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Delete some unused AudioCodingModule methods

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Methods deleted:

  ReceiveFrequency, PlayoutFrequency, ReceiveCodec,
  SetMinimumPlayoutDelay, SetMaximumPlayoutDelay,
  SetBaseMinimumPlayoutDelayMs, GetBaseMinimumPlayoutDelayMs,
  PlayoutTimestamp, FilteredCurrentDelayMs, TargetDelayMs.

Became unused with cl
https://webrtc-review.googlesource.com/c/src/+/111504

Bug: None
Change-Id: Ie50e8e86a622661c3daa9db83a2e66489dcd2d98
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148071
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28918}
This commit is contained in:
Niels Möller
2019-08-13 15:54:15 +02:00
committed by Commit Bot
parent 728a0ee459
commit 5ceb4ac5ed
4 changed files with 12 additions and 188 deletions
Download Patch File Download Diff File Expand all files Collapse all files

91
modules/audio_coding/include/audio_coding_module.h
View File

@ -185,41 +185,10 @@ class AudioCodingModule {
//
virtual int32_t InitializeReceiver() = 0;
///////////////////////////////////////////////////////////////////////////
// int32_t ReceiveFrequency()
// Get sampling frequency of the last received payload.
//
// Return value:
// non-negative the sampling frequency in Hertz.
// -1 if an error has occurred.
//
virtual int32_t ReceiveFrequency() const = 0;
///////////////////////////////////////////////////////////////////////////
// int32_t PlayoutFrequency()
// Get sampling frequency of audio played out.
//
// Return value:
// the sampling frequency in Hertz.
//
virtual int32_t PlayoutFrequency() const = 0;
// Replace any existing decoders with the given payload type -> decoder map.
virtual void SetReceiveCodecs(
const std::map<int, SdpAudioFormat>& codecs) = 0;
///////////////////////////////////////////////////////////////////////////
// absl::optional<std::pair<int, SdpAudioFormat>> ReceiveCodec()
// Get the codec info associated with last received payload.
//
// Return value:
// A payload type and SdpAudioFormat describing the format associated with
// the last received payload.
// An empty Optional if no payload has yet been received.
//
virtual absl::optional<std::pair<int, SdpAudioFormat>> ReceiveCodec()
const = 0;
///////////////////////////////////////////////////////////////////////////
// int32_t IncomingPacket()
// Call this function to insert a parsed RTP packet into ACM.
@ -238,66 +207,6 @@ class AudioCodingModule {
const size_t payload_len_bytes,
const RTPHeader& rtp_header) = 0;
///////////////////////////////////////////////////////////////////////////
// int SetMinimumPlayoutDelay()
// Set a minimum for the playout delay, used for lip-sync. NetEq maintains
// such a delay unless channel condition yields to a higher delay.
//
// Input:
// -time_ms : minimum delay in milliseconds.
//
// Return value:
// -1 if failed to set the delay,
// 0 if the minimum delay is set.
//
virtual int SetMinimumPlayoutDelay(int time_ms) = 0;
///////////////////////////////////////////////////////////////////////////
// int SetMaximumPlayoutDelay()
// Set a maximum for the playout delay
//
// Input:
// -time_ms : maximum delay in milliseconds.
//
// Return value:
// -1 if failed to set the delay,
// 0 if the maximum delay is set.
//
virtual int SetMaximumPlayoutDelay(int time_ms) = 0;
// Sets a base minimum for the playout delay. Base minimum delay sets lower
// bound minimum delay value which is set via SetMinimumPlayoutDelay.
//
// Returns true if value was successfully set, false overwise.
virtual bool SetBaseMinimumPlayoutDelayMs(int delay_ms) = 0;
// Returns current value of base minimum delay in milliseconds.
virtual int GetBaseMinimumPlayoutDelayMs() const = 0;
///////////////////////////////////////////////////////////////////////////
// int32_t PlayoutTimestamp()
// The send timestamp of an RTP packet is associated with the decoded
// audio of the packet in question. This function returns the timestamp of
// the latest audio obtained by calling PlayoutData10ms(), or empty if no
// valid timestamp is available.
//
virtual absl::optional<uint32_t> PlayoutTimestamp() = 0;
///////////////////////////////////////////////////////////////////////////
// int FilteredCurrentDelayMs()
// Returns the current total delay from NetEq (packet buffer and sync buffer)
// in ms, with smoothing applied to even out short-time fluctuations due to
// jitter. The packet buffer part of the delay is not updated during DTX/CNG
// periods.
//
virtual int FilteredCurrentDelayMs() const = 0;
///////////////////////////////////////////////////////////////////////////
// int FilteredCurrentDelayMs()
// Returns the current target delay for NetEq in ms.
//
virtual int TargetDelayMs() const = 0;
///////////////////////////////////////////////////////////////////////////
// int32_t PlayoutData10Ms(
// Get 10 milliseconds of raw audio data for playout, at the given sampling