Change NetEq::GetAudio to use AudioFrame

webrtc/modules/audio_coding/neteq/neteq_impl.cc

Index: webrtc/modules/audio_coding/neteq/neteq_impl.cc
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.cc b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
index 93699821cfbc430cc6b9fa8daa7cbb2326e8ee64..ed04514170a182c024a86fb687e25d12b8011642 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
@@ -148,13 +148,13 @@ int NetEqImpl::InsertSyncPacket(const WebRtcRTPHeader& rtp_header,
   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;
@@ -162,8 +162,7 @@ int NetEqImpl::GetAudio(size_t max_length, int16_t* output_audio,
   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 ||
@@ -739,10 +738,7 @@ int NetEqImpl::InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
   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;
@@ -857,16 +853,22 @@ int NetEqImpl::GetAudioInternal(size_t max_length,
   // 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();
-  }
-  const size_t samples_from_sync =
-      sync_buffer_->GetNextAudioInterleaved(num_output_samples_per_channel,
-                                            output);
-  *num_channels = 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();
+  }
+  audio_frame->Reset();
+  const size_t samples_from_sync = sync_buffer_->GetNextAudioInterleaved(

[ivoc, 2016/03/03 16:28:15]

Does it make sense to rewrite this function to accept an AudioFrame as input?

[hlundin-webrtc, 2016/03/04 09:55:31]

Done. It took some more re-writing.

+      num_output_samples_per_channel, audio_frame->data_);
+  audio_frame->interleaved_ = true;
+  audio_frame->num_channels_ = sync_buffer_->Channels();
+  audio_frame->samples_per_channel_ = output_size_samples_;
+  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|
@@ -882,17 +884,16 @@ int NetEqImpl::GetAudioInternal(size_t max_length,
                   << ") != 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