commit 549d80b979158a3535af299ee6277e77ca0873ca
author henrik.lundin <henrik.lundin@webrtc.org> Thu Aug 25 00:44:24 2016 -0700
committer Commit bot <commit-bot@chromium.org> Thu Aug 25 07:44:32 2016 +0000
tree b0469fec185038d8a7346747ee4c51f41af20ff4
parent fe8f48962aacab4a8437c9d98dfcda225329c68e

NetEq: only update current_rtp_payload_type_ when validated

The current_rtp_payload_type_ should only be updated when the packet is
actually inserted into the packet buffer, since then the payload type
has been validated. This CL removes an unvalidated setting of this value
that happened after SSRC change or upon first packet.

BUG=webrtc:5447

Review-Url: https://codereview.webrtc.org/2270793003
Cr-Commit-Position: refs/heads/master@{#13910}

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 e65466f..78e3112 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
@@ -598,7 +598,6 @@
 
     // Update codecs.
     timestamp_ = main_header.timestamp;
-    current_rtp_payload_type_ = main_header.payloadType;
 
     // Reset timestamp scaling.
     timestamp_scaler_->Reset();
@@ -744,13 +743,10 @@
     new_codec_ = true;
   }
 
-  if (current_rtp_payload_type_ != 0xFF) {
-    const DecoderDatabase::DecoderInfo* dec_info =
-        decoder_database_->GetDecoderInfo(current_rtp_payload_type_);
-    if (!dec_info) {
-      assert(false);  // Already checked that the payload type is known.
-    }
-  }
+  RTC_DCHECK(current_rtp_payload_type_ == 0xFF ||
+             decoder_database_->GetDecoderInfo(current_rtp_payload_type_))
+      << "Payload type " << static_cast<int>(current_rtp_payload_type_)
+      << " is unknown where it shouldn't be";
 
   if (update_sample_rate_and_channels && !packet_buffer_->Empty()) {
     // We do not use |current_rtp_payload_type_| to |set payload_type|, but

webrtc/modules/audio_coding/neteq/neteq_unittest.cc

diff --git a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
index b59046a..976c685 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
@@ -1521,7 +1521,10 @@
   EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_type_);
 
   // Insert some speech packets.
-  for (int i = 0; i < 3; ++i) {
+  const uint32_t first_speech_timestamp = timestamp;
+  int timeout_counter = 0;
+  do {
+    ASSERT_LT(timeout_counter++, 20) << "Test timed out";
     PopulateRtpInfo(seq_no, timestamp, &rtp_info);
     ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
     ++seq_no;
@@ -1530,7 +1533,7 @@
     // Pull audio once.
     ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
     ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
-  }
+  } while (!IsNewerTimestamp(out_frame_.timestamp_, first_speech_timestamp));
   // Verify speech output.
   EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
 }