Remove AsSendChannel/AsReceiveChannel methods
This CL removes a couple more opportunities for client code to interact directly with the MediaChannel implementation classes. No-try because of infra failure. No-Try: true Bug: webrtc:13931 Change-Id: I658b8b04eff11de7831a1933d16d40fc59c3f0fc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288380 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38935}
This commit is contained in:
Harald Alvestrand
WebRTC LUCI CQ
parent
994734f593
commit
d3ac3b668a
@ -450,6 +450,7 @@ rtc_library("rtc_audio_video") {
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deps = [
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":codec",
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":media_channel",
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":media_channel_impl",
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":media_constants",
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":rtc_media_base",
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":rtp_utils",
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@ -74,11 +74,6 @@ class MediaChannel : public MediaSendChannelInterface,
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bool enable_dscp = false);
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virtual ~MediaChannel();
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// Downcasting to the implemented interfaces.
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MediaSendChannelInterface* AsSendChannel() { return this; }
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MediaReceiveChannelInterface* AsReceiveChannel() { return this; }
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// Downcasting to the subclasses.
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virtual VideoMediaChannel* AsVideoChannel() {
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RTC_CHECK_NOTREACHED();
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@ -24,7 +24,6 @@
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#include "call/audio_state.h"
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#include "media/base/codec.h"
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#include "media/base/media_channel.h"
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#include "media/base/media_channel_impl.h"
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#include "media/base/media_config.h"
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#include "media/base/video_common.h"
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#include "rtc_base/system/file_wrapper.h"
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@ -38,6 +37,9 @@ class Call;
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namespace cricket {
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class VideoMediaChannel;
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class VoiceMediaChannel;
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// Checks that the scalability_mode value of each encoding is supported by at
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// least one video codec of the list. If the list is empty, no check is done.
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webrtc::RTCError CheckScalabilityModeValues(
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@ -31,6 +31,7 @@
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#include "call/flexfec_receive_stream.h"
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#include "call/video_receive_stream.h"
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#include "call/video_send_stream.h"
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#include "media/base/media_channel_impl.h"
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#include "media/base/media_engine.h"
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#include "rtc_base/network_route.h"
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#include "rtc_base/synchronization/mutex.h"
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File diff suppressed because it is too large
Load Diff
@ -25,6 +25,7 @@
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#include "api/transport/rtp/rtp_source.h"
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#include "call/audio_state.h"
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#include "call/call.h"
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#include "media/base/media_channel_impl.h"
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#include "media/base/media_engine.h"
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#include "media/base/rtp_utils.h"
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#include "modules/async_audio_processing/async_audio_processing.h"
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@ -229,6 +229,9 @@ class WebRtcVoiceEngineTestFake : public ::testing::TestWithParam<bool> {
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channel_ = engine_->CreateMediaChannel(&call_, cricket::MediaConfig(),
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cricket::AudioOptions(),
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webrtc::CryptoOptions());
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send_channel_ = std::make_unique<cricket::VoiceMediaSendChannel>(channel_);
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receive_channel_ =
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std::make_unique<cricket::VoiceMediaReceiveChannel>(channel_);
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return (channel_ != nullptr);
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}
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@ -247,7 +250,7 @@ class WebRtcVoiceEngineTestFake : public ::testing::TestWithParam<bool> {
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if (!SetupChannel()) {
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return false;
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}
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if (!channel_->AsSendChannel()->AddSendStream(sp)) {
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if (!send_channel_->AddSendStream(sp)) {
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return false;
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}
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if (!use_null_apm_) {
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@ -258,7 +261,7 @@ class WebRtcVoiceEngineTestFake : public ::testing::TestWithParam<bool> {
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bool AddRecvStream(uint32_t ssrc) {
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EXPECT_TRUE(channel_);
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return channel_->AsReceiveChannel()->AddRecvStream(
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return receive_channel_->AddRecvStream(
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cricket::StreamParams::CreateLegacy(ssrc));
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}
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@ -266,15 +269,15 @@ class WebRtcVoiceEngineTestFake : public ::testing::TestWithParam<bool> {
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EXPECT_TRUE(SetupSendStream());
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// Remove stream added in Setup.
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EXPECT_TRUE(call_.GetAudioSendStream(kSsrcX));
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EXPECT_TRUE(channel_->AsSendChannel()->RemoveSendStream(kSsrcX));
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EXPECT_TRUE(send_channel_->RemoveSendStream(kSsrcX));
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// Verify the channel does not exist.
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EXPECT_FALSE(call_.GetAudioSendStream(kSsrcX));
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}
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void DeliverPacket(const void* data, int len) {
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rtc::CopyOnWriteBuffer packet(reinterpret_cast<const uint8_t*>(data), len);
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channel_->AsReceiveChannel()->OnPacketReceived(packet,
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/* packet_time_us */ -1);
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receive_channel_->OnPacketReceived(packet,
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/* packet_time_us */ -1);
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rtc::Thread::Current()->ProcessMessages(0);
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}
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@ -340,7 +343,7 @@ class WebRtcVoiceEngineTestFake : public ::testing::TestWithParam<bool> {
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if (caller) {
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// If this is a caller, local description will be applied and add the
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// send stream.
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EXPECT_TRUE(channel_->AsSendChannel()->AddSendStream(
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EXPECT_TRUE(send_channel_->AddSendStream(
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cricket::StreamParams::CreateLegacy(kSsrcX)));
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}
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@ -356,7 +359,7 @@ class WebRtcVoiceEngineTestFake : public ::testing::TestWithParam<bool> {
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if (!caller) {
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// If this is callee, there's no active send channel yet.
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EXPECT_FALSE(channel_->InsertDtmf(ssrc, 2, 123));
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EXPECT_TRUE(channel_->AsSendChannel()->AddSendStream(
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EXPECT_TRUE(send_channel_->AddSendStream(
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cricket::StreamParams::CreateLegacy(kSsrcX)));
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}
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@ -379,7 +382,7 @@ class WebRtcVoiceEngineTestFake : public ::testing::TestWithParam<bool> {
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// For a caller, the answer will be applied in set remote description
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// where SetSendParameters() is called.
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EXPECT_TRUE(SetupChannel());
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EXPECT_TRUE(channel_->AsSendChannel()->AddSendStream(
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EXPECT_TRUE(send_channel_->AddSendStream(
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cricket::StreamParams::CreateLegacy(kSsrcX)));
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send_parameters_.extmap_allow_mixed = extmap_allow_mixed;
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SetSendParameters(send_parameters_);
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@ -392,7 +395,7 @@ class WebRtcVoiceEngineTestFake : public ::testing::TestWithParam<bool> {
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// where SetExtmapAllowMixed() and AddSendStream() are called.
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EXPECT_TRUE(SetupChannel());
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channel_->SetExtmapAllowMixed(extmap_allow_mixed);
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EXPECT_TRUE(channel_->AsSendChannel()->AddSendStream(
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EXPECT_TRUE(send_channel_->AddSendStream(
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cricket::StreamParams::CreateLegacy(kSsrcX)));
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const webrtc::AudioSendStream::Config& config = GetSendStreamConfig(kSsrcX);
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@ -422,13 +425,11 @@ class WebRtcVoiceEngineTestFake : public ::testing::TestWithParam<bool> {
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// Sets the per-stream maximum bitrate limit for the specified SSRC.
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bool SetMaxBitrateForStream(int32_t ssrc, int bitrate) {
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webrtc::RtpParameters parameters =
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channel_->AsSendChannel()->GetRtpSendParameters(ssrc);
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send_channel_->GetRtpSendParameters(ssrc);
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EXPECT_EQ(1UL, parameters.encodings.size());
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parameters.encodings[0].max_bitrate_bps = bitrate;
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return channel_->AsSendChannel()
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->SetRtpSendParameters(ssrc, parameters)
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.ok();
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return send_channel_->SetRtpSendParameters(ssrc, parameters).ok();
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}
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void SetGlobalMaxBitrate(const cricket::AudioCodec& codec, int bitrate) {
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@ -474,7 +475,7 @@ class WebRtcVoiceEngineTestFake : public ::testing::TestWithParam<bool> {
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// Verify that reading back the parameters gives results
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// consistent with the Set() result.
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webrtc::RtpParameters resulting_parameters =
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channel_->AsSendChannel()->GetRtpSendParameters(kSsrcX);
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send_channel_->GetRtpSendParameters(kSsrcX);
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EXPECT_EQ(1UL, resulting_parameters.encodings.size());
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EXPECT_EQ(expected_result ? stream_max : -1,
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resulting_parameters.encodings[0].max_bitrate_bps);
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@ -535,7 +536,7 @@ class WebRtcVoiceEngineTestFake : public ::testing::TestWithParam<bool> {
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EXPECT_EQ(id, GetSendStreamConfig(kSsrcX).rtp.extensions[0].id);
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// Ensure extension is set properly on new stream.
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EXPECT_TRUE(channel_->AsSendChannel()->AddSendStream(
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EXPECT_TRUE(send_channel_->AddSendStream(
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cricket::StreamParams::CreateLegacy(kSsrcY)));
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EXPECT_NE(call_.GetAudioSendStream(kSsrcX),
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call_.GetAudioSendStream(kSsrcY));
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@ -803,6 +804,8 @@ class WebRtcVoiceEngineTestFake : public ::testing::TestWithParam<bool> {
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cricket::FakeCall call_;
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std::unique_ptr<cricket::WebRtcVoiceEngine> engine_;
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cricket::VoiceMediaChannel* channel_ = nullptr;
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std::unique_ptr<cricket::VoiceMediaSendChannel> send_channel_;
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std::unique_ptr<cricket::VoiceMediaReceiveChannel> receive_channel_;
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cricket::AudioSendParameters send_parameters_;
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cricket::AudioRecvParameters recv_parameters_;
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FakeAudioSource fake_source_;
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@ -821,7 +824,7 @@ TEST_P(WebRtcVoiceEngineTestFake, CreateMediaChannel) {
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// Test that we can add a send stream and that it has the correct defaults.
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TEST_P(WebRtcVoiceEngineTestFake, CreateSendStream) {
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EXPECT_TRUE(SetupChannel());
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EXPECT_TRUE(channel_->AsSendChannel()->AddSendStream(
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EXPECT_TRUE(send_channel_->AddSendStream(
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cricket::StreamParams::CreateLegacy(kSsrcX)));
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const webrtc::AudioSendStream::Config& config = GetSendStreamConfig(kSsrcX);
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EXPECT_EQ(kSsrcX, config.rtp.ssrc);
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@ -1079,7 +1082,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthMultiRateAsCallee) {
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parameters.max_bandwidth_bps = kDesiredBitrate;
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SetSendParameters(parameters);
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EXPECT_TRUE(channel_->AsSendChannel()->AddSendStream(
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EXPECT_TRUE(send_channel_->AddSendStream(
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cricket::StreamParams::CreateLegacy(kSsrcX)));
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EXPECT_EQ(kDesiredBitrate, GetCodecBitrate(kSsrcX));
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@ -1131,13 +1134,12 @@ TEST_P(WebRtcVoiceEngineTestFake, SetMaxBitratePerStream) {
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TEST_P(WebRtcVoiceEngineTestFake, CannotSetMaxBitrateForNonexistentStream) {
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EXPECT_TRUE(SetupChannel());
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webrtc::RtpParameters nonexistent_parameters =
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channel_->AsSendChannel()->GetRtpSendParameters(kSsrcX);
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send_channel_->GetRtpSendParameters(kSsrcX);
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EXPECT_EQ(0u, nonexistent_parameters.encodings.size());
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nonexistent_parameters.encodings.push_back(webrtc::RtpEncodingParameters());
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EXPECT_FALSE(channel_->AsSendChannel()
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->SetRtpSendParameters(kSsrcX, nonexistent_parameters)
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.ok());
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EXPECT_FALSE(
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send_channel_->SetRtpSendParameters(kSsrcX, nonexistent_parameters).ok());
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}
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TEST_P(WebRtcVoiceEngineTestFake,
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@ -1149,25 +1151,22 @@ TEST_P(WebRtcVoiceEngineTestFake,
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EXPECT_TRUE(SetupSendStream());
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webrtc::RtpParameters parameters =
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channel_->AsSendChannel()->GetRtpSendParameters(kSsrcX);
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send_channel_->GetRtpSendParameters(kSsrcX);
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// Two or more encodings should result in failure.
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parameters.encodings.push_back(webrtc::RtpEncodingParameters());
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EXPECT_FALSE(
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channel_->AsSendChannel()->SetRtpSendParameters(kSsrcX, parameters).ok());
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EXPECT_FALSE(send_channel_->SetRtpSendParameters(kSsrcX, parameters).ok());
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// Zero encodings should also fail.
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parameters.encodings.clear();
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EXPECT_FALSE(
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channel_->AsSendChannel()->SetRtpSendParameters(kSsrcX, parameters).ok());
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EXPECT_FALSE(send_channel_->SetRtpSendParameters(kSsrcX, parameters).ok());
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}
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// Changing the SSRC through RtpParameters is not allowed.
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TEST_P(WebRtcVoiceEngineTestFake, CannotSetSsrcInRtpSendParameters) {
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EXPECT_TRUE(SetupSendStream());
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webrtc::RtpParameters parameters =
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channel_->AsSendChannel()->GetRtpSendParameters(kSsrcX);
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send_channel_->GetRtpSendParameters(kSsrcX);
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parameters.encodings[0].ssrc = 0xdeadbeef;
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EXPECT_FALSE(
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channel_->AsSendChannel()->SetRtpSendParameters(kSsrcX, parameters).ok());
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EXPECT_FALSE(send_channel_->SetRtpSendParameters(kSsrcX, parameters).ok());
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}
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// Test that a stream will not be sending if its encoding is made
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@ -1178,20 +1177,18 @@ TEST_P(WebRtcVoiceEngineTestFake, SetRtpParametersEncodingsActive) {
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EXPECT_TRUE(GetSendStream(kSsrcX).IsSending());
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// Get current parameters and change "active" to false.
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webrtc::RtpParameters parameters =
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channel_->AsSendChannel()->GetRtpSendParameters(kSsrcX);
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send_channel_->GetRtpSendParameters(kSsrcX);
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ASSERT_EQ(1u, parameters.encodings.size());
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ASSERT_TRUE(parameters.encodings[0].active);
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parameters.encodings[0].active = false;
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EXPECT_TRUE(
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channel_->AsSendChannel()->SetRtpSendParameters(kSsrcX, parameters).ok());
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EXPECT_TRUE(send_channel_->SetRtpSendParameters(kSsrcX, parameters).ok());
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EXPECT_FALSE(GetSendStream(kSsrcX).IsSending());
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// Now change it back to active and verify we resume sending.
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// This should occur even when other parameters are updated.
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parameters.encodings[0].active = true;
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parameters.encodings[0].max_bitrate_bps = absl::optional<int>(6000);
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EXPECT_TRUE(
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channel_->AsSendChannel()->SetRtpSendParameters(kSsrcX, parameters).ok());
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EXPECT_TRUE(send_channel_->SetRtpSendParameters(kSsrcX, parameters).ok());
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EXPECT_TRUE(GetSendStream(kSsrcX).IsSending());
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}
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@ -1199,18 +1196,16 @@ TEST_P(WebRtcVoiceEngineTestFake, SetRtpParametersAdaptivePtime) {
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EXPECT_TRUE(SetupSendStream());
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// Get current parameters and change "adaptive_ptime" to true.
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webrtc::RtpParameters parameters =
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channel_->AsSendChannel()->GetRtpSendParameters(kSsrcX);
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send_channel_->GetRtpSendParameters(kSsrcX);
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ASSERT_EQ(1u, parameters.encodings.size());
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ASSERT_FALSE(parameters.encodings[0].adaptive_ptime);
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parameters.encodings[0].adaptive_ptime = true;
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EXPECT_TRUE(
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channel_->AsSendChannel()->SetRtpSendParameters(kSsrcX, parameters).ok());
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EXPECT_TRUE(send_channel_->SetRtpSendParameters(kSsrcX, parameters).ok());
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EXPECT_TRUE(GetAudioNetworkAdaptorConfig(kSsrcX));
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EXPECT_EQ(16000, GetSendStreamConfig(kSsrcX).min_bitrate_bps);
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parameters.encodings[0].adaptive_ptime = false;
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EXPECT_TRUE(
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channel_->AsSendChannel()->SetRtpSendParameters(kSsrcX, parameters).ok());
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EXPECT_TRUE(send_channel_->SetRtpSendParameters(kSsrcX, parameters).ok());
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EXPECT_FALSE(GetAudioNetworkAdaptorConfig(kSsrcX));
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EXPECT_EQ(32000, GetSendStreamConfig(kSsrcX).min_bitrate_bps);
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}
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@ -1225,10 +1220,9 @@ TEST_P(WebRtcVoiceEngineTestFake,
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GetAudioNetworkAdaptorConfig(kSsrcX));
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webrtc::RtpParameters parameters =
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channel_->AsSendChannel()->GetRtpSendParameters(kSsrcX);
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send_channel_->GetRtpSendParameters(kSsrcX);
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parameters.encodings[0].adaptive_ptime = false;
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EXPECT_TRUE(
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channel_->AsSendChannel()->SetRtpSendParameters(kSsrcX, parameters).ok());
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EXPECT_TRUE(send_channel_->SetRtpSendParameters(kSsrcX, parameters).ok());
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EXPECT_EQ(send_parameters_.options.audio_network_adaptor_config,
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GetAudioNetworkAdaptorConfig(kSsrcX));
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}
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@ -1246,7 +1240,7 @@ TEST_P(WebRtcVoiceEngineTestFake, RtpParametersArePerStream) {
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SetupForMultiSendStream();
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// Create send streams.
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for (uint32_t ssrc : kSsrcs4) {
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EXPECT_TRUE(channel_->AsSendChannel()->AddSendStream(
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EXPECT_TRUE(send_channel_->AddSendStream(
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cricket::StreamParams::CreateLegacy(ssrc)));
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}
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// Configure one stream to be limited by the stream config, another to be
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@ -1278,7 +1272,7 @@ TEST_P(WebRtcVoiceEngineTestFake, GetRtpSendParametersCodecs) {
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SetSendParameters(parameters);
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webrtc::RtpParameters rtp_parameters =
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channel_->AsSendChannel()->GetRtpSendParameters(kSsrcX);
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send_channel_->GetRtpSendParameters(kSsrcX);
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ASSERT_EQ(2u, rtp_parameters.codecs.size());
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EXPECT_EQ(kOpusCodec.ToCodecParameters(), rtp_parameters.codecs[0]);
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EXPECT_EQ(kPcmuCodec.ToCodecParameters(), rtp_parameters.codecs[1]);
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@ -1291,7 +1285,7 @@ TEST_P(WebRtcVoiceEngineTestFake, GetRtpSendParametersRtcpCname) {
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EXPECT_TRUE(SetupSendStream(params));
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webrtc::RtpParameters rtp_parameters =
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channel_->AsSendChannel()->GetRtpSendParameters(kSsrcX);
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send_channel_->GetRtpSendParameters(kSsrcX);
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EXPECT_STREQ("rtcpcname", rtp_parameters.rtcp.cname.c_str());
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}
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@ -1300,13 +1294,13 @@ TEST_P(WebRtcVoiceEngineTestFake,
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EXPECT_TRUE(SetupSendStream());
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webrtc::RtpParameters rtp_parameters =
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channel_->AsSendChannel()->GetRtpSendParameters(kSsrcX);
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send_channel_->GetRtpSendParameters(kSsrcX);
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rtp_parameters.header_extensions.emplace_back();
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EXPECT_NE(0u, rtp_parameters.header_extensions.size());
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webrtc::RTCError result =
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channel_->AsSendChannel()->SetRtpSendParameters(kSsrcX, rtp_parameters);
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send_channel_->SetRtpSendParameters(kSsrcX, rtp_parameters);
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EXPECT_EQ(webrtc::RTCErrorType::INVALID_MODIFICATION, result.type());
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}
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@ -1314,7 +1308,7 @@ TEST_P(WebRtcVoiceEngineTestFake,
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TEST_P(WebRtcVoiceEngineTestFake, GetRtpSendParametersSsrc) {
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EXPECT_TRUE(SetupSendStream());
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webrtc::RtpParameters rtp_parameters =
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channel_->AsSendChannel()->GetRtpSendParameters(kSsrcX);
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send_channel_->GetRtpSendParameters(kSsrcX);
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ASSERT_EQ(1u, rtp_parameters.encodings.size());
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EXPECT_EQ(kSsrcX, rtp_parameters.encodings[0].ssrc);
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}
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@ -1328,18 +1322,15 @@ TEST_P(WebRtcVoiceEngineTestFake, SetAndGetRtpSendParameters) {
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SetSendParameters(parameters);
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webrtc::RtpParameters initial_params =
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channel_->AsSendChannel()->GetRtpSendParameters(kSsrcX);
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send_channel_->GetRtpSendParameters(kSsrcX);
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||||
// We should be able to set the params we just got.
|
||||
EXPECT_TRUE(channel_->AsSendChannel()
|
||||
->SetRtpSendParameters(kSsrcX, initial_params)
|
||||
.ok());
|
||||
EXPECT_TRUE(send_channel_->SetRtpSendParameters(kSsrcX, initial_params).ok());
|
||||
|
||||
// ... And this shouldn't change the params returned by GetRtpSendParameters.
|
||||
webrtc::RtpParameters new_params =
|
||||
channel_->AsSendChannel()->GetRtpSendParameters(kSsrcX);
|
||||
EXPECT_EQ(initial_params,
|
||||
channel_->AsSendChannel()->GetRtpSendParameters(kSsrcX));
|
||||
send_channel_->GetRtpSendParameters(kSsrcX);
|
||||
EXPECT_EQ(initial_params, send_channel_->GetRtpSendParameters(kSsrcX));
|
||||
}
|
||||
|
||||
// Test that max_bitrate_bps in send stream config gets updated correctly when
|
||||
@ -1351,15 +1342,13 @@ TEST_P(WebRtcVoiceEngineTestFake, SetRtpSendParameterUpdatesMaxBitrate) {
|
||||
SetSendParameters(send_parameters);
|
||||
|
||||
webrtc::RtpParameters rtp_parameters =
|
||||
channel_->AsSendChannel()->GetRtpSendParameters(kSsrcX);
|
||||
send_channel_->GetRtpSendParameters(kSsrcX);
|
||||
// Expect empty on parameters.encodings[0].max_bitrate_bps;
|
||||
EXPECT_FALSE(rtp_parameters.encodings[0].max_bitrate_bps);
|
||||
|
||||
constexpr int kMaxBitrateBps = 6000;
|
||||
rtp_parameters.encodings[0].max_bitrate_bps = kMaxBitrateBps;
|
||||
EXPECT_TRUE(channel_->AsSendChannel()
|
||||
->SetRtpSendParameters(kSsrcX, rtp_parameters)
|
||||
.ok());
|
||||
EXPECT_TRUE(send_channel_->SetRtpSendParameters(kSsrcX, rtp_parameters).ok());
|
||||
|
||||
const int max_bitrate = GetSendStreamConfig(kSsrcX).max_bitrate_bps;
|
||||
EXPECT_EQ(max_bitrate, kMaxBitrateBps);
|
||||
@ -1370,19 +1359,17 @@ TEST_P(WebRtcVoiceEngineTestFake, SetRtpSendParameterUpdatesMaxBitrate) {
|
||||
TEST_P(WebRtcVoiceEngineTestFake, SetRtpSendParameterInvalidBitratePriority) {
|
||||
EXPECT_TRUE(SetupSendStream());
|
||||
webrtc::RtpParameters rtp_parameters =
|
||||
channel_->AsSendChannel()->GetRtpSendParameters(kSsrcX);
|
||||
send_channel_->GetRtpSendParameters(kSsrcX);
|
||||
EXPECT_EQ(1UL, rtp_parameters.encodings.size());
|
||||
EXPECT_EQ(webrtc::kDefaultBitratePriority,
|
||||
rtp_parameters.encodings[0].bitrate_priority);
|
||||
|
||||
rtp_parameters.encodings[0].bitrate_priority = 0;
|
||||
EXPECT_FALSE(channel_->AsSendChannel()
|
||||
->SetRtpSendParameters(kSsrcX, rtp_parameters)
|
||||
.ok());
|
||||
EXPECT_FALSE(
|
||||
send_channel_->SetRtpSendParameters(kSsrcX, rtp_parameters).ok());
|
||||
rtp_parameters.encodings[0].bitrate_priority = -1.0;
|
||||
EXPECT_FALSE(channel_->AsSendChannel()
|
||||
->SetRtpSendParameters(kSsrcX, rtp_parameters)
|
||||
.ok());
|
||||
EXPECT_FALSE(
|
||||
send_channel_->SetRtpSendParameters(kSsrcX, rtp_parameters).ok());
|
||||
}
|
||||
|
||||
// Test that the bitrate_priority in the send stream config gets updated when
|
||||
@ -1390,21 +1377,18 @@ TEST_P(WebRtcVoiceEngineTestFake, SetRtpSendParameterInvalidBitratePriority) {
|
||||
TEST_P(WebRtcVoiceEngineTestFake, SetRtpSendParameterUpdatesBitratePriority) {
|
||||
EXPECT_TRUE(SetupSendStream());
|
||||
webrtc::RtpParameters rtp_parameters =
|
||||
channel_->AsSendChannel()->GetRtpSendParameters(kSsrcX);
|
||||
send_channel_->GetRtpSendParameters(kSsrcX);
|
||||
|
||||
EXPECT_EQ(1UL, rtp_parameters.encodings.size());
|
||||
EXPECT_EQ(webrtc::kDefaultBitratePriority,
|
||||
rtp_parameters.encodings[0].bitrate_priority);
|
||||
double new_bitrate_priority = 2.0;
|
||||
rtp_parameters.encodings[0].bitrate_priority = new_bitrate_priority;
|
||||
EXPECT_TRUE(channel_->AsSendChannel()
|
||||
->SetRtpSendParameters(kSsrcX, rtp_parameters)
|
||||
.ok());
|
||||
EXPECT_TRUE(send_channel_->SetRtpSendParameters(kSsrcX, rtp_parameters).ok());
|
||||
|
||||
// The priority should get set for both the audio channel's rtp parameters
|
||||
// and the audio send stream's audio config.
|
||||
EXPECT_EQ(new_bitrate_priority, channel_->AsSendChannel()
|
||||
->GetRtpSendParameters(kSsrcX)
|
||||
EXPECT_EQ(new_bitrate_priority, send_channel_->GetRtpSendParameters(kSsrcX)
|
||||
.encodings[0]
|
||||
.bitrate_priority);
|
||||
EXPECT_EQ(new_bitrate_priority, GetSendStreamConfig(kSsrcX).bitrate_priority);
|
||||
@ -1832,7 +1816,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecEnableNackAsCallee) {
|
||||
// NACK should be enabled even with no send stream.
|
||||
EXPECT_EQ(kRtpHistoryMs, GetRecvStreamConfig(kSsrcX).rtp.nack.rtp_history_ms);
|
||||
|
||||
EXPECT_TRUE(channel_->AsSendChannel()->AddSendStream(
|
||||
EXPECT_TRUE(send_channel_->AddSendStream(
|
||||
cricket::StreamParams::CreateLegacy(kSsrcX)));
|
||||
}
|
||||
|
||||
@ -2078,7 +2062,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsCNandDTMFAsCallee) {
|
||||
parameters.codecs[2].id = 97; // narrowband CN
|
||||
parameters.codecs[3].id = 98; // DTMF
|
||||
SetSendParameters(parameters);
|
||||
EXPECT_TRUE(channel_->AsSendChannel()->AddSendStream(
|
||||
EXPECT_TRUE(send_channel_->AddSendStream(
|
||||
cricket::StreamParams::CreateLegacy(kSsrcX)));
|
||||
|
||||
const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
|
||||
@ -2252,7 +2236,7 @@ TEST_P(WebRtcVoiceEngineTestFake, CreateAndDeleteMultipleSendStreams) {
|
||||
SetSend(true);
|
||||
|
||||
for (uint32_t ssrc : kSsrcs4) {
|
||||
EXPECT_TRUE(channel_->AsSendChannel()->AddSendStream(
|
||||
EXPECT_TRUE(send_channel_->AddSendStream(
|
||||
cricket::StreamParams::CreateLegacy(ssrc)));
|
||||
SetAudioSend(ssrc, true, &fake_source_);
|
||||
// Verify that we are in a sending state for all the created streams.
|
||||
@ -2262,9 +2246,9 @@ TEST_P(WebRtcVoiceEngineTestFake, CreateAndDeleteMultipleSendStreams) {
|
||||
|
||||
// Delete the send streams.
|
||||
for (uint32_t ssrc : kSsrcs4) {
|
||||
EXPECT_TRUE(channel_->AsSendChannel()->RemoveSendStream(ssrc));
|
||||
EXPECT_TRUE(send_channel_->RemoveSendStream(ssrc));
|
||||
EXPECT_FALSE(call_.GetAudioSendStream(ssrc));
|
||||
EXPECT_FALSE(channel_->AsSendChannel()->RemoveSendStream(ssrc));
|
||||
EXPECT_FALSE(send_channel_->RemoveSendStream(ssrc));
|
||||
}
|
||||
EXPECT_EQ(0u, call_.GetAudioSendStreams().size());
|
||||
}
|
||||
@ -2275,7 +2259,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsWithMultipleSendStreams) {
|
||||
|
||||
// Create send streams.
|
||||
for (uint32_t ssrc : kSsrcs4) {
|
||||
EXPECT_TRUE(channel_->AsSendChannel()->AddSendStream(
|
||||
EXPECT_TRUE(send_channel_->AddSendStream(
|
||||
cricket::StreamParams::CreateLegacy(ssrc)));
|
||||
}
|
||||
|
||||
@ -2315,7 +2299,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendWithMultipleSendStreams) {
|
||||
|
||||
// Create the send channels and they should be a "not sending" date.
|
||||
for (uint32_t ssrc : kSsrcs4) {
|
||||
EXPECT_TRUE(channel_->AsSendChannel()->AddSendStream(
|
||||
EXPECT_TRUE(send_channel_->AddSendStream(
|
||||
cricket::StreamParams::CreateLegacy(ssrc)));
|
||||
SetAudioSend(ssrc, true, &fake_source_);
|
||||
EXPECT_FALSE(GetSendStream(ssrc).IsSending());
|
||||
@ -2342,7 +2326,7 @@ TEST_P(WebRtcVoiceEngineTestFake, GetStatsWithMultipleSendStreams) {
|
||||
|
||||
// Create send streams.
|
||||
for (uint32_t ssrc : kSsrcs4) {
|
||||
EXPECT_TRUE(channel_->AsSendChannel()->AddSendStream(
|
||||
EXPECT_TRUE(send_channel_->AddSendStream(
|
||||
cricket::StreamParams::CreateLegacy(ssrc)));
|
||||
}
|
||||
|
||||
@ -2377,7 +2361,7 @@ TEST_P(WebRtcVoiceEngineTestFake, GetStatsWithMultipleSendStreams) {
|
||||
// Remove the kSsrcY stream. No receiver stats.
|
||||
{
|
||||
cricket::VoiceMediaInfo info;
|
||||
EXPECT_TRUE(channel_->AsReceiveChannel()->RemoveRecvStream(kSsrcY));
|
||||
EXPECT_TRUE(receive_channel_->RemoveRecvStream(kSsrcY));
|
||||
EXPECT_CALL(*adm_, GetPlayoutUnderrunCount()).WillOnce(Return(0));
|
||||
EXPECT_EQ(true,
|
||||
channel_->GetStats(&info, /*get_and_clear_legacy_stats=*/true));
|
||||
@ -2438,8 +2422,8 @@ TEST_P(WebRtcVoiceEngineTestFake, PlayoutWithMultipleStreams) {
|
||||
EXPECT_TRUE(GetRecvStream(kSsrcZ).started());
|
||||
|
||||
// Now remove the recv streams.
|
||||
EXPECT_TRUE(channel_->AsReceiveChannel()->RemoveRecvStream(kSsrcZ));
|
||||
EXPECT_TRUE(channel_->AsReceiveChannel()->RemoveRecvStream(kSsrcY));
|
||||
EXPECT_TRUE(receive_channel_->RemoveRecvStream(kSsrcZ));
|
||||
EXPECT_TRUE(receive_channel_->RemoveRecvStream(kSsrcY));
|
||||
}
|
||||
|
||||
TEST_P(WebRtcVoiceEngineTestFake, SetAudioNetworkAdaptorViaOptions) {
|
||||
@ -2529,7 +2513,7 @@ TEST_P(WebRtcVoiceEngineTestFake, GetStats) {
|
||||
// Remove the kSsrcY stream. No receiver stats.
|
||||
{
|
||||
cricket::VoiceMediaInfo info;
|
||||
EXPECT_TRUE(channel_->AsReceiveChannel()->RemoveRecvStream(kSsrcY));
|
||||
EXPECT_TRUE(receive_channel_->RemoveRecvStream(kSsrcY));
|
||||
EXPECT_CALL(*adm_, GetPlayoutUnderrunCount()).WillOnce(Return(0));
|
||||
EXPECT_EQ(true,
|
||||
channel_->GetStats(&info, /*get_and_clear_legacy_stats=*/true));
|
||||
@ -2567,7 +2551,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendSsrcWithMultipleStreams) {
|
||||
TEST_P(WebRtcVoiceEngineTestFake, SetSendSsrcAfterCreatingReceiveChannel) {
|
||||
EXPECT_TRUE(SetupChannel());
|
||||
EXPECT_TRUE(AddRecvStream(kSsrcY));
|
||||
EXPECT_TRUE(channel_->AsSendChannel()->AddSendStream(
|
||||
EXPECT_TRUE(send_channel_->AddSendStream(
|
||||
cricket::StreamParams::CreateLegacy(kSsrcX)));
|
||||
EXPECT_TRUE(call_.GetAudioSendStream(kSsrcX));
|
||||
EXPECT_EQ(kSsrcX, GetRecvStreamConfig(kSsrcY).rtp.local_ssrc);
|
||||
@ -2630,9 +2614,9 @@ TEST_P(WebRtcVoiceEngineTestFake, RecvWithMultipleStreams) {
|
||||
EXPECT_EQ(s3.received_packets(), 1);
|
||||
EXPECT_TRUE(s3.VerifyLastPacket(packets[3], sizeof(packets[3])));
|
||||
|
||||
EXPECT_TRUE(channel_->AsReceiveChannel()->RemoveRecvStream(ssrc3));
|
||||
EXPECT_TRUE(channel_->AsReceiveChannel()->RemoveRecvStream(ssrc2));
|
||||
EXPECT_TRUE(channel_->AsReceiveChannel()->RemoveRecvStream(ssrc1));
|
||||
EXPECT_TRUE(receive_channel_->RemoveRecvStream(ssrc3));
|
||||
EXPECT_TRUE(receive_channel_->RemoveRecvStream(ssrc2));
|
||||
EXPECT_TRUE(receive_channel_->RemoveRecvStream(ssrc1));
|
||||
}
|
||||
|
||||
// Test that receiving on an unsignaled stream works (a stream is created).
|
||||
@ -2655,7 +2639,7 @@ TEST_P(WebRtcVoiceEngineTestFake, RecvUnsignaledSsrcWithSignaledStreamId) {
|
||||
EXPECT_TRUE(SetupChannel());
|
||||
cricket::StreamParams unsignaled_stream;
|
||||
unsignaled_stream.set_stream_ids({kSyncLabel});
|
||||
ASSERT_TRUE(channel_->AsReceiveChannel()->AddRecvStream(unsignaled_stream));
|
||||
ASSERT_TRUE(receive_channel_->AddRecvStream(unsignaled_stream));
|
||||
// The stream shouldn't have been created at this point because it doesn't
|
||||
// have any SSRCs.
|
||||
EXPECT_EQ(0u, call_.GetAudioReceiveStreams().size());
|
||||
@ -2669,8 +2653,8 @@ TEST_P(WebRtcVoiceEngineTestFake, RecvUnsignaledSsrcWithSignaledStreamId) {
|
||||
|
||||
// Remset the unsignaled stream to clear the cached parameters. If a new
|
||||
// default unsignaled receive stream is created it will not have a sync group.
|
||||
channel_->AsReceiveChannel()->ResetUnsignaledRecvStream();
|
||||
channel_->AsReceiveChannel()->RemoveRecvStream(kSsrc1);
|
||||
receive_channel_->ResetUnsignaledRecvStream();
|
||||
receive_channel_->RemoveRecvStream(kSsrc1);
|
||||
|
||||
DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
|
||||
|
||||
@ -2699,7 +2683,7 @@ TEST_P(WebRtcVoiceEngineTestFake,
|
||||
ASSERT_EQ(receivers1.size(), 2u);
|
||||
|
||||
// Should remove all default streams.
|
||||
channel_->AsReceiveChannel()->ResetUnsignaledRecvStream();
|
||||
receive_channel_->ResetUnsignaledRecvStream();
|
||||
const auto& receivers2 = call_.GetAudioReceiveStreams();
|
||||
EXPECT_EQ(0u, receivers2.size());
|
||||
}
|
||||
@ -2828,7 +2812,7 @@ TEST_P(WebRtcVoiceEngineTestFake, AddRecvStreamAfterUnsignaled_Updates) {
|
||||
stream_params.ssrcs.push_back(1);
|
||||
stream_params.set_stream_ids({new_stream_id});
|
||||
|
||||
EXPECT_TRUE(channel_->AsReceiveChannel()->AddRecvStream(stream_params));
|
||||
EXPECT_TRUE(receive_channel_->AddRecvStream(stream_params));
|
||||
EXPECT_EQ(1u, streams.size());
|
||||
// The audio receive stream should not have been recreated.
|
||||
EXPECT_EQ(audio_receive_stream_id, streams.front()->id());
|
||||
@ -3226,7 +3210,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetOutputVolume) {
|
||||
EXPECT_FALSE(channel_->SetOutputVolume(kSsrcY, 0.5));
|
||||
cricket::StreamParams stream;
|
||||
stream.ssrcs.push_back(kSsrcY);
|
||||
EXPECT_TRUE(channel_->AsReceiveChannel()->AddRecvStream(stream));
|
||||
EXPECT_TRUE(receive_channel_->AddRecvStream(stream));
|
||||
EXPECT_DOUBLE_EQ(1, GetRecvStream(kSsrcY).gain());
|
||||
EXPECT_TRUE(channel_->SetOutputVolume(kSsrcY, 3));
|
||||
EXPECT_DOUBLE_EQ(3, GetRecvStream(kSsrcY).gain());
|
||||
@ -3268,18 +3252,15 @@ TEST_P(WebRtcVoiceEngineTestFake, SetOutputVolumeUnsignaledRecvStream) {
|
||||
|
||||
TEST_P(WebRtcVoiceEngineTestFake, BaseMinimumPlayoutDelayMs) {
|
||||
EXPECT_TRUE(SetupChannel());
|
||||
EXPECT_FALSE(receive_channel_->SetBaseMinimumPlayoutDelayMs(kSsrcY, 200));
|
||||
EXPECT_FALSE(
|
||||
channel_->AsReceiveChannel()->SetBaseMinimumPlayoutDelayMs(kSsrcY, 200));
|
||||
EXPECT_FALSE(channel_->AsReceiveChannel()
|
||||
->GetBaseMinimumPlayoutDelayMs(kSsrcY)
|
||||
.has_value());
|
||||
receive_channel_->GetBaseMinimumPlayoutDelayMs(kSsrcY).has_value());
|
||||
|
||||
cricket::StreamParams stream;
|
||||
stream.ssrcs.push_back(kSsrcY);
|
||||
EXPECT_TRUE(channel_->AsReceiveChannel()->AddRecvStream(stream));
|
||||
EXPECT_TRUE(receive_channel_->AddRecvStream(stream));
|
||||
EXPECT_EQ(0, GetRecvStream(kSsrcY).base_mininum_playout_delay_ms());
|
||||
EXPECT_TRUE(
|
||||
channel_->AsReceiveChannel()->SetBaseMinimumPlayoutDelayMs(kSsrcY, 300));
|
||||
EXPECT_TRUE(receive_channel_->SetBaseMinimumPlayoutDelayMs(kSsrcY, 300));
|
||||
EXPECT_EQ(300, GetRecvStream(kSsrcY).base_mininum_playout_delay_ms());
|
||||
}
|
||||
|
||||
@ -3290,70 +3271,57 @@ TEST_P(WebRtcVoiceEngineTestFake,
|
||||
|
||||
// Spawn an unsignaled stream by sending a packet - delay should be 0.
|
||||
DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
|
||||
EXPECT_EQ(0, channel_->AsReceiveChannel()
|
||||
->GetBaseMinimumPlayoutDelayMs(kSsrc1)
|
||||
.value_or(-1));
|
||||
EXPECT_EQ(
|
||||
0, receive_channel_->GetBaseMinimumPlayoutDelayMs(kSsrc1).value_or(-1));
|
||||
// Check that it doesn't provide default values for unknown ssrc.
|
||||
EXPECT_FALSE(channel_->AsReceiveChannel()
|
||||
->GetBaseMinimumPlayoutDelayMs(kSsrcY)
|
||||
.has_value());
|
||||
EXPECT_FALSE(
|
||||
receive_channel_->GetBaseMinimumPlayoutDelayMs(kSsrcY).has_value());
|
||||
|
||||
// Check that default value for unsignaled streams is 0.
|
||||
EXPECT_EQ(0, channel_->AsReceiveChannel()
|
||||
->GetBaseMinimumPlayoutDelayMs(kSsrc0)
|
||||
.value_or(-1));
|
||||
EXPECT_EQ(
|
||||
0, receive_channel_->GetBaseMinimumPlayoutDelayMs(kSsrc0).value_or(-1));
|
||||
|
||||
// Should remember the delay 100 which will be set on new unsignaled streams,
|
||||
// and also set the delay to 100 on existing unsignaled streams.
|
||||
EXPECT_TRUE(
|
||||
channel_->AsReceiveChannel()->SetBaseMinimumPlayoutDelayMs(kSsrc0, 100));
|
||||
EXPECT_EQ(100, channel_->AsReceiveChannel()
|
||||
->GetBaseMinimumPlayoutDelayMs(kSsrc0)
|
||||
.value_or(-1));
|
||||
EXPECT_TRUE(receive_channel_->SetBaseMinimumPlayoutDelayMs(kSsrc0, 100));
|
||||
EXPECT_EQ(
|
||||
100, receive_channel_->GetBaseMinimumPlayoutDelayMs(kSsrc0).value_or(-1));
|
||||
// Check that it doesn't provide default values for unknown ssrc.
|
||||
EXPECT_FALSE(channel_->AsReceiveChannel()
|
||||
->GetBaseMinimumPlayoutDelayMs(kSsrcY)
|
||||
.has_value());
|
||||
EXPECT_FALSE(
|
||||
receive_channel_->GetBaseMinimumPlayoutDelayMs(kSsrcY).has_value());
|
||||
|
||||
// Spawn an unsignaled stream by sending a packet - delay should be 100.
|
||||
unsigned char pcmuFrame2[sizeof(kPcmuFrame)];
|
||||
memcpy(pcmuFrame2, kPcmuFrame, sizeof(kPcmuFrame));
|
||||
rtc::SetBE32(&pcmuFrame2[8], kSsrcX);
|
||||
DeliverPacket(pcmuFrame2, sizeof(pcmuFrame2));
|
||||
EXPECT_EQ(100, channel_->AsReceiveChannel()
|
||||
->GetBaseMinimumPlayoutDelayMs(kSsrcX)
|
||||
.value_or(-1));
|
||||
EXPECT_EQ(
|
||||
100, receive_channel_->GetBaseMinimumPlayoutDelayMs(kSsrcX).value_or(-1));
|
||||
|
||||
// Setting delay with SSRC=0 should affect all unsignaled streams.
|
||||
EXPECT_TRUE(
|
||||
channel_->AsReceiveChannel()->SetBaseMinimumPlayoutDelayMs(kSsrc0, 300));
|
||||
EXPECT_TRUE(receive_channel_->SetBaseMinimumPlayoutDelayMs(kSsrc0, 300));
|
||||
if (kMaxUnsignaledRecvStreams > 1) {
|
||||
EXPECT_EQ(300, channel_->AsReceiveChannel()
|
||||
->GetBaseMinimumPlayoutDelayMs(kSsrc1)
|
||||
.value_or(-1));
|
||||
EXPECT_EQ(
|
||||
300,
|
||||
receive_channel_->GetBaseMinimumPlayoutDelayMs(kSsrc1).value_or(-1));
|
||||
}
|
||||
EXPECT_EQ(300, channel_->AsReceiveChannel()
|
||||
->GetBaseMinimumPlayoutDelayMs(kSsrcX)
|
||||
.value_or(-1));
|
||||
EXPECT_EQ(
|
||||
300, receive_channel_->GetBaseMinimumPlayoutDelayMs(kSsrcX).value_or(-1));
|
||||
|
||||
// Setting delay on an individual stream affects only that.
|
||||
EXPECT_TRUE(
|
||||
channel_->AsReceiveChannel()->SetBaseMinimumPlayoutDelayMs(kSsrcX, 400));
|
||||
EXPECT_TRUE(receive_channel_->SetBaseMinimumPlayoutDelayMs(kSsrcX, 400));
|
||||
if (kMaxUnsignaledRecvStreams > 1) {
|
||||
EXPECT_EQ(300, channel_->AsReceiveChannel()
|
||||
->GetBaseMinimumPlayoutDelayMs(kSsrc1)
|
||||
.value_or(-1));
|
||||
EXPECT_EQ(
|
||||
300,
|
||||
receive_channel_->GetBaseMinimumPlayoutDelayMs(kSsrc1).value_or(-1));
|
||||
}
|
||||
EXPECT_EQ(400, channel_->AsReceiveChannel()
|
||||
->GetBaseMinimumPlayoutDelayMs(kSsrcX)
|
||||
.value_or(-1));
|
||||
EXPECT_EQ(300, channel_->AsReceiveChannel()
|
||||
->GetBaseMinimumPlayoutDelayMs(kSsrc0)
|
||||
.value_or(-1));
|
||||
EXPECT_EQ(
|
||||
400, receive_channel_->GetBaseMinimumPlayoutDelayMs(kSsrcX).value_or(-1));
|
||||
EXPECT_EQ(
|
||||
300, receive_channel_->GetBaseMinimumPlayoutDelayMs(kSsrc0).value_or(-1));
|
||||
// Check that it doesn't provide default values for unknown ssrc.
|
||||
EXPECT_FALSE(channel_->AsReceiveChannel()
|
||||
->GetBaseMinimumPlayoutDelayMs(kSsrcY)
|
||||
.has_value());
|
||||
EXPECT_FALSE(
|
||||
receive_channel_->GetBaseMinimumPlayoutDelayMs(kSsrcY).has_value());
|
||||
}
|
||||
|
||||
TEST_P(WebRtcVoiceEngineTestFake, SetsSyncGroupFromStreamId) {
|
||||
@ -3365,9 +3333,9 @@ TEST_P(WebRtcVoiceEngineTestFake, SetsSyncGroupFromStreamId) {
|
||||
sp.set_stream_ids({kStreamId});
|
||||
// Creating two channels to make sure that sync label is set properly for both
|
||||
// the default voice channel and following ones.
|
||||
EXPECT_TRUE(channel_->AsReceiveChannel()->AddRecvStream(sp));
|
||||
EXPECT_TRUE(receive_channel_->AddRecvStream(sp));
|
||||
sp.ssrcs[0] += 1;
|
||||
EXPECT_TRUE(channel_->AsReceiveChannel()->AddRecvStream(sp));
|
||||
EXPECT_TRUE(receive_channel_->AddRecvStream(sp));
|
||||
|
||||
ASSERT_EQ(2u, call_.GetAudioReceiveStreams().size());
|
||||
EXPECT_EQ(kStreamId,
|
||||
@ -3390,7 +3358,7 @@ TEST_P(WebRtcVoiceEngineTestFake, ConfiguresAudioReceiveStreamRtpExtensions) {
|
||||
EXPECT_TRUE(SetupSendStream());
|
||||
SetSendParameters(send_parameters_);
|
||||
for (uint32_t ssrc : ssrcs) {
|
||||
EXPECT_TRUE(channel_->AsReceiveChannel()->AddRecvStream(
|
||||
EXPECT_TRUE(receive_channel_->AddRecvStream(
|
||||
cricket::StreamParams::CreateLegacy(ssrc)));
|
||||
}
|
||||
|
||||
@ -3452,8 +3420,8 @@ TEST_P(WebRtcVoiceEngineTestFake, DeliverAudioPacket_Call) {
|
||||
const cricket::FakeAudioReceiveStream* s =
|
||||
call_.GetAudioReceiveStream(kAudioSsrc);
|
||||
EXPECT_EQ(0, s->received_packets());
|
||||
channel_->AsReceiveChannel()->OnPacketReceived(kPcmuPacket,
|
||||
/* packet_time_us */ -1);
|
||||
receive_channel_->OnPacketReceived(kPcmuPacket,
|
||||
/* packet_time_us */ -1);
|
||||
rtc::Thread::Current()->ProcessMessages(0);
|
||||
|
||||
EXPECT_EQ(1, s->received_packets());
|
||||
@ -3465,7 +3433,7 @@ TEST_P(WebRtcVoiceEngineTestFake, AssociateFirstSendChannel_SendCreatedFirst) {
|
||||
EXPECT_TRUE(SetupSendStream());
|
||||
EXPECT_TRUE(AddRecvStream(kSsrcY));
|
||||
EXPECT_EQ(kSsrcX, GetRecvStreamConfig(kSsrcY).rtp.local_ssrc);
|
||||
EXPECT_TRUE(channel_->AsSendChannel()->AddSendStream(
|
||||
EXPECT_TRUE(send_channel_->AddSendStream(
|
||||
cricket::StreamParams::CreateLegacy(kSsrcZ)));
|
||||
EXPECT_EQ(kSsrcX, GetRecvStreamConfig(kSsrcY).rtp.local_ssrc);
|
||||
EXPECT_TRUE(AddRecvStream(kSsrcW));
|
||||
@ -3475,12 +3443,12 @@ TEST_P(WebRtcVoiceEngineTestFake, AssociateFirstSendChannel_SendCreatedFirst) {
|
||||
TEST_P(WebRtcVoiceEngineTestFake, AssociateFirstSendChannel_RecvCreatedFirst) {
|
||||
EXPECT_TRUE(SetupRecvStream());
|
||||
EXPECT_EQ(0xFA17FA17u, GetRecvStreamConfig(kSsrcX).rtp.local_ssrc);
|
||||
EXPECT_TRUE(channel_->AsSendChannel()->AddSendStream(
|
||||
EXPECT_TRUE(send_channel_->AddSendStream(
|
||||
cricket::StreamParams::CreateLegacy(kSsrcY)));
|
||||
EXPECT_EQ(kSsrcY, GetRecvStreamConfig(kSsrcX).rtp.local_ssrc);
|
||||
EXPECT_TRUE(AddRecvStream(kSsrcZ));
|
||||
EXPECT_EQ(kSsrcY, GetRecvStreamConfig(kSsrcZ).rtp.local_ssrc);
|
||||
EXPECT_TRUE(channel_->AsSendChannel()->AddSendStream(
|
||||
EXPECT_TRUE(send_channel_->AddSendStream(
|
||||
cricket::StreamParams::CreateLegacy(kSsrcW)));
|
||||
EXPECT_EQ(kSsrcY, GetRecvStreamConfig(kSsrcX).rtp.local_ssrc);
|
||||
EXPECT_EQ(kSsrcY, GetRecvStreamConfig(kSsrcZ).rtp.local_ssrc);
|
||||
@ -3529,7 +3497,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetRawAudioSinkUnsignaledRecvStream) {
|
||||
EXPECT_NE(nullptr, GetRecvStream(kSsrc1).sink());
|
||||
|
||||
// If we remove and add a default stream, it should get the same sink.
|
||||
EXPECT_TRUE(channel_->AsReceiveChannel()->RemoveRecvStream(kSsrc1));
|
||||
EXPECT_TRUE(receive_channel_->RemoveRecvStream(kSsrc1));
|
||||
DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
|
||||
EXPECT_NE(nullptr, GetRecvStream(kSsrc1).sink());
|
||||
|
||||
@ -3579,13 +3547,13 @@ TEST_P(WebRtcVoiceEngineTestFake, OnReadyToSendSignalsNetworkState) {
|
||||
EXPECT_EQ(webrtc::kNetworkUp,
|
||||
call_.GetNetworkState(webrtc::MediaType::VIDEO));
|
||||
|
||||
channel_->AsSendChannel()->OnReadyToSend(false);
|
||||
send_channel_->OnReadyToSend(false);
|
||||
EXPECT_EQ(webrtc::kNetworkDown,
|
||||
call_.GetNetworkState(webrtc::MediaType::AUDIO));
|
||||
EXPECT_EQ(webrtc::kNetworkUp,
|
||||
call_.GetNetworkState(webrtc::MediaType::VIDEO));
|
||||
|
||||
channel_->AsSendChannel()->OnReadyToSend(true);
|
||||
send_channel_->OnReadyToSend(true);
|
||||
EXPECT_EQ(webrtc::kNetworkUp,
|
||||
call_.GetNetworkState(webrtc::MediaType::AUDIO));
|
||||
EXPECT_EQ(webrtc::kNetworkUp,
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user