Adapt to WebRTC r6818

Voice detection and level estimation support have been removed in
WebRTC audio processing.

Voice detection is disabled by macro ENABLE_WEBRTC_VOICEDETECTION.

Level estimation is disabled by macro ENABLE_WEBRTC_LEVELESTIMATION.

Library/TeamTalkLib/avstream/MediaPlayback.cpp

@@ -199,12 +199,12 @@ bool MediaPlayback::SetupSpeexPreprocess(bool enableagc, const SpeexAGC& agc,
 bool MediaPlayback::SetupWebRTCPreprocess(const webrtc::AudioProcessing::Config& webrtc)
 {
     if (!m_apm)
-        m_apm.reset(webrtc::AudioProcessingBuilder().Create());
+        m_apm = webrtc::AudioProcessingBuilder().Create();
 
     m_apm->ApplyConfig(webrtc);
     if (m_apm->Initialize() != webrtc::AudioProcessing::kNoError)
     {
-        m_apm.reset();
+        m_apm.release();
         return false;
     }
     return true;

Library/TeamTalkLib/avstream/MediaPlayback.h

@@ -134,7 +134,7 @@ class MediaPlayback : public soundsystem::StreamPlayer
     std::shared_ptr<SpeexPreprocess> m_preprocess_left, m_preprocess_right;
 #endif
 #if defined(ENABLE_WEBRTC)
-    std::unique_ptr<webrtc::AudioProcessing> m_apm;
+    rtc::scoped_refptr<webrtc::AudioProcessing> m_apm;
 #endif
     bool m_last_callback = false;
     ACE_Future<bool> m_drained;

Library/TeamTalkLib/avstream/SoundLoopback.cpp

@@ -120,7 +120,7 @@ bool SoundLoopback::StartTest(int inputdevid, int outputdevid,
 #if defined(ENABLE_WEBRTC)
     if (IsEnabled(apm_cfg))
     {
-        m_apm.reset(webrtc::AudioProcessingBuilder().Create());
+        m_apm = webrtc::AudioProcessingBuilder().Create();
         if (!m_apm)
         {
             StopTest();
@@ -214,7 +214,7 @@ bool SoundLoopback::StartDuplexTest(int inputdevid, int outputdevid,
 #if defined(ENABLE_WEBRTC)
     if (IsEnabled(apm_cfg))
     {
-        m_apm.reset(webrtc::AudioProcessingBuilder().Create());
+        m_apm = webrtc::AudioProcessingBuilder().Create();
         if (!m_apm)
         {
             StopTest();
@@ -254,7 +254,7 @@ bool SoundLoopback::StopTest()
 #endif
 
 #if defined(ENABLE_WEBRTC)
-    m_apm.reset();
+    m_apm.release();
 #endif
     m_preprocess_buffer_left.clear();
     m_preprocess_buffer_right.clear();

Library/TeamTalkLib/avstream/SoundLoopback.h

@@ -111,7 +111,7 @@ class SoundLoopback
     SpeexPreprocess m_preprocess_left, m_preprocess_right;
 #endif
 #if defined(ENABLE_WEBRTC)
-    std::unique_ptr<webrtc::AudioProcessing> m_apm;
+    rtc::scoped_refptr<webrtc::AudioProcessing> m_apm;
     std::vector<short> m_prev_buffer;
 #endif
     std::vector<short> m_preprocess_buffer_left, m_preprocess_buffer_right;

Library/TeamTalkLib/avstream/WebRTCPreprocess.cpp

@@ -30,8 +30,9 @@
 // webrtc::GainControlImpl queries this feature. Field trials is
 // excluded by passing rtc_exclude_field_trial_default=true to GN.
 namespace webrtc { namespace field_trial {
-std::string FindFullName(const std::string& trial)
+std::string FindFullName(absl::string_view trial_)
 {
+    std::string trial(trial_);;
 #if defined(UNICODE)
     ACE_TString str = LocalToUnicode(trial.c_str());
 #else
@@ -82,7 +83,7 @@ int WebRTCPreprocess(webrtc::AudioProcessing& apm, const media::AudioFrame& infr
     }
 
     // AudioProcessingStats
-    int output_rms_dbfs = 0;
+    int output_rms_dbfs = 0;
     bool voice_detected = false;
 
     int in_index = 0, out_index = 0, n = 0;
@@ -116,9 +117,11 @@ int WebRTCPreprocess(webrtc::AudioProcessing& apm, const media::AudioFrame& infr
         if (stats)
         {
             auto wstats = apm.GetStatistics();
+#if defined(ENABLE_WEBRTC_VOICEDETECTION)
             output_rms_dbfs += wstats.output_rms_dbfs.value_or(0);
             assert(!wstats.output_rms_dbfs.has_value() || wstats.output_rms_dbfs.value() <= 127);
             assert(!wstats.output_rms_dbfs.has_value() || wstats.output_rms_dbfs.value() >= 0);
+#endif
             voice_detected |= wstats.voice_detected.value_or(false);
         }
 
@@ -129,7 +132,9 @@ int WebRTCPreprocess(webrtc::AudioProcessing& apm, const media::AudioFrame& infr
 
     if (stats && n > 0)
     {
+#if defined(ENABLE_WEBRTC_VOICEDETECTION)
         stats->output_rms_dbfs = output_rms_dbfs / n;
+#endif
         stats->voice_detected = voice_detected;
     }
 

Library/TeamTalkLib/avstream/WebRTCPreprocess.h

@@ -25,10 +25,10 @@
 #define WEBRTCPREPROCESS_H
 
 #include <mystd/MyStd.h>
+#include <codec/MediaUtil.h>
 
 #include <audio_processing.h>
 
-#include <codec/MediaUtil.h>
 
 bool IsEnabled(const webrtc::AudioProcessing::Config& cfg);
 

Library/TeamTalkLib/bin/dll/Convert.cpp

@@ -1067,17 +1067,23 @@ void Convert(const WebRTCAudioPreprocessor& webrtc, webrtc::AudioProcessing::Con
         break;
     }
 
+#if defined(ENABLE_WEBRTC_VOICEDETECTION)
     result.voice_detection.enabled = webrtc.voicedetection.bEnable;
-
+#endif
+
     result.gain_controller2.enabled = webrtc.gaincontroller2.bEnable;
     result.gain_controller2.fixed_digital.gain_db = webrtc.gaincontroller2.fixeddigital.fGainDB;
     result.gain_controller2.adaptive_digital.enabled = webrtc.gaincontroller2.adaptivedigital.bEnable;
+#if defined(ENABLE_WEBRTC_GAINCONTROLLER2)
     result.gain_controller2.adaptive_digital.initial_saturation_margin_db = webrtc.gaincontroller2.adaptivedigital.fInitialSaturationMarginDB;
     result.gain_controller2.adaptive_digital.extra_saturation_margin_db = webrtc.gaincontroller2.adaptivedigital.fExtraSaturationMarginDB;
+#endif
     result.gain_controller2.adaptive_digital.max_gain_change_db_per_second = webrtc.gaincontroller2.adaptivedigital.fMaxGainChangeDBPerSecond;
     result.gain_controller2.adaptive_digital.max_output_noise_level_dbfs = webrtc.gaincontroller2.adaptivedigital.fMaxOutputNoiseLevelDBFS;
 
+#if defined(ENABLE_WEBRTC_LEVELESTIMATION)
     result.level_estimation.enabled = webrtc.levelestimation.bEnable;
+#endif
 }
 
 void Convert(const webrtc::AudioProcessing::Config& cfg, WebRTCAudioPreprocessor& result)
@@ -1086,21 +1092,27 @@ void Convert(const webrtc::AudioProcessing::Config& cfg, WebRTCAudioPreprocessor
     result.preamplifier.fFixedGainFactor = cfg.pre_amplifier.fixed_gain_factor;
 
     result.echocanceller.bEnable = cfg.echo_canceller.enabled;
-    
+
     result.noisesuppression.bEnable = cfg.noise_suppression.enabled;
     result.noisesuppression.nLevel = cfg.noise_suppression.level;
 
+#if defined(ENABLE_WEBRTC_VOICEDETECTION)
     result.voicedetection.bEnable = cfg.voice_detection.enabled;
+#endif
 
     result.gaincontroller2.bEnable = cfg.gain_controller2.enabled;
     result.gaincontroller2.fixeddigital.fGainDB = cfg.gain_controller2.fixed_digital.gain_db;
     result.gaincontroller2.adaptivedigital.bEnable = cfg.gain_controller2.adaptive_digital.enabled;
+#if defined(ENABLE_WEBRTC_GAINCONTROLLER2)
     result.gaincontroller2.adaptivedigital.fInitialSaturationMarginDB = cfg.gain_controller2.adaptive_digital.initial_saturation_margin_db;
     result.gaincontroller2.adaptivedigital.fExtraSaturationMarginDB = cfg.gain_controller2.adaptive_digital.extra_saturation_margin_db;
+#endif
     result.gaincontroller2.adaptivedigital.fMaxGainChangeDBPerSecond = cfg.gain_controller2.adaptive_digital.max_gain_change_db_per_second;
     result.gaincontroller2.adaptivedigital.fMaxOutputNoiseLevelDBFS = cfg.gain_controller2.adaptive_digital.max_output_noise_level_dbfs;
 
+#if defined(ENABLE_WEBRTC_LEVELESTIMATION)
     result.levelestimation.bEnable = cfg.level_estimation.enabled;
+#endif
 }
 #endif
 

Library/TeamTalkLib/teamtalk/client/AudioThread.cpp

@@ -189,7 +189,7 @@ void AudioThread::StopEncoder()
 #endif
 
 #if defined(ENABLE_WEBRTC)
-    m_apm.reset();
+    m_apm.release();
     m_aps.reset();
 #endif
 
@@ -235,7 +235,7 @@ bool AudioThread::UpdatePreprocessor(const teamtalk::AudioPreprocessor& preproce
 #if defined(ENABLE_WEBRTC)
     if (preprocess.preprocessor != AUDIOPREPROCESSOR_WEBRTC)
     {
-        m_apm.reset();
+        m_apm.release();
         m_aps.reset();
     }
 #endif
@@ -267,11 +267,11 @@ bool AudioThread::UpdatePreprocessor(const teamtalk::AudioPreprocessor& preproce
         }
 
         if (!m_apm)
-            m_apm.reset(webrtc::AudioProcessingBuilder().Create());
+            m_apm = webrtc::AudioProcessingBuilder().Create();
         m_apm->ApplyConfig(preprocess.webrtc);
         if (m_apm->Initialize() != webrtc::AudioProcessing::kNoError)
         {
-            m_apm.reset();
+            m_apm.release();
             MYTRACE(ACE_TEXT("Failed to initialize WebRTC audio preprocessor\n"));
             return false;
         }
@@ -424,7 +424,7 @@ void AudioThread::QueueAudio(ACE_Message_Block* mb_audio)
 
 bool AudioThread::IsVoiceActive()
 {
-#if defined(ENABLE_WEBRTC)
+#if defined(ENABLE_WEBRTC_VOICEDETECTION)
     std::unique_lock<std::recursive_mutex> g(m_preprocess_lock);
 
     if (m_apm && m_apm->GetConfig().voice_detection.enabled)
@@ -440,14 +440,13 @@ bool AudioThread::IsVoiceActive()
 
 int AudioThread::GetCurrentVoiceLevel()
 {
-#if defined(ENABLE_WEBRTC)
+#if defined(ENABLE_WEBRTC_LEVELESTIMATION)
     std::unique_lock<std::recursive_mutex> g(m_preprocess_lock);
 
     if (m_apm)
     {
         assert(m_aps);
-        auto cfg = m_apm->GetConfig();
-        if (cfg.level_estimation.enabled)
+        if (m_apm->GetConfig().level_estimation.enabled)
         {
             // WebRTC's maximum value for dB from digital full scale
             float value = 127.f - m_aps->output_rms_dbfs.value_or(0);
@@ -693,13 +692,15 @@ void AudioThread::PreprocessWebRTC(media::AudioFrame& audblock, bool& vad)
         MYTRACE(ACE_TEXT("WebRTC failed to process audio\n"));
     }
 
+#if defined(ENABLE_WEBRTC_VOICEDETECTION)
     vad = m_apm->GetConfig().voice_detection.enabled;
     if (vad)
     {
         assert(m_aps);
         if (m_aps->voice_detected.value_or(false))
             m_lastActive = ACE_OS::gettimeofday();
     }
+#endif
 }
 #endif
 

Library/TeamTalkLib/teamtalk/client/AudioThread.h

@@ -112,7 +112,7 @@ class AudioThread : protected ACE_Task<ACE_MT_SYNCH>
     std::unique_ptr<SpeexPreprocess> m_preprocess_left, m_preprocess_right;
 #endif
 #if defined(ENABLE_WEBRTC)
-    std::unique_ptr<webrtc::AudioProcessing> m_apm;
+    rtc::scoped_refptr<webrtc::AudioProcessing> m_apm;
     std::unique_ptr<webrtc::AudioProcessingStats> m_aps;
 #endif
 #if defined(ENABLE_SPEEX)