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Remove unused API on WebRtcVoiceEngine.

Browse Source 
BUG=1695
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46209004

Cr-Commit-Position: refs/heads/master@{#9153}
This commit is contained in:
Fredrik Solenberg 2015-05-07 16:01:26 +02:00
parent 2ea71c3279
commit 6179b89e53
4 changed files with 2 additions and 203 deletions
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29
talk/media/webrtc/fakewebrtcvoiceengine.h
View File

@ -41,7 +41,6 @@
#include "webrtc/base/gunit.h"
#include "webrtc/base/stringutils.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/video_engine/include/vie_network.h"
namespace cricket {
@ -203,8 +202,6 @@ class FakeWebRtcVoiceEngine
dtmf_type(106),
red_type(117),
nack_max_packets(0),
vie_network(NULL),
video_channel(-1),
send_ssrc(0),
send_audio_level_ext_(-1),
receive_audio_level_ext_(-1),
@ -235,8 +232,6 @@ class FakeWebRtcVoiceEngine
int dtmf_type;
int red_type;
int nack_max_packets;
webrtc::ViENetwork* vie_network;
int video_channel;
uint32 send_ssrc;
int send_audio_level_ext_;
int receive_audio_level_ext_;
@ -332,16 +327,6 @@ class FakeWebRtcVoiceEngine
int GetNACKMaxPackets(int channel) {
return channels_[channel]->nack_max_packets;
}
webrtc::ViENetwork* GetViENetwork(int channel) {
WEBRTC_ASSERT_CHANNEL(channel);
// WARNING: This pointer is for verification purposes only. Calling
// functions on it may result in undefined behavior!
return channels_[channel]->vie_network;
}
int GetVideoChannel(int channel) {
WEBRTC_ASSERT_CHANNEL(channel);
return channels_[channel]->video_channel;
}
const webrtc::PacketTime& GetLastRtpPacketTime(int channel) {
WEBRTC_ASSERT_CHANNEL(channel);
return channels_[channel]->last_rtp_packet_time;
@ -1028,19 +1013,9 @@ class FakeWebRtcVoiceEngine
unsigned short payloadSize));
WEBRTC_STUB(GetLastRemoteTimeStamp, (int channel,
uint32_t* lastRemoteTimeStamp));
WEBRTC_FUNC(SetVideoEngineBWETarget, (int channel,
WEBRTC_STUB(SetVideoEngineBWETarget, (int channel,
webrtc::ViENetwork* vie_network,
int video_channel)) {
WEBRTC_CHECK_CHANNEL(channel);
channels_[channel]->vie_network = vie_network;
channels_[channel]->video_channel = video_channel;
if (vie_network) {
// The interface is released here to avoid leaks. A test should not
// attempt to call functions on the interface stored in the channel.
vie_network->Release();
}
return 0;
}
int video_channel));
// webrtc::VoEVideoSync
WEBRTC_STUB(GetPlayoutBufferSize, (int& bufferMs));

57
talk/media/webrtc/webrtcvoiceengine.cc
View File

@ -1887,8 +1887,6 @@ WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel(WebRtcVoiceEngine *engine)
typing_noise_detected_(false),
desired_send_(SEND_NOTHING),
send_(SEND_NOTHING),
shared_bwe_vie_(NULL),
shared_bwe_vie_channel_(-1),
call_(nullptr),
default_receive_ssrc_(0) {
engine->RegisterChannel(this);
@ -1900,7 +1898,6 @@ WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel(WebRtcVoiceEngine *engine)
WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel() {
LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel "
<< voe_channel();
SetupSharedBandwidthEstimation(NULL, -1);
DCHECK(receive_streams_.empty() || call_);
// Remove any remaining send streams, the default channel will be deleted
@ -2003,11 +2000,6 @@ bool WebRtcVoiceMediaChannel::SetOptions(const AudioOptions& options) {
}
}
// Force update of Video Engine BWE forwarding to reflect experiment setting.
if (!SetupSharedBandwidthEstimation(shared_bwe_vie_,
shared_bwe_vie_channel_)) {
return false;
}
SetCall(call_);
LOG(LS_INFO) << "Set voice channel options. Current options: "
@ -2767,9 +2759,6 @@ bool WebRtcVoiceMediaChannel::AddRecvStream(const StreamParams& sp) {
receive_channels_.insert(std::make_pair(
default_receive_ssrc_,
new WebRtcVoiceChannelRenderer(voe_channel(), audio_transport)));
if (!SetupSharedBweOnChannel(voe_channel())) {
return false;
}
return SetPlayout(voe_channel(), playout_);
}
@ -2857,11 +2846,6 @@ bool WebRtcVoiceMediaChannel::ConfigureRecvChannel(int channel) {
return false;
}
// Set up channel to be able to forward incoming packets to video engine BWE.
if (!SetupSharedBweOnChannel(channel)) {
return false;
}
return SetPlayout(channel, playout_);
}
@ -3639,34 +3623,12 @@ int WebRtcVoiceMediaChannel::GetSendChannelNum(uint32 ssrc) {
return -1;
}
bool WebRtcVoiceMediaChannel::SetupSharedBandwidthEstimation(
webrtc::VideoEngine* vie, int vie_channel) {
shared_bwe_vie_ = vie;
shared_bwe_vie_channel_ = vie_channel;
if (!SetupSharedBweOnChannel(voe_channel())) {
return false;
}
for (ChannelMap::iterator it = receive_channels_.begin();
it != receive_channels_.end(); ++it) {
if (!SetupSharedBweOnChannel(it->second->channel())) {
return false;
}
}
return true;
}
void WebRtcVoiceMediaChannel::SetCall(webrtc::Call* call) {
DCHECK(thread_checker_.CalledOnValidThread());
DCHECK(!call || !shared_bwe_vie_);
DCHECK(!call || shared_bwe_vie_channel_ == -1);
for (const auto& it : receive_channels_) {
TryRemoveAudioRecvStream(it.first);
}
call_ = call;
for (const auto& it : receive_channels_) {
TryAddAudioRecvStream(it.first);
}
@ -3821,25 +3783,6 @@ bool WebRtcVoiceMediaChannel::SetHeaderExtension(ExtensionSetterFunction setter,
return true;
}
bool WebRtcVoiceMediaChannel::SetupSharedBweOnChannel(int voe_channel) {
webrtc::ViENetwork* vie_network = NULL;
int vie_channel = -1;
if (options_.combined_audio_video_bwe.GetWithDefaultIfUnset(false) &&
shared_bwe_vie_ != NULL && shared_bwe_vie_channel_ != -1) {
vie_network = webrtc::ViENetwork::GetInterface(shared_bwe_vie_);
vie_channel = shared_bwe_vie_channel_;
}
if (engine()->voe()->rtp()->SetVideoEngineBWETarget(voe_channel, vie_network,
vie_channel) == -1) {
LOG_RTCERR3(SetVideoEngineBWETarget, voe_channel, vie_network, vie_channel);
if (vie_network != NULL) {
// Don't fail if we're tearing down.
return false;
}
}
return true;
}
void WebRtcVoiceMediaChannel::TryAddAudioRecvStream(uint32 ssrc) {
DCHECK(thread_checker_.CalledOnValidThread());
// If we are hooked up to a webrtc::Call, create an AudioReceiveStream too.

9
talk/media/webrtc/webrtcvoiceengine.h
View File

@ -394,8 +394,6 @@ class WebRtcVoiceMediaChannel
int GetReceiveChannelNum(uint32 ssrc);
int GetSendChannelNum(uint32 ssrc);
bool SetupSharedBandwidthEstimation(webrtc::VideoEngine* vie,
int vie_channel);
void SetCall(webrtc::Call* call);
protected:
int GetLastEngineError() { return engine()->GetLastEngineError(); }
@ -439,8 +437,6 @@ class WebRtcVoiceMediaChannel
bool SetHeaderExtension(ExtensionSetterFunction setter, int channel_id,
const RtpHeaderExtension* extension);
bool SetupSharedBweOnChannel(int voe_channel);
void TryAddAudioRecvStream(uint32 ssrc);
void TryRemoveAudioRecvStream(uint32 ssrc);
@ -468,11 +464,6 @@ class WebRtcVoiceMediaChannel
bool typing_noise_detected_;
SendFlags desired_send_;
SendFlags send_;
// shared_bwe_vie_ and shared_bwe_vie_channel_ together identifies a WebRTC
// VideoEngine channel that this voice channel should forward incoming packets
// to for Bandwidth Estimation purposes.
webrtc::VideoEngine* shared_bwe_vie_;
int shared_bwe_vie_channel_;
webrtc::Call* call_;
// send_channels_ contains the channels which are being used for sending.

110
talk/media/webrtc/webrtcvoiceengine_unittest.cc
View File

@ -3410,116 +3410,6 @@ TEST(WebRtcVoiceEngineTest, CoInitialize) {
}
#endif
TEST_F(WebRtcVoiceEngineTestFake, ChangeCombinedAudioVideoBweOption) {
// Test that changing the combined_audio_video_bwe option results in the
// expected state changes in VoiceEngine.
cricket::ViEWrapper vie;
const int kVieCh = 667;
EXPECT_TRUE(SetupEngine());
cricket::WebRtcVoiceMediaChannel* media_channel =
static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_);
EXPECT_TRUE(media_channel->SetupSharedBandwidthEstimation(vie.engine(),
kVieCh));
EXPECT_TRUE(media_channel->AddRecvStream(
cricket::StreamParams::CreateLegacy(2)));
int recv_ch = voe_.GetLastChannel();
// Combined BWE should not be set up yet.
EXPECT_EQ(NULL, voe_.GetViENetwork(recv_ch));
EXPECT_EQ(-1, voe_.GetVideoChannel(recv_ch));
// Enable combined BWE option - now it should be set up.
cricket::AudioOptions options;
options.combined_audio_video_bwe.Set(true);
EXPECT_TRUE(media_channel->SetOptions(options));
EXPECT_EQ(vie.network(), voe_.GetViENetwork(recv_ch));
EXPECT_EQ(kVieCh, voe_.GetVideoChannel(recv_ch));
// Disable combined BWE option - should be disabled again.
options.combined_audio_video_bwe.Set(false);
EXPECT_TRUE(media_channel->SetOptions(options));
EXPECT_EQ(NULL, voe_.GetViENetwork(recv_ch));
EXPECT_EQ(-1, voe_.GetVideoChannel(recv_ch));
EXPECT_TRUE(media_channel->SetupSharedBandwidthEstimation(NULL, -1));
}
TEST_F(WebRtcVoiceEngineTestFake, SetupSharedBandwidthEstimation) {
// Test that calling SetupSharedBandwidthEstimation() on the voice media
// channel results in the expected state changes in VoiceEngine.
cricket::ViEWrapper vie1;
cricket::ViEWrapper vie2;
const int kVieCh1 = 667;
const int kVieCh2 = 70;
EXPECT_TRUE(SetupEngine());
cricket::WebRtcVoiceMediaChannel* media_channel =
static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_);
cricket::AudioOptions options;
options.combined_audio_video_bwe.Set(true);
EXPECT_TRUE(media_channel->SetOptions(options));
EXPECT_TRUE(media_channel->AddRecvStream(
cricket::StreamParams::CreateLegacy(2)));
int recv_ch = voe_.GetLastChannel();
// Combined BWE should not be set up yet.
EXPECT_EQ(NULL, voe_.GetViENetwork(recv_ch));
EXPECT_EQ(-1, voe_.GetVideoChannel(recv_ch));
// Register - should be enabled.
EXPECT_TRUE(media_channel->SetupSharedBandwidthEstimation(vie1.engine(),
kVieCh1));
EXPECT_EQ(vie1.network(), voe_.GetViENetwork(recv_ch));
EXPECT_EQ(kVieCh1, voe_.GetVideoChannel(recv_ch));
// Re-register - should still be enabled.
EXPECT_TRUE(media_channel->SetupSharedBandwidthEstimation(vie2.engine(),
kVieCh2));
EXPECT_EQ(vie2.network(), voe_.GetViENetwork(recv_ch));
EXPECT_EQ(kVieCh2, voe_.GetVideoChannel(recv_ch));
// Unregister - should be disabled again.
EXPECT_TRUE(media_channel->SetupSharedBandwidthEstimation(NULL, -1));
EXPECT_EQ(NULL, voe_.GetViENetwork(recv_ch));
EXPECT_EQ(-1, voe_.GetVideoChannel(recv_ch));
}
TEST_F(WebRtcVoiceEngineTestFake, ConfigureCombinedBweForNewRecvStreams) {
// Test that adding receive streams after enabling combined bandwidth
// estimation will correctly configure each channel.
cricket::ViEWrapper vie;
const int kVieCh = 667;
EXPECT_TRUE(SetupEngine());
cricket::WebRtcVoiceMediaChannel* media_channel =
static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_);
EXPECT_TRUE(media_channel->SetupSharedBandwidthEstimation(vie.engine(),
kVieCh));
cricket::AudioOptions options;
options.combined_audio_video_bwe.Set(true);
EXPECT_TRUE(media_channel->SetOptions(options));
static const uint32 kSsrcs[] = {1, 2, 3, 4};
int voe_channels[ARRAY_SIZE(kSsrcs)] = {0};
for (unsigned int i = 0; i < ARRAY_SIZE(kSsrcs); ++i) {
EXPECT_TRUE(media_channel->AddRecvStream(
cricket::StreamParams::CreateLegacy(kSsrcs[i])));
int recv_ch = media_channel->GetReceiveChannelNum(kSsrcs[i]);
EXPECT_NE(-1, recv_ch);
voe_channels[i] = recv_ch;
EXPECT_EQ(vie.network(), voe_.GetViENetwork(recv_ch));
EXPECT_EQ(kVieCh, voe_.GetVideoChannel(recv_ch));
}
EXPECT_TRUE(media_channel->SetupSharedBandwidthEstimation(NULL, -1));
for (unsigned int i = 0; i < ARRAY_SIZE(voe_channels); ++i) {
EXPECT_EQ(NULL, voe_.GetViENetwork(voe_channels[i]));
EXPECT_EQ(-1, voe_.GetVideoChannel(voe_channels[i]));
}
}
TEST_F(WebRtcVoiceEngineTestFake, ChangeCombinedBweOption_Call) {
// Test that changing the combined_audio_video_bwe option results in the
// expected state changes on an associated Call.