Use high QP threshold for HW VP8 encoder frame downscaling.
Before HW VP8 downscaling was triggered by frame drops only.
Also reset the encoder when it drops large amount of frames in a row.
BUG=b/26504665
Review URL: https://codereview.webrtc.org/1592883004
Cr-Commit-Position: refs/heads/master@{#11406}
diff --git a/talk/app/webrtc/java/jni/androidmediaencoder_jni.cc b/talk/app/webrtc/java/jni/androidmediaencoder_jni.cc
index 4452e34..05e8145 100644
--- a/talk/app/webrtc/java/jni/androidmediaencoder_jni.cc
+++ b/talk/app/webrtc/java/jni/androidmediaencoder_jni.cc
@@ -78,7 +78,9 @@
#define MAX_ENCODER_Q_SIZE 2
// Maximum allowed latency in ms.
#define MAX_ENCODER_LATENCY_MS 70
-
+// Maximum amount of dropped frames caused by full encoder queue - exceeding
+// this threshold means that encoder probably got stuck and need to be reset.
+#define ENCODER_STALL_FRAMEDROP_THRESHOLD 60
// Logging macros.
#define TAG_ENCODER "MediaCodecVideoEncoder"
@@ -227,7 +229,9 @@
int64_t current_timestamp_us_; // Current frame timestamps in us.
int frames_received_; // Number of frames received by encoder.
int frames_encoded_; // Number of frames encoded by encoder.
- int frames_dropped_; // Number of frames dropped by encoder.
+ int frames_dropped_media_encoder_; // Number of frames dropped by encoder.
+ // Number of dropped frames caused by full queue.
+ int consecutive_full_queue_frame_drops_;
int frames_in_queue_; // Number of frames in encoder queue.
int64_t start_time_ms_; // Start time for statistics.
int current_frames_; // Number of frames in the current statistics interval.
@@ -385,18 +389,15 @@
// always = 127. Note that in SW, QP is that of the user-level range [0,
// 63].
const int kMaxQp = 127;
- // TODO(pbos): Investigate whether high-QP thresholds make sense for VP8.
- // This effectively disables high QP as VP8 QP can't go above this
- // threshold.
- const int kDisabledBadQpThreshold = kMaxQp + 1;
- quality_scaler_.Init(kMaxQp / kLowQpThresholdDenominator,
- kDisabledBadQpThreshold, true);
+ const int kBadQpThreshold = 95;
+ quality_scaler_.Init(
+ kMaxQp / kLowQpThresholdDenominator, kBadQpThreshold, false);
} else if (codecType_ == kVideoCodecH264) {
// H264 QP is in the range [0, 51].
const int kMaxQp = 51;
const int kBadQpThreshold = 40;
- quality_scaler_.Init(kMaxQp / kLowQpThresholdDenominator, kBadQpThreshold,
- false);
+ quality_scaler_.Init(
+ kMaxQp / kLowQpThresholdDenominator, kBadQpThreshold, false);
} else {
// When adding codec support to additional hardware codecs, also configure
// their QP thresholds for scaling.
@@ -444,9 +445,6 @@
int32_t MediaCodecVideoEncoder::SetRates(uint32_t new_bit_rate,
uint32_t frame_rate) {
- if (scale_)
- quality_scaler_.ReportFramerate(frame_rate);
-
return codec_thread_->Invoke<int32_t>(
Bind(&MediaCodecVideoEncoder::SetRatesOnCodecThread,
this,
@@ -512,7 +510,8 @@
yuv_size_ = width_ * height_ * 3 / 2;
frames_received_ = 0;
frames_encoded_ = 0;
- frames_dropped_ = 0;
+ frames_dropped_media_encoder_ = 0;
+ consecutive_full_queue_frame_drops_ = 0;
frames_in_queue_ = 0;
current_timestamp_us_ = 0;
start_time_ms_ = GetCurrentTimeMs();
@@ -634,6 +633,7 @@
ALOGW << "Encoder drop frame - failed callback.";
drop_next_input_frame_ = false;
current_timestamp_us_ += rtc::kNumMicrosecsPerSec / last_set_fps_;
+ frames_dropped_media_encoder_++;
OnDroppedFrame();
return WEBRTC_VIDEO_CODEC_OK;
}
@@ -648,12 +648,22 @@
if (frames_in_queue_ > MAX_ENCODER_Q_SIZE ||
encoder_latency_ms > MAX_ENCODER_LATENCY_MS) {
ALOGD << "Drop frame - encoder is behind by " << encoder_latency_ms <<
- " ms. Q size: " << frames_in_queue_;
+ " ms. Q size: " << frames_in_queue_ << ". Consecutive drops: " <<
+ consecutive_full_queue_frame_drops_;
current_timestamp_us_ += rtc::kNumMicrosecsPerSec / last_set_fps_;
+ consecutive_full_queue_frame_drops_++;
+ if (consecutive_full_queue_frame_drops_ >=
+ ENCODER_STALL_FRAMEDROP_THRESHOLD) {
+ ALOGE << "Encoder got stuck. Reset.";
+ ResetCodecOnCodecThread();
+ return WEBRTC_VIDEO_CODEC_ERROR;
+ }
+ frames_dropped_media_encoder_++;
OnDroppedFrame();
return WEBRTC_VIDEO_CODEC_OK;
}
}
+ consecutive_full_queue_frame_drops_ = 0;
VideoFrame input_frame = frame;
if (scale_) {
@@ -695,8 +705,10 @@
if (j_input_buffer_index == -1) {
// Video codec falls behind - no input buffer available.
ALOGW << "Encoder drop frame - no input buffers available";
- current_timestamp_us_ += rtc::kNumMicrosecsPerSec / last_set_fps_;
frame_rtc_times_ms_.erase(frame_rtc_times_ms_.begin());
+ current_timestamp_us_ += rtc::kNumMicrosecsPerSec / last_set_fps_;
+ frames_dropped_media_encoder_++;
+ OnDroppedFrame();
return WEBRTC_VIDEO_CODEC_OK; // TODO(fischman): see webrtc bug 2887.
}
if (j_input_buffer_index == -2) {
@@ -827,7 +839,7 @@
JNIEnv* jni = AttachCurrentThreadIfNeeded();
ALOGD << "EncoderReleaseOnCodecThread: Frames received: " <<
frames_received_ << ". Encoded: " << frames_encoded_ <<
- ". Dropped: " << frames_dropped_;
+ ". Dropped: " << frames_dropped_media_encoder_;
ScopedLocalRefFrame local_ref_frame(jni);
for (size_t i = 0; i < input_buffers_.size(); ++i)
jni->DeleteGlobalRef(input_buffers_[i]);
@@ -850,6 +862,9 @@
last_set_fps_ == frame_rate) {
return WEBRTC_VIDEO_CODEC_OK;
}
+ if (scale_) {
+ quality_scaler_.ReportFramerate(frame_rate);
+ }
JNIEnv* jni = AttachCurrentThreadIfNeeded();
ScopedLocalRefFrame local_ref_frame(jni);
if (new_bit_rate > 0) {
@@ -1135,7 +1150,6 @@
}
void MediaCodecVideoEncoder::OnDroppedFrame() {
- frames_dropped_++;
// Report dropped frame to quality_scaler_.
if (scale_)
quality_scaler_.ReportDroppedFrame();