Move APM initialization into WebRtcVoiceEngine
TBR=kwiberg@webrtc.org
Bug: webrtc:4690
Change-Id: Icd8590d3f7476c1a841c7e2425d1134d224b1a53
Reviewed-on: https://webrtc-review.googlesource.com/23480
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20855}diff --git a/media/engine/apm_helpers.cc b/media/engine/apm_helpers.cc
index 0f67be1..5c12faa 100644
--- a/media/engine/apm_helpers.cc
+++ b/media/engine/apm_helpers.cc
@@ -18,6 +18,24 @@
namespace webrtc {
namespace apm_helpers {
+void Init(AudioProcessing* apm) {
+ RTC_DCHECK(apm);
+
+ constexpr int kMinVolumeLevel = 0;
+ constexpr int kMaxVolumeLevel = 255;
+
+ // This is the initialization which used to happen in VoEBase::Init(), but
+ // which is not covered by the WVoE::ApplyOptions().
+ if (apm->echo_cancellation()->enable_drift_compensation(false) != 0) {
+ RTC_DLOG(LS_ERROR) << "Failed to disable drift compensation.";
+ }
+ GainControl* gc = apm->gain_control();
+ if (gc->set_analog_level_limits(kMinVolumeLevel, kMaxVolumeLevel) != 0) {
+ RTC_DLOG(LS_ERROR) << "Failed to set analog level limits with minimum: "
+ << kMinVolumeLevel << " and maximum: " << kMaxVolumeLevel;
+ }
+}
+
AgcConfig GetAgcConfig(AudioProcessing* apm) {
RTC_DCHECK(apm);
AgcConfig result;
diff --git a/media/engine/apm_helpers.h b/media/engine/apm_helpers.h
index a4ddd49..caa485f 100644
--- a/media/engine/apm_helpers.h
+++ b/media/engine/apm_helpers.h
@@ -31,6 +31,7 @@
namespace apm_helpers {
+void Init(AudioProcessing* apm);
AgcConfig GetAgcConfig(AudioProcessing* apm);
void SetAgcConfig(AudioProcessing* apm,
const AgcConfig& config);
diff --git a/media/engine/apm_helpers_unittest.cc b/media/engine/apm_helpers_unittest.cc
index 4b537af..c20eb6f 100644
--- a/media/engine/apm_helpers_unittest.cc
+++ b/media/engine/apm_helpers_unittest.cc
@@ -29,6 +29,7 @@
Config config;
config.Set<ExperimentalAgc>(new ExperimentalAgc(false));
apm_ = rtc::scoped_refptr<AudioProcessing>(AudioProcessing::Create(config));
+ apm_helpers::Init(apm());
EXPECT_EQ(0, voe_wrapper_.base()->Init(
&mock_audio_device_, apm_,
MockAudioDecoderFactory::CreateEmptyFactory()));
@@ -102,14 +103,12 @@
TEST(ApmHelpersTest, AgcStatus_DefaultMode) {
TestHelper helper;
GainControl* gc = helper.apm()->gain_control();
+ EXPECT_FALSE(gc->is_enabled());
#if defined(TARGET_IPHONE_SIMULATOR) && TARGET_IPHONE_SIMULATOR
- EXPECT_FALSE(gc->is_enabled());
- EXPECT_EQ(GainControl::kAdaptiveDigital, gc->mode());
+ EXPECT_EQ(GainControl::kAdaptiveAnalog, gc->mode());
#elif defined(WEBRTC_IOS) || defined(WEBRTC_ANDROID)
- EXPECT_FALSE(gc->is_enabled());
EXPECT_EQ(GainControl::kFixedDigital, gc->mode());
#else
- EXPECT_TRUE(gc->is_enabled());
EXPECT_EQ(GainControl::kAdaptiveAnalog, gc->mode());
#endif
}
@@ -266,7 +265,7 @@
// of duplicating all relevant tests from audio_processing_test.cc.
TEST(ApmHelpersTest, HighPassFilter_DefaultMode) {
TestHelper helper;
- EXPECT_TRUE(helper.apm()->high_pass_filter()->is_enabled());
+ EXPECT_FALSE(helper.apm()->high_pass_filter()->is_enabled());
}
// TODO(solenberg): Move this test to a better place - added here for the sake
diff --git a/media/engine/fakewebrtcvoiceengine.h b/media/engine/fakewebrtcvoiceengine.h
index 9c9a428..444afdd 100644
--- a/media/engine/fakewebrtcvoiceengine.h
+++ b/media/engine/fakewebrtcvoiceengine.h
@@ -65,9 +65,8 @@
inited_ = true;
return 0;
}
- WEBRTC_FUNC(Terminate, ()) {
+ void Terminate() override {
inited_ = false;
- return 0;
}
webrtc::voe::TransmitMixer* transmit_mixer() override {
return transmit_mixer_;
diff --git a/media/engine/webrtcvoiceengine.cc b/media/engine/webrtcvoiceengine.cc
index 9630bc4..808c5d2 100644
--- a/media/engine/webrtcvoiceengine.cc
+++ b/media/engine/webrtcvoiceengine.cc
@@ -70,7 +70,6 @@
// See also http://tools.ietf.org/html/draft-jennings-rtcweb-qos-00
const rtc::DiffServCodePoint kAudioDscpValue = rtc::DSCP_EF;
-// Constants from voice_engine_defines.h.
const int kMinTelephoneEventCode = 0; // RFC4733 (Section 2.3.1)
const int kMaxTelephoneEventCode = 255;
@@ -299,7 +298,7 @@
#endif // WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE
RTC_CHECK(adm());
webrtc::adm_helpers::Init(adm());
-
+ webrtc::apm_helpers::Init(apm());
RTC_CHECK_EQ(0, voe_wrapper_->base()->Init(adm(), apm(), decoder_factory_));
transmit_mixer_ = voe_wrapper_->base()->transmit_mixer();
RTC_DCHECK(transmit_mixer_);
@@ -331,11 +330,6 @@
RTC_DCHECK(error);
}
- // Set default audio devices.
-#if !defined(WEBRTC_IOS)
- apm()->Initialize();
-#endif // !WEBRTC_IOS
-
// May be null for VoE injected for testing.
if (voe()->engine()) {
audio_state_ = webrtc::AudioState::Create(
@@ -723,6 +717,7 @@
webrtc::AudioProcessing* WebRtcVoiceEngine::apm() const {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(apm_);
return apm_.get();
}
diff --git a/media/engine/webrtcvoiceengine_unittest.cc b/media/engine/webrtcvoiceengine_unittest.cc
index 6573b7b..b03c8df 100644
--- a/media/engine/webrtcvoiceengine_unittest.cc
+++ b/media/engine/webrtcvoiceengine_unittest.cc
@@ -68,6 +68,16 @@
constexpr int kRtpHistoryMs = 5000;
+constexpr webrtc::GainControl::Mode kDefaultAgcMode =
+#if defined(WEBRTC_IOS) || defined(WEBRTC_ANDROID)
+ webrtc::GainControl::kFixedDigital;
+#else
+ webrtc::GainControl::kAdaptiveAnalog;
+#endif
+
+constexpr webrtc::NoiseSuppression::Level kDefaultNsLevel =
+ webrtc::NoiseSuppression::kHigh;
+
class FakeVoEWrapper : public cricket::VoEWrapper {
public:
explicit FakeVoEWrapper(cricket::FakeWebRtcVoiceEngine* engine)
@@ -137,7 +147,6 @@
EXPECT_CALL(*apm, GetConfig()).WillRepeatedly(ReturnPointee(&apm_config));
EXPECT_CALL(*apm, ApplyConfig(_)).WillRepeatedly(SaveArg<0>(&apm_config));
EXPECT_CALL(*apm, SetExtraOptions(testing::_));
- EXPECT_CALL(*apm, Initialize()).WillOnce(Return(0));
EXPECT_CALL(*apm, DetachAecDump());
StrictMock<MockTransmitMixer> transmit_mixer;
EXPECT_CALL(transmit_mixer, EnableStereoChannelSwapping(false));
@@ -183,12 +192,15 @@
EXPECT_CALL(*apm_, GetConfig()).WillRepeatedly(ReturnPointee(&apm_config_));
EXPECT_CALL(*apm_, ApplyConfig(_)).WillRepeatedly(SaveArg<0>(&apm_config_));
EXPECT_CALL(*apm_, SetExtraOptions(testing::_));
- EXPECT_CALL(*apm_, Initialize()).WillOnce(Return(0));
EXPECT_CALL(*apm_, DetachAecDump());
// Default Options.
EXPECT_CALL(apm_ec_, Enable(true)).WillOnce(Return(0));
EXPECT_CALL(apm_ec_, enable_metrics(true)).WillOnce(Return(0));
+ EXPECT_CALL(apm_ec_, enable_drift_compensation(false)).WillOnce(Return(0));
+ EXPECT_CALL(apm_gc_, set_mode(kDefaultAgcMode)).WillOnce(Return(0));
EXPECT_CALL(apm_gc_, Enable(true)).WillOnce(Return(0));
+ EXPECT_CALL(apm_gc_, set_analog_level_limits(0, 255)).WillOnce(Return(0));
+ EXPECT_CALL(apm_ns_, set_level(kDefaultNsLevel)).WillOnce(Return(0));
EXPECT_CALL(apm_ns_, Enable(true)).WillOnce(Return(0));
EXPECT_CALL(apm_vd_, Enable(true)).WillOnce(Return(0));
EXPECT_CALL(transmit_mixer_, EnableStereoChannelSwapping(false));
@@ -210,6 +222,7 @@
engine_->Init();
send_parameters_.codecs.push_back(kPcmuCodec);
recv_parameters_.codecs.push_back(kPcmuCodec);
+
// Default Options.
EXPECT_TRUE(IsHighPassFilterEnabled());
}
@@ -2221,6 +2234,7 @@
EXPECT_CALL(adm_,
BuiltInAGCIsAvailable()).Times(2).WillRepeatedly(Return(false));
EXPECT_CALL(adm_, SetAGC(true)).Times(2).WillRepeatedly(Return(0));
+ EXPECT_CALL(apm_gc_, set_mode(kDefaultAgcMode)).Times(2).WillOnce(Return(0));
EXPECT_CALL(apm_gc_, Enable(true)).Times(2).WillOnce(Return(0));
send_parameters_.options.tx_agc_target_dbov = 3;
send_parameters_.options.tx_agc_digital_compression_gain = 9;
@@ -2775,6 +2789,7 @@
EXPECT_CALL(adm_, SetAGC(false)).WillOnce(Return(0));
EXPECT_CALL(apm_ec_, Enable(true)).WillOnce(Return(0));
EXPECT_CALL(apm_ec_, enable_metrics(true)).WillOnce(Return(0));
+ EXPECT_CALL(apm_gc_, set_mode(kDefaultAgcMode)).WillOnce(Return(0));
EXPECT_CALL(apm_gc_, Enable(false)).WillOnce(Return(0));
send_parameters_.options.auto_gain_control = false;
SetSendParameters(send_parameters_);
@@ -2783,6 +2798,7 @@
EXPECT_CALL(adm_, SetAGC(true)).WillOnce(Return(0));
EXPECT_CALL(apm_ec_, Enable(true)).WillOnce(Return(0));
EXPECT_CALL(apm_ec_, enable_metrics(true)).WillOnce(Return(0));
+ EXPECT_CALL(apm_gc_, set_mode(kDefaultAgcMode)).WillOnce(Return(0));
EXPECT_CALL(apm_gc_, Enable(true)).WillOnce(Return(0));
send_parameters_.options.auto_gain_control = true;
send_parameters_.options.adjust_agc_delta = rtc::nullopt;
@@ -2792,7 +2808,9 @@
EXPECT_CALL(adm_, SetAGC(true)).WillOnce(Return(0));
EXPECT_CALL(apm_ec_, Enable(true)).WillOnce(Return(0));
EXPECT_CALL(apm_ec_, enable_metrics(true)).WillOnce(Return(0));
+ EXPECT_CALL(apm_gc_, set_mode(kDefaultAgcMode)).WillOnce(Return(0));
EXPECT_CALL(apm_gc_, Enable(true)).WillOnce(Return(0));
+ EXPECT_CALL(apm_ns_, set_level(kDefaultNsLevel)).WillOnce(Return(0));
EXPECT_CALL(apm_ns_, Enable(false)).WillOnce(Return(0));
EXPECT_CALL(apm_vd_, Enable(false)).WillOnce(Return(0));
EXPECT_CALL(transmit_mixer_, EnableStereoChannelSwapping(true));
@@ -2807,7 +2825,9 @@
EXPECT_CALL(adm_, SetAGC(true)).WillOnce(Return(0));
EXPECT_CALL(apm_ec_, Enable(true)).WillOnce(Return(0));
EXPECT_CALL(apm_ec_, enable_metrics(true)).WillOnce(Return(0));
+ EXPECT_CALL(apm_gc_, set_mode(kDefaultAgcMode)).WillOnce(Return(0));
EXPECT_CALL(apm_gc_, Enable(true)).WillOnce(Return(0));
+ EXPECT_CALL(apm_ns_, set_level(kDefaultNsLevel)).WillOnce(Return(0));
EXPECT_CALL(apm_ns_, Enable(false)).WillOnce(Return(0));
EXPECT_CALL(apm_vd_, Enable(false)).WillOnce(Return(0));
EXPECT_CALL(transmit_mixer_, EnableStereoChannelSwapping(true));
@@ -2858,7 +2878,9 @@
EXPECT_CALL(adm_, SetAGC(true)).Times(2).WillRepeatedly(Return(0));
EXPECT_CALL(apm_ec_, Enable(true)).Times(2).WillRepeatedly(Return(0));
EXPECT_CALL(apm_ec_, enable_metrics(true)).Times(2).WillRepeatedly(Return(0));
+ EXPECT_CALL(apm_gc_, set_mode(kDefaultAgcMode)).Times(2).WillOnce(Return(0));
EXPECT_CALL(apm_gc_, Enable(true)).Times(2).WillRepeatedly(Return(0));
+ EXPECT_CALL(apm_ns_, set_level(kDefaultNsLevel)).Times(2).WillOnce(Return(0));
EXPECT_CALL(apm_ns_, Enable(true)).Times(2).WillRepeatedly(Return(0));
EXPECT_TRUE(channel1->SetSendParameters(parameters_options_all));
EXPECT_EQ(parameters_options_all.options, channel1->options());
@@ -2871,7 +2893,9 @@
EXPECT_CALL(adm_, SetAGC(true)).WillOnce(Return(0));
EXPECT_CALL(apm_ec_, Enable(true)).WillOnce(Return(0));
EXPECT_CALL(apm_ec_, enable_metrics(true)).WillOnce(Return(0));
+ EXPECT_CALL(apm_gc_, set_mode(kDefaultAgcMode)).WillOnce(Return(0));
EXPECT_CALL(apm_gc_, Enable(true)).WillOnce(Return(0));
+ EXPECT_CALL(apm_ns_, set_level(kDefaultNsLevel)).WillOnce(Return(0));
EXPECT_CALL(apm_ns_, Enable(false)).WillOnce(Return(0));
EXPECT_TRUE(channel1->SetSendParameters(parameters_options_no_ns));
cricket::AudioOptions expected_options = parameters_options_all.options;
@@ -2886,7 +2910,9 @@
EXPECT_CALL(adm_, SetAGC(false)).WillOnce(Return(0));
EXPECT_CALL(apm_ec_, Enable(true)).WillOnce(Return(0));
EXPECT_CALL(apm_ec_, enable_metrics(true)).WillOnce(Return(0));
+ EXPECT_CALL(apm_gc_, set_mode(kDefaultAgcMode)).WillOnce(Return(0));
EXPECT_CALL(apm_gc_, Enable(false)).WillOnce(Return(0));
+ EXPECT_CALL(apm_ns_, set_level(kDefaultNsLevel)).WillOnce(Return(0));
EXPECT_CALL(apm_ns_, Enable(true)).WillOnce(Return(0));
EXPECT_TRUE(channel2->SetSendParameters(parameters_options_no_agc));
expected_options.echo_cancellation = true;
@@ -2897,21 +2923,27 @@
EXPECT_CALL(adm_, SetAGC(true)).WillOnce(Return(0));
EXPECT_CALL(apm_ec_, Enable(true)).WillOnce(Return(0));
EXPECT_CALL(apm_ec_, enable_metrics(true)).WillOnce(Return(0));
+ EXPECT_CALL(apm_gc_, set_mode(kDefaultAgcMode)).WillOnce(Return(0));
EXPECT_CALL(apm_gc_, Enable(true)).WillOnce(Return(0));
+ EXPECT_CALL(apm_ns_, set_level(kDefaultNsLevel)).WillOnce(Return(0));
EXPECT_CALL(apm_ns_, Enable(true)).WillOnce(Return(0));
EXPECT_TRUE(channel_->SetSendParameters(parameters_options_all));
EXPECT_CALL(adm_, SetAGC(true)).WillOnce(Return(0));
EXPECT_CALL(apm_ec_, Enable(true)).WillOnce(Return(0));
EXPECT_CALL(apm_ec_, enable_metrics(true)).WillOnce(Return(0));
+ EXPECT_CALL(apm_gc_, set_mode(kDefaultAgcMode)).WillOnce(Return(0));
EXPECT_CALL(apm_gc_, Enable(true)).WillOnce(Return(0));
+ EXPECT_CALL(apm_ns_, set_level(kDefaultNsLevel)).WillOnce(Return(0));
EXPECT_CALL(apm_ns_, Enable(false)).WillOnce(Return(0));
channel1->SetSend(true);
EXPECT_CALL(adm_, SetAGC(false)).WillOnce(Return(0));
EXPECT_CALL(apm_ec_, Enable(true)).WillOnce(Return(0));
EXPECT_CALL(apm_ec_, enable_metrics(true)).WillOnce(Return(0));
+ EXPECT_CALL(apm_gc_, set_mode(kDefaultAgcMode)).WillOnce(Return(0));
EXPECT_CALL(apm_gc_, Enable(false)).WillOnce(Return(0));
+ EXPECT_CALL(apm_ns_, set_level(kDefaultNsLevel)).WillOnce(Return(0));
EXPECT_CALL(apm_ns_, Enable(true)).WillOnce(Return(0));
channel2->SetSend(true);
@@ -2923,7 +2955,9 @@
EXPECT_CALL(adm_, SetAGC(false)).WillOnce(Return(0));
EXPECT_CALL(apm_ec_, Enable(true)).WillOnce(Return(0));
EXPECT_CALL(apm_ec_, enable_metrics(true)).WillOnce(Return(0));
+ EXPECT_CALL(apm_gc_, set_mode(kDefaultAgcMode)).WillOnce(Return(0));
EXPECT_CALL(apm_gc_, Enable(false)).WillOnce(Return(0));
+ EXPECT_CALL(apm_ns_, set_level(kDefaultNsLevel)).WillOnce(Return(0));
EXPECT_CALL(apm_ns_, Enable(false)).WillOnce(Return(0));
EXPECT_TRUE(channel2->SetSendParameters(parameters_options_no_agc_nor_ns));
expected_options.echo_cancellation = true;
diff --git a/test/mock_voice_engine.h b/test/mock_voice_engine.h
index 7fec120..57c6b58 100644
--- a/test/mock_voice_engine.h
+++ b/test/mock_voice_engine.h
@@ -95,7 +95,7 @@
AudioProcessing* external_apm,
const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory));
MOCK_METHOD0(transmit_mixer, voe::TransmitMixer*());
- MOCK_METHOD0(Terminate, int());
+ MOCK_METHOD0(Terminate, void());
MOCK_METHOD0(CreateChannel, int());
MOCK_METHOD1(CreateChannel, int(const ChannelConfig& config));
MOCK_METHOD1(DeleteChannel, int(int channel));
diff --git a/voice_engine/BUILD.gn b/voice_engine/BUILD.gn
index 23e4f7a..caafcca 100644
--- a/voice_engine/BUILD.gn
+++ b/voice_engine/BUILD.gn
@@ -28,7 +28,6 @@
"utility.h",
"voe_base_impl.cc",
"voe_base_impl.h",
- "voice_engine_defines.h",
"voice_engine_impl.cc",
"voice_engine_impl.h",
]
diff --git a/voice_engine/channel.cc b/voice_engine/channel.cc
index 20dc87f..bd8dbf4 100644
--- a/voice_engine/channel.cc
+++ b/voice_engine/channel.cc
@@ -57,6 +57,10 @@
constexpr int64_t kMaxRetransmissionWindowMs = 1000;
constexpr int64_t kMinRetransmissionWindowMs = 30;
+// Video Sync.
+constexpr int kVoiceEngineMinMinPlayoutDelayMs = 0;
+constexpr int kVoiceEngineMaxMinPlayoutDelayMs = 10000;
+
} // namespace
const int kTelephoneEventAttenuationdB = 10;
diff --git a/voice_engine/channel.h b/voice_engine/channel.h
index 548b331..8431b01 100644
--- a/voice_engine/channel.h
+++ b/voice_engine/channel.h
@@ -32,7 +32,6 @@
#include "voice_engine/audio_level.h"
#include "voice_engine/include/voe_base.h"
#include "voice_engine/shared_data.h"
-#include "voice_engine/voice_engine_defines.h"
namespace rtc {
class TimestampWrapAroundHandler;
diff --git a/voice_engine/include/voe_base.h b/voice_engine/include/voe_base.h
index 2f74fcc..a995325 100644
--- a/voice_engine/include/voe_base.h
+++ b/voice_engine/include/voe_base.h
@@ -102,8 +102,7 @@
virtual voe::TransmitMixer* transmit_mixer() = 0;
// Terminates all VoiceEngine functions and releases allocated resources.
- // Returns 0.
- virtual int Terminate() = 0;
+ virtual void Terminate() = 0;
// Creates a new channel and allocates the required resources for it.
// The second version accepts a |config| struct which includes an Audio Coding
diff --git a/voice_engine/shared_data.h b/voice_engine/shared_data.h
index d313c24..7978ff7 100644
--- a/voice_engine/shared_data.h
+++ b/voice_engine/shared_data.h
@@ -22,7 +22,6 @@
#include "rtc_base/thread_annotations.h"
#include "rtc_base/thread_checker.h"
#include "voice_engine/channel_manager.h"
-#include "voice_engine/voice_engine_defines.h"
class ProcessThread;
diff --git a/voice_engine/transmit_mixer.h b/voice_engine/transmit_mixer.h
index 6cfe070..42b6212 100644
--- a/voice_engine/transmit_mixer.h
+++ b/voice_engine/transmit_mixer.h
@@ -20,7 +20,6 @@
#include "rtc_base/criticalsection.h"
#include "voice_engine/audio_level.h"
#include "voice_engine/include/voe_base.h"
-#include "voice_engine/voice_engine_defines.h"
#if !defined(WEBRTC_ANDROID) && !defined(WEBRTC_IOS)
#define WEBRTC_VOICE_ENGINE_TYPING_DETECTION 1
diff --git a/voice_engine/utility.cc b/voice_engine/utility.cc
index ecc3e22..9398702 100644
--- a/voice_engine/utility.cc
+++ b/voice_engine/utility.cc
@@ -17,7 +17,6 @@
#include "modules/include/module_common_types.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
-#include "voice_engine/voice_engine_defines.h"
namespace webrtc {
namespace voe {
diff --git a/voice_engine/utility_unittest.cc b/voice_engine/utility_unittest.cc
index cd91d21..c798582 100644
--- a/voice_engine/utility_unittest.cc
+++ b/voice_engine/utility_unittest.cc
@@ -16,7 +16,6 @@
#include "rtc_base/format_macros.h"
#include "test/gtest.h"
#include "voice_engine/utility.h"
-#include "voice_engine/voice_engine_defines.h"
namespace webrtc {
namespace voe {
diff --git a/voice_engine/voe_base_impl.cc b/voice_engine/voe_base_impl.cc
index 695b06d..5d49872 100644
--- a/voice_engine/voe_base_impl.cc
+++ b/voice_engine/voe_base_impl.cc
@@ -55,6 +55,8 @@
RTC_DCHECK(shared_->transmit_mixer() != nullptr);
RTC_DCHECK(shared_->audio_device() != nullptr);
+ constexpr uint32_t kMaxVolumeLevel = 255;
+
uint32_t max_volume = 0;
uint16_t voe_mic_level = 0;
// Check for zero to skip this calculation; the consumer may use this to
@@ -155,57 +157,15 @@
shared_->set_audio_device(audio_device);
shared_->set_audio_processing(audio_processing);
- // Configure AudioProcessing components.
- // TODO(peah): Move this initialization to webrtcvoiceengine.cc.
- if (audio_processing->high_pass_filter()->Enable(true) != 0) {
- RTC_LOG_F(LS_ERROR) << "Failed to enable high pass filter.";
- return -1;
- }
- if (audio_processing->echo_cancellation()->enable_drift_compensation(false) !=
- 0) {
- RTC_LOG_F(LS_ERROR) << "Failed to disable drift compensation.";
- return -1;
- }
- if (audio_processing->noise_suppression()->set_level(kDefaultNsMode) != 0) {
- RTC_LOG_F(LS_ERROR) << "Failed to set noise suppression level: "
- << kDefaultNsMode;
- return -1;
- }
- GainControl* agc = audio_processing->gain_control();
- if (agc->set_analog_level_limits(kMinVolumeLevel, kMaxVolumeLevel) != 0) {
- RTC_LOG_F(LS_ERROR) << "Failed to set analog level limits with minimum: "
- << kMinVolumeLevel
- << " and maximum: " << kMaxVolumeLevel;
- return -1;
- }
- if (agc->set_mode(kDefaultAgcMode) != 0) {
- RTC_LOG_F(LS_ERROR) << "Failed to set mode: " << kDefaultAgcMode;
- return -1;
- }
- if (agc->Enable(kDefaultAgcState) != 0) {
- RTC_LOG_F(LS_ERROR) << "Failed to set agc state: " << kDefaultAgcState;
- return -1;
- }
-
-#ifdef WEBRTC_VOICE_ENGINE_AGC
- bool agc_enabled =
- agc->mode() == GainControl::kAdaptiveAnalog && agc->is_enabled();
- if (shared_->audio_device()->SetAGC(agc_enabled) != 0) {
- RTC_LOG_F(LS_ERROR) << "Failed to set agc to enabled: " << agc_enabled;
- // TODO(ajm): No error return here due to
- // https://code.google.com/p/webrtc/issues/detail?id=1464
- }
-#endif
-
RTC_DCHECK(decoder_factory);
decoder_factory_ = decoder_factory;
return 0;
}
-int VoEBaseImpl::Terminate() {
+void VoEBaseImpl::Terminate() {
rtc::CritScope cs(shared_->crit_sec());
- return TerminateInternal();
+ TerminateInternal();
}
int VoEBaseImpl::CreateChannel() {
@@ -436,7 +396,7 @@
return ret;
}
-int32_t VoEBaseImpl::TerminateInternal() {
+void VoEBaseImpl::TerminateInternal() {
// Delete any remaining channel objects
shared_->channel_manager().DestroyAllChannels();
@@ -446,7 +406,5 @@
shared_->set_audio_device(nullptr);
shared_->set_audio_processing(nullptr);
-
- return 0;
}
} // namespace webrtc
diff --git a/voice_engine/voe_base_impl.h b/voice_engine/voe_base_impl.h
index 4cd6c86..b96be9e 100644
--- a/voice_engine/voe_base_impl.h
+++ b/voice_engine/voe_base_impl.h
@@ -31,7 +31,7 @@
voe::TransmitMixer* transmit_mixer() override {
return shared_->transmit_mixer();
}
- int Terminate() override;
+ void Terminate() override;
int CreateChannel() override;
int CreateChannel(const ChannelConfig& config) override;
@@ -89,7 +89,7 @@
int32_t StopPlayout();
int32_t StartSend();
int32_t StopSend();
- int32_t TerminateInternal();
+ void TerminateInternal();
void GetPlayoutData(int sample_rate, size_t number_of_channels,
size_t number_of_frames, bool feed_data_to_apm,
diff --git a/voice_engine/voe_base_unittest.cc b/voice_engine/voe_base_unittest.cc
index 51fdf4e..56c3d13 100644
--- a/voice_engine/voe_base_unittest.cc
+++ b/voice_engine/voe_base_unittest.cc
@@ -28,7 +28,7 @@
}
~VoEBaseTest() {
- EXPECT_EQ(0, base_->Terminate());
+ base_->Terminate();
EXPECT_EQ(1, base_->Release());
EXPECT_TRUE(VoiceEngine::Delete(voe_));
}
diff --git a/voice_engine/voice_engine_defines.h b/voice_engine/voice_engine_defines.h
index 35f9492..b4d928a 100644
--- a/voice_engine/voice_engine_defines.h
+++ b/voice_engine/voice_engine_defines.h
@@ -8,43 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-/*
- * This file contains common constants for VoiceEngine, as well as
- * platform specific settings.
- */
-
-#ifndef VOICE_ENGINE_VOICE_ENGINE_DEFINES_H_
-#define VOICE_ENGINE_VOICE_ENGINE_DEFINES_H_
-
-#include "modules/audio_processing/include/audio_processing.h"
-
-namespace webrtc {
-
-// VolumeControl
-enum { kMinVolumeLevel = 0 };
-enum { kMaxVolumeLevel = 255 };
-
-// Audio processing
-const NoiseSuppression::Level kDefaultNsMode = NoiseSuppression::kModerate;
-const GainControl::Mode kDefaultAgcMode =
-#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
- GainControl::kAdaptiveDigital;
-#else
- GainControl::kAdaptiveAnalog;
-#endif
-const bool kDefaultAgcState =
-#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
- false;
-#else
- true;
-#endif
-
-// VideoSync
-// Lowest minimum playout delay
-enum { kVoiceEngineMinMinPlayoutDelayMs = 0 };
-// Highest minimum playout delay
-enum { kVoiceEngineMaxMinPlayoutDelayMs = 10000 };
-
-} // namespace webrtc
-
-#endif // VOICE_ENGINE_VOICE_ENGINE_DEFINES_H_
+// TODO(solenberg): Remove this file once downstream dependencies are removed.
