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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-21 11:44:51 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-21 11:44:51 +0000 |
commit | 9e3c08db40b8916968b9f30096c7be3f00ce9647 (patch) | |
tree | a68f146d7fa01f0134297619fbe7e33db084e0aa /third_party/libwebrtc/test/encoder_settings.cc | |
parent | Initial commit. (diff) | |
download | thunderbird-9e3c08db40b8916968b9f30096c7be3f00ce9647.tar.xz thunderbird-9e3c08db40b8916968b9f30096c7be3f00ce9647.zip |
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/test/encoder_settings.cc')
-rw-r--r-- | third_party/libwebrtc/test/encoder_settings.cc | 149 |
1 files changed, 149 insertions, 0 deletions
diff --git a/third_party/libwebrtc/test/encoder_settings.cc b/third_party/libwebrtc/test/encoder_settings.cc new file mode 100644 index 0000000000..f5b298b107 --- /dev/null +++ b/third_party/libwebrtc/test/encoder_settings.cc @@ -0,0 +1,149 @@ +/* + * Copyright (c) 2014 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 "test/encoder_settings.h" + +#include <algorithm> + +#include "api/scoped_refptr.h" +#include "api/video_codecs/sdp_video_format.h" +#include "call/rtp_config.h" +#include "rtc_base/checks.h" + +namespace webrtc { +namespace test { + +const size_t DefaultVideoStreamFactory::kMaxNumberOfStreams; +const int DefaultVideoStreamFactory::kMaxBitratePerStream[] = {150000, 450000, + 1500000}; +const int DefaultVideoStreamFactory::kDefaultMinBitratePerStream[] = { + 30000, 200000, 700000}; + +// static +std::vector<VideoStream> CreateVideoStreams( + int width, + int height, + const webrtc::VideoEncoderConfig& encoder_config) { + RTC_DCHECK(encoder_config.number_of_streams <= + DefaultVideoStreamFactory::kMaxNumberOfStreams); + + std::vector<VideoStream> stream_settings(encoder_config.number_of_streams); + + int bitrate_left_bps = 0; + if (encoder_config.max_bitrate_bps > 0) { + bitrate_left_bps = encoder_config.max_bitrate_bps; + } else { + for (size_t stream_num = 0; stream_num < encoder_config.number_of_streams; + ++stream_num) { + bitrate_left_bps += + DefaultVideoStreamFactory::kMaxBitratePerStream[stream_num]; + } + } + + for (size_t i = 0; i < encoder_config.number_of_streams; ++i) { + stream_settings[i].width = + (i + 1) * width / encoder_config.number_of_streams; + stream_settings[i].height = + (i + 1) * height / encoder_config.number_of_streams; + stream_settings[i].max_framerate = 30; + stream_settings[i].max_qp = 56; + stream_settings[i].min_bitrate_bps = + DefaultVideoStreamFactory::kDefaultMinBitratePerStream[i]; + + // Use configured values instead of default values if set. + const VideoStream stream = (i < encoder_config.simulcast_layers.size()) + ? encoder_config.simulcast_layers[i] + : VideoStream(); + + int max_bitrate_bps = + stream.max_bitrate_bps > 0 + ? stream.max_bitrate_bps + : DefaultVideoStreamFactory::kMaxBitratePerStream[i]; + max_bitrate_bps = std::min(bitrate_left_bps, max_bitrate_bps); + + int target_bitrate_bps = stream.target_bitrate_bps > 0 + ? stream.target_bitrate_bps + : max_bitrate_bps; + target_bitrate_bps = std::min(max_bitrate_bps, target_bitrate_bps); + + if (stream.min_bitrate_bps > 0) { + RTC_DCHECK_LE(stream.min_bitrate_bps, target_bitrate_bps); + stream_settings[i].min_bitrate_bps = stream.min_bitrate_bps; + } + if (stream.max_framerate > 0) { + stream_settings[i].max_framerate = stream.max_framerate; + } + if (stream.num_temporal_layers) { + RTC_DCHECK_GE(*stream.num_temporal_layers, 1); + stream_settings[i].num_temporal_layers = stream.num_temporal_layers; + } + if (stream.scale_resolution_down_by >= 1.0) { + stream_settings[i].width = width / stream.scale_resolution_down_by; + stream_settings[i].height = height / stream.scale_resolution_down_by; + } + stream_settings[i].scalability_mode = stream.scalability_mode; + stream_settings[i].target_bitrate_bps = target_bitrate_bps; + stream_settings[i].max_bitrate_bps = max_bitrate_bps; + stream_settings[i].active = + encoder_config.number_of_streams == 1 || stream.active; + + bitrate_left_bps -= stream_settings[i].target_bitrate_bps; + } + + stream_settings[encoder_config.number_of_streams - 1].max_bitrate_bps += + bitrate_left_bps; + stream_settings[0].bitrate_priority = encoder_config.bitrate_priority; + + return stream_settings; +} + +DefaultVideoStreamFactory::DefaultVideoStreamFactory() {} + +std::vector<VideoStream> DefaultVideoStreamFactory::CreateEncoderStreams( + int frame_width, + int frame_height, + const webrtc::VideoEncoderConfig& encoder_config) { + return CreateVideoStreams(frame_width, frame_height, encoder_config); +} + +void FillEncoderConfiguration(VideoCodecType codec_type, + size_t num_streams, + VideoEncoderConfig* configuration) { + RTC_DCHECK_LE(num_streams, DefaultVideoStreamFactory::kMaxNumberOfStreams); + + configuration->codec_type = codec_type; + configuration->number_of_streams = num_streams; + configuration->video_stream_factory = + rtc::make_ref_counted<DefaultVideoStreamFactory>(); + configuration->max_bitrate_bps = 0; + configuration->frame_drop_enabled = true; + configuration->simulcast_layers = std::vector<VideoStream>(num_streams); + for (size_t i = 0; i < num_streams; ++i) { + configuration->max_bitrate_bps += + DefaultVideoStreamFactory::kMaxBitratePerStream[i]; + } +} + +VideoReceiveStreamInterface::Decoder CreateMatchingDecoder( + int payload_type, + const std::string& payload_name) { + VideoReceiveStreamInterface::Decoder decoder; + decoder.payload_type = payload_type; + decoder.video_format = SdpVideoFormat(payload_name); + return decoder; +} + +VideoReceiveStreamInterface::Decoder CreateMatchingDecoder( + const VideoSendStream::Config& config) { + return CreateMatchingDecoder(config.rtp.payload_type, + config.rtp.payload_name); +} + +} // namespace test +} // namespace webrtc |