Removed sync packet support from NetEq.
I could not find a single place it was used, outside of the unittests
for the sync packet support itself.
Review-Url: https://codereview.webrtc.org/2309303002
Cr-Commit-Position: refs/heads/master@{#14130}
diff --git a/webrtc/modules/audio_coding/neteq/include/neteq.h b/webrtc/modules/audio_coding/neteq/include/neteq.h
index 9420cdb..cae56b3 100644
--- a/webrtc/modules/audio_coding/neteq/include/neteq.h
+++ b/webrtc/modules/audio_coding/neteq/include/neteq.h
@@ -128,8 +128,7 @@
kDecodedTooMuch,
kFrameSplitError,
kRedundancySplitError,
- kPacketBufferCorruption,
- kSyncPacketNotAccepted
+ kPacketBufferCorruption
};
// Creates a new NetEq object, with parameters set in |config|. The |config|
@@ -149,18 +148,6 @@
rtc::ArrayView<const uint8_t> payload,
uint32_t receive_timestamp) = 0;
- // Inserts a sync-packet into packet queue. Sync-packets are decoded to
- // silence and are intended to keep AV-sync intact in an event of long packet
- // losses when Video NACK is enabled but Audio NACK is not. Clients of NetEq
- // might insert sync-packet when they observe that buffer level of NetEq is
- // decreasing below a certain threshold, defined by the application.
- // Sync-packets should have the same payload type as the last audio payload
- // type, i.e. they cannot have DTMF or CNG payload type, nor a codec change
- // can be implied by inserting a sync-packet.
- // Returns kOk on success, kFail on failure.
- virtual int InsertSyncPacket(const WebRtcRTPHeader& rtp_header,
- uint32_t receive_timestamp) = 0;
-
// Instructs NetEq to deliver 10 ms of audio data. The data is written to
// |audio_frame|. All data in |audio_frame| is wiped; |data_|, |speech_type_|,
// |num_channels_|, |sample_rate_hz_|, |samples_per_channel_|, and