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8a9f3a8f53
This reverts commitf9e3bdd2ce. Reason for revert: reland with fix Original change's description: > Revert "Remove dependency of video_replay on TestADM." > > This reverts commit01716663a9. > > Reason for revert: breaking CallPerfTest > https://ci.chromium.org/ui/p/webrtc/builders/perf/Perf%20Android32%20(R%20Pixel5)/967/overview > > Original change's description: > > Remove dependency of video_replay on TestADM. > > > > This should remove requirement to build TestADM in chromium build. > > > > Bug: b/272350185, webrtc:15081 > > Change-Id: Iceb8862aa81099c22bd378ae692229f01ab3314c > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302380 > > Reviewed-by: Henrik Andreassson <henrika@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Artem Titov <titovartem@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39934} > > Bug: b/272350185, webrtc:15081 > Change-Id: I73aa0fd3c3d8c244d20e5f29f5792a4c7d7e4165 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303160 > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Owners-Override: Jeremy Leconte <jleconte@google.com> > Commit-Queue: Jeremy Leconte <jleconte@google.com> > Cr-Commit-Position: refs/heads/main@{#39939} Bug: b/272350185, webrtc:15081 Change-Id: I360ef3e140e60fc21d622480d1f3326e40a76f58 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303400 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Auto-Submit: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39946}
86 lines
2.6 KiB
C++
86 lines
2.6 KiB
C++
/*
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* Copyright (c) 2017 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 "audio/test/audio_end_to_end_test.h"
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#include <algorithm>
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#include <memory>
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#include "api/task_queue/task_queue_base.h"
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#include "call/fake_network_pipe.h"
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#include "call/simulated_network.h"
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#include "modules/audio_device/include/test_audio_device.h"
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#include "system_wrappers/include/sleep.h"
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#include "test/gtest.h"
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#include "test/video_test_constants.h"
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namespace webrtc {
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namespace test {
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namespace {
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constexpr int kSampleRate = 48000;
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} // namespace
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AudioEndToEndTest::AudioEndToEndTest()
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: EndToEndTest(VideoTestConstants::kDefaultTimeout) {}
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size_t AudioEndToEndTest::GetNumVideoStreams() const {
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return 0;
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}
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size_t AudioEndToEndTest::GetNumAudioStreams() const {
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return 1;
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}
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size_t AudioEndToEndTest::GetNumFlexfecStreams() const {
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return 0;
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}
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std::unique_ptr<TestAudioDeviceModule::Capturer>
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AudioEndToEndTest::CreateCapturer() {
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return TestAudioDeviceModule::CreatePulsedNoiseCapturer(32000, kSampleRate);
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}
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std::unique_ptr<TestAudioDeviceModule::Renderer>
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AudioEndToEndTest::CreateRenderer() {
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return TestAudioDeviceModule::CreateDiscardRenderer(kSampleRate);
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}
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void AudioEndToEndTest::OnFakeAudioDevicesCreated(
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AudioDeviceModule* send_audio_device,
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AudioDeviceModule* recv_audio_device) {
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send_audio_device_ = send_audio_device;
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}
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void AudioEndToEndTest::ModifyAudioConfigs(
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AudioSendStream::Config* send_config,
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std::vector<AudioReceiveStreamInterface::Config>* receive_configs) {
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// Large bitrate by default.
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const webrtc::SdpAudioFormat kDefaultFormat("opus", 48000, 2,
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{{"stereo", "1"}});
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send_config->send_codec_spec = AudioSendStream::Config::SendCodecSpec(
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test::VideoTestConstants::kAudioSendPayloadType, kDefaultFormat);
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send_config->min_bitrate_bps = 32000;
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send_config->max_bitrate_bps = 32000;
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}
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void AudioEndToEndTest::OnAudioStreamsCreated(
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AudioSendStream* send_stream,
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const std::vector<AudioReceiveStreamInterface*>& receive_streams) {
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ASSERT_NE(nullptr, send_stream);
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ASSERT_EQ(1u, receive_streams.size());
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ASSERT_NE(nullptr, receive_streams[0]);
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send_stream_ = send_stream;
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receive_stream_ = receive_streams[0];
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}
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} // namespace test
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} // namespace webrtc
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