Bitrate controller for VideoToolbox encoder.

Also fixes a crash on encoder Release.

BUG=webrtc:4081

Review URL: https://codereview.webrtc.org/1660963002

Cr-Commit-Position: refs/heads/master@{#11729}
This commit is contained in:
tkchin
2016-02-23 22:49:42 -08:00
committed by Commit bot
parent 0ed85b2ee3
commit f75d008235
21 changed files with 580 additions and 79 deletions

View File

@ -138,6 +138,8 @@ static_library("rtc_base_approved") {
"platform_thread_types.h",
"random.cc",
"random.h",
"rate_statistics.cc",
"rate_statistics.h",
"refcount.h",
"safe_conversions.h",
"safe_conversions_impl.h",

View File

@ -106,6 +106,8 @@
'platform_thread_types.h',
'random.cc',
'random.h',
'rate_statistics.cc',
'rate_statistics.h',
'ratetracker.cc',
'ratetracker.h',
'refcount.h',

View File

@ -85,6 +85,7 @@
'proxy_unittest.cc',
'proxydetect_unittest.cc',
'random_unittest.cc',
'rate_statistics_unittest.cc',
'ratelimiter_unittest.cc',
'ratetracker_unittest.cc',
'referencecountedsingletonfactory_unittest.cc',

View File

@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/modules/remote_bitrate_estimator/rate_statistics.h"
#include "webrtc/base/rate_statistics.h"
#include <assert.h>
@ -20,11 +20,9 @@ RateStatistics::RateStatistics(uint32_t window_size_ms, float scale)
accumulated_count_(0),
oldest_time_(0),
oldest_index_(0),
scale_(scale / (num_buckets_ - 1)) {
}
scale_(scale / (num_buckets_ - 1)) {}
RateStatistics::~RateStatistics() {
}
RateStatistics::~RateStatistics() {}
void RateStatistics::Reset() {
accumulated_count_ = 0;

View File

@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_RATE_STATISTICS_H_
#define WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_RATE_STATISTICS_H_
#ifndef WEBRTC_BASE_RATE_STATISTICS_H_
#define WEBRTC_BASE_RATE_STATISTICS_H_
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/typedefs.h"
@ -50,4 +50,4 @@ class RateStatistics {
};
} // namespace webrtc
#endif // WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_RATE_STATISTICS_H_
#endif // WEBRTC_BASE_RATE_STATISTICS_H_

View File

@ -9,7 +9,7 @@
*/
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/remote_bitrate_estimator/rate_statistics.h"
#include "webrtc/base/rate_statistics.h"
namespace {

View File

@ -277,7 +277,6 @@
'remote_bitrate_estimator/include/mock/mock_remote_bitrate_estimator.h',
'remote_bitrate_estimator/inter_arrival_unittest.cc',
'remote_bitrate_estimator/overuse_detector_unittest.cc',
'remote_bitrate_estimator/rate_statistics_unittest.cc',
'remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time_unittest.cc',
'remote_bitrate_estimator/remote_bitrate_estimator_single_stream_unittest.cc',
'remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc',
@ -360,6 +359,7 @@
'video_coding/codecs/vp8/simulcast_unittest.h',
'video_coding/codecs/vp9/screenshare_layers_unittest.cc',
'video_coding/include/mock/mock_vcm_callbacks.h',
'video_coding/bitrate_adjuster_unittest.cc',
'video_coding/decoding_state_unittest.cc',
'video_coding/jitter_buffer_unittest.cc',
'video_coding/jitter_estimator_tests.cc',

View File

@ -10,8 +10,6 @@ source_set("remote_bitrate_estimator") {
sources = [
"include/bwe_defines.h",
"include/remote_bitrate_estimator.h",
"rate_statistics.cc",
"rate_statistics.h",
]
configs += [ "../../:common_inherited_config" ]

View File

@ -17,12 +17,12 @@
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/random.h"
#include "webrtc/base/rate_statistics.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/remote_bitrate_estimator/inter_arrival.h"
#include "webrtc/modules/remote_bitrate_estimator/overuse_detector.h"
#include "webrtc/modules/remote_bitrate_estimator/overuse_estimator.h"
#include "webrtc/modules/remote_bitrate_estimator/rate_statistics.h"
#include "webrtc/test/field_trial.h"
namespace webrtc {

View File

@ -30,8 +30,6 @@
'overuse_detector.h',
'overuse_estimator.cc',
'overuse_estimator.h',
'rate_statistics.cc',
'rate_statistics.h',
'remote_bitrate_estimator_abs_send_time.cc',
'remote_bitrate_estimator_abs_send_time.h',
'remote_bitrate_estimator_single_stream.cc',

View File

@ -17,6 +17,7 @@
#include "webrtc/base/checks.h"
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/rate_statistics.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread_checker.h"
#include "webrtc/modules/remote_bitrate_estimator/aimd_rate_control.h"
@ -24,7 +25,6 @@
#include "webrtc/modules/remote_bitrate_estimator/inter_arrival.h"
#include "webrtc/modules/remote_bitrate_estimator/overuse_detector.h"
#include "webrtc/modules/remote_bitrate_estimator/overuse_estimator.h"
#include "webrtc/modules/remote_bitrate_estimator/rate_statistics.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
namespace webrtc {

View File

@ -14,9 +14,9 @@
#include <map>
#include <vector>
#include "webrtc/base/rate_statistics.h"
#include "webrtc/modules/remote_bitrate_estimator/aimd_rate_control.h"
#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "webrtc/modules/remote_bitrate_estimator/rate_statistics.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
namespace webrtc {

View File

@ -10,6 +10,8 @@ import("../../build/webrtc.gni")
source_set("video_coding") {
sources = [
"bitrate_adjuster.cc",
"bitrate_adjuster.h",
"codec_database.cc",
"codec_database.h",
"codec_timer.cc",

View File

@ -0,0 +1,160 @@
/*
* Copyright 2016 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/include/bitrate_adjuster.h"
#include <cmath>
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/system_wrappers/include/clock.h"
namespace webrtc {
// Update bitrate at most once every second.
const uint32_t BitrateAdjuster::kBitrateUpdateIntervalMs = 1000;
// Update bitrate at most once every 30 frames.
const uint32_t BitrateAdjuster::kBitrateUpdateFrameInterval = 30;
// 10 percent of original.
const float BitrateAdjuster::kBitrateTolerancePct = .1f;
const float BitrateAdjuster::kBytesPerMsToBitsPerSecond = 8 * 1000;
BitrateAdjuster::BitrateAdjuster(Clock* clock,
float min_adjusted_bitrate_pct,
float max_adjusted_bitrate_pct)
: clock_(clock),
min_adjusted_bitrate_pct_(min_adjusted_bitrate_pct),
max_adjusted_bitrate_pct_(max_adjusted_bitrate_pct),
bitrate_tracker_(1.5 * kBitrateUpdateIntervalMs,
kBytesPerMsToBitsPerSecond) {
Reset();
}
void BitrateAdjuster::SetTargetBitrateBps(uint32_t bitrate_bps) {
rtc::CritScope cs(&crit_);
// If the change in target bitrate is large, update the adjusted bitrate
// immediately since it's likely we have gained or lost a sizeable amount of
// bandwidth and we'll want to respond quickly.
// If the change in target bitrate fits within the existing tolerance of
// encoder output, wait for the next adjustment time to preserve
// existing penalties and not forcibly reset the adjusted bitrate to target.
// However, if we received many small deltas within an update time
// window and one of them exceeds the tolerance when compared to the last
// target we updated against, treat it as a large change in target bitrate.
if (!IsWithinTolerance(bitrate_bps, target_bitrate_bps_) ||
!IsWithinTolerance(bitrate_bps, last_adjusted_target_bitrate_bps_)) {
adjusted_bitrate_bps_ = bitrate_bps;
last_adjusted_target_bitrate_bps_ = bitrate_bps;
}
target_bitrate_bps_ = bitrate_bps;
}
uint32_t BitrateAdjuster::GetTargetBitrateBps() const {
rtc::CritScope cs(&crit_);
return target_bitrate_bps_;
}
uint32_t BitrateAdjuster::GetAdjustedBitrateBps() const {
rtc::CritScope cs(&crit_);
return adjusted_bitrate_bps_;
}
uint32_t BitrateAdjuster::GetEstimatedBitrateBps() {
rtc::CritScope cs(&crit_);
return bitrate_tracker_.Rate(clock_->TimeInMilliseconds());
}
void BitrateAdjuster::Update(size_t frame_size) {
rtc::CritScope cs(&crit_);
uint32_t current_time_ms = clock_->TimeInMilliseconds();
bitrate_tracker_.Update(frame_size, current_time_ms);
UpdateBitrate(current_time_ms);
}
bool BitrateAdjuster::IsWithinTolerance(uint32_t bitrate_bps,
uint32_t target_bitrate_bps) {
if (target_bitrate_bps == 0) {
return false;
}
float delta = std::abs(static_cast<float>(bitrate_bps) -
static_cast<float>(target_bitrate_bps));
float delta_pct = delta / target_bitrate_bps;
return delta_pct < kBitrateTolerancePct;
}
uint32_t BitrateAdjuster::GetMinAdjustedBitrateBps() const {
return min_adjusted_bitrate_pct_ * target_bitrate_bps_;
}
uint32_t BitrateAdjuster::GetMaxAdjustedBitrateBps() const {
return max_adjusted_bitrate_pct_ * target_bitrate_bps_;
}
// Only safe to call this after Update calls have stopped
void BitrateAdjuster::Reset() {
rtc::CritScope cs(&crit_);
target_bitrate_bps_ = 0;
adjusted_bitrate_bps_ = 0;
last_adjusted_target_bitrate_bps_ = 0;
last_bitrate_update_time_ms_ = 0;
frames_since_last_update_ = 0;
bitrate_tracker_.Reset();
}
void BitrateAdjuster::UpdateBitrate(uint32_t current_time_ms) {
uint32_t time_since_last_update_ms =
current_time_ms - last_bitrate_update_time_ms_;
// Don't attempt to update bitrate unless enough time and frames have passed.
++frames_since_last_update_;
if (time_since_last_update_ms < kBitrateUpdateIntervalMs ||
frames_since_last_update_ < kBitrateUpdateFrameInterval) {
return;
}
float estimated_bitrate_bps = bitrate_tracker_.Rate(current_time_ms);
float target_bitrate_bps = target_bitrate_bps_;
float error = target_bitrate_bps - estimated_bitrate_bps;
// Adjust if we've overshot by any amount or if we've undershot too much.
if (estimated_bitrate_bps > target_bitrate_bps ||
error > kBitrateTolerancePct * target_bitrate_bps) {
// Adjust the bitrate by a fraction of the error.
float adjustment = .5 * error;
float adjusted_bitrate_bps = target_bitrate_bps + adjustment;
// Clamp the adjustment.
float min_bitrate_bps = GetMinAdjustedBitrateBps();
float max_bitrate_bps = GetMaxAdjustedBitrateBps();
adjusted_bitrate_bps = std::max(adjusted_bitrate_bps, min_bitrate_bps);
adjusted_bitrate_bps = std::min(adjusted_bitrate_bps, max_bitrate_bps);
// Set the adjustment if it's not already set.
float last_adjusted_bitrate_bps = adjusted_bitrate_bps_;
if (adjusted_bitrate_bps != last_adjusted_bitrate_bps) {
LOG(LS_VERBOSE) << "Adjusting encoder bitrate:"
<< "\n target_bitrate:"
<< static_cast<uint32_t>(target_bitrate_bps)
<< "\n estimated_bitrate:"
<< static_cast<uint32_t>(estimated_bitrate_bps)
<< "\n last_adjusted_bitrate:"
<< static_cast<uint32_t>(last_adjusted_bitrate_bps)
<< "\n adjusted_bitrate:"
<< static_cast<uint32_t>(adjusted_bitrate_bps);
adjusted_bitrate_bps_ = adjusted_bitrate_bps;
}
}
last_bitrate_update_time_ms_ = current_time_ms;
frames_since_last_update_ = 0;
last_adjusted_target_bitrate_bps_ = target_bitrate_bps_;
}
} // namespace webrtc

View File

@ -0,0 +1,168 @@
/*
* Copyright 2016 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/gtest/include/gtest/gtest.h"
#include "webrtc/modules/video_coding/include/bitrate_adjuster.h"
#include "webrtc/system_wrappers/include/clock.h"
namespace webrtc {
class BitrateAdjusterTest : public ::testing::Test {
public:
BitrateAdjusterTest()
: clock_(0),
adjuster_(&clock_, kMinAdjustedBitratePct, kMaxAdjustedBitratePct) {}
// Simulate an output bitrate for one update cycle of BitrateAdjuster.
void SimulateBitrateBps(uint32_t bitrate_bps) {
const uint32_t update_interval_ms =
BitrateAdjuster::kBitrateUpdateIntervalMs;
const uint32_t update_frame_interval =
BitrateAdjuster::kBitrateUpdateFrameInterval;
// Round up frame interval so we get one cycle passes.
const uint32_t frame_interval_ms =
(update_interval_ms + update_frame_interval - 1) /
update_frame_interval;
const size_t frame_size_bytes =
(bitrate_bps * frame_interval_ms) / (8 * 1000);
for (size_t i = 0; i < update_frame_interval; ++i) {
clock_.AdvanceTimeMilliseconds(frame_interval_ms);
adjuster_.Update(frame_size_bytes);
}
}
uint32_t GetTargetBitrateBpsPct(float pct) {
return pct * adjuster_.GetTargetBitrateBps();
}
void VerifyAdjustment() {
// The adjusted bitrate should be between the estimated bitrate and the
// target bitrate within clamp.
uint32_t target_bitrate_bps = adjuster_.GetTargetBitrateBps();
uint32_t adjusted_bitrate_bps = adjuster_.GetAdjustedBitrateBps();
uint32_t estimated_bitrate_bps = adjuster_.GetEstimatedBitrateBps();
uint32_t adjusted_lower_bound_bps =
GetTargetBitrateBpsPct(kMinAdjustedBitratePct);
uint32_t adjusted_upper_bound_bps =
GetTargetBitrateBpsPct(kMaxAdjustedBitratePct);
EXPECT_LE(adjusted_bitrate_bps, adjusted_upper_bound_bps);
EXPECT_GE(adjusted_bitrate_bps, adjusted_lower_bound_bps);
if (estimated_bitrate_bps > target_bitrate_bps) {
EXPECT_LT(adjusted_bitrate_bps, target_bitrate_bps);
}
}
protected:
static const float kMinAdjustedBitratePct;
static const float kMaxAdjustedBitratePct;
SimulatedClock clock_;
BitrateAdjuster adjuster_;
};
const float BitrateAdjusterTest::kMinAdjustedBitratePct = .5f;
const float BitrateAdjusterTest::kMaxAdjustedBitratePct = .95f;
TEST_F(BitrateAdjusterTest, VaryingBitrates) {
const uint32_t target_bitrate_bps = 640000;
adjuster_.SetTargetBitrateBps(target_bitrate_bps);
// Grossly overshoot for a little while. Adjusted bitrate should decrease.
uint32_t actual_bitrate_bps = 2 * target_bitrate_bps;
uint32_t last_adjusted_bitrate_bps = 0;
uint32_t adjusted_bitrate_bps = 0;
SimulateBitrateBps(actual_bitrate_bps);
VerifyAdjustment();
last_adjusted_bitrate_bps = adjuster_.GetAdjustedBitrateBps();
SimulateBitrateBps(actual_bitrate_bps);
VerifyAdjustment();
adjusted_bitrate_bps = adjuster_.GetAdjustedBitrateBps();
EXPECT_LT(adjusted_bitrate_bps, last_adjusted_bitrate_bps);
last_adjusted_bitrate_bps = adjusted_bitrate_bps;
// After two cycles we should've stabilized and hit the lower bound.
EXPECT_EQ(GetTargetBitrateBpsPct(kMinAdjustedBitratePct),
adjusted_bitrate_bps);
// Simulate encoder settling down. Adjusted bitrate should increase.
SimulateBitrateBps(target_bitrate_bps);
adjusted_bitrate_bps = adjuster_.GetAdjustedBitrateBps();
VerifyAdjustment();
EXPECT_GT(adjusted_bitrate_bps, last_adjusted_bitrate_bps);
last_adjusted_bitrate_bps = adjusted_bitrate_bps;
SimulateBitrateBps(target_bitrate_bps);
adjusted_bitrate_bps = adjuster_.GetAdjustedBitrateBps();
VerifyAdjustment();
EXPECT_GT(adjusted_bitrate_bps, last_adjusted_bitrate_bps);
last_adjusted_bitrate_bps = adjusted_bitrate_bps;
// After two cycles we should've stabilized and hit the upper bound.
EXPECT_EQ(GetTargetBitrateBpsPct(kMaxAdjustedBitratePct),
adjusted_bitrate_bps);
}
// Tests that large changes in target bitrate will result in immediate change
// in adjusted bitrate.
TEST_F(BitrateAdjusterTest, LargeTargetDelta) {
uint32_t target_bitrate_bps = 640000;
adjuster_.SetTargetBitrateBps(target_bitrate_bps);
EXPECT_EQ(target_bitrate_bps, adjuster_.GetAdjustedBitrateBps());
float delta_pct = BitrateAdjuster::kBitrateTolerancePct * 2;
target_bitrate_bps = (1 + delta_pct) * target_bitrate_bps;
adjuster_.SetTargetBitrateBps(target_bitrate_bps);
EXPECT_EQ(target_bitrate_bps, adjuster_.GetAdjustedBitrateBps());
target_bitrate_bps = (1 - delta_pct) * target_bitrate_bps;
adjuster_.SetTargetBitrateBps(target_bitrate_bps);
EXPECT_EQ(target_bitrate_bps, adjuster_.GetAdjustedBitrateBps());
}
// Tests that small changes in target bitrate within tolerance will not affect
// adjusted bitrate immediately.
TEST_F(BitrateAdjusterTest, SmallTargetDelta) {
const uint32_t initial_target_bitrate_bps = 640000;
uint32_t target_bitrate_bps = initial_target_bitrate_bps;
adjuster_.SetTargetBitrateBps(target_bitrate_bps);
EXPECT_EQ(initial_target_bitrate_bps, adjuster_.GetAdjustedBitrateBps());
float delta_pct = BitrateAdjuster::kBitrateTolerancePct / 2;
target_bitrate_bps = (1 + delta_pct) * target_bitrate_bps;
adjuster_.SetTargetBitrateBps(target_bitrate_bps);
EXPECT_EQ(initial_target_bitrate_bps, adjuster_.GetAdjustedBitrateBps());
target_bitrate_bps = (1 - delta_pct) * target_bitrate_bps;
adjuster_.SetTargetBitrateBps(target_bitrate_bps);
EXPECT_EQ(initial_target_bitrate_bps, adjuster_.GetAdjustedBitrateBps());
}
TEST_F(BitrateAdjusterTest, SmallTargetDeltaOverflow) {
const uint32_t initial_target_bitrate_bps = 640000;
uint32_t target_bitrate_bps = initial_target_bitrate_bps;
adjuster_.SetTargetBitrateBps(target_bitrate_bps);
EXPECT_EQ(initial_target_bitrate_bps, adjuster_.GetAdjustedBitrateBps());
float delta_pct = BitrateAdjuster::kBitrateTolerancePct / 2;
target_bitrate_bps = (1 + delta_pct) * target_bitrate_bps;
adjuster_.SetTargetBitrateBps(target_bitrate_bps);
EXPECT_EQ(initial_target_bitrate_bps, adjuster_.GetAdjustedBitrateBps());
// 1.05 * 1.05 is 1.1 which is greater than tolerance for the initial target
// bitrate. Since we didn't advance the clock the adjuster never updated.
target_bitrate_bps = (1 + delta_pct) * target_bitrate_bps;
adjuster_.SetTargetBitrateBps(target_bitrate_bps);
EXPECT_EQ(target_bitrate_bps, adjuster_.GetAdjustedBitrateBps());
}
} // namespace webrtc

View File

@ -164,6 +164,10 @@ int H264VideoToolboxDecoder::RegisterDecodeCompleteCallback(
}
int H264VideoToolboxDecoder::Release() {
// Need to invalidate the session so that callbacks no longer occur and it
// is safe to null out the callback.
DestroyDecompressionSession();
SetVideoFormat(nullptr);
callback_ = nullptr;
return WEBRTC_VIDEO_CODEC_OK;
}

View File

@ -21,6 +21,7 @@
#include "webrtc/base/logging.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_nalu.h"
#include "webrtc/system_wrappers/include/clock.h"
namespace internal {
@ -66,6 +67,22 @@ void SetVTSessionProperty(VTSessionRef session,
}
}
// Convenience function for setting a VT property.
void SetVTSessionProperty(VTSessionRef session,
CFStringRef key,
uint32_t value) {
int64_t value_64 = value;
CFNumberRef cfNum =
CFNumberCreate(kCFAllocatorDefault, kCFNumberSInt64Type, &value_64);
OSStatus status = VTSessionSetProperty(session, key, cfNum);
CFRelease(cfNum);
if (status != noErr) {
std::string key_string = CFStringToString(key);
LOG(LS_ERROR) << "VTSessionSetProperty failed to set: " << key_string
<< " to " << value << ": " << status;
}
}
// Convenience function for setting a VT property.
void SetVTSessionProperty(VTSessionRef session, CFStringRef key, bool value) {
CFBooleanRef cf_bool = (value) ? kCFBooleanTrue : kCFBooleanFalse;
@ -93,20 +110,21 @@ void SetVTSessionProperty(VTSessionRef session,
// Struct that we pass to the encoder per frame to encode. We receive it again
// in the encoder callback.
struct FrameEncodeParams {
FrameEncodeParams(webrtc::EncodedImageCallback* cb,
FrameEncodeParams(webrtc::H264VideoToolboxEncoder* e,
const webrtc::CodecSpecificInfo* csi,
int32_t w,
int32_t h,
int64_t rtms,
uint32_t ts)
: callback(cb), width(w), height(h), render_time_ms(rtms), timestamp(ts) {
: encoder(e), width(w), height(h), render_time_ms(rtms), timestamp(ts) {
if (csi) {
codec_specific_info = *csi;
} else {
codec_specific_info.codecType = webrtc::kVideoCodecH264;
}
}
webrtc::EncodedImageCallback* callback;
webrtc::H264VideoToolboxEncoder* encoder;
webrtc::CodecSpecificInfo codec_specific_info;
int32_t width;
int32_t height;
@ -153,7 +171,7 @@ bool CopyVideoFrameToPixelBuffer(const webrtc::VideoFrame& frame,
}
// This is the callback function that VideoToolbox calls when encode is
// complete.
// complete. From inspection this happens on its own queue.
void VTCompressionOutputCallback(void* encoder,
void* params,
OSStatus status,
@ -161,54 +179,27 @@ void VTCompressionOutputCallback(void* encoder,
CMSampleBufferRef sample_buffer) {
rtc::scoped_ptr<FrameEncodeParams> encode_params(
reinterpret_cast<FrameEncodeParams*>(params));
if (status != noErr) {
LOG(LS_ERROR) << "H264 encoding failed.";
return;
}
if (info_flags & kVTEncodeInfo_FrameDropped) {
LOG(LS_INFO) << "H264 encode dropped frame.";
}
bool is_keyframe = false;
CFArrayRef attachments =
CMSampleBufferGetSampleAttachmentsArray(sample_buffer, 0);
if (attachments != nullptr && CFArrayGetCount(attachments)) {
CFDictionaryRef attachment =
static_cast<CFDictionaryRef>(CFArrayGetValueAtIndex(attachments, 0));
is_keyframe =
!CFDictionaryContainsKey(attachment, kCMSampleAttachmentKey_NotSync);
}
// Convert the sample buffer into a buffer suitable for RTP packetization.
// TODO(tkchin): Allocate buffers through a pool.
rtc::scoped_ptr<rtc::Buffer> buffer(new rtc::Buffer());
rtc::scoped_ptr<webrtc::RTPFragmentationHeader> header;
if (!H264CMSampleBufferToAnnexBBuffer(sample_buffer, is_keyframe,
buffer.get(), header.accept())) {
return;
}
webrtc::EncodedImage frame(buffer->data(), buffer->size(), buffer->size());
frame._encodedWidth = encode_params->width;
frame._encodedHeight = encode_params->height;
frame._completeFrame = true;
frame._frameType =
is_keyframe ? webrtc::kVideoFrameKey : webrtc::kVideoFrameDelta;
frame.capture_time_ms_ = encode_params->render_time_ms;
frame._timeStamp = encode_params->timestamp;
int result = encode_params->callback->Encoded(
frame, &(encode_params->codec_specific_info), header.get());
if (result != 0) {
LOG(LS_ERROR) << "Encoded callback failed: " << result;
}
encode_params->encoder->OnEncodedFrame(
status, info_flags, sample_buffer, encode_params->codec_specific_info,
encode_params->width, encode_params->height,
encode_params->render_time_ms, encode_params->timestamp);
}
} // namespace internal
namespace webrtc {
// .5 is set as a mininum to prevent overcompensating for large temporary
// overshoots. We don't want to degrade video quality too badly.
// .95 is set to prevent oscillations. When a lower bitrate is set on the
// encoder than previously set, its output seems to have a brief period of
// drastically reduced bitrate, so we want to avoid that. In steady state
// conditions, 0.95 seems to give us better overall bitrate over long periods
// of time.
H264VideoToolboxEncoder::H264VideoToolboxEncoder()
: callback_(nullptr), compression_session_(nullptr) {}
: callback_(nullptr),
compression_session_(nullptr),
bitrate_adjuster_(Clock::GetRealTimeClock(), .5, .95) {}
H264VideoToolboxEncoder::~H264VideoToolboxEncoder() {
DestroyCompressionSession();
@ -224,7 +215,8 @@ int H264VideoToolboxEncoder::InitEncode(const VideoCodec* codec_settings,
width_ = codec_settings->width;
height_ = codec_settings->height;
// We can only set average bitrate on the HW encoder.
bitrate_ = codec_settings->startBitrate * 1000;
target_bitrate_bps_ = codec_settings->startBitrate;
bitrate_adjuster_.SetTargetBitrateBps(target_bitrate_bps_);
// TODO(tkchin): Try setting payload size via
// kVTCompressionPropertyKey_MaxH264SliceBytes.
@ -287,8 +279,12 @@ int H264VideoToolboxEncoder::Encode(
}
rtc::scoped_ptr<internal::FrameEncodeParams> encode_params;
encode_params.reset(new internal::FrameEncodeParams(
callback_, codec_specific_info, width_, height_,
input_image.render_time_ms(), input_image.timestamp()));
this, codec_specific_info, width_, height_, input_image.render_time_ms(),
input_image.timestamp()));
// Update the bitrate if needed.
SetBitrateBps(bitrate_adjuster_.GetAdjustedBitrateBps());
VTCompressionSessionEncodeFrame(
compression_session_, pixel_buffer, presentation_time_stamp,
kCMTimeInvalid, frame_properties, encode_params.release(), nullptr);
@ -315,20 +311,20 @@ int H264VideoToolboxEncoder::SetChannelParameters(uint32_t packet_loss,
int H264VideoToolboxEncoder::SetRates(uint32_t new_bitrate_kbit,
uint32_t frame_rate) {
bitrate_ = new_bitrate_kbit * 1000;
if (compression_session_) {
internal::SetVTSessionProperty(compression_session_,
kVTCompressionPropertyKey_AverageBitRate,
bitrate_);
}
target_bitrate_bps_ = 1000 * new_bitrate_kbit;
bitrate_adjuster_.SetTargetBitrateBps(target_bitrate_bps_);
SetBitrateBps(bitrate_adjuster_.GetAdjustedBitrateBps());
return WEBRTC_VIDEO_CODEC_OK;
}
int H264VideoToolboxEncoder::Release() {
// Need to reset so that the session is invalidated and won't use the
// callback anymore. Do not remove callback until the session is invalidated
// since async encoder callbacks can occur until invalidation.
int ret = ResetCompressionSession();
callback_ = nullptr;
// Need to reset to that the session is invalidated and won't use the
// callback anymore.
return ResetCompressionSession();
return ret;
}
int H264VideoToolboxEncoder::ResetCompressionSession() {
@ -389,11 +385,10 @@ void H264VideoToolboxEncoder::ConfigureCompressionSession() {
internal::SetVTSessionProperty(compression_session_,
kVTCompressionPropertyKey_ProfileLevel,
kVTProfileLevel_H264_Baseline_AutoLevel);
internal::SetVTSessionProperty(
compression_session_, kVTCompressionPropertyKey_AverageBitRate, bitrate_);
internal::SetVTSessionProperty(compression_session_,
kVTCompressionPropertyKey_AllowFrameReordering,
false);
SetEncoderBitrateBps(target_bitrate_bps_);
// TODO(tkchin): Look at entropy mode and colorspace matrices.
// TODO(tkchin): Investigate to see if there's any way to make this work.
// May need it to interop with Android. Currently this call just fails.
@ -423,6 +418,73 @@ const char* H264VideoToolboxEncoder::ImplementationName() const {
return "VideoToolbox";
}
void H264VideoToolboxEncoder::SetBitrateBps(uint32_t bitrate_bps) {
if (encoder_bitrate_bps_ != bitrate_bps) {
SetEncoderBitrateBps(bitrate_bps);
}
}
void H264VideoToolboxEncoder::SetEncoderBitrateBps(uint32_t bitrate_bps) {
if (compression_session_) {
internal::SetVTSessionProperty(compression_session_,
kVTCompressionPropertyKey_AverageBitRate,
bitrate_bps);
encoder_bitrate_bps_ = bitrate_bps;
}
}
void H264VideoToolboxEncoder::OnEncodedFrame(
OSStatus status,
VTEncodeInfoFlags info_flags,
CMSampleBufferRef sample_buffer,
CodecSpecificInfo codec_specific_info,
int32_t width,
int32_t height,
int64_t render_time_ms,
uint32_t timestamp) {
if (status != noErr) {
LOG(LS_ERROR) << "H264 encode failed.";
return;
}
if (info_flags & kVTEncodeInfo_FrameDropped) {
LOG(LS_INFO) << "H264 encode dropped frame.";
}
bool is_keyframe = false;
CFArrayRef attachments =
CMSampleBufferGetSampleAttachmentsArray(sample_buffer, 0);
if (attachments != nullptr && CFArrayGetCount(attachments)) {
CFDictionaryRef attachment =
static_cast<CFDictionaryRef>(CFArrayGetValueAtIndex(attachments, 0));
is_keyframe =
!CFDictionaryContainsKey(attachment, kCMSampleAttachmentKey_NotSync);
}
// Convert the sample buffer into a buffer suitable for RTP packetization.
// TODO(tkchin): Allocate buffers through a pool.
rtc::scoped_ptr<rtc::Buffer> buffer(new rtc::Buffer());
rtc::scoped_ptr<webrtc::RTPFragmentationHeader> header;
if (!H264CMSampleBufferToAnnexBBuffer(sample_buffer, is_keyframe,
buffer.get(), header.accept())) {
return;
}
webrtc::EncodedImage frame(buffer->data(), buffer->size(), buffer->size());
frame._encodedWidth = width;
frame._encodedHeight = height;
frame._completeFrame = true;
frame._frameType =
is_keyframe ? webrtc::kVideoFrameKey : webrtc::kVideoFrameDelta;
frame.capture_time_ms_ = render_time_ms;
frame._timeStamp = timestamp;
int result = callback_->Encoded(frame, &codec_specific_info, header.get());
if (result != 0) {
LOG(LS_ERROR) << "Encode callback failed: " << result;
return;
}
bitrate_adjuster_.Update(frame._size);
}
} // namespace webrtc
#endif // defined(WEBRTC_VIDEO_TOOLBOX_SUPPORTED)

View File

@ -13,6 +13,7 @@
#define WEBRTC_MODULES_VIDEO_CODING_CODECS_H264_H264_VIDEO_TOOLBOX_ENCODER_H_
#include "webrtc/modules/video_coding/codecs/h264/include/h264.h"
#include "webrtc/modules/video_coding/include/bitrate_adjuster.h"
#if defined(WEBRTC_VIDEO_TOOLBOX_SUPPORTED)
@ -50,14 +51,27 @@ class H264VideoToolboxEncoder : public H264Encoder {
const char* ImplementationName() const override;
void OnEncodedFrame(OSStatus status,
VTEncodeInfoFlags info_flags,
CMSampleBufferRef sample_buffer,
CodecSpecificInfo codec_specific_info,
int32_t width,
int32_t height,
int64_t render_time_ms,
uint32_t timestamp);
private:
int ResetCompressionSession();
void ConfigureCompressionSession();
void DestroyCompressionSession();
void SetBitrateBps(uint32_t bitrate_bps);
void SetEncoderBitrateBps(uint32_t bitrate_bps);
webrtc::EncodedImageCallback* callback_;
EncodedImageCallback* callback_;
VTCompressionSessionRef compression_session_;
int32_t bitrate_; // Bitrate in bits per second.
BitrateAdjuster bitrate_adjuster_;
uint32_t target_bitrate_bps_;
uint32_t encoder_bitrate_bps_;
int32_t width_;
int32_t height_;
}; // H264VideoToolboxEncoder

View File

@ -0,0 +1,90 @@
/*
* Copyright 2016 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_INCLUDE_BITRATE_ADJUSTER_H_
#define WEBRTC_MODULES_VIDEO_CODING_INCLUDE_BITRATE_ADJUSTER_H_
#include <functional>
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/gtest_prod_util.h"
#include "webrtc/base/rate_statistics.h"
namespace webrtc {
class Clock;
// Certain hardware encoders tend to consistently overshoot the bitrate that
// they are configured to encode at. This class estimates an adjusted bitrate
// that when set on the encoder will produce the desired bitrate.
class BitrateAdjuster {
public:
// min_adjusted_bitrate_pct and max_adjusted_bitrate_pct are the lower and
// upper bound outputted adjusted bitrates as a percentage of the target
// bitrate.
BitrateAdjuster(Clock* clock,
float min_adjusted_bitrate_pct,
float max_adjusted_bitrate_pct);
virtual ~BitrateAdjuster() {}
static const uint32_t kBitrateUpdateIntervalMs;
static const uint32_t kBitrateUpdateFrameInterval;
static const float kBitrateTolerancePct;
static const float kBytesPerMsToBitsPerSecond;
// Sets the desired bitrate in bps (bits per second).
// Should be called at least once before Update.
void SetTargetBitrateBps(uint32_t bitrate_bps);
uint32_t GetTargetBitrateBps() const;
// Returns the adjusted bitrate in bps.
uint32_t GetAdjustedBitrateBps() const;
// Returns what we think the current bitrate is.
uint32_t GetEstimatedBitrateBps();
// This should be called after each frame is encoded. The timestamp at which
// it is called is used to estimate the output bitrate of the encoder.
// Should be called from only one thread.
void Update(size_t frame_size);
private:
// Returns true if the bitrate is within kBitrateTolerancePct of bitrate_bps.
bool IsWithinTolerance(uint32_t bitrate_bps, uint32_t target_bitrate_bps);
// Returns smallest possible adjusted value.
uint32_t GetMinAdjustedBitrateBps() const EXCLUSIVE_LOCKS_REQUIRED(crit_);
// Returns largest possible adjusted value.
uint32_t GetMaxAdjustedBitrateBps() const EXCLUSIVE_LOCKS_REQUIRED(crit_);
void Reset();
void UpdateBitrate(uint32_t current_time_ms) EXCLUSIVE_LOCKS_REQUIRED(crit_);
rtc::CriticalSection crit_;
Clock* const clock_;
const float min_adjusted_bitrate_pct_;
const float max_adjusted_bitrate_pct_;
// The bitrate we want.
volatile uint32_t target_bitrate_bps_ GUARDED_BY(crit_);
// The bitrate we use to get what we want.
volatile uint32_t adjusted_bitrate_bps_ GUARDED_BY(crit_);
// The target bitrate that the adjusted bitrate was computed from.
volatile uint32_t last_adjusted_target_bitrate_bps_ GUARDED_BY(crit_);
// Used to estimate bitrate.
RateStatistics bitrate_tracker_ GUARDED_BY(crit_);
// The last time we tried to adjust the bitrate.
uint32_t last_bitrate_update_time_ms_ GUARDED_BY(crit_);
// The number of frames since the last time we tried to adjust the bitrate.
uint32_t frames_since_last_update_ GUARDED_BY(crit_);
};
} // namespace webrtc
#endif // WEBRTC_MODULES_VIDEO_CODING_INCLUDE_BITRATE_ADJUSTER_H_

View File

@ -22,6 +22,7 @@
],
'sources': [
# interfaces
'include/bitrate_adjuster.h',
'include/video_coding.h',
'include/video_coding_defines.h',
@ -54,6 +55,7 @@
'video_coding_impl.h',
# sources
'bitrate_adjuster.cc',
'codec_database.cc',
'codec_timer.cc',
'content_metrics_processing.cc',

View File

@ -14,11 +14,11 @@
#include <string>
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/rate_statistics.h"
#include "webrtc/base/ratetracker.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/common_types.h"
#include "webrtc/frame_callback.h"
#include "webrtc/modules/remote_bitrate_estimator/rate_statistics.h"
#include "webrtc/modules/video_coding/include/video_coding_defines.h"
#include "webrtc/video/report_block_stats.h"
#include "webrtc/video/vie_channel.h"