Moved codec-specific audio packet splitting into decoders.
There's still some code run specifically for Opus w/ FEC. It will be
addressed in a separate CL.
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2326003002
Cr-Commit-Position: refs/heads/master@{#14319}
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.cc b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
index ca20e5b..98588f4 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
@@ -658,21 +658,6 @@
}
}
- // Split payloads into smaller chunks. This also verifies that all payloads
- // are of a known payload type.
- ret = payload_splitter_->SplitAudio(&packet_list, *decoder_database_);
- if (ret != PayloadSplitter::kOK) {
- PacketBuffer::DeleteAllPackets(&packet_list);
- switch (ret) {
- case PayloadSplitter::kUnknownPayloadType:
- return kUnknownRtpPayloadType;
- case PayloadSplitter::kFrameSplitError:
- return kFrameSplitError;
- default:
- return kOtherError;
- }
- }
-
// Update bandwidth estimate, if the packet is not comfort noise.
if (!packet_list.empty() &&
!decoder_database_->IsComfortNoise(main_header.payloadType)) {
@@ -710,7 +695,7 @@
const RTPHeader& original_header = packet->header;
for (auto& result : results) {
RTC_DCHECK(result.frame);
- // Reuse the packet if possible
+ // Reuse the packet if possible.
if (!packet) {
packet.reset(new Packet);
packet->header = original_header;