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050ffefd85
This CL extends logging related to HW->SW fallbacks on the encoder side in WebRTC. The goal is to make it easier to track down the different steps taken when setting up the video encoder and why/when HW encoding fails. Current logs are added on several lines which makes regexp searching difficult. This CL adds all related information on one line instead. Three new search tags are also added VSE (VideoStreamEncoder), VESFW (VideoEncoderSoftwareFallbackWrapper) and SEA (SimulcastEncoderAdapter). The idea is to allow searching for the tags to see correlated logs. It has been verified that these added logs also show up in WebRTC logs in Meet. Logs from the GPU process are not included due to the sandboxed nature which makes it much more complex to add to the native WebRTC log. I think that these simple logs will provide value as is. Example: https://gist.github.com/henrik-and/41946f7f0b10774241bd14d7687f770b Bug: b/322132132 Change-Id: Iec58c9741a9dd6bab3236a88e9a6e45440f5d980 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339260 Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41733}
43 lines
1.7 KiB
C++
43 lines
1.7 KiB
C++
/*
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* Copyright (c) 2024 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/video_coding/include/video_error_codes.h"
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namespace webrtc {
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const char* WebRtcVideoCodecErrorToString(int32_t error_code) {
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switch (error_code) {
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case WEBRTC_VIDEO_CODEC_TARGET_BITRATE_OVERSHOOT:
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return "WEBRTC_VIDEO_CODEC_TARGET_BITRATE_OVERSHOOT";
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case WEBRTC_VIDEO_CODEC_OK_REQUEST_KEYFRAME:
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return "WEBRTC_VIDEO_CODEC_OK_REQUEST_KEYFRAME";
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case WEBRTC_VIDEO_CODEC_NO_OUTPUT:
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return "WEBRTC_VIDEO_CODEC_NO_OUTPUT";
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case WEBRTC_VIDEO_CODEC_ERROR:
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return "WEBRTC_VIDEO_CODEC_ERROR";
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case WEBRTC_VIDEO_CODEC_MEMORY:
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return "WEBRTC_VIDEO_CODEC_MEMORY";
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case WEBRTC_VIDEO_CODEC_ERR_PARAMETER:
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return "WEBRTC_VIDEO_CODEC_ERR_PARAMETER";
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case WEBRTC_VIDEO_CODEC_TIMEOUT:
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return "WEBRTC_VIDEO_CODEC_TIMEOUT";
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case WEBRTC_VIDEO_CODEC_UNINITIALIZED:
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return "WEBRTC_VIDEO_CODEC_UNINITIALIZED";
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case WEBRTC_VIDEO_CODEC_FALLBACK_SOFTWARE:
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return "WEBRTC_VIDEO_CODEC_FALLBACK_SOFTWARE";
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case WEBRTC_VIDEO_CODEC_ERR_SIMULCAST_PARAMETERS_NOT_SUPPORTED:
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return "WEBRTC_VIDEO_CODEC_ERR_SIMULCAST_PARAMETERS_NOT_SUPPORTED";
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case WEBRTC_VIDEO_CODEC_ENCODER_FAILURE:
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return "WEBRTC_VIDEO_CODEC_ENCODER_FAILURE";
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default:
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return "WEBRTC_VIDEO_CODEC_UNKNOWN";
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}
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}
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} // namespace webrtc
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