Move VP9 frame rate controller to separate class.

Bug: webrtc:9669
Change-Id: I6f30587778e9783182af11d2410464024918e171
Reviewed-on: https://webrtc-review.googlesource.com/96201
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24487}
This commit is contained in:
Sergey Silkin
2018-08-29 01:05:26 +02:00
committed by Commit Bot
parent 5a998d7246
commit ae3144c51f
6 changed files with 230 additions and 47 deletions

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@ -234,6 +234,8 @@ rtc_source_set("video_coding_utility") {
"utility/default_video_bitrate_allocator.h", "utility/default_video_bitrate_allocator.h",
"utility/frame_dropper.cc", "utility/frame_dropper.cc",
"utility/frame_dropper.h", "utility/frame_dropper.h",
"utility/framerate_controller.cc",
"utility/framerate_controller.h",
"utility/ivf_file_writer.cc", "utility/ivf_file_writer.cc",
"utility/ivf_file_writer.h", "utility/ivf_file_writer.h",
"utility/moving_average.cc", "utility/moving_average.cc",
@ -799,6 +801,7 @@ if (rtc_include_tests) {
"timing_unittest.cc", "timing_unittest.cc",
"utility/default_video_bitrate_allocator_unittest.cc", "utility/default_video_bitrate_allocator_unittest.cc",
"utility/frame_dropper_unittest.cc", "utility/frame_dropper_unittest.cc",
"utility/framerate_controller_unittest.cc",
"utility/ivf_file_writer_unittest.cc", "utility/ivf_file_writer_unittest.cc",
"utility/mock/mock_frame_dropper.h", "utility/mock/mock_frame_dropper.h",
"utility/moving_average_unittest.cc", "utility/moving_average_unittest.cc",

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@ -157,8 +157,7 @@ VP9EncoderImpl::VP9EncoderImpl(const cricket::VideoCodec& codec)
num_spatial_layers_(0), num_spatial_layers_(0),
is_svc_(false), is_svc_(false),
inter_layer_pred_(InterLayerPredMode::kOn), inter_layer_pred_(InterLayerPredMode::kOn),
output_framerate_(1000.0, 1000.0), framerate_controller_(kMaxScreenSharingFramerateFps),
last_encoded_frame_rtp_timestamp_(0),
is_flexible_mode_(false) { is_flexible_mode_(false) {
memset(&codec_, 0, sizeof(codec_)); memset(&codec_, 0, sizeof(codec_));
memset(&svc_params_, 0, sizeof(vpx_svc_extra_cfg_t)); memset(&svc_params_, 0, sizeof(vpx_svc_extra_cfg_t));
@ -358,11 +357,9 @@ int VP9EncoderImpl::InitEncode(const VideoCodec* inst,
num_temporal_layers_ = 1; num_temporal_layers_ = 1;
// Init framerate controller. // Init framerate controller.
output_framerate_.Reset();
if (codec_.mode == VideoCodecMode::kScreensharing) { if (codec_.mode == VideoCodecMode::kScreensharing) {
target_framerate_fps_ = kMaxScreenSharingFramerateFps; framerate_controller_.Reset();
} else { framerate_controller_.SetTargetRate(kMaxScreenSharingFramerateFps);
target_framerate_fps_.reset();
} }
is_svc_ = (num_spatial_layers_ > 1 || num_temporal_layers_ > 1); is_svc_ = (num_spatial_layers_ > 1 || num_temporal_layers_ > 1);
@ -672,7 +669,8 @@ int VP9EncoderImpl::Encode(const VideoFrame& input_image,
} }
if (VideoCodecMode::kScreensharing == codec_.mode && !force_key_frame_) { if (VideoCodecMode::kScreensharing == codec_.mode && !force_key_frame_) {
if (DropFrame(input_image.timestamp())) { if (framerate_controller_.DropFrame(1000 * input_image.timestamp() /
kVideoPayloadTypeFrequency)) {
return WEBRTC_VIDEO_CODEC_OK; return WEBRTC_VIDEO_CODEC_OK;
} }
} }
@ -734,8 +732,10 @@ int VP9EncoderImpl::Encode(const VideoFrame& input_image,
} }
RTC_CHECK_GT(codec_.maxFramerate, 0); RTC_CHECK_GT(codec_.maxFramerate, 0);
uint32_t duration = uint32_t target_framerate_fps = codec_.mode == VideoCodecMode::kScreensharing
90000 / target_framerate_fps_.value_or(codec_.maxFramerate); ? kMaxScreenSharingFramerateFps
: codec_.maxFramerate;
uint32_t duration = 90000 / target_framerate_fps;
const vpx_codec_err_t rv = vpx_codec_encode(encoder_, raw_, timestamp_, const vpx_codec_err_t rv = vpx_codec_encode(encoder_, raw_, timestamp_,
duration, flags, VPX_DL_REALTIME); duration, flags, VPX_DL_REALTIME);
if (rv != VPX_CODEC_OK) { if (rv != VPX_CODEC_OK) {
@ -1064,45 +1064,14 @@ void VP9EncoderImpl::DeliverBufferedFrame(bool end_of_picture) {
&frag_info); &frag_info);
encoded_image_._length = 0; encoded_image_._length = 0;
if (end_of_picture) { if (end_of_picture && codec_.mode == VideoCodecMode::kScreensharing) {
const uint32_t timestamp_ms = const uint32_t timestamp_ms =
1000 * encoded_image_.Timestamp() / kVideoPayloadTypeFrequency; 1000 * encoded_image_.Timestamp() / kVideoPayloadTypeFrequency;
output_framerate_.Update(1, timestamp_ms); framerate_controller_.AddFrame(timestamp_ms);
last_encoded_frame_rtp_timestamp_ = encoded_image_.Timestamp();
} }
} }
} }
bool VP9EncoderImpl::DropFrame(uint32_t rtp_timestamp) {
if (target_framerate_fps_) {
if (rtp_timestamp < last_encoded_frame_rtp_timestamp_) {
// Timestamp has wrapped around. Reset framerate statistic.
output_framerate_.Reset();
return false;
}
const uint32_t timestamp_ms =
1000 * rtp_timestamp / kVideoPayloadTypeFrequency;
const uint32_t framerate_fps =
output_framerate_.Rate(timestamp_ms).value_or(0);
if (framerate_fps > *target_framerate_fps_) {
return true;
}
// Primarily check if frame interval is too short using frame timestamps,
// as if they are correct they won't be affected by queuing in webrtc.
const uint32_t expected_frame_interval =
kVideoPayloadTypeFrequency / *target_framerate_fps_;
const uint32_t ts_diff = rtp_timestamp - last_encoded_frame_rtp_timestamp_;
if (ts_diff < 85 * expected_frame_interval / 100) {
return true;
}
}
return false;
}
int VP9EncoderImpl::SetChannelParameters(uint32_t packet_loss, int64_t rtt) { int VP9EncoderImpl::SetChannelParameters(uint32_t packet_loss, int64_t rtt) {
return WEBRTC_VIDEO_CODEC_OK; return WEBRTC_VIDEO_CODEC_OK;
} }

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@ -20,7 +20,7 @@
#include "media/base/vp9_profile.h" #include "media/base/vp9_profile.h"
#include "modules/video_coding/codecs/vp9/vp9_frame_buffer_pool.h" #include "modules/video_coding/codecs/vp9/vp9_frame_buffer_pool.h"
#include "rtc_base/rate_statistics.h" #include "modules/video_coding/utility/framerate_controller.h"
#include "vpx/vp8cx.h" #include "vpx/vp8cx.h"
#include "vpx/vpx_decoder.h" #include "vpx/vpx_decoder.h"
@ -82,7 +82,7 @@ class VP9EncoderImpl : public VP9Encoder {
void DeliverBufferedFrame(bool end_of_picture); void DeliverBufferedFrame(bool end_of_picture);
bool DropFrame(uint32_t rtp_timestamp); bool DropFrame(uint8_t spatial_idx, uint32_t rtp_timestamp);
// Determine maximum target for Intra frames // Determine maximum target for Intra frames
// //
@ -117,9 +117,7 @@ class VP9EncoderImpl : public VP9Encoder {
InterLayerPredMode inter_layer_pred_; InterLayerPredMode inter_layer_pred_;
// Framerate controller. // Framerate controller.
absl::optional<float> target_framerate_fps_; FramerateController framerate_controller_;
RateStatistics output_framerate_;
uint32_t last_encoded_frame_rtp_timestamp_;
// Used for flexible mode. // Used for flexible mode.
bool is_flexible_mode_; bool is_flexible_mode_;

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@ -0,0 +1,81 @@
/*
* Copyright (c) 2018 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 "modules/video_coding/utility/framerate_controller.h"
#include "rtc_base/checks.h"
namespace webrtc {
FramerateController::FramerateController(float target_framerate_fps)
: min_frame_interval_ms_(0), framerate_estimator_(1000.0, 1000.0) {
SetTargetRate(target_framerate_fps);
}
void FramerateController::SetTargetRate(float target_framerate_fps) {
if (target_framerate_fps_ != target_framerate_fps) {
framerate_estimator_.Reset();
if (last_timestamp_ms_) {
framerate_estimator_.Update(1, *last_timestamp_ms_);
}
const size_t target_frame_interval_ms = 1000 / target_framerate_fps;
target_framerate_fps_ = target_framerate_fps;
min_frame_interval_ms_ = 85 * target_frame_interval_ms / 100;
}
}
float FramerateController::GetTargetRate() {
return *target_framerate_fps_;
}
void FramerateController::Reset() {
framerate_estimator_.Reset();
last_timestamp_ms_.reset();
}
bool FramerateController::DropFrame(uint32_t timestamp_ms) const {
if (timestamp_ms < last_timestamp_ms_) {
// Timestamp jumps backward. We can't make adequate drop decision. Don't
// drop this frame. Stats will be reset in AddFrame().
return false;
}
if (Rate(timestamp_ms).value_or(*target_framerate_fps_) >
target_framerate_fps_) {
return true;
}
if (last_timestamp_ms_) {
const int64_t diff_ms =
static_cast<int64_t>(timestamp_ms) - *last_timestamp_ms_;
if (diff_ms < min_frame_interval_ms_) {
return true;
}
}
return false;
}
void FramerateController::AddFrame(uint32_t timestamp_ms) {
if (timestamp_ms < last_timestamp_ms_) {
// Timestamp jumps backward.
Reset();
}
framerate_estimator_.Update(1, timestamp_ms);
last_timestamp_ms_ = timestamp_ms;
}
absl::optional<float> FramerateController::Rate(uint32_t timestamp_ms) const {
return framerate_estimator_.Rate(timestamp_ms);
}
} // namespace webrtc

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@ -0,0 +1,44 @@
/*
* Copyright (c) 2018 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 MODULES_VIDEO_CODING_UTILITY_FRAMERATE_CONTROLLER_H_
#define MODULES_VIDEO_CODING_UTILITY_FRAMERATE_CONTROLLER_H_
#include "absl/types/optional.h"
#include "rtc_base/rate_statistics.h"
namespace webrtc {
class FramerateController {
public:
explicit FramerateController(float target_framerate_fps);
void SetTargetRate(float target_framerate_fps);
float GetTargetRate();
// Advices user to drop next frame in order to reach target framerate.
bool DropFrame(uint32_t timestamp_ms) const;
void AddFrame(uint32_t timestamp_ms);
void Reset();
private:
absl::optional<float> Rate(uint32_t timestamp_ms) const;
absl::optional<float> target_framerate_fps_;
absl::optional<uint32_t> last_timestamp_ms_;
uint32_t min_frame_interval_ms_;
RateStatistics framerate_estimator_;
};
} // namespace webrtc
#endif // MODULES_VIDEO_CODING_UTILITY_FRAMERATE_CONTROLLER_H_

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@ -0,0 +1,88 @@
/*
* Copyright (c) 2018 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 "modules/video_coding/utility/framerate_controller.h"
#include "test/gtest.h"
namespace webrtc {
TEST(FramerateController, KeepTargetFramerate) {
const float input_framerate_fps = 20;
const float target_framerate_fps = 5;
const float max_abs_framerate_error_fps = target_framerate_fps * 0.1f;
const size_t input_duration_secs = 3;
const size_t num_input_frames = input_duration_secs * input_framerate_fps;
FramerateController framerate_controller(target_framerate_fps);
size_t num_dropped_frames = 0;
for (size_t frame_num = 0; frame_num < num_input_frames; ++frame_num) {
const uint32_t timestamp_ms =
static_cast<uint32_t>(1000 * frame_num / input_framerate_fps);
if (framerate_controller.DropFrame(timestamp_ms)) {
++num_dropped_frames;
} else {
framerate_controller.AddFrame(timestamp_ms);
}
}
const float output_framerate_fps =
static_cast<float>(num_input_frames - num_dropped_frames) /
input_duration_secs;
EXPECT_NEAR(output_framerate_fps, target_framerate_fps,
max_abs_framerate_error_fps);
}
TEST(FramerateController, DoNotDropAnyFramesIfTargerEqualsInput) {
const float input_framerate_fps = 30;
const size_t input_duration_secs = 3;
const size_t num_input_frames = input_duration_secs * input_framerate_fps;
FramerateController framerate_controller(input_framerate_fps);
size_t num_dropped_frames = 0;
for (size_t frame_num = 0; frame_num < num_input_frames; ++frame_num) {
const uint32_t timestamp_ms =
static_cast<uint32_t>(1000 * frame_num / input_framerate_fps);
if (framerate_controller.DropFrame(timestamp_ms)) {
++num_dropped_frames;
} else {
framerate_controller.AddFrame(timestamp_ms);
}
}
EXPECT_EQ(num_dropped_frames, 0U);
}
TEST(FramerateController, DoNotDropFrameWhenTimestampJumpsBackward) {
FramerateController framerate_controller(30);
ASSERT_FALSE(framerate_controller.DropFrame(66));
framerate_controller.AddFrame(66);
EXPECT_FALSE(framerate_controller.DropFrame(33));
}
TEST(FramerateController, DropFrameIfItIsTooCloseToPreviousFrame) {
FramerateController framerate_controller(30);
ASSERT_FALSE(framerate_controller.DropFrame(33));
framerate_controller.AddFrame(33);
EXPECT_TRUE(framerate_controller.DropFrame(34));
}
TEST(FramerateController, FrameDroppingStartsFromSecondInputFrame) {
const float input_framerate_fps = 23;
const float target_framerate_fps = 19;
const uint32_t input_frame_duration_ms =
static_cast<uint32_t>(1000 / input_framerate_fps);
FramerateController framerate_controller(target_framerate_fps);
ASSERT_FALSE(framerate_controller.DropFrame(1 * input_frame_duration_ms));
framerate_controller.AddFrame(1 * input_frame_duration_ms);
EXPECT_TRUE(framerate_controller.DropFrame(2 * input_frame_duration_ms));
}
} // namespace webrtc