Remove voe::TransmitMixer
TransmitMixer's functionality is moved into the AudioTransportProxy
owned by AudioState. This removes the need for an AudioTransport
implementation in VoEBaseImpl, which means that the proxy is no longer
a proxy, hence AudioTransportProxy is renamed to AudioTransportImpl.
In the short term, AudioState needs to know which AudioDeviceModule is
used, so it is added in AudioState::Config. AudioTransportImpl needs
to know which AudioSendStream:s are currently enabled to send, so
AudioState maintains a map of them, which is reduced into a simple
vector for AudioTransportImpl.
To encode and transmit audio,
AudioSendStream::OnAudioData(std::unique_ptr<AudioFrame> audio_frame)
is introduced, which is used in both the Chromium and standalone use
cases. This removes the need for two different instances of
voe::Channel::ProcessAndEncodeAudio(), so there is now only one,
taking an AudioFrame as argument. Callers need to allocate their own
AudioFrame:s, which is wasteful but not a regression since this was
already happening in the voe::Channel functions.
Most of the logic changed resides in
AudioTransportImpl::RecordedDataIsAvailable(), where two strange
things were found:
1. The clock drift parameter was ineffective since
apm->echo_cancellation()->enable_drift_compensation(false) is
called during initialization.
2. The output parameter 'new_mic_volume' was never set - instead it
was returned as a result, causing the ADM to never update the
analog mic gain
(https://cs.chromium.org/chromium/src/third_party/webrtc/voice_engine/voe_base_impl.cc?q=voe_base_impl.cc&dr&l=100).
Besides this, tests are updated, and some dead code is removed which
was found in the process.
Bug: webrtc:4690, webrtc:8591
Change-Id: I789d5296bf5efb7299a5ee05a4f3ce6abf9124b2
Reviewed-on: https://webrtc-review.googlesource.com/26681
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21301}diff --git a/audio/audio_send_stream_unittest.cc b/audio/audio_send_stream_unittest.cc
index 145a8e2..f605637 100644
--- a/audio/audio_send_stream_unittest.cc
+++ b/audio/audio_send_stream_unittest.cc
@@ -18,6 +18,7 @@
#include "call/fake_rtp_transport_controller_send.h"
#include "call/rtp_transport_controller_send_interface.h"
#include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
+#include "modules/audio_device/include/mock_audio_device.h"
#include "modules/audio_mixer/audio_mixer_impl.h"
#include "modules/audio_processing/include/audio_processing_statistics.h"
#include "modules/audio_processing/include/mock_audio_processing.h"
@@ -33,7 +34,6 @@
#include "test/mock_audio_encoder_factory.h"
#include "test/mock_voe_channel_proxy.h"
#include "test/mock_voice_engine.h"
-#include "voice_engine/transmit_mixer.h"
namespace webrtc {
namespace test {
@@ -58,9 +58,6 @@
const double kEchoReturnLossEnhancement = 101;
const double kResidualEchoLikelihood = -1.0f;
const double kResidualEchoLikelihoodMax = 23.0f;
-const int32_t kSpeechInputLevel = 96;
-const double kTotalInputEnergy = 0.25;
-const double kTotalInputDuration = 0.5;
const CallStatistics kCallStats = {
1345, 1678, 1901, 1234, 112, 13456, 17890, 1567, -1890, -1123};
const ReportBlock kReportBlock = {456, 780, 123, 567, 890, 132, 143, 13354};
@@ -85,14 +82,6 @@
uint32_t max_padding_bitrate_bps));
};
-class MockTransmitMixer : public voe::TransmitMixer {
- public:
- MOCK_CONST_METHOD0(AudioLevelFullRange, int16_t());
- MOCK_CONST_METHOD0(GetTotalInputEnergy, double());
- MOCK_CONST_METHOD0(GetTotalInputDuration, double());
- MOCK_CONST_METHOD0(typing_noise_detected, bool());
-};
-
std::unique_ptr<MockAudioEncoder> SetupAudioEncoderMock(
int payload_type,
const SdpAudioFormat& format) {
@@ -151,12 +140,12 @@
audio_encoder_(nullptr) {
using testing::Invoke;
- EXPECT_CALL(voice_engine_, audio_transport());
-
AudioState::Config config;
config.voice_engine = &voice_engine_;
config.audio_mixer = AudioMixerImpl::Create();
config.audio_processing = audio_processing_;
+ config.audio_device_module =
+ new rtc::RefCountedObject<MockAudioDeviceModule>();
audio_state_ = AudioState::Create(config);
SetupDefaultChannelProxy(audio_bwe_enabled);
@@ -301,17 +290,6 @@
.WillRepeatedly(Return(report_blocks));
EXPECT_CALL(*channel_proxy_, GetANAStatistics())
.WillRepeatedly(Return(ANAStats()));
- EXPECT_CALL(voice_engine_, transmit_mixer())
- .WillRepeatedly(Return(&transmit_mixer_));
-
- EXPECT_CALL(transmit_mixer_, AudioLevelFullRange())
- .WillRepeatedly(Return(kSpeechInputLevel));
- EXPECT_CALL(transmit_mixer_, GetTotalInputEnergy())
- .WillRepeatedly(Return(kTotalInputEnergy));
- EXPECT_CALL(transmit_mixer_, GetTotalInputDuration())
- .WillRepeatedly(Return(kTotalInputDuration));
- EXPECT_CALL(transmit_mixer_, typing_noise_detected())
- .WillRepeatedly(Return(true));
audio_processing_stats_.echo_return_loss = kEchoReturnLoss;
audio_processing_stats_.echo_return_loss_enhancement =
@@ -334,7 +312,6 @@
AudioSendStream::Config stream_config_;
testing::StrictMock<MockVoEChannelProxy>* channel_proxy_ = nullptr;
rtc::scoped_refptr<MockAudioProcessing> audio_processing_;
- MockTransmitMixer transmit_mixer_;
AudioProcessingStats audio_processing_stats_;
SimulatedClock simulated_clock_;
PacketRouter packet_router_;
@@ -447,9 +424,9 @@
(kIsacCodec.plfreq / 1000)),
stats.jitter_ms);
EXPECT_EQ(kCallStats.rttMs, stats.rtt_ms);
- EXPECT_EQ(static_cast<int32_t>(kSpeechInputLevel), stats.audio_level);
- EXPECT_EQ(kTotalInputEnergy, stats.total_input_energy);
- EXPECT_EQ(kTotalInputDuration, stats.total_input_duration);
+ EXPECT_EQ(0, stats.audio_level);
+ EXPECT_EQ(0, stats.total_input_energy);
+ EXPECT_EQ(0, stats.total_input_duration);
EXPECT_EQ(kEchoDelayMedian, stats.apm_statistics.delay_median_ms);
EXPECT_EQ(kEchoDelayStdDev, stats.apm_statistics.delay_standard_deviation_ms);
EXPECT_EQ(kEchoReturnLoss, stats.apm_statistics.echo_return_loss);
@@ -461,7 +438,7 @@
stats.apm_statistics.residual_echo_likelihood);
EXPECT_EQ(kResidualEchoLikelihoodMax,
stats.apm_statistics.residual_echo_likelihood_recent_max);
- EXPECT_TRUE(stats.typing_noise_detected);
+ EXPECT_FALSE(stats.typing_noise_detected);
}
TEST(AudioSendStreamTest, SendCodecAppliesAudioNetworkAdaptor) {
@@ -594,7 +571,5 @@
}
send_stream.Reconfigure(new_config);
}
-
-
} // namespace test
} // namespace webrtc