NetEq: Flush and reset if the speech and cng sample rates mismatch

webrtc/modules/audio_coding/neteq/packet_buffer.cc

Index: webrtc/modules/audio_coding/neteq/packet_buffer.cc
diff --git a/webrtc/modules/audio_coding/neteq/packet_buffer.cc b/webrtc/modules/audio_coding/neteq/packet_buffer.cc
index 0e512b64d1277a43c91a03f3e230b5e08265ab2a..77c4578a6f4b343258b098f2659490c98a576252 100644
--- a/webrtc/modules/audio_coding/neteq/packet_buffer.cc
+++ b/webrtc/modules/audio_coding/neteq/packet_buffer.cc
@@ -22,7 +22,7 @@
 #include "webrtc/modules/audio_coding/neteq/tick_timer.h"
 
 namespace webrtc {
-
+namespace {
 // Predicate used when inserting packets in the buffer list.
 // Operator() returns true when |packet| goes before |new_packet|.
 class NewTimestampIsLarger {
@@ -38,6 +38,19 @@ class NewTimestampIsLarger {
   const Packet* new_packet_;
 };
 
+// Returns true if either payload types is empty; or if both payload types are

[minyue-webrtc, 2016/09/01 09:35:32]

"Returns true if either payload types is empty" can be moved out of the function
to match the function name better.

and can avoid  "rtc::Optional<uint8_t>(packet->header.payloadType)" as the first
argument for the use of this function

[hlundin-webrtc, 2016/09/01 10:31:10]

Done.

+// known to the decoder database, and have the same sample rate.
+bool EqualSampleRates(rtc::Optional<uint8_t> pt1,
+                      rtc::Optional<uint8_t> pt2,
+                      const DecoderDatabase& decoder_database) {
+  if (!pt1 || !pt2)
+    return true;
+  auto di1 = decoder_database.GetDecoderInfo(*pt1);
+  auto di2 = decoder_database.GetDecoderInfo(*pt2);
+  return di1 && di2 && di1->SampleRateHz() == di2->SampleRateHz();
+}
+}  // namespace
+
 PacketBuffer::PacketBuffer(size_t max_number_of_packets,
                            const TickTimer* tick_timer)
     : max_number_of_packets_(max_number_of_packets), tick_timer_(tick_timer) {}
@@ -126,8 +139,10 @@ int PacketBuffer::InsertPacketList(
           rtc::Optional<uint8_t>(packet->header.payloadType);
     } else if (!decoder_database.IsDtmf(packet->header.payloadType)) {
       // This must be speech.
-      if (*current_rtp_payload_type &&
-          **current_rtp_payload_type != packet->header.payloadType) {
+      if ((*current_rtp_payload_type &&
+           **current_rtp_payload_type != packet->header.payloadType) ||
+          !EqualSampleRates(rtc::Optional<uint8_t>(packet->header.payloadType),
+                            *current_cng_rtp_payload_type, decoder_database)) {
         *current_cng_rtp_payload_type = rtc::Optional<uint8_t>();
         Flush();
         flushed = true;