Files
platform-external-webrtc/api/video/test/video_bitrate_allocation_unittest.cc
Stefan Holmer f70446874a Reland "Move allocation and rtp conversion logic out of payload router."
This reverts commit c2406e4eaf7703c6c64d21318186adda791e09fd.

Reason for revert: Reland by removing the conflict with the broken CL.

Original change's description:
> Revert "Move allocation and rtp conversion logic out of payload router."
> 
> This reverts commit 1da4d79ba3275b3fa48cad3b2c0949e0d3b7afe7.
> 
> Reason for revert: Need to revert https://webrtc-review.googlesource.com/c/src/+/88220
> 
> This causes a merge conflict. So need to revert this first.
> 
> Original change's description:
> > Move allocation and rtp conversion logic out of payload router.
> > 
> > Makes it easier to write tests, and allows for moving rtp module
> > ownership into the payload router in the future.
> > 
> > The RtpPayloadParams class is split into declaration and definition and
> > moved into separate files.
> > 
> > Bug: webrtc:9517
> > Change-Id: I8700628edff19abcacfe8d3a20e4ba7476f712ad
> > Reviewed-on: https://webrtc-review.googlesource.com/88564
> > Commit-Queue: Stefan Holmer <stefan@webrtc.org>
> > Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#23983}
> 
> TBR=sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org
> 
> Change-Id: I342c4bf483d975c87c706fe7f76f44e2dc60fe4c
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9517
> Reviewed-on: https://webrtc-review.googlesource.com/88821
> Reviewed-by: JT Teh <jtteh@webrtc.org>
> Commit-Queue: JT Teh <jtteh@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23991}

TBR=sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,lliuu@webrtc.org,jtteh@webrtc.org,tkchin@webrtc.org

Change-Id: I154145cdbc668feee86dbe78860147a6954fee6c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9517
Reviewed-on: https://webrtc-review.googlesource.com/89020
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23996}
2018-07-17 08:17:44 +00:00

64 lines
2.1 KiB
C++

/*
* Copyright (c) 2018 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 <memory>
#include <string>
#include "api/video/video_bitrate_allocation.h"
#include "test/gtest.h"
namespace webrtc {
TEST(VideoBitrateAllocation, SimulcastTargetBitrate) {
VideoBitrateAllocation bitrate;
bitrate.SetBitrate(0, 0, 10000);
bitrate.SetBitrate(0, 1, 20000);
bitrate.SetBitrate(1, 0, 40000);
bitrate.SetBitrate(1, 1, 80000);
VideoBitrateAllocation layer0_bitrate;
layer0_bitrate.SetBitrate(0, 0, 10000);
layer0_bitrate.SetBitrate(0, 1, 20000);
VideoBitrateAllocation layer1_bitrate;
layer1_bitrate.SetBitrate(0, 0, 40000);
layer1_bitrate.SetBitrate(0, 1, 80000);
std::vector<absl::optional<VideoBitrateAllocation>> layer_allocations =
bitrate.GetSimulcastAllocations();
EXPECT_EQ(layer0_bitrate, layer_allocations[0]);
EXPECT_EQ(layer1_bitrate, layer_allocations[1]);
}
TEST(VideoBitrateAllocation, SimulcastTargetBitrateWithInactiveStream) {
// Create bitrate allocation with bitrate only for the first and third stream.
VideoBitrateAllocation bitrate;
bitrate.SetBitrate(0, 0, 10000);
bitrate.SetBitrate(0, 1, 20000);
bitrate.SetBitrate(2, 0, 40000);
bitrate.SetBitrate(2, 1, 80000);
VideoBitrateAllocation layer0_bitrate;
layer0_bitrate.SetBitrate(0, 0, 10000);
layer0_bitrate.SetBitrate(0, 1, 20000);
VideoBitrateAllocation layer2_bitrate;
layer2_bitrate.SetBitrate(0, 0, 40000);
layer2_bitrate.SetBitrate(0, 1, 80000);
std::vector<absl::optional<VideoBitrateAllocation>> layer_allocations =
bitrate.GetSimulcastAllocations();
EXPECT_EQ(layer0_bitrate, layer_allocations[0]);
EXPECT_FALSE(layer_allocations[1]);
EXPECT_EQ(layer2_bitrate, layer_allocations[2]);
}
} // namespace webrtc