Change NetEq::GetAudio to use AudioFrame

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 130 matching lines @@
   int error =
       InsertPacketInternal(rtp_header, kSyncPayload, receive_timestamp, true);
 
   if (error != 0) {
     error_code_ = error;
     return kFail;
   }
   return kOK;
 }
 
-int NetEqImpl::GetAudio(size_t max_length, int16_t* output_audio,
-                        size_t* samples_per_channel, size_t* num_channels,
-                        NetEqOutputType* type) {
+int NetEqImpl::GetAudio(AudioFrame* audio_frame, NetEqOutputType* type) {
   TRACE_EVENT0("webrtc", "NetEqImpl::GetAudio");
   rtc::CritScope lock(&crit_sect_);
-  int error = GetAudioInternal(max_length, output_audio, samples_per_channel,
-                               num_channels);
+  int error = GetAudioInternal(audio_frame);
+  RTC_DCHECK_EQ(
+      audio_frame->sample_rate_hz_,
+      rtc::checked_cast<int>(audio_frame->samples_per_channel_ * 100));
   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);
+  last_output_sample_rate_hz_ = audio_frame->sample_rate_hz_;
   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,
                                    const std::string& name,
@@ skipping 555 matching lines @@
   } else if (delay_manager_->last_pack_cng_or_dtmf() == -1) {
     // This is first "normal" packet after CNG or DTMF.
     // Reset packet time counter and measure time until next packet,
     // but don't update statistics.
     delay_manager_->set_last_pack_cng_or_dtmf(0);
     delay_manager_->ResetPacketIatCount();
   }
   return 0;
 }
 
-int NetEqImpl::GetAudioInternal(size_t max_length,
-                                int16_t* output,
-                                size_t* samples_per_channel,
-                                size_t* num_channels) {
+int NetEqImpl::GetAudioInternal(AudioFrame* audio_frame) {
   PacketList packet_list;
   DtmfEvent dtmf_event;
   Operations operation;
   bool play_dtmf;
   int return_value = GetDecision(&operation, &packet_list, &dtmf_event,
                                  &play_dtmf);
   if (return_value != 0) {
     last_mode_ = kModeError;
     return return_value;
   }
@@ skipping 94 matching lines @@
   if (last_mode_ != kModeRfc3389Cng) {
     comfort_noise_->Reset();
   }
 
   // Copy from |algorithm_buffer| to |sync_buffer_|.
   sync_buffer_->PushBack(*algorithm_buffer_);
 
   // Extract data from |sync_buffer_| to |output|.
   size_t num_output_samples_per_channel = output_size_samples_;
   size_t num_output_samples = output_size_samples_ * sync_buffer_->Channels();
-  if (num_output_samples > max_length) {
-    LOG(LS_WARNING) << "Output array is too short. " << max_length << " < " <<
-        output_size_samples_ << " * " << sync_buffer_->Channels();
-    num_output_samples = max_length;
-    num_output_samples_per_channel = max_length / sync_buffer_->Channels();
+  if (num_output_samples > AudioFrame::kMaxDataSizeSamples) {
+    LOG(LS_WARNING) << "Output array is too short. "
+                    << AudioFrame::kMaxDataSizeSamples << " < "
+                    << output_size_samples_ << " * "
+                    << sync_buffer_->Channels();
+    num_output_samples = AudioFrame::kMaxDataSizeSamples;
+    num_output_samples_per_channel =
+        AudioFrame::kMaxDataSizeSamples / sync_buffer_->Channels();
   }
-  const size_t samples_from_sync =
-      sync_buffer_->GetNextAudioInterleaved(num_output_samples_per_channel,
-                                            output);
-  *num_channels = sync_buffer_->Channels();
+  sync_buffer_->GetNextAudioInterleaved(num_output_samples_per_channel,
+                                        audio_frame);
+  audio_frame->sample_rate_hz_ = fs_hz_;
   if (sync_buffer_->FutureLength() < expand_->overlap_length()) {
     // The sync buffer should always contain |overlap_length| samples, but now
     // too many samples have been extracted. Reinstall the |overlap_length|
     // lookahead by moving the index.
     const size_t missing_lookahead_samples =
         expand_->overlap_length() - sync_buffer_->FutureLength();
     RTC_DCHECK_GE(sync_buffer_->next_index(), missing_lookahead_samples);
     sync_buffer_->set_next_index(sync_buffer_->next_index() -
                                  missing_lookahead_samples);
   }
-  if (samples_from_sync != output_size_samples_) {
-    LOG(LS_ERROR) << "samples_from_sync (" << samples_from_sync
+  if (audio_frame->samples_per_channel_ != output_size_samples_) {
+    LOG(LS_ERROR) << "audio_frame->samples_per_channel_ ("
+                  << audio_frame->samples_per_channel_
                   << ") != output_size_samples_ (" << output_size_samples_
                   << ")";
     // TODO(minyue): treatment of under-run, filling zeros
-    memset(output, 0, num_output_samples * sizeof(int16_t));
-    *samples_per_channel = output_size_samples_;
+    memset(audio_frame->data_, 0, num_output_samples * sizeof(int16_t));
     return kSampleUnderrun;
   }
-  *samples_per_channel = output_size_samples_;
 
   // Should always have overlap samples left in the |sync_buffer_|.
   RTC_DCHECK_GE(sync_buffer_->FutureLength(), expand_->overlap_length());
 
   if (play_dtmf) {
-    return_value = DtmfOverdub(dtmf_event, sync_buffer_->Channels(), output);
+    return_value =
+        DtmfOverdub(dtmf_event, sync_buffer_->Channels(), audio_frame->data_);
   }
 
   // Update the background noise parameters if last operation wrote data
   // straight from the decoder to the |sync_buffer_|. That is, none of the
   // operations that modify the signal can be followed by a parameter update.
   if ((last_mode_ == kModeNormal) ||
       (last_mode_ == kModeAccelerateFail) ||
       (last_mode_ == kModePreemptiveExpandFail) ||
       (last_mode_ == kModeRfc3389Cng) ||
       (last_mode_ == kModeCodecInternalCng)) {
@@ skipping 1137 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