Correct the calculation of discard rate.
Bug: webrtc:7903
Change-Id: Ib5d6fd882a994dd542b616e5fe1c75710346dd31
Reviewed-on: https://chromium-review.googlesource.com/575057
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19101}diff --git a/webrtc/modules/audio_coding/neteq/packet_buffer.h b/webrtc/modules/audio_coding/neteq/packet_buffer.h
index 806ca70..afd5f04 100644
--- a/webrtc/modules/audio_coding/neteq/packet_buffer.h
+++ b/webrtc/modules/audio_coding/neteq/packet_buffer.h
@@ -52,7 +52,7 @@
// the packet object.
// Returns PacketBuffer::kOK on success, PacketBuffer::kFlushed if the buffer
// was flushed due to overfilling.
- virtual int InsertPacket(Packet&& packet);
+ virtual int InsertPacket(Packet&& packet, StatisticsCalculator* stats);
// Inserts a list of packets into the buffer. The buffer will take over
// ownership of the packet objects.
@@ -66,7 +66,8 @@
PacketList* packet_list,
const DecoderDatabase& decoder_database,
rtc::Optional<uint8_t>* current_rtp_payload_type,
- rtc::Optional<uint8_t>* current_cng_rtp_payload_type);
+ rtc::Optional<uint8_t>* current_cng_rtp_payload_type,
+ StatisticsCalculator* stats);
// Gets the timestamp for the first packet in the buffer and writes it to the
// output variable |next_timestamp|.