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Remove AudioCodingModule::IncomingPayload

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This method is no longer in use.

Bug: webrtc:3520
Change-Id: Ie1419fa95e6ef482e9adc1ed7af57af2c3510a65
Reviewed-on: https://webrtc-review.googlesource.com/4667
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20047}
This commit is contained in:
Henrik Lundin
2017-09-29 13:34:53 +02:00
committed by Commit Bot
parent a86ac6d198
commit d4a790fbea
2 changed files with 0 additions and 73 deletions
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29
modules/audio_coding/include/audio_coding_module.h
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@ -587,35 +587,6 @@ class AudioCodingModule {
const size_t payload_len_bytes,
const WebRtcRTPHeader& rtp_info) = 0;
///////////////////////////////////////////////////////////////////////////
// int32_t IncomingPayload()
// Call this API to push incoming payloads when there is no rtp-info.
// The rtp-info will be created in ACM. One usage for this API is when
// pre-encoded files are pushed in ACM
//
// Inputs:
// -incoming_payload : received payload.
// -payload_len_byte : the length, in bytes, of the received payload.
// -payload_type : the payload-type. This specifies which codec has
// to be used to decode the payload.
// -timestamp : send timestamp of the payload. ACM starts with
// a random value and increment it by the
// packet-size, which is given when the codec in
// question is registered by RegisterReceiveCodec().
// Therefore, it is essential to have the timestamp
// if the frame-size differ from the registered
// value or if the incoming payload contains DTX
// packets.
//
// Return value:
// -1 if failed to push in the payload
// 0 if payload is successfully pushed in.
//
virtual int32_t IncomingPayload(const uint8_t* incoming_payload,
const size_t payload_len_byte,
const uint8_t payload_type,
const uint32_t timestamp = 0) = 0;
///////////////////////////////////////////////////////////////////////////
// int SetMinimumPlayoutDelay()
// Set a minimum for the playout delay, used for lip-sync. NetEq maintains