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Remove latency-based frame dropping on Android.

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This logic currently prevents loopback calls on Nexus 5X when it's
slightly overloaded to maintain input framerate since encoding at ~25fps
with one framedrop results in >70ms between frames naturally.

With this change applied Nexus 5X can maintain ~25fps both in and out
without building excessive latency (>2 frames) (this is now covered by
CPU adaptation outside the codec wrapper).

BUG=
R=glaznev@webrtc.org

Review URL: https://codereview.webrtc.org/1854413004 .

Cr-Commit-Position: refs/heads/master@{#12317}
This commit is contained in:
Peter Boström
2016-04-11 16:42:40 +02:00
parent c7bdf8a729
commit f7704d197b
1 changed files with 17 additions and 24 deletions
Download Patch File Download Diff File Expand all files Collapse all files

41
webrtc/api/java/jni/androidmediaencoder_jni.cc
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@ -62,8 +62,6 @@ namespace webrtc_jni {
#define MAX_ALLOWED_VIDEO_FPS 60 #define MAX_ALLOWED_VIDEO_FPS 60
// Maximum allowed frames in encoder input queue. // Maximum allowed frames in encoder input queue.
#define MAX_ENCODER_Q_SIZE 2 #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 // Maximum amount of dropped frames caused by full encoder queue - exceeding
// this threshold means that encoder probably got stuck and need to be reset. // this threshold means that encoder probably got stuck and need to be reset.
#define ENCODER_STALL_FRAMEDROP_THRESHOLD 60 #define ENCODER_STALL_FRAMEDROP_THRESHOLD 60
@ -642,29 +640,24 @@ int32_t MediaCodecVideoEncoder::EncodeOnCodecThread(
RTC_CHECK(frame_types->size() == 1) << "Unexpected stream count"; RTC_CHECK(frame_types->size() == 1) << "Unexpected stream count";
// Check if we accumulated too many frames in encoder input buffers // Check if we accumulated too many frames in encoder input buffers and drop
// or the encoder latency exceeds 70 ms and drop frame if so. // frame if so.
if (frames_in_queue_ > 0 && last_input_timestamp_ms_ >= 0) { if (frames_in_queue_ > MAX_ENCODER_Q_SIZE) {
int encoder_latency_ms = last_input_timestamp_ms_ - ALOGD << "Already " << frames_in_queue_ << " frames in the queue, dropping"
last_output_timestamp_ms_; << ". TS: " << (int)(current_timestamp_us_ / 1000)
if (frames_in_queue_ > MAX_ENCODER_Q_SIZE || << ". Fps: " << last_set_fps_
encoder_latency_ms > MAX_ENCODER_LATENCY_MS) { << ". Consecutive drops: " << consecutive_full_queue_frame_drops_;
ALOGD << "Drop frame - encoder is behind by " << encoder_latency_ms << current_timestamp_us_ += rtc::kNumMicrosecsPerSec / last_set_fps_;
" ms. Q size: " << frames_in_queue_ << ". TS: " << consecutive_full_queue_frame_drops_++;
(int)(current_timestamp_us_ / 1000) << ". Fps: " << last_set_fps_ << if (consecutive_full_queue_frame_drops_ >=
". Consecutive drops: " << consecutive_full_queue_frame_drops_ ; ENCODER_STALL_FRAMEDROP_THRESHOLD) {
current_timestamp_us_ += rtc::kNumMicrosecsPerSec / last_set_fps_; ALOGE << "Encoder got stuck. Reset.";
consecutive_full_queue_frame_drops_++; ResetCodecOnCodecThread();
if (consecutive_full_queue_frame_drops_ >= return WEBRTC_VIDEO_CODEC_ERROR;
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;
} }
frames_dropped_media_encoder_++;
OnDroppedFrame();
return WEBRTC_VIDEO_CODEC_OK;
} }
consecutive_full_queue_frame_drops_ = 0; consecutive_full_queue_frame_drops_ = 0;