This CL should reduce the number of timeouts for dequeueInputBuffer() which results in the log "MediaCodecVideo: dequeueInputBuffer error" followed by software fallback for VP8/VP9 and codec restart for H264.
A timeout always happen for dequeueInputBuffer() when frames_received_ > frames_decoded_ + num_input_buffers. The following code tries to drain the decoder before enqueuing more input buffers:
// Try to drain the decoder and wait until output is not too
// much behind the input.
if (frames_received_ > frames_decoded_ + max_pending_frames_) {
ALOGV("Received: %d. Decoded: %d. Wait for output...",
frames_received_, frames_decoded_);
if (!DeliverPendingOutputs(jni, kMediaCodecTimeoutMs,
true /* dropFrames */)) {
ALOGE << "DeliverPendingOutputs error";
return ProcessHWErrorOnCodecThread();
}
if (frames_received_ > frames_decoded_ + max_pending_frames_) {
ALOGE << "Output buffer dequeue timeout";
return ProcessHWErrorOnCodecThread();
}
...
}
However, for H264, |max_pending_frames_| can currently be larger than the number of input buffers so that the code above is never executed. This CL limits |max_pending_frames_| to the number of input buffers.
TBR=glaznev
BUG=b/24867188,b/24864151
Review URL: https://codereview.webrtc.org/1394303005
Cr-Commit-Position: refs/heads/master@{#10273}
WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.
Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.
The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others. This page is maintained by the Google Chrome team.
Development
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://chromium.googlesource.com/external/webrtc
- Samples and reference apps: https://github.com/webrtc
- Mailing list: http://groups.google.com/group/discuss-webrtc
- Continuous build: http://build.chromium.org/p/client.webrtc
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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