtual ~PacketManipulatorTest() {}
+
+ void SetUp() {
+ PacketRelatedTest::SetUp();
+ }
+
+ void TearDown() {
+ PacketRelatedTest::TearDown();
+ }
+
+ void VerifyPacketLoss(int expected_nbr_packets_dropped,
+ int actual_nbr_packets_dropped,
+ size_t expected_packet_data_length,
+ uint8_t* expected_packet_data,
+ EncodedImage& actual_image) {
+ EXPECT_EQ(expected_nbr_packets_dropped, actual_nbr_packets_dropped);
+ EXPECT_EQ(expected_packet_data_length, image_._length);
+ EXPECT_EQ(0, memcmp(expected_packet_data, actual_image._buffer,
+ expected_packet_data_length));
+ }
+};
+
+TEST_F(PacketManipulatorTest, Constructor) {
+ PacketManipulatorImpl manipulator(&packet_reader_, no_drop_config_, false);
+}
+
+TEST_F(PacketManipulatorTest, DropNone) {
+ PacketManipulatorImpl manipulator(&packet_reader_, no_drop_config_, false);
+ int nbr_packets_dropped = manipulator.ManipulatePackets(&image_);
+ VerifyPacketLoss(0, nbr_packets_dropped, kPacketDataLength,
+ packet_data_, image_);
+}
+
+TEST_F(PacketManipulatorTest, UniformDropNoneSmallFrame) {
+ size_t data_length = 400; // smaller than the packet size
+ image_._length = data_length;
+ PacketManipulatorImpl manipulator(&packet_reader_, no_drop_config_, false);
+ int nbr_packets_dropped = manipulator.ManipulatePackets(&image_);
+
+ VerifyPacketLoss(0, nbr_packets_dropped, data_length,
+ packet_data_, image_);
+}
+
+TEST_F(PacketManipulatorTest, UniformDropAll) {
+ PacketManipulatorImpl manipulator(&packet_reader_, drop_config_, false);
+ int nbr_packets_dropped = manipulator.ManipulatePackets(&image_);
+ VerifyPacketLoss(kPacketDataNumberOfPackets, nbr_packets_dropped,
+ 0, packet_data_, image_);
+}
+
+// Use our customized test class to make the second packet being lost
+TEST_F(PacketManipulatorTest, UniformDropSinglePacket) {
+ drop_config_.packet_loss_probability = 0.5;
+ PredictivePacketManipulator manipulator(&packet_reader_, drop_config_);
+ manipulator.AddRandomResult(1.0);
+ manipulator.AddRandomResult(0.3); // less than 0.5 will cause packet loss
+ manipulator.AddRandomResult(1.0);
+
+ // Execute the test target method:
+ int nbr_packets_dropped = manipulator.ManipulatePackets(&image_);
+
+ // Since we setup the predictive packet manipulator, it will throw away the
+ // second packet. The third packet is also lost because when we have lost one,
+ // the remains shall also be discarded (in the current implementation).
+ VerifyPacketLoss(2, nbr_packets_dropped, kPacketSizeInBytes, packet1_,
+ image_);
+}
+
+// Use our customized test class to make the second packet being lost
+TEST_F(PacketManipulatorTest, BurstDropNinePackets) {
+ // Create a longer packet data structure (10 packets)
+ const int kNbrPackets = 10;
+ const size_t kDataLength = kPacketSizeInBytes * kNbrPackets;
+ uint8_t data[kDataLength];
+ uint8_t* data_pointer = data;
+ // Fill with 0s, 1s and so on to be able to easily verify which were dropped:
+ for (int i = 0; i < kNbrPackets; ++i) {
+ memset(data_pointer + i * kPacketSizeInBytes, i, kPacketSizeInBytes);
+ }
+ // Overwrite the defaults from the test fixture:
+ image_._buffer = data;
+ image_._length = kDataLength;
+ image_._size = kDataLength;
+
+ drop_config_.packet_loss_probability = 0.5;
+ drop_config_.packet_loss_burst_length = 5;
+ drop_config_.packet_loss_mode = kBurst;
+ PredictivePacketManipulator manipulator(&packet_reader_, drop_config_);
+ manipulator.AddRandomResult(1.0);
+ manipulator.AddRandomResult(0.3); // less than 0.5 will cause packet loss
+ for (int i = 0; i < kNbrPackets - 2; ++i) {
+ manipulator.AddRandomResult(1.0);
+ }
+
+ // Execute the test target method:
+ int nbr_packets_dropped = manipulator.ManipulatePackets(&image_);
+
+ // Should discard every packet after the first one.
+ VerifyPacketLoss(9, nbr_packets_dropped, kPacketSizeInBytes, data, image_);
+}
+
+} // namespace test
+} // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_coding/codecs/test/predictive_packet_manipulator.cc b/chromium/third_party/webrtc/modules/video_coding/codecs/test/predictive_packet_manipulator.cc
new file mode 100644
index 00000000000..c92cfa48a73
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/video_coding/codecs/test/predictive_packet_manipulator.cc
@@ -0,0 +1,48 @@
+/*
+ * Copyright (c) 2012 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.
+ */
+
+#include "webrtc/modules/video_coding/codecs/test/predictive_packet_manipulator.h"
+
+#include <assert.h>
+#include <stdio.h>
+
+#include "webrtc/test/testsupport/packet_reader.h"
+
+namespace webrtc {
+namespace test {
+
+PredictivePacketManipulator::PredictivePacketManipulator(
+ PacketReader* packet_reader, const NetworkingConfig& config)
+ : PacketManipulatorImpl(packet_reader, config, false) {
+}
+
+PredictivePacketManipulator::~PredictivePacketManipulator() {
+}
+
+
+void PredictivePacketManipulator::AddRandomResult(double result) {
+ assert(result >= 0.0 && result <= 1.0);
+ random_results_.push(result);
+}
+
+double PredictivePacketManipulator::RandomUniform() {
+ if(random_results_.size() == 0u) {
+ fprintf(stderr, "No more stored results, please make sure AddRandomResult()"
+ "is called same amount of times you're going to invoke the "
+ "RandomUniform() function, i.e. once per packet.\n");
+ assert(false);
+ }
+ double result = random_results_.front();
+ random_results_.pop();
+ return result;
+}
+
+} // namespace test
+} // namespace webrtcc
diff --git a/chromium/third_party/webrtc/modules/video_coding/codecs/test/predictive_packet_manipulator.h b/chromium/third_party/webrtc/modules/video_coding/codecs/test/predictive_packet_manipulator.h
new file mode 100644
index 00000000000..082712d870d
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/video_coding/codecs/test/predictive_packet_manipulator.h
@@ -0,0 +1,45 @@
+/*
+ * Copyright (c) 2012 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_MODULES_VIDEO_CODING_CODECS_TEST_PREDICTIVE_PACKET_MANIPULATOR_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_PREDICTIVE_PACKET_MANIPULATOR_H_
+
+#include <queue>
+
+#include "webrtc/modules/video_coding/codecs/test/packet_manipulator.h"
+#include "webrtc/test/testsupport/packet_reader.h"
+
+namespace webrtc {
+namespace test {
+
+// Predictive packet manipulator that allows for setup of the result of
+// the random invocations.
+class PredictivePacketManipulator : public PacketManipulatorImpl {
+ public:
+ PredictivePacketManipulator(PacketReader* packet_reader,
+ const NetworkingConfig& config);
+ virtual ~PredictivePacketManipulator();
+ // Adds a result. You must add at least the same number of results as the
+ // expected calls to the RandomUniform method. The results are added to a
+ // FIFO queue so they will be returned in the same order they were added.
+ // Result parameter must be 0.0 to 1.0.
+ void AddRandomResult(double result);
+ protected:
+ // Returns a uniformly distributed random value between 0.0 and 1.0
+ double RandomUniform() override;
+
+ private:
+ std::queue<double> random_results_;
+};
+
+} // namespace test
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_PREDICTIVE_PACKET_MANIPULATOR_H_
diff --git a/chromium/third_party/webrtc/modules/video_coding/codecs/test/stats.cc b/chromium/third_party/webrtc/modules/video_coding/codecs/test/stats.cc
new file mode 100644
index 00000000000..91a2f3c5f41
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/video_coding/codecs/test/stats.cc
@@ -0,0 +1,190 @@
+/*
+ * Copyright (c) 2012 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.
+ */
+
+#include "webrtc/modules/video_coding/codecs/test/stats.h"
+
+#include <assert.h>
+#include <stdio.h>
+
+#include <algorithm> // min_element, max_element
+
+#include "webrtc/base/format_macros.h"
+
+namespace webrtc {
+namespace test {
+
+FrameStatistic::FrameStatistic()
+ : encoding_successful(false),
+ decoding_successful(false),
+ encode_return_code(0),
+ decode_return_code(0),
+ encode_time_in_us(0),
+ decode_time_in_us(0),
+ frame_number(0),
+ packets_dropped(0),
+ total_packets(0),
+ bit_rate_in_kbps(0),
+ encoded_frame_length_in_bytes(0),
+ frame_type(kDeltaFrame) {}
+
+Stats::Stats() {}
+
+Stats::~Stats() {}
+
+bool LessForEncodeTime(const FrameStatistic& s1, const FrameStatistic& s2) {
+ return s1.encode_time_in_us < s2.encode_time_in_us;
+}
+
+bool LessForDecodeTime(const FrameStatistic& s1, const FrameStatistic& s2) {
+ return s1.decode_time_in_us < s2.decode_time_in_us;
+}
+
+bool LessForEncodedSize(const FrameStatistic& s1, const FrameStatistic& s2) {
+ return s1.encoded_frame_length_in_bytes < s2.encoded_frame_length_in_bytes;
+}
+
+bool LessForBitRate(const FrameStatistic& s1, const FrameStatistic& s2) {
+ return s1.bit_rate_in_kbps < s2.bit_rate_in_kbps;
+}
+
+FrameStatistic& Stats::NewFrame(int frame_number) {
+ assert(frame_number >= 0);
+ FrameStatistic stat;
+ stat.frame_number = frame_number;
+ stats_.push_back(stat);
+ return stats_[frame_number];
+}
+
+void Stats::PrintSummary() {
+ printf("Processing summary:\n");
+ if (stats_.size() == 0) {
+ printf("No frame statistics have been logged yet.\n");
+ return;
+ }
+
+ // Calculate min, max, average and total encoding time
+ int total_encoding_time_in_us = 0;
+ int total_decoding_time_in_us = 0;
+ size_t total_encoded_frames_lengths = 0;
+ size_t total_encoded_key_frames_lengths = 0;
+ size_t total_encoded_nonkey_frames_lengths = 0;
+ size_t nbr_keyframes = 0;
+ size_t nbr_nonkeyframes = 0;
+
+ for (FrameStatisticsIterator it = stats_.begin();
+ it != stats_.end(); ++it) {
+ total_encoding_time_in_us += it->encode_time_in_us;
+ total_decoding_time_in_us += it->decode_time_in_us;
+ total_encoded_frames_lengths += it->encoded_frame_length_in_bytes;
+ if (it->frame_type == webrtc::kKeyFrame) {
+ total_encoded_key_frames_lengths += it->encoded_frame_length_in_bytes;
+ nbr_keyframes++;
+ } else {
+ total_encoded_nonkey_frames_lengths += it->encoded_frame_length_in_bytes;
+ nbr_nonkeyframes++;
+ }
+ }
+
+ FrameStatisticsIterator frame;
+
+ // ENCODING
+ printf("Encoding time:\n");
+ frame = std::min_element(stats_.begin(),
+ stats_.end(), LessForEncodeTime);
+ printf(" Min : %7d us (frame %d)\n",
+ frame->encode_time_in_us, frame->frame_number);
+
+ frame = std::max_element(stats_.begin(),
+ stats_.end(), LessForEncodeTime);
+ printf(" Max : %7d us (frame %d)\n",
+ frame->encode_time_in_us, frame->frame_number);
+
+ printf(" Average : %7d us\n",
+ static_cast<int>(total_encoding_time_in_us / stats_.size()));
+
+ // DECODING
+ printf("Decoding time:\n");
+ // only consider frames that were successfully decoded (packet loss may cause
+ // failures)
+ std::vector<FrameStatistic> decoded_frames;
+ for (std::vector<FrameStatistic>::iterator it = stats_.begin();
+ it != stats_.end(); ++it) {
+ if (it->decoding_successful) {
+ decoded_frames.push_back(*it);
+ }
+ }
+ if (decoded_frames.size() == 0) {
+ printf("No successfully decoded frames exist in this statistics.\n");
+ } else {
+ frame = std::min_element(decoded_frames.begin(),
+ decoded_frames.end(), LessForDecodeTime);
+ printf(" Min : %7d us (frame %d)\n",
+ frame->decode_time_in_us, frame->frame_number);
+
+ frame = std::max_element(decoded_frames.begin(),
+ decoded_frames.end(), LessForDecodeTime);
+ printf(" Max : %7d us (frame %d)\n",
+ frame->decode_time_in_us, frame->frame_number);
+
+ printf(" Average : %7d us\n",
+ static_cast<int>(total_decoding_time_in_us / decoded_frames.size()));
+ printf(" Failures: %d frames failed to decode.\n",
+ static_cast<int>(stats_.size() - decoded_frames.size()));
+ }
+
+ // SIZE
+ printf("Frame sizes:\n");
+ frame = std::min_element(stats_.begin(),
+ stats_.end(), LessForEncodedSize);
+ printf(" Min : %7" PRIuS " bytes (frame %d)\n",
+ frame->encoded_frame_length_in_bytes, frame->frame_number);
+
+ frame = std::max_element(stats_.begin(),
+ stats_.end(), LessForEncodedSize);
+ printf(" Max : %7" PRIuS " bytes (frame %d)\n",
+ frame->encoded_frame_length_in_bytes, frame->frame_number);
+
+ printf(" Average : %7" PRIuS " bytes\n",
+ total_encoded_frames_lengths / stats_.size());
+ if (nbr_keyframes > 0) {
+ printf(" Average key frame size : %7" PRIuS " bytes (%" PRIuS
+ " keyframes)\n",
+ total_encoded_key_frames_lengths / nbr_keyframes, nbr_keyframes);
+ }
+ if (nbr_nonkeyframes > 0) {
+ printf(" Average non-key frame size: %7" PRIuS " bytes (%" PRIuS
+ " frames)\n",
+ total_encoded_nonkey_frames_lengths / nbr_nonkeyframes,
+ nbr_nonkeyframes);
+ }
+
+ // BIT RATE
+ printf("Bit rates:\n");
+ frame = std::min_element(stats_.begin(),
+ stats_.end(), LessForBitRate);
+ printf(" Min bit rate: %7d kbps (frame %d)\n",
+ frame->bit_rate_in_kbps, frame->frame_number);
+
+ frame = std::max_element(stats_.begin(),
+ stats_.end(), LessForBitRate);
+ printf(" Max bit rate: %7d kbps (frame %d)\n",
+ frame->bit_rate_in_kbps, frame->frame_number);
+
+ printf("\n");
+ printf("Total encoding time : %7d ms.\n",
+ total_encoding_time_in_us / 1000);
+ printf("Total decoding time : %7d ms.\n",
+ total_decoding_time_in_us / 1000);
+ printf("Total processing time: %7d ms.\n",
+ (total_encoding_time_in_us + total_decoding_time_in_us) / 1000);
+}
+
+} // namespace test
+} // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_coding/codecs/test/stats.h b/chromium/third_party/webrtc/modules/video_coding/codecs/test/stats.h
new file mode 100644
index 00000000000..8dc8f159fef
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/video_coding/codecs/test/stats.h
@@ -0,0 +1,70 @@
+/*
+ * Copyright (c) 2011 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_MODULES_VIDEO_CODING_CODECS_TEST_STATS_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_STATS_H_
+
+#include <vector>
+
+#include "webrtc/common_video/interface/video_image.h"
+
+namespace webrtc {
+namespace test {
+
+// Contains statistics of a single frame that has been processed.
+struct FrameStatistic {
+ FrameStatistic();
+
+ bool encoding_successful;
+ bool decoding_successful;
+ int encode_return_code;
+ int decode_return_code;
+ int encode_time_in_us;
+ int decode_time_in_us;
+ int frame_number;
+ // How many packets were discarded of the encoded frame data (if any).
+ int packets_dropped;
+ size_t total_packets;
+
+ // Current bit rate. Calculated out of the size divided with the time
+ // interval per frame.
+ int bit_rate_in_kbps;
+
+ // Copied from EncodedImage
+ size_t encoded_frame_length_in_bytes;
+ webrtc::VideoFrameType frame_type;
+};
+
+// Handles statistics from a single video processing run.
+// Contains calculation methods for interesting metrics from these stats.
+class Stats {
+ public:
+ typedef std::vector<FrameStatistic>::iterator FrameStatisticsIterator;
+
+ Stats();
+ virtual ~Stats();
+
+ // Add a new statistic data object.
+ // The frame number must be incrementing and start at zero in order to use
+ // it as an index for the frame_statistics_ vector.
+ // Returns the newly created statistic object.
+ FrameStatistic& NewFrame(int frame_number);
+
+ // Prints a summary of all the statistics that have been gathered during the
+ // processing
+ void PrintSummary();
+
+ std::vector<FrameStatistic> stats_;
+};
+
+} // namespace test
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_STATS_H_
diff --git a/chromium/third_party/webrtc/modules/video_coding/codecs/test/stats_unittest.cc b/chromium/third_party/webrtc/modules/video_coding/codecs/test/stats_unittest.cc
new file mode 100644
index 00000000000..a2d27e71d6c
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/video_coding/codecs/test/stats_unittest.cc
@@ -0,0 +1,64 @@
+/*
+ * Copyright (c) 2011 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.
+ */
+
+#include "webrtc/modules/video_coding/codecs/test/stats.h"
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+namespace test {
+
+class StatsTest: public testing::Test {
+ protected:
+ StatsTest() {
+ }
+
+ virtual ~StatsTest() {
+ }
+
+ void SetUp() {
+ stats_ = new Stats();
+ }
+
+ void TearDown() {
+ delete stats_;
+ }
+
+ Stats* stats_;
+};
+
+// Test empty object
+TEST_F(StatsTest, Uninitialized) {
+ EXPECT_EQ(0u, stats_->stats_.size());
+ stats_->PrintSummary(); // should not crash
+}
+
+// Add single frame stats and verify
+TEST_F(StatsTest, AddOne) {
+ stats_->NewFrame(0u);
+ FrameStatistic* frameStat = &stats_->stats_[0];
+ EXPECT_EQ(0, frameStat->frame_number);
+}
+
+// Add multiple frame stats and verify
+TEST_F(StatsTest, AddMany) {
+ int nbr_of_frames = 1000;
+ for (int i = 0; i < nbr_of_frames; ++i) {
+ FrameStatistic& frameStat = stats_->NewFrame(i);
+ EXPECT_EQ(i, frameStat.frame_number);
+ }
+ EXPECT_EQ(nbr_of_frames, static_cast<int>(stats_->stats_.size()));
+
+ stats_->PrintSummary(); // should not crash
+}
+
+} // namespace test
+} // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_coding/codecs/test/videoprocessor.cc b/chromium/third_party/webrtc/modules/video_coding/codecs/test/videoprocessor.cc
new file mode 100644
index 00000000000..10a3ff22598
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/video_coding/codecs/test/videoprocessor.cc
@@ -0,0 +1,415 @@
+/*
+ * Copyright (c) 2012 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.
+ */
+
+#include "webrtc/modules/video_coding/codecs/test/videoprocessor.h"
+
+#include <assert.h>
+#include <string.h>
+
+#include <limits>
+#include <vector>
+
+#include "webrtc/system_wrappers/interface/cpu_info.h"
+
+namespace webrtc {
+namespace test {
+
+TestConfig::TestConfig()
+ : name(""),
+ description(""),
+ test_number(0),
+ input_filename(""),
+ output_filename(""),
+ output_dir("out"),
+ networking_config(),
+ exclude_frame_types(kExcludeOnlyFirstKeyFrame),
+ frame_length_in_bytes(0),
+ use_single_core(false),
+ keyframe_interval(0),
+ codec_settings(NULL),
+ verbose(true) {}
+
+TestConfig::~TestConfig() {}
+
+VideoProcessorImpl::VideoProcessorImpl(webrtc::VideoEncoder* encoder,
+ webrtc::VideoDecoder* decoder,
+ FrameReader* frame_reader,
+ FrameWriter* frame_writer,
+ PacketManipulator* packet_manipulator,
+ const TestConfig& config,
+ Stats* stats)
+ : encoder_(encoder),
+ decoder_(decoder),
+ frame_reader_(frame_reader),
+ frame_writer_(frame_writer),
+ packet_manipulator_(packet_manipulator),
+ config_(config),
+ stats_(stats),
+ encode_callback_(NULL),
+ decode_callback_(NULL),
+ source_buffer_(NULL),
+ first_key_frame_has_been_excluded_(false),
+ last_frame_missing_(false),
+ initialized_(false),
+ encoded_frame_size_(0),
+ prev_time_stamp_(0),
+ num_dropped_frames_(0),
+ num_spatial_resizes_(0),
+ last_encoder_frame_width_(0),
+ last_encoder_frame_height_(0),
+ scaler_() {
+ assert(encoder);
+ assert(decoder);
+ assert(frame_reader);
+ assert(frame_writer);
+ assert(packet_manipulator);
+ assert(stats);
+}
+
+bool VideoProcessorImpl::Init() {
+ // Calculate a factor used for bit rate calculations:
+ bit_rate_factor_ = config_.codec_settings->maxFramerate * 0.001 * 8; // bits
+
+ // Initialize data structures used by the encoder/decoder APIs
+ size_t frame_length_in_bytes = frame_reader_->FrameLength();
+ source_buffer_ = new uint8_t[frame_length_in_bytes];
+ last_successful_frame_buffer_ = new uint8_t[frame_length_in_bytes];
+ // Set fixed properties common for all frames.
+ // To keep track of spatial resize actions by encoder.
+ last_encoder_frame_width_ = config_.codec_settings->width;
+ last_encoder_frame_height_ = config_.codec_settings->height;
+
+ // Setup required callbacks for the encoder/decoder:
+ encode_callback_ = new VideoProcessorEncodeCompleteCallback(this);
+ decode_callback_ = new VideoProcessorDecodeCompleteCallback(this);
+ int32_t register_result =
+ encoder_->RegisterEncodeCompleteCallback(encode_callback_);
+ if (register_result != WEBRTC_VIDEO_CODEC_OK) {
+ fprintf(stderr, "Failed to register encode complete callback, return code: "
+ "%d\n", register_result);
+ return false;
+ }
+ register_result = decoder_->RegisterDecodeCompleteCallback(decode_callback_);
+ if (register_result != WEBRTC_VIDEO_CODEC_OK) {
+ fprintf(stderr, "Failed to register decode complete callback, return code: "
+ "%d\n", register_result);
+ return false;
+ }
+ // Init the encoder and decoder
+ uint32_t nbr_of_cores = 1;
+ if (!config_.use_single_core) {
+ nbr_of_cores = CpuInfo::DetectNumberOfCores();
+ }
+ int32_t init_result =
+ encoder_->InitEncode(config_.codec_settings, nbr_of_cores,
+ config_.networking_config.max_payload_size_in_bytes);
+ if (init_result != WEBRTC_VIDEO_CODEC_OK) {
+ fprintf(stderr, "Failed to initialize VideoEncoder, return code: %d\n",
+ init_result);
+ return false;
+ }
+ init_result = decoder_->InitDecode(config_.codec_settings, nbr_of_cores);
+ if (init_result != WEBRTC_VIDEO_CODEC_OK) {
+ fprintf(stderr, "Failed to initialize VideoDecoder, return code: %d\n",
+ init_result);
+ return false;
+ }
+
+ if (config_.verbose) {
+ printf("Video Processor:\n");
+ printf(" #CPU cores used : %d\n", nbr_of_cores);
+ printf(" Total # of frames: %d\n", frame_reader_->NumberOfFrames());
+ printf(" Codec settings:\n");
+ printf(" Start bitrate : %d kbps\n",
+ config_.codec_settings->startBitrate);
+ printf(" Width : %d\n", config_.codec_settings->width);
+ printf(" Height : %d\n", config_.codec_settings->height);
+ }
+ initialized_ = true;
+ return true;
+}
+
+VideoProcessorImpl::~VideoProcessorImpl() {
+ delete[] source_buffer_;
+ delete[] last_successful_frame_buffer_;
+ encoder_->RegisterEncodeCompleteCallback(NULL);
+ delete encode_callback_;
+ decoder_->RegisterDecodeCompleteCallback(NULL);
+ delete decode_callback_;
+}
+
+
+void VideoProcessorImpl::SetRates(int bit_rate, int frame_rate) {
+ int set_rates_result = encoder_->SetRates(bit_rate, frame_rate);
+ assert(set_rates_result >= 0);
+ if (set_rates_result < 0) {
+ fprintf(stderr, "Failed to update encoder with new rate %d, "
+ "return code: %d\n", bit_rate, set_rates_result);
+ }
+ num_dropped_frames_ = 0;
+ num_spatial_resizes_ = 0;
+}
+
+size_t VideoProcessorImpl::EncodedFrameSize() {
+ return encoded_frame_size_;
+}
+
+int VideoProcessorImpl::NumberDroppedFrames() {
+ return num_dropped_frames_;
+}
+
+int VideoProcessorImpl::NumberSpatialResizes() {
+ return num_spatial_resizes_;
+}
+
+bool VideoProcessorImpl::ProcessFrame(int frame_number) {
+ assert(frame_number >=0);
+ if (!initialized_) {
+ fprintf(stderr, "Attempting to use uninitialized VideoProcessor!\n");
+ return false;
+ }
+ // |prev_time_stamp_| is used for getting number of dropped frames.
+ if (frame_number == 0) {
+ prev_time_stamp_ = -1;
+ }
+ if (frame_reader_->ReadFrame(source_buffer_)) {
+ // Copy the source frame to the newly read frame data.
+ source_frame_.CreateFrame(source_buffer_,
+ config_.codec_settings->width,
+ config_.codec_settings->height,
+ kVideoRotation_0);
+
+ // Ensure we have a new statistics data object we can fill:
+ FrameStatistic& stat = stats_->NewFrame(frame_number);
+
+ encode_start_ = TickTime::Now();
+ // Use the frame number as "timestamp" to identify frames
+ source_frame_.set_timestamp(frame_number);
+
+ // Decide if we're going to force a keyframe:
+ std::vector<VideoFrameType> frame_types(1, kDeltaFrame);
+ if (config_.keyframe_interval > 0 &&
+ frame_number % config_.keyframe_interval == 0) {
+ frame_types[0] = kKeyFrame;
+ }
+
+ // For dropped frames, we regard them as zero size encoded frames.
+ encoded_frame_size_ = 0;
+
+ int32_t encode_result = encoder_->Encode(source_frame_, NULL, &frame_types);
+
+ if (encode_result != WEBRTC_VIDEO_CODEC_OK) {
+ fprintf(stderr, "Failed to encode frame %d, return code: %d\n",
+ frame_number, encode_result);
+ }
+ stat.encode_return_code = encode_result;
+ return true;
+ } else {
+ return false; // we've reached the last frame
+ }
+}
+
+void VideoProcessorImpl::FrameEncoded(const EncodedImage& encoded_image) {
+ // Timestamp is frame number, so this gives us #dropped frames.
+ int num_dropped_from_prev_encode = encoded_image._timeStamp -
+ prev_time_stamp_ - 1;
+ num_dropped_frames_ += num_dropped_from_prev_encode;
+ prev_time_stamp_ = encoded_image._timeStamp;
+ if (num_dropped_from_prev_encode > 0) {
+ // For dropped frames, we write out the last decoded frame to avoid getting
+ // out of sync for the computation of PSNR and SSIM.
+ for (int i = 0; i < num_dropped_from_prev_encode; i++) {
+ frame_writer_->WriteFrame(last_successful_frame_buffer_);
+ }
+ }
+ // Frame is not dropped, so update the encoded frame size
+ // (encoder callback is only called for non-zero length frames).
+ encoded_frame_size_ = encoded_image._length;
+
+ TickTime encode_stop = TickTime::Now();
+ int frame_number = encoded_image._timeStamp;
+ FrameStatistic& stat = stats_->stats_[frame_number];
+ stat.encode_time_in_us = GetElapsedTimeMicroseconds(encode_start_,
+ encode_stop);
+ stat.encoding_successful = true;
+ stat.encoded_frame_length_in_bytes = encoded_image._length;
+ stat.frame_number = encoded_image._timeStamp;
+ stat.frame_type = encoded_image._frameType;
+ stat.bit_rate_in_kbps = encoded_image._length * bit_rate_factor_;
+ stat.total_packets = encoded_image._length /
+ config_.networking_config.packet_size_in_bytes + 1;
+
+ // Perform packet loss if criteria is fullfilled:
+ bool exclude_this_frame = false;
+ // Only keyframes can be excluded
+ if (encoded_image._frameType == kKeyFrame) {
+ switch (config_.exclude_frame_types) {
+ case kExcludeOnlyFirstKeyFrame:
+ if (!first_key_frame_has_been_excluded_) {
+ first_key_frame_has_been_excluded_ = true;
+ exclude_this_frame = true;
+ }
+ break;
+ case kExcludeAllKeyFrames:
+ exclude_this_frame = true;
+ break;
+ default:
+ assert(false);
+ }
+ }
+ rtc::scoped_ptr<uint8_t[]> copied_buffer(new uint8_t[encoded_image._length]);
+ memcpy(copied_buffer.get(), encoded_image._buffer, encoded_image._length);
+ EncodedImage copied_image;
+ memcpy(&copied_image, &encoded_image, sizeof(copied_image));
+ copied_image._size = copied_image._length;
+ copied_image._buffer = copied_buffer.get();
+ if (!exclude_this_frame) {
+ stat.packets_dropped =
+ packet_manipulator_->ManipulatePackets(&copied_image);
+ }
+
+ // Keep track of if frames are lost due to packet loss so we can tell
+ // this to the encoder (this is handled by the RTP logic in the full stack)
+ decode_start_ = TickTime::Now();
+ // TODO(kjellander): Pass fragmentation header to the decoder when
+ // CL 172001 has been submitted and PacketManipulator supports this.
+ int32_t decode_result =
+ decoder_->Decode(copied_image, last_frame_missing_, NULL);
+ stat.decode_return_code = decode_result;
+ if (decode_result != WEBRTC_VIDEO_CODEC_OK) {
+ // Write the last successful frame the output file to avoid getting it out
+ // of sync with the source file for SSIM and PSNR comparisons:
+ frame_writer_->WriteFrame(last_successful_frame_buffer_);
+ }
+ // save status for losses so we can inform the decoder for the next frame:
+ last_frame_missing_ = copied_image._length == 0;
+}
+
+void VideoProcessorImpl::FrameDecoded(const I420VideoFrame& image) {
+ TickTime decode_stop = TickTime::Now();
+ int frame_number = image.timestamp();
+ // Report stats
+ FrameStatistic& stat = stats_->stats_[frame_number];
+ stat.decode_time_in_us = GetElapsedTimeMicroseconds(decode_start_,
+ decode_stop);
+ stat.decoding_successful = true;
+
+ // Check for resize action (either down or up):
+ if (static_cast<int>(image.width()) != last_encoder_frame_width_ ||
+ static_cast<int>(image.height()) != last_encoder_frame_height_ ) {
+ ++num_spatial_resizes_;
+ last_encoder_frame_width_ = image.width();
+ last_encoder_frame_height_ = image.height();
+ }
+ // Check if codec size is different from native/original size, and if so,
+ // upsample back to original size: needed for PSNR and SSIM computations.
+ if (image.width() != config_.codec_settings->width ||
+ image.height() != config_.codec_settings->height) {
+ I420VideoFrame up_image;
+ int ret_val = scaler_.Set(image.width(), image.height(),
+ config_.codec_settings->width,
+ config_.codec_settings->height,
+ kI420, kI420, kScaleBilinear);
+ assert(ret_val >= 0);
+ if (ret_val < 0) {
+ fprintf(stderr, "Failed to set scalar for frame: %d, return code: %d\n",
+ frame_number, ret_val);
+ }
+ ret_val = scaler_.Scale(image, &up_image);
+ assert(ret_val >= 0);
+ if (ret_val < 0) {
+ fprintf(stderr, "Failed to scale frame: %d, return code: %d\n",
+ frame_number, ret_val);
+ }
+ // TODO(mikhal): Extracting the buffer for now - need to update test.
+ size_t length = CalcBufferSize(kI420, up_image.width(), up_image.height());
+ rtc::scoped_ptr<uint8_t[]> image_buffer(new uint8_t[length]);
+ int extracted_length = ExtractBuffer(up_image, length, image_buffer.get());
+ assert(extracted_length > 0);
+ // Update our copy of the last successful frame:
+ memcpy(last_successful_frame_buffer_, image_buffer.get(), extracted_length);
+ bool write_success = frame_writer_->WriteFrame(image_buffer.get());
+ assert(write_success);
+ if (!write_success) {
+ fprintf(stderr, "Failed to write frame %d to disk!", frame_number);
+ }
+ } else { // No resize.
+ // Update our copy of the last successful frame:
+ // TODO(mikhal): Add as a member function, so won't be allocated per frame.
+ size_t length = CalcBufferSize(kI420, image.width(), image.height());
+ rtc::scoped_ptr<uint8_t[]> image_buffer(new uint8_t[length]);
+ int extracted_length = ExtractBuffer(image, length, image_buffer.get());
+ assert(extracted_length > 0);
+ memcpy(last_successful_frame_buffer_, image_buffer.get(), extracted_length);
+
+ bool write_success = frame_writer_->WriteFrame(image_buffer.get());
+ assert(write_success);
+ if (!write_success) {
+ fprintf(stderr, "Failed to write frame %d to disk!", frame_number);
+ }
+ }
+}
+
+int VideoProcessorImpl::GetElapsedTimeMicroseconds(
+ const webrtc::TickTime& start, const webrtc::TickTime& stop) {
+ uint64_t encode_time = (stop - start).Microseconds();
+ assert(encode_time <
+ static_cast<unsigned int>(std::numeric_limits<int>::max()));
+ return static_cast<int>(encode_time);
+}
+
+const char* ExcludeFrameTypesToStr(ExcludeFrameTypes e) {
+ switch (e) {
+ case kExcludeOnlyFirstKeyFrame:
+ return "ExcludeOnlyFirstKeyFrame";
+ case kExcludeAllKeyFrames:
+ return "ExcludeAllKeyFrames";
+ default:
+ assert(false);
+ return "Unknown";
+ }
+}
+
+const char* VideoCodecTypeToStr(webrtc::VideoCodecType e) {
+ switch (e) {
+ case kVideoCodecVP8:
+ return "VP8";
+ case kVideoCodecI420:
+ return "I420";
+ case kVideoCodecRED:
+ return "RED";
+ case kVideoCodecULPFEC:
+ return "ULPFEC";
+ case kVideoCodecUnknown:
+ return "Unknown";
+ default:
+ assert(false);
+ return "Unknown";
+ }
+}
+
+// Callbacks
+int32_t
+VideoProcessorImpl::VideoProcessorEncodeCompleteCallback::Encoded(
+ const EncodedImage& encoded_image,
+ const webrtc::CodecSpecificInfo* codec_specific_info,
+ const webrtc::RTPFragmentationHeader* fragmentation) {
+ video_processor_->FrameEncoded(encoded_image); // Forward to parent class.
+ return 0;
+}
+int32_t
+VideoProcessorImpl::VideoProcessorDecodeCompleteCallback::Decoded(
+ I420VideoFrame& image) {
+ video_processor_->FrameDecoded(image); // forward to parent class
+ return 0;
+}
+
+} // namespace test
+} // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_coding/codecs/test/videoprocessor.h b/chromium/third_party/webrtc/modules/video_coding/codecs/test/videoprocessor.h
new file mode 100644
index 00000000000..cf55b75e61b
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/video_coding/codecs/test/videoprocessor.h
@@ -0,0 +1,254 @@
+/*
+ * Copyright (c) 2012 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_MODULES_VIDEO_CODING_CODECS_TEST_VIDEOPROCESSOR_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_VIDEOPROCESSOR_H_
+
+#include <string>
+
+#include "webrtc/common_video/libyuv/include/scaler.h"
+#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
+#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
+#include "webrtc/modules/video_coding/codecs/test/packet_manipulator.h"
+#include "webrtc/modules/video_coding/codecs/test/stats.h"
+#include "webrtc/system_wrappers/interface/tick_util.h"
+#include "webrtc/test/testsupport/frame_reader.h"
+#include "webrtc/test/testsupport/frame_writer.h"
+#include "webrtc/video_frame.h"
+
+namespace webrtc {
+namespace test {
+
+// Defines which frame types shall be excluded from packet loss and when.
+enum ExcludeFrameTypes {
+ // Will exclude the first keyframe in the video sequence from packet loss.
+ // Following keyframes will be targeted for packet loss.
+ kExcludeOnlyFirstKeyFrame,
+ // Exclude all keyframes from packet loss, no matter where in the video
+ // sequence they occur.
+ kExcludeAllKeyFrames
+};
+// Returns a string representation of the enum value.
+const char* ExcludeFrameTypesToStr(ExcludeFrameTypes e);
+
+// Test configuration for a test run
+struct TestConfig {
+ TestConfig();
+ ~TestConfig();
+
+ // Name of the test. This is purely metadata and does not affect
+ // the test in any way.
+ std::string name;
+
+ // More detailed description of the test. This is purely metadata and does
+ // not affect the test in any way.
+ std::string description;
+
+ // Number of this test. Useful if multiple runs of the same test with
+ // different configurations shall be managed.
+ int test_number;
+
+ // File to process for the test. This must be a video file in the YUV format.
+ std::string input_filename;
+
+ // File to write to during processing for the test. Will be a video file
+ // in the YUV format.
+ std::string output_filename;
+
+ // Path to the directory where encoded files will be put
+ // (absolute or relative to the executable). Default: "out".
+ std::string output_dir;
+
+ // Configurations related to networking.
+ NetworkingConfig networking_config;
+
+ // Decides how the packet loss simulations shall exclude certain frames
+ // from packet loss. Default: kExcludeOnlyFirstKeyFrame.
+ ExcludeFrameTypes exclude_frame_types;
+
+ // The length of a single frame of the input video file. This value is
+ // calculated out of the width and height according to the video format
+ // specification. Must be set before processing.
+ size_t frame_length_in_bytes;
+
+ // Force the encoder and decoder to use a single core for processing.
+ // Using a single core is necessary to get a deterministic behavior for the
+ // encoded frames - using multiple cores will produce different encoded frames
+ // since multiple cores are competing to consume the byte budget for each
+ // frame in parallel.
+ // If set to false, the maximum number of available cores will be used.
+ // Default: false.
+ bool use_single_core;
+
+ // If set to a value >0 this setting forces the encoder to create a keyframe
+ // every Nth frame. Note that the encoder may create a keyframe in other
+ // locations in addition to the interval that is set using this parameter.
+ // Forcing key frames may also affect encoder planning optimizations in
+ // a negative way, since it will suddenly be forced to produce an expensive
+ // key frame.
+ // Default: 0.
+ int keyframe_interval;
+
+ // The codec settings to use for the test (target bitrate, video size,
+ // framerate and so on). This struct must be created and filled in using
+ // the VideoCodingModule::Codec() method.
+ webrtc::VideoCodec* codec_settings;
+
+ // If printing of information to stdout shall be performed during processing.
+ bool verbose;
+};
+
+// Returns a string representation of the enum value.
+const char* VideoCodecTypeToStr(webrtc::VideoCodecType e);
+
+// Handles encoding/decoding of video using the VideoEncoder/VideoDecoder
+// interfaces. This is done in a sequential manner in order to be able to
+// measure times properly.
+// The class processes a frame at the time for the configured input file.
+// It maintains state of where in the source input file the processing is at.
+//
+// Regarding packet loss: Note that keyframes are excluded (first or all
+// depending on the ExcludeFrameTypes setting). This is because if key frames
+// would be altered, all the following delta frames would be pretty much
+// worthless. VP8 has an error-resilience feature that makes it able to handle
+// packet loss in key non-first keyframes, which is why only the first is
+// excluded by default.
+// Packet loss in such important frames is handled on a higher level in the
+// Video Engine, where signaling would request a retransmit of the lost packets,
+// since they're so important.
+//
+// Note this class is not thread safe in any way and is meant for simple testing
+// purposes.
+class VideoProcessor {
+ public:
+ virtual ~VideoProcessor() {}
+
+ // Performs initial calculations about frame size, sets up callbacks etc.
+ // Returns false if an error has occurred, in addition to printing to stderr.
+ virtual bool Init() = 0;
+
+ // Processes a single frame. Returns true as long as there's more frames
+ // available in the source clip.
+ // Frame number must be an integer >=0.
+ virtual bool ProcessFrame(int frame_number) = 0;
+
+ // Updates the encoder with the target bit rate and the frame rate.
+ virtual void SetRates(int bit_rate, int frame_rate) = 0;
+
+ // Return the size of the encoded frame in bytes. Dropped frames by the
+ // encoder are regarded as zero size.
+ virtual size_t EncodedFrameSize() = 0;
+
+ // Return the number of dropped frames.
+ virtual int NumberDroppedFrames() = 0;
+
+ // Return the number of spatial resizes.
+ virtual int NumberSpatialResizes() = 0;
+};
+
+class VideoProcessorImpl : public VideoProcessor {
+ public:
+ VideoProcessorImpl(webrtc::VideoEncoder* encoder,
+ webrtc::VideoDecoder* decoder,
+ FrameReader* frame_reader,
+ FrameWriter* frame_writer,
+ PacketManipulator* packet_manipulator,
+ const TestConfig& config,
+ Stats* stats);
+ virtual ~VideoProcessorImpl();
+ bool Init() override;
+ bool ProcessFrame(int frame_number) override;
+
+ private:
+ // Invoked by the callback when a frame has completed encoding.
+ void FrameEncoded(const webrtc::EncodedImage& encodedImage);
+ // Invoked by the callback when a frame has completed decoding.
+ void FrameDecoded(const webrtc::I420VideoFrame& image);
+ // Used for getting a 32-bit integer representing time
+ // (checks the size is within signed 32-bit bounds before casting it)
+ int GetElapsedTimeMicroseconds(const webrtc::TickTime& start,
+ const webrtc::TickTime& stop);
+ // Updates the encoder with the target bit rate and the frame rate.
+ void SetRates(int bit_rate, int frame_rate) override;
+ // Return the size of the encoded frame in bytes.
+ size_t EncodedFrameSize() override;
+ // Return the number of dropped frames.
+ int NumberDroppedFrames() override;
+ // Return the number of spatial resizes.
+ int NumberSpatialResizes() override;
+
+ webrtc::VideoEncoder* encoder_;
+ webrtc::VideoDecoder* decoder_;
+ FrameReader* frame_reader_;
+ FrameWriter* frame_writer_;
+ PacketManipulator* packet_manipulator_;
+ const TestConfig& config_;
+ Stats* stats_;
+
+ EncodedImageCallback* encode_callback_;
+ DecodedImageCallback* decode_callback_;
+ // Buffer used for reading the source video file:
+ uint8_t* source_buffer_;
+ // Keep track of the last successful frame, since we need to write that
+ // when decoding fails:
+ uint8_t* last_successful_frame_buffer_;
+ webrtc::I420VideoFrame source_frame_;
+ // To keep track of if we have excluded the first key frame from packet loss:
+ bool first_key_frame_has_been_excluded_;
+ // To tell the decoder previous frame have been dropped due to packet loss:
+ bool last_frame_missing_;
+ // If Init() has executed successfully.
+ bool initialized_;
+ size_t encoded_frame_size_;
+ int prev_time_stamp_;
+ int num_dropped_frames_;
+ int num_spatial_resizes_;
+ int last_encoder_frame_width_;
+ int last_encoder_frame_height_;
+ Scaler scaler_;
+
+ // Statistics
+ double bit_rate_factor_; // multiply frame length with this to get bit rate
+ webrtc::TickTime encode_start_;
+ webrtc::TickTime decode_start_;
+
+ // Callback class required to implement according to the VideoEncoder API.
+ class VideoProcessorEncodeCompleteCallback
+ : public webrtc::EncodedImageCallback {
+ public:
+ explicit VideoProcessorEncodeCompleteCallback(VideoProcessorImpl* vp)
+ : video_processor_(vp) {}
+ int32_t Encoded(
+ const webrtc::EncodedImage& encoded_image,
+ const webrtc::CodecSpecificInfo* codec_specific_info,
+ const webrtc::RTPFragmentationHeader* fragmentation) override;
+
+ private:
+ VideoProcessorImpl* video_processor_;
+ };
+
+ // Callback class required to implement according to the VideoDecoder API.
+ class VideoProcessorDecodeCompleteCallback
+ : public webrtc::DecodedImageCallback {
+ public:
+ explicit VideoProcessorDecodeCompleteCallback(VideoProcessorImpl* vp)
+ : video_processor_(vp) {
+ }
+ int32_t Decoded(webrtc::I420VideoFrame& image) override;
+
+ private:
+ VideoProcessorImpl* video_processor_;
+ };
+};
+
+} // namespace test
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_VIDEOPROCESSOR_H_
diff --git a/chromium/third_party/webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest.cc b/chromium/third_party/webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest.cc
new file mode 100644
index 00000000000..6c0e1254fca
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest.cc
@@ -0,0 +1,936 @@
+/*
+ * Copyright (c) 2012 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.
+ */
+
+#include <math.h>
+
+#include "testing/gtest/include/gtest/gtest.h"
+
+#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
+#include "webrtc/modules/video_coding/codecs/test/packet_manipulator.h"
+#include "webrtc/modules/video_coding/codecs/test/videoprocessor.h"
+#include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
+#include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h"
+#include "webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.h"
+#include "webrtc/modules/video_coding/main/interface/video_coding.h"
+#include "webrtc/test/testsupport/fileutils.h"
+#include "webrtc/test/testsupport/frame_reader.h"
+#include "webrtc/test/testsupport/frame_writer.h"
+#include "webrtc/test/testsupport/gtest_disable.h"
+#include "webrtc/test/testsupport/metrics/video_metrics.h"
+#include "webrtc/test/testsupport/packet_reader.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+// Maximum number of rate updates (i.e., calls to encoder to change bitrate
+// and/or frame rate) for the current tests.
+const int kMaxNumRateUpdates = 3;
+
+const int kPercTargetvsActualMismatch = 20;
+const int kBaseKeyFrameInterval = 3000;
+
+// Codec and network settings.
+struct CodecConfigPars {
+ VideoCodecType codec_type;
+ float packet_loss;
+ int num_temporal_layers;
+ int key_frame_interval;
+ bool error_concealment_on;
+ bool denoising_on;
+ bool frame_dropper_on;
+ bool spatial_resize_on;
+};
+
+// Quality metrics.
+struct QualityMetrics {
+ double minimum_avg_psnr;
+ double minimum_min_psnr;
+ double minimum_avg_ssim;
+ double minimum_min_ssim;
+};
+
+// The sequence of bitrate and frame rate changes for the encoder, the frame
+// number where the changes are made, and the total number of frames for the
+// test.
+struct RateProfile {
+ int target_bit_rate[kMaxNumRateUpdates];
+ int input_frame_rate[kMaxNumRateUpdates];
+ int frame_index_rate_update[kMaxNumRateUpdates + 1];
+ int num_frames;
+};
+
+// Metrics for the rate control. The rate mismatch metrics are defined as
+// percentages.|max_time_hit_target| is defined as number of frames, after a
+// rate update is made to the encoder, for the encoder to reach within
+// |kPercTargetvsActualMismatch| of new target rate. The metrics are defined for
+// each rate update sequence.
+struct RateControlMetrics {
+ int max_num_dropped_frames;
+ int max_key_frame_size_mismatch;
+ int max_delta_frame_size_mismatch;
+ int max_encoding_rate_mismatch;
+ int max_time_hit_target;
+ int num_spatial_resizes;
+};
+
+
+// Sequence used is foreman (CIF): may be better to use VGA for resize test.
+const int kCIFWidth = 352;
+const int kCIFHeight = 288;
+const int kNbrFramesShort = 100; // Some tests are run for shorter sequence.
+const int kNbrFramesLong = 299;
+
+// Parameters from VP8 wrapper, which control target size of key frames.
+const float kInitialBufferSize = 0.5f;
+const float kOptimalBufferSize = 0.6f;
+const float kScaleKeyFrameSize = 0.5f;
+
+// Integration test for video processor. Encodes+decodes a clip and
+// writes it to the output directory. After completion, quality metrics
+// (PSNR and SSIM) and rate control metrics are computed to verify that the
+// quality and encoder response is acceptable. The rate control tests allow us
+// to verify the behavior for changing bitrate, changing frame rate, frame
+// dropping/spatial resize, and temporal layers. The limits for the rate
+// control metrics are set to be fairly conservative, so failure should only
+// happen when some significant regression or breakdown occurs.
+class VideoProcessorIntegrationTest: public testing::Test {
+ protected:
+ VideoEncoder* encoder_;
+ VideoDecoder* decoder_;
+ webrtc::test::FrameReader* frame_reader_;
+ webrtc::test::FrameWriter* frame_writer_;
+ webrtc::test::PacketReader packet_reader_;
+ webrtc::test::PacketManipulator* packet_manipulator_;
+ webrtc::test::Stats stats_;
+ webrtc::test::TestConfig config_;
+ VideoCodec codec_settings_;
+ webrtc::test::VideoProcessor* processor_;
+
+ // Quantities defined/updated for every encoder rate update.
+ // Some quantities defined per temporal layer (at most 3 layers in this test).
+ int num_frames_per_update_[3];
+ float sum_frame_size_mismatch_[3];
+ float sum_encoded_frame_size_[3];
+ float encoding_bitrate_[3];
+ float per_frame_bandwidth_[3];
+ float bit_rate_layer_[3];
+ float frame_rate_layer_[3];
+ int num_frames_total_;
+ float sum_encoded_frame_size_total_;
+ float encoding_bitrate_total_;
+ float perc_encoding_rate_mismatch_;
+ int num_frames_to_hit_target_;
+ bool encoding_rate_within_target_;
+ int bit_rate_;
+ int frame_rate_;
+ int layer_;
+ float target_size_key_frame_initial_;
+ float target_size_key_frame_;
+ float sum_key_frame_size_mismatch_;
+ int num_key_frames_;
+ float start_bitrate_;
+
+ // Codec and network settings.
+ VideoCodecType codec_type_;
+ float packet_loss_;
+ int num_temporal_layers_;
+ int key_frame_interval_;
+ bool error_concealment_on_;
+ bool denoising_on_;
+ bool frame_dropper_on_;
+ bool spatial_resize_on_;
+
+
+ VideoProcessorIntegrationTest() {}
+ virtual ~VideoProcessorIntegrationTest() {}
+
+ void SetUpCodecConfig() {
+ if (codec_type_ == kVideoCodecVP8) {
+ encoder_ = VP8Encoder::Create();
+ decoder_ = VP8Decoder::Create();
+ VideoCodingModule::Codec(kVideoCodecVP8, &codec_settings_);
+ } else if (codec_type_ == kVideoCodecVP9) {
+ encoder_ = VP9Encoder::Create();
+ decoder_ = VP9Decoder::Create();
+ VideoCodingModule::Codec(kVideoCodecVP9, &codec_settings_);
+ }
+
+ // CIF is currently used for all tests below.
+ // Setup the TestConfig struct for processing of a clip in CIF resolution.
+ config_.input_filename =
+ webrtc::test::ResourcePath("foreman_cif", "yuv");
+
+ // Generate an output filename in a safe way.
+ config_.output_filename = webrtc::test::TempFilename(
+ webrtc::test::OutputPath(), "videoprocessor_integrationtest");
+ config_.frame_length_in_bytes = CalcBufferSize(kI420,
+ kCIFWidth, kCIFHeight);
+ config_.verbose = false;
+ // Only allow encoder/decoder to use single core, for predictability.
+ config_.use_single_core = true;
+ // Key frame interval and packet loss are set for each test.
+ config_.keyframe_interval = key_frame_interval_;
+ config_.networking_config.packet_loss_probability = packet_loss_;
+
+ // Configure codec settings.
+ config_.codec_settings = &codec_settings_;
+ config_.codec_settings->startBitrate = start_bitrate_;
+ config_.codec_settings->width = kCIFWidth;
+ config_.codec_settings->height = kCIFHeight;
+
+ // These features may be set depending on the test.
+ switch (config_.codec_settings->codecType) {
+ case kVideoCodecVP8:
+ config_.codec_settings->codecSpecific.VP8.errorConcealmentOn =
+ error_concealment_on_;
+ config_.codec_settings->codecSpecific.VP8.denoisingOn =
+ denoising_on_;
+ config_.codec_settings->codecSpecific.VP8.numberOfTemporalLayers =
+ num_temporal_layers_;
+ config_.codec_settings->codecSpecific.VP8.frameDroppingOn =
+ frame_dropper_on_;
+ config_.codec_settings->codecSpecific.VP8.automaticResizeOn =
+ spatial_resize_on_;
+ config_.codec_settings->codecSpecific.VP8.keyFrameInterval =
+ kBaseKeyFrameInterval;
+ break;
+ case kVideoCodecVP9:
+ config_.codec_settings->codecSpecific.VP9.denoisingOn =
+ denoising_on_;
+ config_.codec_settings->codecSpecific.VP9.numberOfTemporalLayers =
+ num_temporal_layers_;
+ config_.codec_settings->codecSpecific.VP9.frameDroppingOn =
+ frame_dropper_on_;
+ config_.codec_settings->codecSpecific.VP9.keyFrameInterval =
+ kBaseKeyFrameInterval;
+ break;
+ default:
+ assert(false);
+ break;
+ }
+ frame_reader_ =
+ new webrtc::test::FrameReaderImpl(config_.input_filename,
+ config_.frame_length_in_bytes);
+ frame_writer_ =
+ new webrtc::test::FrameWriterImpl(config_.output_filename,
+ config_.frame_length_in_bytes);
+ ASSERT_TRUE(frame_reader_->Init());
+ ASSERT_TRUE(frame_writer_->Init());
+
+ packet_manipulator_ = new webrtc::test::PacketManipulatorImpl(
+ &packet_reader_, config_.networking_config, config_.verbose);
+ processor_ = new webrtc::test::VideoProcessorImpl(encoder_, decoder_,
+ frame_reader_,
+ frame_writer_,
+ packet_manipulator_,
+ config_, &stats_);
+ ASSERT_TRUE(processor_->Init());
+ }
+
+ // Reset quantities after each encoder update, update the target
+ // per-frame bandwidth.
+ void ResetRateControlMetrics(int num_frames) {
+ for (int i = 0; i < num_temporal_layers_; i++) {
+ num_frames_per_update_[i] = 0;
+ sum_frame_size_mismatch_[i] = 0.0f;
+ sum_encoded_frame_size_[i] = 0.0f;
+ encoding_bitrate_[i] = 0.0f;
+ // Update layer per-frame-bandwidth.
+ per_frame_bandwidth_[i] = static_cast<float>(bit_rate_layer_[i]) /
+ static_cast<float>(frame_rate_layer_[i]);
+ }
+ // Set maximum size of key frames, following setting in the VP8 wrapper.
+ float max_key_size = kScaleKeyFrameSize * kOptimalBufferSize * frame_rate_;
+ // We don't know exact target size of the key frames (except for first one),
+ // but the minimum in libvpx is ~|3 * per_frame_bandwidth| and maximum is
+ // set by |max_key_size_ * per_frame_bandwidth|. Take middle point/average
+ // as reference for mismatch. Note key frames always correspond to base
+ // layer frame in this test.
+ target_size_key_frame_ = 0.5 * (3 + max_key_size) * per_frame_bandwidth_[0];
+ num_frames_total_ = 0;
+ sum_encoded_frame_size_total_ = 0.0f;
+ encoding_bitrate_total_ = 0.0f;
+ perc_encoding_rate_mismatch_ = 0.0f;
+ num_frames_to_hit_target_ = num_frames;
+ encoding_rate_within_target_ = false;
+ sum_key_frame_size_mismatch_ = 0.0;
+ num_key_frames_ = 0;
+ }
+
+ // For every encoded frame, update the rate control metrics.
+ void UpdateRateControlMetrics(int frame_num, VideoFrameType frame_type) {
+ float encoded_size_kbits = processor_->EncodedFrameSize() * 8.0f / 1000.0f;
+ // Update layer data.
+ // Update rate mismatch relative to per-frame bandwidth for delta frames.
+ if (frame_type == kDeltaFrame) {
+ // TODO(marpan): Should we count dropped (zero size) frames in mismatch?
+ sum_frame_size_mismatch_[layer_] += fabs(encoded_size_kbits -
+ per_frame_bandwidth_[layer_]) /
+ per_frame_bandwidth_[layer_];
+ } else {
+ float target_size = (frame_num == 1) ? target_size_key_frame_initial_ :
+ target_size_key_frame_;
+ sum_key_frame_size_mismatch_ += fabs(encoded_size_kbits - target_size) /
+ target_size;
+ num_key_frames_ += 1;
+ }
+ sum_encoded_frame_size_[layer_] += encoded_size_kbits;
+ // Encoding bitrate per layer: from the start of the update/run to the
+ // current frame.
+ encoding_bitrate_[layer_] = sum_encoded_frame_size_[layer_] *
+ frame_rate_layer_[layer_] /
+ num_frames_per_update_[layer_];
+ // Total encoding rate: from the start of the update/run to current frame.
+ sum_encoded_frame_size_total_ += encoded_size_kbits;
+ encoding_bitrate_total_ = sum_encoded_frame_size_total_ * frame_rate_ /
+ num_frames_total_;
+ perc_encoding_rate_mismatch_ = 100 * fabs(encoding_bitrate_total_ -
+ bit_rate_) / bit_rate_;
+ if (perc_encoding_rate_mismatch_ < kPercTargetvsActualMismatch &&
+ !encoding_rate_within_target_) {
+ num_frames_to_hit_target_ = num_frames_total_;
+ encoding_rate_within_target_ = true;
+ }
+ }
+
+ // Verify expected behavior of rate control and print out data.
+ void VerifyRateControl(int update_index,
+ int max_key_frame_size_mismatch,
+ int max_delta_frame_size_mismatch,
+ int max_encoding_rate_mismatch,
+ int max_time_hit_target,
+ int max_num_dropped_frames,
+ int num_spatial_resizes) {
+ int num_dropped_frames = processor_->NumberDroppedFrames();
+ int num_resize_actions = processor_->NumberSpatialResizes();
+ printf("For update #: %d,\n "
+ " Target Bitrate: %d,\n"
+ " Encoding bitrate: %f,\n"
+ " Frame rate: %d \n",
+ update_index, bit_rate_, encoding_bitrate_total_, frame_rate_);
+ printf(" Number of frames to approach target rate = %d, \n"
+ " Number of dropped frames = %d, \n"
+ " Number of spatial resizes = %d, \n",
+ num_frames_to_hit_target_, num_dropped_frames, num_resize_actions);
+ EXPECT_LE(perc_encoding_rate_mismatch_, max_encoding_rate_mismatch);
+ if (num_key_frames_ > 0) {
+ int perc_key_frame_size_mismatch = 100 * sum_key_frame_size_mismatch_ /
+ num_key_frames_;
+ printf(" Number of Key frames: %d \n"
+ " Key frame rate mismatch: %d \n",
+ num_key_frames_, perc_key_frame_size_mismatch);
+ EXPECT_LE(perc_key_frame_size_mismatch, max_key_frame_size_mismatch);
+ }
+ printf("\n");
+ printf("Rates statistics for Layer data \n");
+ for (int i = 0; i < num_temporal_layers_ ; i++) {
+ printf("Layer #%d \n", i);
+ int perc_frame_size_mismatch = 100 * sum_frame_size_mismatch_[i] /
+ num_frames_per_update_[i];
+ int perc_encoding_rate_mismatch = 100 * fabs(encoding_bitrate_[i] -
+ bit_rate_layer_[i]) /
+ bit_rate_layer_[i];
+ printf(" Target Layer Bit rate: %f \n"
+ " Layer frame rate: %f, \n"
+ " Layer per frame bandwidth: %f, \n"
+ " Layer Encoding bit rate: %f, \n"
+ " Layer Percent frame size mismatch: %d, \n"
+ " Layer Percent encoding rate mismatch = %d, \n"
+ " Number of frame processed per layer = %d \n",
+ bit_rate_layer_[i], frame_rate_layer_[i], per_frame_bandwidth_[i],
+ encoding_bitrate_[i], perc_frame_size_mismatch,
+ perc_encoding_rate_mismatch, num_frames_per_update_[i]);
+ EXPECT_LE(perc_frame_size_mismatch, max_delta_frame_size_mismatch);
+ EXPECT_LE(perc_encoding_rate_mismatch, max_encoding_rate_mismatch);
+ }
+ printf("\n");
+ EXPECT_LE(num_frames_to_hit_target_, max_time_hit_target);
+ EXPECT_LE(num_dropped_frames, max_num_dropped_frames);
+ EXPECT_EQ(num_resize_actions, num_spatial_resizes);
+ }
+
+ // Layer index corresponding to frame number, for up to 3 layers.
+ void LayerIndexForFrame(int frame_number) {
+ if (num_temporal_layers_ == 1) {
+ layer_ = 0;
+ } else if (num_temporal_layers_ == 2) {
+ // layer 0: 0 2 4 ...
+ // layer 1: 1 3
+ if (frame_number % 2 == 0) {
+ layer_ = 0;
+ } else {
+ layer_ = 1;
+ }
+ } else if (num_temporal_layers_ == 3) {
+ // layer 0: 0 4 8 ...
+ // layer 1: 2 6
+ // layer 2: 1 3 5 7
+ if (frame_number % 4 == 0) {
+ layer_ = 0;
+ } else if ((frame_number + 2) % 4 == 0) {
+ layer_ = 1;
+ } else if ((frame_number + 1) % 2 == 0) {
+ layer_ = 2;
+ }
+ } else {
+ assert(false); // Only up to 3 layers.
+ }
+ }
+
+ // Set the bitrate and frame rate per layer, for up to 3 layers.
+ void SetLayerRates() {
+ assert(num_temporal_layers_<= 3);
+ for (int i = 0; i < num_temporal_layers_; i++) {
+ float bit_rate_ratio =
+ kVp8LayerRateAlloction[num_temporal_layers_ - 1][i];
+ if (i > 0) {
+ float bit_rate_delta_ratio = kVp8LayerRateAlloction
+ [num_temporal_layers_ - 1][i] -
+ kVp8LayerRateAlloction[num_temporal_layers_ - 1][i - 1];
+ bit_rate_layer_[i] = bit_rate_ * bit_rate_delta_ratio;
+ } else {
+ bit_rate_layer_[i] = bit_rate_ * bit_rate_ratio;
+ }
+ frame_rate_layer_[i] = frame_rate_ / static_cast<float>(
+ 1 << (num_temporal_layers_ - 1));
+ }
+ if (num_temporal_layers_ == 3) {
+ frame_rate_layer_[2] = frame_rate_ / 2.0f;
+ }
+ }
+
+ VideoFrameType FrameType(int frame_number) {
+ if (frame_number == 0 || ((frame_number) % key_frame_interval_ == 0 &&
+ key_frame_interval_ > 0)) {
+ return kKeyFrame;
+ } else {
+ return kDeltaFrame;
+ }
+ }
+
+ void TearDown() {
+ delete processor_;
+ delete packet_manipulator_;
+ delete frame_writer_;
+ delete frame_reader_;
+ delete decoder_;
+ delete encoder_;
+ }
+
+ // Processes all frames in the clip and verifies the result.
+ void ProcessFramesAndVerify(QualityMetrics quality_metrics,
+ RateProfile rate_profile,
+ CodecConfigPars process,
+ RateControlMetrics* rc_metrics) {
+ // Codec/config settings.
+ codec_type_ = process.codec_type;
+ start_bitrate_ = rate_profile.target_bit_rate[0];
+ packet_loss_ = process.packet_loss;
+ key_frame_interval_ = process.key_frame_interval;
+ num_temporal_layers_ = process.num_temporal_layers;
+ error_concealment_on_ = process.error_concealment_on;
+ denoising_on_ = process.denoising_on;
+ frame_dropper_on_ = process.frame_dropper_on;
+ spatial_resize_on_ = process.spatial_resize_on;
+ SetUpCodecConfig();
+ // Update the layers and the codec with the initial rates.
+ bit_rate_ = rate_profile.target_bit_rate[0];
+ frame_rate_ = rate_profile.input_frame_rate[0];
+ SetLayerRates();
+ // Set the initial target size for key frame.
+ target_size_key_frame_initial_ = 0.5 * kInitialBufferSize *
+ bit_rate_layer_[0];
+ processor_->SetRates(bit_rate_, frame_rate_);
+ // Process each frame, up to |num_frames|.
+ int num_frames = rate_profile.num_frames;
+ int update_index = 0;
+ ResetRateControlMetrics(
+ rate_profile.frame_index_rate_update[update_index + 1]);
+ int frame_number = 0;
+ VideoFrameType frame_type = kDeltaFrame;
+ while (processor_->ProcessFrame(frame_number) &&
+ frame_number < num_frames) {
+ // Get the layer index for the frame |frame_number|.
+ LayerIndexForFrame(frame_number);
+ frame_type = FrameType(frame_number);
+ // Counter for whole sequence run.
+ ++frame_number;
+ // Counters for each rate update.
+ ++num_frames_per_update_[layer_];
+ ++num_frames_total_;
+ UpdateRateControlMetrics(frame_number, frame_type);
+ // If we hit another/next update, verify stats for current state and
+ // update layers and codec with new rates.
+ if (frame_number ==
+ rate_profile.frame_index_rate_update[update_index + 1]) {
+ VerifyRateControl(
+ update_index,
+ rc_metrics[update_index].max_key_frame_size_mismatch,
+ rc_metrics[update_index].max_delta_frame_size_mismatch,
+ rc_metrics[update_index].max_encoding_rate_mismatch,
+ rc_metrics[update_index].max_time_hit_target,
+ rc_metrics[update_index].max_num_dropped_frames,
+ rc_metrics[update_index].num_spatial_resizes);
+ // Update layer rates and the codec with new rates.
+ ++update_index;
+ bit_rate_ = rate_profile.target_bit_rate[update_index];
+ frame_rate_ = rate_profile.input_frame_rate[update_index];
+ SetLayerRates();
+ ResetRateControlMetrics(rate_profile.
+ frame_index_rate_update[update_index + 1]);
+ processor_->SetRates(bit_rate_, frame_rate_);
+ }
+ }
+ VerifyRateControl(
+ update_index,
+ rc_metrics[update_index].max_key_frame_size_mismatch,
+ rc_metrics[update_index].max_delta_frame_size_mismatch,
+ rc_metrics[update_index].max_encoding_rate_mismatch,
+ rc_metrics[update_index].max_time_hit_target,
+ rc_metrics[update_index].max_num_dropped_frames,
+ rc_metrics[update_index].num_spatial_resizes);
+ EXPECT_EQ(num_frames, frame_number);
+ EXPECT_EQ(num_frames + 1, static_cast<int>(stats_.stats_.size()));
+
+ // Release encoder and decoder to make sure they have finished processing:
+ EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Release());
+ EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, decoder_->Release());
+ // Close the files before we start using them for SSIM/PSNR calculations.
+ frame_reader_->Close();
+ frame_writer_->Close();
+
+ // TODO(marpan): should compute these quality metrics per SetRates update.
+ webrtc::test::QualityMetricsResult psnr_result, ssim_result;
+ EXPECT_EQ(0, webrtc::test::I420MetricsFromFiles(
+ config_.input_filename.c_str(),
+ config_.output_filename.c_str(),
+ config_.codec_settings->width,
+ config_.codec_settings->height,
+ &psnr_result,
+ &ssim_result));
+ printf("PSNR avg: %f, min: %f SSIM avg: %f, min: %f\n",
+ psnr_result.average, psnr_result.min,
+ ssim_result.average, ssim_result.min);
+ stats_.PrintSummary();
+ EXPECT_GT(psnr_result.average, quality_metrics.minimum_avg_psnr);
+ EXPECT_GT(psnr_result.min, quality_metrics.minimum_min_psnr);
+ EXPECT_GT(ssim_result.average, quality_metrics.minimum_avg_ssim);
+ EXPECT_GT(ssim_result.min, quality_metrics.minimum_min_ssim);
+ if (!remove(config_.output_filename.c_str())) {
+ fprintf(stderr, "Failed to remove temporary file!");
+ }
+ }
+};
+
+void SetRateProfilePars(RateProfile* rate_profile,
+ int update_index,
+ int bit_rate,
+ int frame_rate,
+ int frame_index_rate_update) {
+ rate_profile->target_bit_rate[update_index] = bit_rate;
+ rate_profile->input_frame_rate[update_index] = frame_rate;
+ rate_profile->frame_index_rate_update[update_index] = frame_index_rate_update;
+}
+
+void SetCodecParameters(CodecConfigPars* process_settings,
+ VideoCodecType codec_type,
+ float packet_loss,
+ int key_frame_interval,
+ int num_temporal_layers,
+ bool error_concealment_on,
+ bool denoising_on,
+ bool frame_dropper_on,
+ bool spatial_resize_on) {
+ process_settings->codec_type = codec_type;
+ process_settings->packet_loss = packet_loss;
+ process_settings->key_frame_interval = key_frame_interval;
+ process_settings->num_temporal_layers = num_temporal_layers,
+ process_settings->error_concealment_on = error_concealment_on;
+ process_settings->denoising_on = denoising_on;
+ process_settings->frame_dropper_on = frame_dropper_on;
+ process_settings->spatial_resize_on = spatial_resize_on;
+}
+
+void SetQualityMetrics(QualityMetrics* quality_metrics,
+ double minimum_avg_psnr,
+ double minimum_min_psnr,
+ double minimum_avg_ssim,
+ double minimum_min_ssim) {
+ quality_metrics->minimum_avg_psnr = minimum_avg_psnr;
+ quality_metrics->minimum_min_psnr = minimum_min_psnr;
+ quality_metrics->minimum_avg_ssim = minimum_avg_ssim;
+ quality_metrics->minimum_min_ssim = minimum_min_ssim;
+}
+
+void SetRateControlMetrics(RateControlMetrics* rc_metrics,
+ int update_index,
+ int max_num_dropped_frames,
+ int max_key_frame_size_mismatch,
+ int max_delta_frame_size_mismatch,
+ int max_encoding_rate_mismatch,
+ int max_time_hit_target,
+ int num_spatial_resizes) {
+ rc_metrics[update_index].max_num_dropped_frames = max_num_dropped_frames;
+ rc_metrics[update_index].max_key_frame_size_mismatch =
+ max_key_frame_size_mismatch;
+ rc_metrics[update_index].max_delta_frame_size_mismatch =
+ max_delta_frame_size_mismatch;
+ rc_metrics[update_index].max_encoding_rate_mismatch =
+ max_encoding_rate_mismatch;
+ rc_metrics[update_index].max_time_hit_target = max_time_hit_target;
+ rc_metrics[update_index].num_spatial_resizes = num_spatial_resizes;
+}
+
+// VP9: Run with no packet loss and fixed bitrate. Quality should be very high.
+// One key frame (first frame only) in sequence. Setting |key_frame_interval|
+// to -1 below means no periodic key frames in test.
+TEST_F(VideoProcessorIntegrationTest, Process0PercentPacketLossVP9) {
+ // Bitrate and frame rate profile.
+ RateProfile rate_profile;
+ SetRateProfilePars(&rate_profile, 0, 500, 30, 0);
+ rate_profile.frame_index_rate_update[1] = kNbrFramesShort + 1;
+ rate_profile.num_frames = kNbrFramesShort;
+ // Codec/network settings.
+ CodecConfigPars process_settings;
+ SetCodecParameters(&process_settings, kVideoCodecVP9, 0.0f, -1, 1, false,
+ false, true, false);
+ // Metrics for expected quality.
+ QualityMetrics quality_metrics;
+ SetQualityMetrics(&quality_metrics, 37.0, 36.0, 0.93, 0.92);
+ // Metrics for rate control.
+ RateControlMetrics rc_metrics[1];
+ SetRateControlMetrics(rc_metrics, 0, 0, 40, 20, 10, 20, 0);
+ ProcessFramesAndVerify(quality_metrics,
+ rate_profile,
+ process_settings,
+ rc_metrics);
+}
+
+// VP9: Run with 5% packet loss and fixed bitrate. Quality should be a bit
+// lower. One key frame (first frame only) in sequence.
+TEST_F(VideoProcessorIntegrationTest, Process5PercentPacketLossVP9) {
+ // Bitrate and frame rate profile.
+ RateProfile rate_profile;
+ SetRateProfilePars(&rate_profile, 0, 500, 30, 0);
+ rate_profile.frame_index_rate_update[1] = kNbrFramesShort + 1;
+ rate_profile.num_frames = kNbrFramesShort;
+ // Codec/network settings.
+ CodecConfigPars process_settings;
+ SetCodecParameters(&process_settings, kVideoCodecVP9, 0.05f, -1, 1, false,
+ false, true, false);
+ // Metrics for expected quality.
+ QualityMetrics quality_metrics;
+ SetQualityMetrics(&quality_metrics, 17.0, 14.0, 0.45, 0.36);
+ // Metrics for rate control.
+ RateControlMetrics rc_metrics[1];
+ SetRateControlMetrics(rc_metrics, 0, 0, 40, 20, 10, 20, 0);
+ ProcessFramesAndVerify(quality_metrics,
+ rate_profile,
+ process_settings,
+ rc_metrics);
+}
+
+
+// VP9: Run with no packet loss, with varying bitrate (3 rate updates):
+// low to high to medium. Check that quality and encoder response to the new
+// target rate/per-frame bandwidth (for each rate update) is within limits.
+// One key frame (first frame only) in sequence.
+TEST_F(VideoProcessorIntegrationTest, ProcessNoLossChangeBitRateVP9) {
+ // Bitrate and frame rate profile.
+ RateProfile rate_profile;
+ SetRateProfilePars(&rate_profile, 0, 200, 30, 0);
+ SetRateProfilePars(&rate_profile, 1, 700, 30, 100);
+ SetRateProfilePars(&rate_profile, 2, 500, 30, 200);
+ rate_profile.frame_index_rate_update[3] = kNbrFramesLong + 1;
+ rate_profile.num_frames = kNbrFramesLong;
+ // Codec/network settings.
+ CodecConfigPars process_settings;
+ SetCodecParameters(&process_settings, kVideoCodecVP9, 0.0f, -1, 1, false,
+ false, true, false);
+ // Metrics for expected quality.
+ QualityMetrics quality_metrics;
+ SetQualityMetrics(&quality_metrics, 35.9, 30.0, 0.90, 0.85);
+ // Metrics for rate control.
+ RateControlMetrics rc_metrics[3];
+ SetRateControlMetrics(rc_metrics, 0, 0, 30, 20, 20, 30, 0);
+ SetRateControlMetrics(rc_metrics, 1, 2, 0, 20, 20, 60, 0);
+ SetRateControlMetrics(rc_metrics, 2, 0, 0, 25, 20, 40, 0);
+ ProcessFramesAndVerify(quality_metrics,
+ rate_profile,
+ process_settings,
+ rc_metrics);
+}
+
+// VP9: Run with no packet loss, with an update (decrease) in frame rate.
+// Lower frame rate means higher per-frame-bandwidth, so easier to encode.
+// At the low bitrate in this test, this means better rate control after the
+// update(s) to lower frame rate. So expect less frame drops, and max values
+// for the rate control metrics can be lower. One key frame (first frame only).
+// Note: quality after update should be higher but we currently compute quality
+// metrics averaged over whole sequence run.
+TEST_F(VideoProcessorIntegrationTest,
+ ProcessNoLossChangeFrameRateFrameDropVP9) {
+ config_.networking_config.packet_loss_probability = 0;
+ // Bitrate and frame rate profile.
+ RateProfile rate_profile;
+ SetRateProfilePars(&rate_profile, 0, 100, 24, 0);
+ SetRateProfilePars(&rate_profile, 1, 100, 15, 100);
+ SetRateProfilePars(&rate_profile, 2, 100, 10, 200);
+ rate_profile.frame_index_rate_update[3] = kNbrFramesLong + 1;
+ rate_profile.num_frames = kNbrFramesLong;
+ // Codec/network settings.
+ CodecConfigPars process_settings;
+ SetCodecParameters(&process_settings, kVideoCodecVP9, 0.0f, -1, 1, false,
+ false, true, false);
+ // Metrics for expected quality.
+ QualityMetrics quality_metrics;
+ SetQualityMetrics(&quality_metrics, 31.5, 18.0, 0.80, 0.44);
+ // Metrics for rate control.
+ RateControlMetrics rc_metrics[3];
+ SetRateControlMetrics(rc_metrics, 0, 35, 50, 70, 15, 45, 0);
+ SetRateControlMetrics(rc_metrics, 1, 10, 0, 40, 10, 30, 0);
+ SetRateControlMetrics(rc_metrics, 2, 5, 0, 30, 5, 20, 0);
+ ProcessFramesAndVerify(quality_metrics,
+ rate_profile,
+ process_settings,
+ rc_metrics);
+}
+
+// VP9: Run with no packet loss and denoiser on. One key frame (first frame).
+TEST_F(VideoProcessorIntegrationTest, ProcessNoLossDenoiserOnVP9) {
+ // Bitrate and frame rate profile.
+ RateProfile rate_profile;
+ SetRateProfilePars(&rate_profile, 0, 500, 30, 0);
+ rate_profile.frame_index_rate_update[1] = kNbrFramesShort + 1;
+ rate_profile.num_frames = kNbrFramesShort;
+ // Codec/network settings.
+ CodecConfigPars process_settings;
+ SetCodecParameters(&process_settings, kVideoCodecVP9, 0.0f, -1, 1, false,
+ true, true, false);
+ // Metrics for expected quality.
+ QualityMetrics quality_metrics;
+ SetQualityMetrics(&quality_metrics, 36.8, 35.8, 0.92, 0.91);
+ // Metrics for rate control.
+ RateControlMetrics rc_metrics[1];
+ SetRateControlMetrics(rc_metrics, 0, 0, 40, 20, 10, 20, 0);
+ ProcessFramesAndVerify(quality_metrics,
+ rate_profile,
+ process_settings,
+ rc_metrics);
+}
+
+// TODO(marpan): Add temporal layer test for VP9, once changes are in
+// vp9 wrapper for this.
+
+// VP8: Run with no packet loss and fixed bitrate. Quality should be very high.
+// One key frame (first frame only) in sequence. Setting |key_frame_interval|
+// to -1 below means no periodic key frames in test.
+TEST_F(VideoProcessorIntegrationTest, ProcessZeroPacketLoss) {
+ // Bitrate and frame rate profile.
+ RateProfile rate_profile;
+ SetRateProfilePars(&rate_profile, 0, 500, 30, 0);
+ rate_profile.frame_index_rate_update[1] = kNbrFramesShort + 1;
+ rate_profile.num_frames = kNbrFramesShort;
+ // Codec/network settings.
+ CodecConfigPars process_settings;
+ SetCodecParameters(&process_settings, kVideoCodecVP8, 0.0f, -1, 1, false,
+ true, true, false);
+ // Metrics for expected quality.
+ QualityMetrics quality_metrics;
+ SetQualityMetrics(&quality_metrics, 34.95, 33.0, 0.90, 0.89);
+ // Metrics for rate control.
+ RateControlMetrics rc_metrics[1];
+ SetRateControlMetrics(rc_metrics, 0, 0, 40, 20, 10, 15, 0);
+ ProcessFramesAndVerify(quality_metrics,
+ rate_profile,
+ process_settings,
+ rc_metrics);
+}
+
+// VP8: Run with 5% packet loss and fixed bitrate. Quality should be a bit
+// lower. One key frame (first frame only) in sequence.
+TEST_F(VideoProcessorIntegrationTest, Process5PercentPacketLoss) {
+ // Bitrate and frame rate profile.
+ RateProfile rate_profile;
+ SetRateProfilePars(&rate_profile, 0, 500, 30, 0);
+ rate_profile.frame_index_rate_update[1] = kNbrFramesShort + 1;
+ rate_profile.num_frames = kNbrFramesShort;
+ // Codec/network settings.
+ CodecConfigPars process_settings;
+ SetCodecParameters(&process_settings, kVideoCodecVP8, 0.05f, -1, 1, false,
+ true, true, false);
+ // Metrics for expected quality.
+ QualityMetrics quality_metrics;
+ SetQualityMetrics(&quality_metrics, 20.0, 16.0, 0.60, 0.40);
+ // Metrics for rate control.
+ RateControlMetrics rc_metrics[1];
+ SetRateControlMetrics(rc_metrics, 0, 0, 40, 20, 10, 15, 0);
+ ProcessFramesAndVerify(quality_metrics,
+ rate_profile,
+ process_settings,
+ rc_metrics);
+}
+
+// VP8: Run with 10% packet loss and fixed bitrate. Quality should be lower.
+// One key frame (first frame only) in sequence.
+TEST_F(VideoProcessorIntegrationTest, Process10PercentPacketLoss) {
+ // Bitrate and frame rate profile.
+ RateProfile rate_profile;
+ SetRateProfilePars(&rate_profile, 0, 500, 30, 0);
+ rate_profile.frame_index_rate_update[1] = kNbrFramesShort + 1;
+ rate_profile.num_frames = kNbrFramesShort;
+ // Codec/network settings.
+ CodecConfigPars process_settings;
+ SetCodecParameters(&process_settings, kVideoCodecVP8, 0.1f, -1, 1, false,
+ true, true, false);
+ // Metrics for expected quality.
+ QualityMetrics quality_metrics;
+ SetQualityMetrics(&quality_metrics, 19.0, 16.0, 0.50, 0.35);
+ // Metrics for rate control.
+ RateControlMetrics rc_metrics[1];
+ SetRateControlMetrics(rc_metrics, 0, 0, 40, 20, 10, 15, 0);
+ ProcessFramesAndVerify(quality_metrics,
+ rate_profile,
+ process_settings,
+ rc_metrics);
+}
+
+// The tests below are currently disabled for Android. For ARM, the encoder
+// uses |cpu_speed| = 12, as opposed to default |cpu_speed| <= 6 for x86,
+// which leads to significantly different quality. The quality and rate control
+// settings in the tests below are defined for encoder speed setting
+// |cpu_speed| <= ~6. A number of settings would need to be significantly
+// modified for the |cpu_speed| = 12 case. For now, keep the tests below
+// disabled on Android. Some quality parameter in the above test has been
+// adjusted to also pass for |cpu_speed| <= 12.
+
+// VP8: Run with no packet loss, with varying bitrate (3 rate updates):
+// low to high to medium. Check that quality and encoder response to the new
+// target rate/per-frame bandwidth (for each rate update) is within limits.
+// One key frame (first frame only) in sequence.
+TEST_F(VideoProcessorIntegrationTest,
+ DISABLED_ON_ANDROID(ProcessNoLossChangeBitRateVP8)) {
+ // Bitrate and frame rate profile.
+ RateProfile rate_profile;
+ SetRateProfilePars(&rate_profile, 0, 200, 30, 0);
+ SetRateProfilePars(&rate_profile, 1, 800, 30, 100);
+ SetRateProfilePars(&rate_profile, 2, 500, 30, 200);
+ rate_profile.frame_index_rate_update[3] = kNbrFramesLong + 1;
+ rate_profile.num_frames = kNbrFramesLong;
+ // Codec/network settings.
+ CodecConfigPars process_settings;
+ SetCodecParameters(&process_settings, kVideoCodecVP8, 0.0f, -1, 1, false,
+ true, true, false);
+ // Metrics for expected quality.
+ QualityMetrics quality_metrics;
+ SetQualityMetrics(&quality_metrics, 34.0, 32.0, 0.85, 0.80);
+ // Metrics for rate control.
+ RateControlMetrics rc_metrics[3];
+ SetRateControlMetrics(rc_metrics, 0, 0, 45, 20, 10, 15, 0);
+ SetRateControlMetrics(rc_metrics, 1, 0, 0, 25, 20, 10, 0);
+ SetRateControlMetrics(rc_metrics, 2, 0, 0, 25, 15, 10, 0);
+ ProcessFramesAndVerify(quality_metrics,
+ rate_profile,
+ process_settings,
+ rc_metrics);
+}
+
+// VP8: Run with no packet loss, with an update (decrease) in frame rate.
+// Lower frame rate means higher per-frame-bandwidth, so easier to encode.
+// At the bitrate in this test, this means better rate control after the
+// update(s) to lower frame rate. So expect less frame drops, and max values
+// for the rate control metrics can be lower. One key frame (first frame only).
+// Note: quality after update should be higher but we currently compute quality
+// metrics averaged over whole sequence run.
+TEST_F(VideoProcessorIntegrationTest,
+ DISABLED_ON_ANDROID(ProcessNoLossChangeFrameRateFrameDropVP8)) {
+ config_.networking_config.packet_loss_probability = 0;
+ // Bitrate and frame rate profile.
+ RateProfile rate_profile;
+ SetRateProfilePars(&rate_profile, 0, 80, 24, 0);
+ SetRateProfilePars(&rate_profile, 1, 80, 15, 100);
+ SetRateProfilePars(&rate_profile, 2, 80, 10, 200);
+ rate_profile.frame_index_rate_update[3] = kNbrFramesLong + 1;
+ rate_profile.num_frames = kNbrFramesLong;
+ // Codec/network settings.
+ CodecConfigPars process_settings;
+ SetCodecParameters(&process_settings, kVideoCodecVP8, 0.0f, -1, 1, false,
+ true, true, false);
+ // Metrics for expected quality.
+ QualityMetrics quality_metrics;
+ SetQualityMetrics(&quality_metrics, 31.0, 22.0, 0.80, 0.65);
+ // Metrics for rate control.
+ RateControlMetrics rc_metrics[3];
+ SetRateControlMetrics(rc_metrics, 0, 40, 20, 75, 15, 60, 0);
+ SetRateControlMetrics(rc_metrics, 1, 10, 0, 25, 10, 35, 0);
+ SetRateControlMetrics(rc_metrics, 2, 0, 0, 20, 10, 15, 0);
+ ProcessFramesAndVerify(quality_metrics,
+ rate_profile,
+ process_settings,
+ rc_metrics);
+}
+
+// Run with no packet loss, at low bitrate. During this time we should've
+// resized once.
+TEST_F(VideoProcessorIntegrationTest,
+ DISABLED_ON_ANDROID(ProcessNoLossSpatialResizeFrameDropVP8)) {
+ config_.networking_config.packet_loss_probability = 0;
+ // Bitrate and frame rate profile.
+ RateProfile rate_profile;
+ SetRateProfilePars(&rate_profile, 0, 50, 30, 0);
+ rate_profile.frame_index_rate_update[1] = kNbrFramesLong + 1;
+ rate_profile.num_frames = kNbrFramesLong;
+ // Codec/network settings.
+ CodecConfigPars process_settings;
+ SetCodecParameters(&process_settings, kVideoCodecVP8, 0.0f, kNbrFramesLong,
+ 1, false, true, true, true);
+ // Metrics for expected quality.
+ QualityMetrics quality_metrics;
+ SetQualityMetrics(&quality_metrics, 25.0, 15.0, 0.70, 0.40);
+ // Metrics for rate control.
+ RateControlMetrics rc_metrics[1];
+ SetRateControlMetrics(rc_metrics, 0, 160, 60, 120, 20, 70, 1);
+ ProcessFramesAndVerify(quality_metrics,
+ rate_profile,
+ process_settings,
+ rc_metrics);
+}
+
+// VP8: Run with no packet loss, with 3 temporal layers, with a rate update in
+// the middle of the sequence. The max values for the frame size mismatch and
+// encoding rate mismatch are applied to each layer.
+// No dropped frames in this test, and internal spatial resizer is off.
+// One key frame (first frame only) in sequence, so no spatial resizing.
+TEST_F(VideoProcessorIntegrationTest,
+ DISABLED_ON_ANDROID(ProcessNoLossTemporalLayersVP8)) {
+ config_.networking_config.packet_loss_probability = 0;
+ // Bitrate and frame rate profile.
+ RateProfile rate_profile;
+ SetRateProfilePars(&rate_profile, 0, 200, 30, 0);
+ SetRateProfilePars(&rate_profile, 1, 400, 30, 150);
+ rate_profile.frame_index_rate_update[2] = kNbrFramesLong + 1;
+ rate_profile.num_frames = kNbrFramesLong;
+ // Codec/network settings.
+ CodecConfigPars process_settings;
+ SetCodecParameters(&process_settings, kVideoCodecVP8, 0.0f, -1, 3, false,
+ true, true, false);
+ // Metrics for expected quality.
+ QualityMetrics quality_metrics;
+ SetQualityMetrics(&quality_metrics, 32.5, 30.0, 0.85, 0.80);
+ // Metrics for rate control.
+ RateControlMetrics rc_metrics[2];
+ SetRateControlMetrics(rc_metrics, 0, 0, 20, 30, 10, 10, 0);
+ SetRateControlMetrics(rc_metrics, 1, 0, 0, 30, 15, 10, 0);
+ ProcessFramesAndVerify(quality_metrics,
+ rate_profile,
+ process_settings,
+ rc_metrics);
+}
+} // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_coding/codecs/test/videoprocessor_unittest.cc b/chromium/third_party/webrtc/modules/video_coding/codecs/test/videoprocessor_unittest.cc
new file mode 100644
index 00000000000..88b5467f1f0
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/video_coding/codecs/test/videoprocessor_unittest.cc
@@ -0,0 +1,99 @@
+/*
+ * Copyright (c) 2012 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.
+ */
+
+#include "testing/gmock/include/gmock/gmock.h"
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/modules/video_coding/codecs/interface/mock/mock_video_codec_interface.h"
+#include "webrtc/modules/video_coding/codecs/test/mock/mock_packet_manipulator.h"
+#include "webrtc/modules/video_coding/codecs/test/videoprocessor.h"
+#include "webrtc/modules/video_coding/main/interface/video_coding.h"
+#include "webrtc/test/testsupport/mock/mock_frame_reader.h"
+#include "webrtc/test/testsupport/mock/mock_frame_writer.h"
+#include "webrtc/test/testsupport/packet_reader.h"
+#include "webrtc/test/testsupport/unittest_utils.h"
+#include "webrtc/typedefs.h"
+
+using ::testing::_;
+using ::testing::AtLeast;
+using ::testing::Return;
+
+namespace webrtc {
+namespace test {
+
+// Very basic testing for VideoProcessor. It's mostly tested by running the
+// video_quality_measurement program.
+class VideoProcessorTest: public testing::Test {
+ protected:
+ MockVideoEncoder encoder_mock_;
+ MockVideoDecoder decoder_mock_;
+ MockFrameReader frame_reader_mock_;
+ MockFrameWriter frame_writer_mock_;
+ MockPacketManipulator packet_manipulator_mock_;
+ Stats stats_;
+ TestConfig config_;
+ VideoCodec codec_settings_;
+
+ VideoProcessorTest() {}
+ virtual ~VideoProcessorTest() {}
+ void SetUp() {
+ // Get a codec configuration struct and configure it.
+ VideoCodingModule::Codec(kVideoCodecVP8, &codec_settings_);
+ config_.codec_settings = &codec_settings_;
+ config_.codec_settings->startBitrate = 100;
+ config_.codec_settings->width = 352;
+ config_.codec_settings->height = 288;
+ }
+ void TearDown() {}
+
+ void ExpectInit() {
+ EXPECT_CALL(encoder_mock_, InitEncode(_, _, _))
+ .Times(1);
+ EXPECT_CALL(encoder_mock_, RegisterEncodeCompleteCallback(_))
+ .Times(AtLeast(1));
+ EXPECT_CALL(decoder_mock_, InitDecode(_, _))
+ .Times(1);
+ EXPECT_CALL(decoder_mock_, RegisterDecodeCompleteCallback(_))
+ .Times(AtLeast(1));
+ EXPECT_CALL(frame_reader_mock_, NumberOfFrames())
+ .WillOnce(Return(1));
+ EXPECT_CALL(frame_reader_mock_, FrameLength())
+ .WillOnce(Return(152064));
+ }
+};
+
+TEST_F(VideoProcessorTest, Init) {
+ ExpectInit();
+ VideoProcessorImpl video_processor(&encoder_mock_, &decoder_mock_,
+ &frame_reader_mock_,
+ &frame_writer_mock_,
+ &packet_manipulator_mock_, config_,
+ &stats_);
+ ASSERT_TRUE(video_processor.Init());
+}
+
+TEST_F(VideoProcessorTest, ProcessFrame) {
+ ExpectInit();
+ EXPECT_CALL(encoder_mock_, Encode(_, _, _))
+ .Times(1);
+ EXPECT_CALL(frame_reader_mock_, ReadFrame(_))
+ .WillOnce(Return(true));
+ // Since we don't return any callback from the mock, the decoder will not
+ // be more than initialized...
+ VideoProcessorImpl video_processor(&encoder_mock_, &decoder_mock_,
+ &frame_reader_mock_,
+ &frame_writer_mock_,
+ &packet_manipulator_mock_, config_,
+ &stats_);
+ ASSERT_TRUE(video_processor.Init());
+ video_processor.ProcessFrame(0);
+}
+
+} // namespace test
+} // namespace webrtc