mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-20 09:07:52 +01:00
259a497632
This reverts commit6c2c13af06. Reason for revert: Intend to investigate and fix perf problems. Original change's description: > Revert "Reland "Move rtp-specific config out of EncoderSettings."" > > This reverts commit04dd176862. > > Reason for revert: Regression in ramp up perf tests. > > Original change's description: > > Reland "Move rtp-specific config out of EncoderSettings." > > > > This is a reland ofbc900cb1d1> > > > Original change's description: > > > Move rtp-specific config out of EncoderSettings. > > > > > > In VideoSendStream::Config, move payload_name and payload_type from > > > EncoderSettings to Rtp. > > > > > > EncoderSettings now contains configuration for VideoStreamEncoder only, > > > and should perhaps be renamed in a follow up cl. It's no longer > > > passed as an argument to VideoCodecInitializer::SetupCodec. > > > > > > The latter then needs a different way to know the codec type, > > > which is provided by a new codec_type member in VideoEncoderConfig. > > > > > > Bug: webrtc:8830 > > > Change-Id: Ifcc691aef1ee6a95e43c0452c5e630d92a511cd6 > > > Reviewed-on: https://webrtc-review.googlesource.com/62062 > > > Commit-Queue: Niels Moller <nisse@webrtc.org> > > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org> > > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#22532} > > > > Bug: webrtc:8830 > > Change-Id: If88ef7d57cdaa4fae3c7b2a97ea5a6e1b833e019 > > Reviewed-on: https://webrtc-review.googlesource.com/63721 > > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > Commit-Queue: Niels Moller <nisse@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#22595} > > TBR=brandtr@webrtc.org,magjed@webrtc.org,nisse@webrtc.org,stefan@webrtc.org > > Bug: webrtc:8830,chromium:827080 > Change-Id: Iaaf146de91ec5c0d741b8efdf143f7e173084fef > Reviewed-on: https://webrtc-review.googlesource.com/65520 > Commit-Queue: Niels Moller <nisse@webrtc.org> > Reviewed-by: Niels Moller <nisse@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22677} TBR=brandtr@webrtc.org,magjed@webrtc.org,nisse@webrtc.org,stefan@webrtc.org # Not skipping CQ checks because original CL landed > 1 day ago. Bug: webrtc:8830, chromium:827080 Change-Id: I9b62987bf5daced90dfeb3ebb6739c80117c487f Reviewed-on: https://webrtc-review.googlesource.com/66862 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22751}
59 lines
1.9 KiB
C++
59 lines
1.9 KiB
C++
/*
|
|
* 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.
|
|
*/
|
|
#ifndef TEST_ENCODER_SETTINGS_H_
|
|
#define TEST_ENCODER_SETTINGS_H_
|
|
|
|
#include <vector>
|
|
|
|
#include "call/video_receive_stream.h"
|
|
#include "call/video_send_stream.h"
|
|
|
|
namespace webrtc {
|
|
namespace test {
|
|
|
|
class DefaultVideoStreamFactory
|
|
: public VideoEncoderConfig::VideoStreamFactoryInterface {
|
|
public:
|
|
DefaultVideoStreamFactory();
|
|
|
|
static const size_t kMaxNumberOfStreams = 3;
|
|
// Defined as {150000, 450000, 1500000};
|
|
static const int kMaxBitratePerStream[];
|
|
// Defined as {50000, 200000, 700000};
|
|
static const int kDefaultMinBitratePerStream[];
|
|
|
|
private:
|
|
std::vector<VideoStream> CreateEncoderStreams(
|
|
int width,
|
|
int height,
|
|
const VideoEncoderConfig& encoder_config) override;
|
|
};
|
|
|
|
// Creates |encoder_config.number_of_streams| VideoStreams where index
|
|
// |encoder_config.number_of_streams -1| have width = |width|, height =
|
|
// |height|. The total max bitrate of all VideoStreams is
|
|
// |encoder_config.max_bitrate_bps|.
|
|
std::vector<VideoStream> CreateVideoStreams(
|
|
int width,
|
|
int height,
|
|
const webrtc::VideoEncoderConfig& encoder_config);
|
|
|
|
void FillEncoderConfiguration(VideoCodecType codec_type, size_t num_streams,
|
|
VideoEncoderConfig* configuration);
|
|
|
|
VideoReceiveStream::Decoder CreateMatchingDecoder(
|
|
int payload_type, const std::string& payload_name);
|
|
|
|
VideoReceiveStream::Decoder CreateMatchingDecoder(
|
|
const VideoSendStream::Config& config);
|
|
} // namespace test
|
|
} // namespace webrtc
|
|
|
|
#endif // TEST_ENCODER_SETTINGS_H_
|