Upstream version 7.36.149.0

[platform/framework/web/crosswalk.git] / src / third_party / webrtc / modules / audio_coding / main / source / audio_coding_module_impl.cc

diff --git a/src/third_party/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc b/src/third_party/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc
index 3802733..5bdbd01 100644 (file)
--- a/src/third_party/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc
+++ b/src/third_party/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"
@@ -1272,11 +1273,7 @@ int32_t AudioCodingModuleImpl::Add10MsData(
     return -1;
   }
 
-  // Allow for 8, 16, 32 and 48kHz input audio.
-  if ((audio_frame.sample_rate_hz_ != 8000)
-      && (audio_frame.sample_rate_hz_ != 16000)
-      && (audio_frame.sample_rate_hz_ != 32000)
-      && (audio_frame.sample_rate_hz_ != 48000)) {
+  if (audio_frame.sample_rate_hz_ > 48000) {
     assert(false);
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
                  "Cannot Add 10 ms audio, input frequency not valid");
@@ -1443,7 +1440,7 @@ int AudioCodingModuleImpl::PreprocessToAddData(const AudioFrame& in_frame,
 
     if (preprocess_frame_.samples_per_channel_ < 0) {
       WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
-                   "Cannot add 10 ms audio, resmapling failed");
+                   "Cannot add 10 ms audio, resampling failed");
       return -1;
     }
     preprocess_frame_.sample_rate_hz_ = send_codec_inst_.plfreq;
@@ -2253,18 +2250,10 @@ int32_t AudioCodingModuleImpl::PlayoutData10Ms(
   int decoded_seq_num;
   uint32_t decoded_timestamp;
   bool update_nack =
-      neteq_.DecodedRtpInfo(&decoded_seq_num, &decoded_timestamp) &&
-      nack_enabled_;  // Update NACK only if it is enabled.
-  audio_frame->num_channels_ = audio_frame_.num_channels_;
-  audio_frame->vad_activity_ = audio_frame_.vad_activity_;
-  audio_frame->speech_type_ = audio_frame_.speech_type_;
+      neteq_.DecodedRtpInfo(&decoded_seq_num, &decoded_timestamp);
 
-  stereo_mode = (audio_frame_.num_channels_ > 1);
-
-  // For stereo playout:
   // Master and Slave samples are interleaved starting with Master.
-  const uint16_t receive_freq =
-      static_cast<uint16_t>(audio_frame_.sample_rate_hz_);
+  uint16_t receive_freq;
   bool tone_detected = false;
   int16_t last_detected_tone;
   int16_t tone;
@@ -2273,7 +2262,17 @@ int32_t AudioCodingModuleImpl::PlayoutData10Ms(
   {
     CriticalSectionScoped lock(acm_crit_sect_);
 
-    if (update_nack) {
+    audio_frame->num_channels_ = audio_frame_.num_channels_;
+    audio_frame->vad_activity_ = audio_frame_.vad_activity_;
+    audio_frame->speech_type_ = audio_frame_.speech_type_;
+
+    stereo_mode = (audio_frame_.num_channels_ > 1);
+
+    receive_freq = static_cast<uint16_t>(audio_frame_.sample_rate_hz_);
+    // Update call statistics.
+    call_stats_.DecodedByNetEq(audio_frame->speech_type_);
+
+    if (nack_enabled_ && update_nack) {
       assert(nack_.get());
       nack_->UpdateLastDecodedPacket(decoded_seq_num, decoded_timestamp);
     }
@@ -2879,6 +2878,9 @@ bool AudioCodingModuleImpl::GetSilence(int desired_sample_rate_hz,
     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;
@@ -3026,6 +3028,12 @@ void AudioCodingModuleImpl::DisableNack() {
   nack_enabled_ = false;
 }
 
+void AudioCodingModuleImpl::GetDecodingCallStatistics(
+      AudioDecodingCallStats* call_stats) const {
+  CriticalSectionScoped lock(acm_crit_sect_);
+  *call_stats = call_stats_.GetDecodingStatistics();
+}
+
 }  // namespace acm1
 
 }  // namespace webrtc