Upstream version 7.36.149.0

[platform/framework/web/crosswalk.git] / src / third_party / webrtc / modules / audio_coding / main / acm2 / audio_coding_module_impl.cc

diff --git a/src/third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc b/src/third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
index 4c64e07..6133d23 100644 (file)
--- a/src/third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
+++ b/src/third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
@@ -114,9 +114,10 @@ static int TimestampLessThan(uint32_t t1, uint32_t t2) {
 
 }  // namespace
 
-AudioCodingModuleImpl::AudioCodingModuleImpl(int id)
+AudioCodingModuleImpl::AudioCodingModuleImpl(
+    const AudioCodingModule::Config& config)
     : packetization_callback_(NULL),
-      id_(id),
+      id_(config.id),
       expected_codec_ts_(0xD87F3F9F),
       expected_in_ts_(0xD87F3F9F),
       send_codec_inst_(),
@@ -131,6 +132,7 @@ AudioCodingModuleImpl::AudioCodingModuleImpl(int id)
       stereo_send_(false),
       current_send_codec_idx_(-1),
       send_codec_registered_(false),
+      receiver_(config),
       acm_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
       vad_callback_(NULL),
       is_first_red_(true),
@@ -159,8 +161,6 @@ AudioCodingModuleImpl::AudioCodingModuleImpl(int id)
     mirror_codec_idx_[i] = -1;
   }
 
-  receiver_.set_id(id_);
-
   // Allocate memory for RED.
   red_buffer_ = new uint8_t[MAX_PAYLOAD_SIZE_BYTE];
 
@@ -201,7 +201,7 @@ AudioCodingModuleImpl::AudioCodingModuleImpl(int id)
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
                  "Cannot initialize receiver");
   }
-  WEBRTC_TRACE(webrtc::kTraceMemory, webrtc::kTraceAudioCoding, id, "Created");
+  WEBRTC_TRACE(webrtc::kTraceMemory, webrtc::kTraceAudioCoding, id_, "Created");
 }
 
 AudioCodingModuleImpl::~AudioCodingModuleImpl() {
@@ -1205,11 +1205,7 @@ int 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");
@@ -1365,13 +1361,17 @@ int AudioCodingModuleImpl::PreprocessToAddData(const AudioFrame& in_frame,
     // The result of the resampler is written to output frame.
     dest_ptr_audio = preprocess_frame_.data_;
 
-    preprocess_frame_.samples_per_channel_ = resampler_.Resample10Msec(
-        src_ptr_audio, in_frame.sample_rate_hz_, send_codec_inst_.plfreq,
-        preprocess_frame_.num_channels_, dest_ptr_audio);
+    preprocess_frame_.samples_per_channel_ =
+        resampler_.Resample10Msec(src_ptr_audio,
+                                  in_frame.sample_rate_hz_,
+                                  send_codec_inst_.plfreq,
+                                  preprocess_frame_.num_channels_,
+                                  AudioFrame::kMaxDataSizeSamples,
+                                  dest_ptr_audio);
 
     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;
@@ -1710,7 +1710,9 @@ int AudioCodingModuleImpl::PlayoutData10Ms(int desired_freq_hz,
   }
 
   audio_frame->id_ = id_;
-  audio_frame->energy_ = 0;
+  // The energy must be -1 in order to have the energy calculated later on in
+  // the AudioConferenceMixer module.
+  audio_frame->energy_ = static_cast<uint32_t>(-1);
   audio_frame->timestamp_ = 0;
   return 0;
 }
@@ -1976,10 +1978,6 @@ int AudioCodingModuleImpl::LeastRequiredDelayMs() const {
   return receiver_.LeastRequiredDelayMs();
 }
 
-const char* AudioCodingModuleImpl::Version() const {
-  return kExperimentalAcmVersion;
-}
-
 void AudioCodingModuleImpl::GetDecodingCallStatistics(
       AudioDecodingCallStats* call_stats) const {
   receiver_.GetDecodingCallStatistics(call_stats);