Update talk to 58174641 together with http://review.webrtc.org/4319005/.
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5287 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc b/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
index 5da42ad..ac92198 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
@@ -19,6 +19,7 @@
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h"
+#include "webrtc/modules/audio_coding/main/acm2/call_statistics.h"
#include "webrtc/modules/audio_coding/main/acm2/nack.h"
#include "webrtc/modules/audio_coding/neteq4/interface/audio_decoder.h"
#include "webrtc/modules/audio_coding/neteq4/interface/neteq.h"
@@ -461,6 +462,7 @@
audio_frame->vad_activity_ = previous_audio_activity_;
SetAudioFrameActivityAndType(vad_enabled_, type, audio_frame);
previous_audio_activity_ = audio_frame->vad_activity_;
+ call_stats_.DecodedByNetEq(audio_frame->speech_type_);
return 0;
}
@@ -761,6 +763,9 @@
return false;
}
+ // Update statistics.
+ call_stats_.DecodedBySilenceGenerator();
+
// Set the values if already got a packet, otherwise set to default values.
if (last_audio_decoder_ >= 0) {
current_sample_rate_hz_ = ACMCodecDB::database_[last_audio_decoder_].plfreq;
@@ -832,6 +837,12 @@
}
}
+void AcmReceiver::GetDecodingCallStatistics(
+ AudioDecodingCallStats* stats) const {
+ CriticalSectionScoped lock(neteq_crit_sect_);
+ *stats = call_stats_.GetDecodingStatistics();
+}
+
} // namespace acm2
} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver.h b/webrtc/modules/audio_coding/main/acm2/acm_receiver.h
index 9267c1e..81eb520 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_receiver.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_receiver.h
@@ -18,6 +18,7 @@
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h"
+#include "webrtc/modules/audio_coding/main/acm2/call_statistics.h"
#include "webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h"
#include "webrtc/modules/audio_coding/neteq4/interface/neteq.h"
#include "webrtc/modules/interface/module_common_types.h"
@@ -320,6 +321,10 @@
//
NetEqBackgroundNoiseMode BackgroundNoiseModeForTest() const;
+ //
+ // Get statistics of calls to GetAudio().
+ void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const;
+
private:
int PayloadType2CodecIndex(uint8_t payload_type) const;
@@ -361,6 +366,8 @@
// initial delay is set.
scoped_ptr<InitialDelayManager::SyncStream> missing_packets_sync_stream_;
scoped_ptr<InitialDelayManager::SyncStream> late_packets_sync_stream_;
+
+ CallStatistics call_stats_;
};
} // namespace acm2
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module.gypi b/webrtc/modules/audio_coding/main/acm2/audio_coding_module.gypi
index 0754126..f51c3bf 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module.gypi
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module.gypi
@@ -84,6 +84,8 @@
'audio_coding_module.cc',
'audio_coding_module_impl.cc',
'audio_coding_module_impl.h',
+ 'call_statistics.cc',
+ 'call_statistics.h',
'initial_delay_manager.cc',
'initial_delay_manager.h',
'nack.cc',
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
index ce05218..4c64e07 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
@@ -20,6 +20,7 @@
#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h"
+#include "webrtc/modules/audio_coding/main/acm2/call_statistics.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
#include "webrtc/system_wrappers/interface/trace.h"
@@ -1979,6 +1980,11 @@
return kExperimentalAcmVersion;
}
+void AudioCodingModuleImpl::GetDecodingCallStatistics(
+ AudioDecodingCallStats* call_stats) const {
+ receiver_.GetDecodingCallStatistics(call_stats);
+}
+
} // namespace acm2
} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
index 17fab39..bc4ea0f 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
@@ -228,6 +228,8 @@
std::vector<uint16_t> GetNackList(int round_trip_time_ms) const;
+ void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const;
+
private:
int UnregisterReceiveCodecSafe(int payload_type);
diff --git a/webrtc/modules/audio_coding/main/acm2/call_statistics.cc b/webrtc/modules/audio_coding/main/acm2/call_statistics.cc
new file mode 100644
index 0000000..9153325
--- /dev/null
+++ b/webrtc/modules/audio_coding/main/acm2/call_statistics.cc
@@ -0,0 +1,55 @@
+/*
+ * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_coding/main/acm2/call_statistics.h"
+
+#include <cassert>
+
+namespace webrtc {
+
+namespace acm2 {
+
+void CallStatistics::DecodedByNetEq(AudioFrame::SpeechType speech_type) {
+ ++decoding_stat_.calls_to_neteq;
+ switch (speech_type) {
+ case AudioFrame::kNormalSpeech: {
+ ++decoding_stat_.decoded_normal;
+ break;
+ }
+ case AudioFrame::kPLC: {
+ ++decoding_stat_.decoded_plc;
+ break;
+ }
+ case AudioFrame::kCNG: {
+ ++decoding_stat_.decoded_cng;
+ break;
+ }
+ case AudioFrame::kPLCCNG: {
+ ++decoding_stat_.decoded_plc_cng;
+ break;
+ }
+ case AudioFrame::kUndefined: {
+ // If the audio is decoded by NetEq, |kUndefined| is not an option.
+ assert(false);
+ }
+ }
+}
+
+void CallStatistics::DecodedBySilenceGenerator() {
+ ++decoding_stat_.calls_to_silence_generator;
+}
+
+const AudioDecodingCallStats& CallStatistics::GetDecodingStatistics() const {
+ return decoding_stat_;
+}
+
+} // namespace acm2
+
+} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/acm2/call_statistics.h b/webrtc/modules/audio_coding/main/acm2/call_statistics.h
new file mode 100644
index 0000000..2aece0f
--- /dev/null
+++ b/webrtc/modules/audio_coding/main/acm2/call_statistics.h
@@ -0,0 +1,63 @@
+/*
+ * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_CALL_STATISTICS_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_CALL_STATISTICS_H_
+
+#include "webrtc/common_types.h"
+#include "webrtc/modules/interface/module_common_types.h"
+
+//
+// This class is for book keeping of calls to ACM. It is not useful to log API
+// calls which are supposed to be called every 10ms, e.g. PlayoutData10Ms(),
+// however, it is useful to know the number of such calls in a given time
+// interval. The current implementation covers calls to PlayoutData10Ms() with
+// detailed accounting of the decoded speech type.
+//
+// Thread Safety
+// =============
+// Please note that this class in not thread safe. The class must be protected
+// if different APIs are called from different threads.
+//
+
+namespace webrtc {
+
+namespace acm2 {
+
+class CallStatistics {
+ public:
+ CallStatistics() {}
+ ~CallStatistics() {}
+
+ // Call this method to indicate that NetEq engaged in decoding. |speech_type|
+ // is the audio-type according to NetEq.
+ void DecodedByNetEq(AudioFrame::SpeechType speech_type);
+
+ // Call this method to indicate that a decoding call resulted in generating
+ // silence, i.e. call to NetEq is bypassed and the output audio is zero.
+ void DecodedBySilenceGenerator();
+
+ // Get statistics for decoding. The statistics include the number of calls to
+ // NetEq and silence generator, as well as the type of speech pulled of off
+ // NetEq, c.f. declaration of AudioDecodingCallStats for detailed description.
+ const AudioDecodingCallStats& GetDecodingStatistics() const;
+
+ private:
+ // Reset the decoding statistics.
+ void ResetDecodingStatistics();
+
+ AudioDecodingCallStats decoding_stat_;
+};
+
+} // namespace acm2
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_CALL_STATISTICS_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/call_statistics_unittest.cc b/webrtc/modules/audio_coding/main/acm2/call_statistics_unittest.cc
new file mode 100644
index 0000000..61aadd7
--- /dev/null
+++ b/webrtc/modules/audio_coding/main/acm2/call_statistics_unittest.cc
@@ -0,0 +1,55 @@
+/*
+ * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "gtest/gtest.h"
+#include "webrtc/modules/audio_coding/main/acm2/call_statistics.h"
+
+namespace webrtc {
+
+namespace acm2 {
+
+TEST(CallStatisticsTest, InitializedZero) {
+ CallStatistics call_stats;
+ AudioDecodingCallStats stats;
+
+ stats = call_stats.GetDecodingStatistics();
+ EXPECT_EQ(0, stats.calls_to_neteq);
+ EXPECT_EQ(0, stats.calls_to_silence_generator);
+ EXPECT_EQ(0, stats.decoded_normal);
+ EXPECT_EQ(0, stats.decoded_cng);
+ EXPECT_EQ(0, stats.decoded_plc);
+ EXPECT_EQ(0, stats.decoded_plc_cng);
+}
+
+TEST(CallStatisticsTest, AllCalls) {
+ CallStatistics call_stats;
+ AudioDecodingCallStats stats;
+
+ call_stats.DecodedBySilenceGenerator();
+ call_stats.DecodedByNetEq(AudioFrame::kNormalSpeech);
+ call_stats.DecodedByNetEq(AudioFrame::kPLC);
+ call_stats.DecodedByNetEq(AudioFrame::kPLCCNG);
+ call_stats.DecodedByNetEq(AudioFrame::kCNG);
+
+ stats = call_stats.GetDecodingStatistics();
+ EXPECT_EQ(4, stats.calls_to_neteq);
+ EXPECT_EQ(1, stats.calls_to_silence_generator);
+ EXPECT_EQ(1, stats.decoded_normal);
+ EXPECT_EQ(1, stats.decoded_cng);
+ EXPECT_EQ(1, stats.decoded_plc);
+ EXPECT_EQ(1, stats.decoded_plc_cng);
+}
+
+} // namespace acm2
+
+} // namespace webrtc
+
+
+
diff --git a/webrtc/modules/audio_coding/main/interface/audio_coding_module.h b/webrtc/modules/audio_coding/main/interface/audio_coding_module.h
index f8b9690..db45add 100644
--- a/webrtc/modules/audio_coding/main/interface/audio_coding_module.h
+++ b/webrtc/modules/audio_coding/main/interface/audio_coding_module.h
@@ -931,6 +931,9 @@
// is returned.
//
virtual std::vector<uint16_t> GetNackList(int round_trip_time_ms) const = 0;
+
+ virtual void GetDecodingCallStatistics(
+ AudioDecodingCallStats* call_stats) const = 0;
};
struct AudioCodingModuleFactory {
diff --git a/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc b/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc
index 1e71a04..556f530 100644
--- a/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc
+++ b/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc
@@ -18,6 +18,7 @@
#include "webrtc/engine_configurations.h"
#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/call_statistics.h"
#include "webrtc/modules/audio_coding/main/source/acm_dtmf_detection.h"
#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h"
#include "webrtc/modules/audio_coding/main/source/acm_resampler.h"
@@ -2273,6 +2274,9 @@
{
CriticalSectionScoped lock(acm_crit_sect_);
+ // Update call statistics.
+ call_stats_.DecodedByNetEq(audio_frame->speech_type_);
+
if (update_nack) {
assert(nack_.get());
nack_->UpdateLastDecodedPacket(decoded_seq_num, decoded_timestamp);
@@ -2879,6 +2883,9 @@
return false;
}
+ // Record call to silence generator.
+ call_stats_.DecodedBySilenceGenerator();
+
// We stop accumulating packets, if the number of packets or the total size
// exceeds a threshold.
int max_num_packets;
@@ -3030,6 +3037,12 @@
return kLegacyAcmVersion;
}
+void AudioCodingModuleImpl::GetDecodingCallStatistics(
+ AudioDecodingCallStats* call_stats) const {
+ CriticalSectionScoped lock(acm_crit_sect_);
+ *call_stats = call_stats_.GetDecodingStatistics();
+}
+
} // namespace acm1
} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.h b/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.h
index 7acde17..f0b22f1 100644
--- a/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.h
+++ b/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.h
@@ -19,6 +19,7 @@
#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h"
#include "webrtc/modules/audio_coding/main/source/acm_neteq.h"
#include "webrtc/modules/audio_coding/main/source/acm_resampler.h"
+#include "webrtc/modules/audio_coding/main/acm2/call_statistics.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
namespace webrtc {
@@ -303,6 +304,8 @@
// Disable NACK.
void DisableNack();
+ void GetDecodingCallStatistics(AudioDecodingCallStats* call_stats) const;
+
private:
// Change required states after starting to receive the codec corresponding
// to |index|.
@@ -441,6 +444,8 @@
Clock* clock_;
scoped_ptr<acm2::Nack> nack_;
bool nack_enabled_;
+
+ acm2::CallStatistics call_stats_;
};
} // namespace acm1
diff --git a/webrtc/modules/audio_coding/neteq/webrtc_neteq.c b/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
index de1ccd1..fad690d 100644
--- a/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+++ b/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
@@ -1104,14 +1104,6 @@
/* If CN or internal CNG */
*outputType = kOutputCNG;
-#ifdef NETEQ_VAD
- }
- else if ( NetEqMainInst->DSPinst.VADInst.VADDecision == 0 )
- {
- /* post-decode VAD says passive speaker */
- *outputType = kOutputVADPassive;
-#endif /* NETEQ_VAD */
-
}
else if ((NetEqMainInst->DSPinst.w16_mode == MODE_EXPAND)
&& (NetEqMainInst->DSPinst.ExpandInst.w16_expandMuteFactor == 0))
@@ -1125,6 +1117,14 @@
/* PLC mode */
*outputType = kOutputPLC;
+#ifdef NETEQ_VAD
+ }
+ else if ( NetEqMainInst->DSPinst.VADInst.VADDecision == 0 )
+ {
+ /* post-decode VAD says passive speaker */
+ *outputType = kOutputVADPassive;
+#endif /* NETEQ_VAD */
+
}
else
{
diff --git a/webrtc/modules/audio_coding/neteq4/neteq_impl.cc b/webrtc/modules/audio_coding/neteq4/neteq_impl.cc
index 73ca5e4..fb27af2 100644
--- a/webrtc/modules/audio_coding/neteq4/neteq_impl.cc
+++ b/webrtc/modules/audio_coding/neteq4/neteq_impl.cc
@@ -1887,13 +1887,13 @@
assert(expand_.get());
if (last_mode_ == kModeCodecInternalCng || last_mode_ == kModeRfc3389Cng) {
return kOutputCNG;
- } else if (vad_->running() && !vad_->active_speech()) {
- return kOutputVADPassive;
} else if (last_mode_ == kModeExpand && expand_->MuteFactor(0) == 0) {
// Expand mode has faded down to background noise only (very long expand).
return kOutputPLCtoCNG;
} else if (last_mode_ == kModeExpand) {
return kOutputPLC;
+ } else if (vad_->running() && !vad_->active_speech()) {
+ return kOutputVADPassive;
} else {
return kOutputNormal;
}