1 files changed, 17 insertions, 37 deletions

diff --git a/webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t_impl.h b/webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t_impl.h
index a986bc4..2e43fd3 100644
--- a/webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t_impl.h
+++ b/webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t_impl.h
@@ -8,29 +8,26 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_IMPL_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_IMPL_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_IMPL_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_IMPL_H_
 
-#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h"
-
-#include "webrtc/base/checks.h"
+#include "rtc_base/checks.h"
 
 namespace webrtc {
 
 template <typename T>
-AudioDecoderIsacT<T>::AudioDecoderIsacT()
-    : AudioDecoderIsacT(nullptr) {}
+bool AudioDecoderIsacT<T>::Config::IsOk() const {
+  return (sample_rate_hz == 16000 || sample_rate_hz == 32000);
+}
 
 template <typename T>
-AudioDecoderIsacT<T>::AudioDecoderIsacT(LockedIsacBandwidthInfo* bwinfo)
-    : bwinfo_(bwinfo), decoder_sample_rate_hz_(-1) {
+AudioDecoderIsacT<T>::AudioDecoderIsacT(const Config& config)
+    : sample_rate_hz_(config.sample_rate_hz) {
+  RTC_CHECK(config.IsOk()) << "Unsupported sample rate "
+                           << config.sample_rate_hz;
   RTC_CHECK_EQ(0, T::Create(&isac_state_));
   T::DecoderInit(isac_state_);
-  if (bwinfo_) {
-    IsacBandwidthInfo bi;
-    T::GetBandwidthInfo(isac_state_, &bi);
-    bwinfo_->Set(bi);
-  }
+  RTC_CHECK_EQ(0, T::SetDecSampRate(isac_state_, sample_rate_hz_));
 }
 
 template <typename T>
@@ -44,12 +41,7 @@ int AudioDecoderIsacT<T>::DecodeInternal(const uint8_t* encoded,
                                          int sample_rate_hz,
                                          int16_t* decoded,
                                          SpeechType* speech_type) {
-  RTC_CHECK(sample_rate_hz == 16000 || sample_rate_hz == 32000)
-      << "Unsupported sample rate " << sample_rate_hz;
-  if (sample_rate_hz != decoder_sample_rate_hz_) {
-    RTC_CHECK_EQ(0, T::SetDecSampRate(isac_state_, sample_rate_hz));
-    decoder_sample_rate_hz_ = sample_rate_hz;
-  }
+  RTC_CHECK_EQ(sample_rate_hz_, sample_rate_hz);
   int16_t temp_type = 1;  // Default is speech.
   int ret =
       T::DecodeInternal(isac_state_, encoded, encoded_len, decoded, &temp_type);
@@ -73,25 +65,13 @@ void AudioDecoderIsacT<T>::Reset() {
 }
 
 template <typename T>
-int AudioDecoderIsacT<T>::IncomingPacket(const uint8_t* payload,
-                                         size_t payload_len,
-                                         uint16_t rtp_sequence_number,
-                                         uint32_t rtp_timestamp,
-                                         uint32_t arrival_timestamp) {
-  int ret = T::UpdateBwEstimate(isac_state_, payload, payload_len,
-                                rtp_sequence_number, rtp_timestamp,
-                                arrival_timestamp);
-  if (bwinfo_) {
-    IsacBandwidthInfo bwinfo;
-    T::GetBandwidthInfo(isac_state_, &bwinfo);
-    bwinfo_->Set(bwinfo);
-  }
-  return ret;
+int AudioDecoderIsacT<T>::ErrorCode() {
+  return T::GetErrorCode(isac_state_);
 }
 
 template <typename T>
-int AudioDecoderIsacT<T>::ErrorCode() {
-  return T::GetErrorCode(isac_state_);
+int AudioDecoderIsacT<T>::SampleRateHz() const {
+  return sample_rate_hz_;
 }
 
 template <typename T>
@@ -101,4 +81,4 @@ size_t AudioDecoderIsacT<T>::Channels() const {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_IMPL_H_
+#endif  // MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_IMPL_H_