NetEq: Add new method last_output_sample_rate_hz

webrtc/modules/audio_coding/neteq/neteq_impl.cc

 /*
  *  Copyright (c) 2012 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.
  */
 
@@ skipping 88 matching lines @@
       nack_enabled_(false) {
   LOG(LS_INFO) << "NetEq config: " << config.ToString();
   int fs = config.sample_rate_hz;
   if (fs != 8000 && fs != 16000 && fs != 32000 && fs != 48000) {
     LOG(LS_ERROR) << "Sample rate " << fs << " Hz not supported. " <<
         "Changing to 8000 Hz.";
     fs = 8000;
   }
   fs_hz_ = fs;
   fs_mult_ = fs / 8000;
+  last_output_sample_rate_hz_ = fs;
   output_size_samples_ = static_cast<size_t>(kOutputSizeMs * 8 * fs_mult_);
   decoder_frame_length_ = 3 * output_size_samples_;
   WebRtcSpl_Init();
   if (create_components) {
     SetSampleRateAndChannels(fs, 1);  // Default is 1 channel.
   }
   RTC_DCHECK(!vad_->enabled());
   if (config.enable_post_decode_vad) {
     vad_->Enable();
   }
@@ skipping 34 matching lines @@
   CriticalSectionScoped lock(crit_sect_.get());
   int error = GetAudioInternal(max_length, output_audio, samples_per_channel,
                                num_channels);
   if (error != 0) {
     error_code_ = error;
     return kFail;
   }
   if (type) {
     *type = LastOutputType();
   }
+  last_output_sample_rate_hz_ =
+      rtc::checked_cast<int>(*samples_per_channel * 100);
+  RTC_DCHECK(last_output_sample_rate_hz_ == 8000 ||
+             last_output_sample_rate_hz_ == 16000 ||
+             last_output_sample_rate_hz_ == 32000 ||
+             last_output_sample_rate_hz_ == 48000)
+      << "Unexpected sample rate " << last_output_sample_rate_hz_;
   return kOK;
 }
 
 int NetEqImpl::RegisterPayloadType(NetEqDecoder codec,
                                    uint8_t rtp_payload_type) {
   CriticalSectionScoped lock(crit_sect_.get());
   LOG(LS_VERBOSE) << "RegisterPayloadType "
                   << static_cast<int>(rtp_payload_type) << " "
                   << static_cast<int>(codec);
   int ret = decoder_database_->RegisterPayload(rtp_payload_type, codec);
@@ skipping 179 matching lines @@
   CriticalSectionScoped lock(crit_sect_.get());
   if (first_packet_) {
     // We don't have a valid RTP timestamp until we have decoded our first
     // RTP packet.
     return false;
   }
   *timestamp = timestamp_scaler_->ToExternal(playout_timestamp_);
   return true;
 }
 
+int NetEqImpl::last_output_sample_rate_hz() const {
+  CriticalSectionScoped lock(crit_sect_.get());
+  return last_output_sample_rate_hz_;
+}
+
 int NetEqImpl::SetTargetNumberOfChannels() {
   return kNotImplemented;
 }
 
 int NetEqImpl::SetTargetSampleRate() {
   return kNotImplemented;
 }
 
 int NetEqImpl::LastError() const {
   CriticalSectionScoped lock(crit_sect_.get());
@@ skipping 1655 matching lines @@
 
 void NetEqImpl::CreateDecisionLogic() {
   decision_logic_.reset(DecisionLogic::Create(fs_hz_, output_size_samples_,
                                               playout_mode_,
                                               decoder_database_.get(),
                                               *packet_buffer_.get(),
                                               delay_manager_.get(),
                                               buffer_level_filter_.get()));
 }
 }  // namespace webrtc