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Early initialize recording on the ADM from WebRtcVoiceMediaChannel.

Browse Source 
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1827263002

Cr-Commit-Position: refs/heads/master@{#12369}
This commit is contained in:
solenberg 2016-04-14 13:56:37 -07:00 committed by Commit bot
parent 39c5e6a27f
commit d53a3f9758
6 changed files with 43 additions and 22 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

3
webrtc/api/test/fakeaudiocapturemodule.cc
View File

@ -205,8 +205,7 @@ int32_t FakeAudioCaptureModule::InitRecording() {
}
bool FakeAudioCaptureModule::RecordingIsInitialized() const {
ASSERT(false);
return 0;
return rec_is_initialized_;
}
int32_t FakeAudioCaptureModule::StartPlayout() {

11
webrtc/media/engine/webrtcvoiceengine.cc
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@ -1855,9 +1855,18 @@ void WebRtcVoiceMediaChannel::SetSend(bool send) {
return;
}
// Apply channel specific options when channel is enabled for sending.
// Apply channel specific options, and initialize the ADM for recording (this
// may take time on some platforms, e.g. Android).
if (send) {
engine()->ApplyOptions(options_);
// InitRecording() may return an error if the ADM is already recording.
if (!engine()->adm()->RecordingIsInitialized() &&
!engine()->adm()->Recording()) {
if (engine()->adm()->InitRecording() != 0) {
LOG(LS_WARNING) << "Failed to initialize recording";
}
}
}
// Change the settings on each send channel.

1
webrtc/media/engine/webrtcvoiceengine.h
View File

@ -235,6 +235,7 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
return send_codec_spec_.codec_inst.pltype != -1;
}
bool ValidateRtpParameters(const webrtc::RtpParameters& parameters);
void SetupRecording();
rtc::ThreadChecker worker_thread_checker_;

44
webrtc/media/engine/webrtcvoiceengine_unittest.cc
View File

@ -164,6 +164,16 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
return GetRecvStream(ssrc).GetConfig();
}
void SetSend(cricket::VoiceMediaChannel* channel, bool enable) {
ASSERT_TRUE(channel);
if (enable) {
EXPECT_CALL(adm_, RecordingIsInitialized()).WillOnce(Return(false));
EXPECT_CALL(adm_, Recording()).WillOnce(Return(false));
EXPECT_CALL(adm_, InitRecording()).WillOnce(Return(0));
}
channel->SetSend(enable);
}
void TestInsertDtmf(uint32_t ssrc, bool caller) {
EXPECT_TRUE(SetupChannel());
if (caller) {
@ -175,7 +185,7 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
// Test we can only InsertDtmf when the other side supports telephone-event.
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
channel_->SetSend(true);
SetSend(channel_, true);
EXPECT_FALSE(channel_->CanInsertDtmf());
EXPECT_FALSE(channel_->InsertDtmf(ssrc, 1, 111));
send_parameters_.codecs.push_back(kTelephoneEventCodec);
@ -2154,9 +2164,9 @@ TEST_F(WebRtcVoiceEngineTestFake, RecvAbsoluteSendTimeHeaderExtensions) {
TEST_F(WebRtcVoiceEngineTestFake, Send) {
EXPECT_TRUE(SetupSendStream());
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
channel_->SetSend(true);
SetSend(channel_, true);
EXPECT_TRUE(GetSendStream(kSsrc1).IsSending());
channel_->SetSend(false);
SetSend(channel_, false);
EXPECT_FALSE(GetSendStream(kSsrc1).IsSending());
}
@ -2166,7 +2176,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SendStateWithAndWithoutSource) {
EXPECT_TRUE(SetupSendStream());
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
EXPECT_TRUE(channel_->SetAudioSend(kSsrc1, true, nullptr, nullptr));
channel_->SetSend(true);
SetSend(channel_, true);
EXPECT_FALSE(GetSendStream(kSsrc1).IsSending());
EXPECT_TRUE(channel_->SetAudioSend(kSsrc1, true, nullptr, &fake_source_));
EXPECT_TRUE(GetSendStream(kSsrc1).IsSending());
@ -2180,7 +2190,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SendStateWhenStreamsAreRecreated) {
EXPECT_FALSE(GetSendStream(kSsrc1).IsSending());
// Turn on sending.
channel_->SetSend(true);
SetSend(channel_, true);
EXPECT_TRUE(GetSendStream(kSsrc1).IsSending());
// Changing RTP header extensions will recreate the AudioSendStream.
@ -2190,7 +2200,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SendStateWhenStreamsAreRecreated) {
EXPECT_TRUE(GetSendStream(kSsrc1).IsSending());
// Turn off sending.
channel_->SetSend(false);
SetSend(channel_, false);
EXPECT_FALSE(GetSendStream(kSsrc1).IsSending());
// Changing RTP header extensions will recreate the AudioSendStream.
@ -2215,7 +2225,7 @@ TEST_F(WebRtcVoiceEngineTestFake, CreateAndDeleteMultipleSendStreams) {
SetupForMultiSendStream();
// Set the global state for sending.
channel_->SetSend(true);
SetSend(channel_, true);
for (uint32_t ssrc : kSsrcs4) {
EXPECT_TRUE(channel_->AddSendStream(
@ -2286,14 +2296,14 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendWithMultipleSendStreams) {
}
// Set the global state for starting sending.
channel_->SetSend(true);
SetSend(channel_, true);
for (uint32_t ssrc : kSsrcs4) {
// Verify that we are in a sending state for all the send streams.
EXPECT_TRUE(GetSendStream(ssrc).IsSending());
}
// Set the global state for stopping sending.
channel_->SetSend(false);
SetSend(channel_, false);
for (uint32_t ssrc : kSsrcs4) {
// Verify that we are in a stop state for all the send streams.
EXPECT_FALSE(GetSendStream(ssrc).IsSending());
@ -2372,7 +2382,7 @@ TEST_F(WebRtcVoiceEngineTestFake, PlayoutWithMultipleStreams) {
EXPECT_TRUE(
channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(kSsrc2)));
int channel_num2 = voe_.GetLastChannel();
channel_->SetSend(true);
SetSend(channel_, true);
EXPECT_TRUE(GetSendStream(kSsrc1).IsSending());
// Make sure only the new stream is played out.
@ -2388,7 +2398,7 @@ TEST_F(WebRtcVoiceEngineTestFake, PlayoutWithMultipleStreams) {
EXPECT_TRUE(voe_.GetPlayout(channel_num3));
// Stop sending.
channel_->SetSend(false);
SetSend(channel_, false);
EXPECT_FALSE(GetSendStream(kSsrc1).IsSending());
// Stop playout.
@ -2420,11 +2430,11 @@ TEST_F(WebRtcVoiceEngineTestFake, CodianSend) {
EXPECT_EQ(0, agc_config.targetLeveldBOv);
send_parameters_.options = options_adjust_agc;
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
channel_->SetSend(true);
SetSend(channel_, true);
EXPECT_TRUE(GetSendStream(kSsrc1).IsSending());
EXPECT_EQ(0, voe_.GetAgcConfig(agc_config));
EXPECT_EQ(agc_config.targetLeveldBOv, 10); // level was attenuated
channel_->SetSend(false);
SetSend(channel_, false);
EXPECT_FALSE(GetSendStream(kSsrc1).IsSending());
EXPECT_EQ(0, voe_.GetAgcConfig(agc_config));
}
@ -2499,7 +2509,7 @@ TEST_F(WebRtcVoiceEngineTestFake, GetStats) {
// Start sending - this affects some reported stats.
{
cricket::VoiceMediaInfo info;
channel_->SetSend(true);
SetSend(channel_, true);
EXPECT_EQ(true, channel_->GetStats(&info));
VerifyVoiceSenderInfo(info.senders[0], true);
}
@ -2755,7 +2765,7 @@ TEST_F(WebRtcVoiceEngineTestFake, InsertDtmfOnSendStreamAsCallee) {
TEST_F(WebRtcVoiceEngineTestFake, TestSetPlayoutError) {
EXPECT_TRUE(SetupSendStream());
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
channel_->SetSend(true);
SetSend(channel_, true);
EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(2)));
EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(3)));
EXPECT_TRUE(channel_->SetPlayout(true));
@ -3036,7 +3046,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SetOptionOverridesViaChannels) {
EXPECT_TRUE(agc_enabled);
EXPECT_TRUE(ns_enabled);
channel1->SetSend(true);
SetSend(channel1.get(), true);
voe_.GetEcStatus(ec_enabled, ec_mode);
voe_.GetAgcStatus(agc_enabled, agc_mode);
voe_.GetNsStatus(ns_enabled, ns_mode);
@ -3044,7 +3054,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SetOptionOverridesViaChannels) {
EXPECT_TRUE(agc_enabled);
EXPECT_FALSE(ns_enabled);
channel2->SetSend(true);
SetSend(channel2.get(), true);
voe_.GetEcStatus(ec_enabled, ec_mode);
voe_.GetAgcStatus(agc_enabled, agc_mode);
voe_.GetNsStatus(ns_enabled, ns_mode);

1
webrtc/voice_engine/channel_proxy.h
View File

@ -70,7 +70,6 @@ class ChannelProxy {
virtual bool SetSendTelephoneEventPayloadType(int payload_type);
virtual bool SendTelephoneEventOutband(int event, int duration_ms);
virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink);
private:

5
webrtc/voice_engine/voe_base_impl.cc
View File

@ -620,11 +620,14 @@ int32_t VoEBaseImpl::StopPlayout() {
}
int32_t VoEBaseImpl::StartSend() {
if (!shared_->audio_device()->Recording()) {
if (!shared_->audio_device()->RecordingIsInitialized() &&
!shared_->audio_device()->Recording()) {
if (shared_->audio_device()->InitRecording() != 0) {
LOG_F(LS_ERROR) << "Failed to initialize recording";
return -1;
}
}
if (!shared_->audio_device()->Recording()) {
if (shared_->audio_device()->StartRecording() != 0) {
LOG_F(LS_ERROR) << "Failed to start recording";
return -1;