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Remove no-op and unused methods from AudioCodingModule

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This CL removes the following no-op and/or unused methods from
AudioCodingModule and AudioCodingModuleImpl:

ConfigISACBandwidthEstimator
DecoderEstimatedBandwidth
IsInternalDTXReplacedWithWebRtc
REDPayloadISAC
ReplaceInternalDTXWithWebRtc
ResetDecoder
ResetEncoder
SendBitrate
SetReceivedEstimatedBandwidth

R=henrik.lundin@webrtc.org

Review URL: https://codereview.webrtc.org/1308283003 .

Cr-Commit-Position: refs/heads/master@{#9773}
This commit is contained in:
Karl Wiberg
2015-08-25 09:37:04 +02:00
parent 7ef9d9104d
commit dd00f113a9
8 changed files with 9 additions and 334 deletions
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125
webrtc/modules/audio_coding/main/interface/audio_coding_module.h
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@ -193,17 +193,6 @@ class AudioCodingModule {
// Sender
//
///////////////////////////////////////////////////////////////////////////
// int32_t ResetEncoder()
// This API resets the states of encoder. All the encoder settings, such as
// send-codec or VAD/DTX, will be preserved.
//
// Return value:
// -1 if failed to initialize,
// 0 if succeeded.
//
virtual int32_t ResetEncoder() = 0;
///////////////////////////////////////////////////////////////////////////
// int32_t RegisterSendCodec()
// Registers a codec, specified by |send_codec|, as sending codec.
@ -261,39 +250,11 @@ class AudioCodingModule {
//
virtual int32_t SendFrequency() const = 0;
///////////////////////////////////////////////////////////////////////////
// int32_t Bitrate()
// Get encoding bit-rate in bits per second.
//
// Return value:
// positive; encoding rate in bits/sec,
// -1 if an error is happened.
//
virtual int32_t SendBitrate() const = 0;
///////////////////////////////////////////////////////////////////////////
// Sets the bitrate to the specified value in bits/sec. If the value is not
// supported by the codec, it will choose another appropriate value.
virtual void SetBitRate(int bitrate_bps) = 0;
///////////////////////////////////////////////////////////////////////////
// int32_t SetReceivedEstimatedBandwidth()
// Set available bandwidth [bits/sec] of the up-link channel.
// This information is used for traffic shaping, and is currently only
// supported if iSAC is the send codec.
//
// Input:
// -bw : bandwidth in bits/sec estimated for
// up-link.
// Return value
// -1 if error occurred in setting the bandwidth,
// 0 bandwidth is set successfully.
//
// TODO(henrik.lundin) Unused. Remove?
virtual int32_t SetReceivedEstimatedBandwidth(
const int32_t bw) = 0;
///////////////////////////////////////////////////////////////////////////
// int32_t RegisterTransportCallback()
// Register a transport callback which will be called to deliver
// the encoded buffers whenever Process() is called and a
@ -465,39 +426,6 @@ class AudioCodingModule {
virtual int32_t VAD(bool* dtx_enabled, bool* vad_enabled,
ACMVADMode* vad_mode) const = 0;
///////////////////////////////////////////////////////////////////////////
// int32_t ReplaceInternalDTXWithWebRtc()
// Used to replace codec internal DTX scheme with WebRtc.
//
// Input:
// -use_webrtc_dtx : if false (default) the codec built-in DTX/VAD
// scheme is used, otherwise the internal DTX is
// replaced with WebRtc DTX/VAD.
//
// Return value:
// -1 if failed to replace codec internal DTX with WebRtc,
// 0 if succeeded.
//
virtual int32_t ReplaceInternalDTXWithWebRtc(
const bool use_webrtc_dtx = false) = 0;
///////////////////////////////////////////////////////////////////////////
// int32_t IsInternalDTXReplacedWithWebRtc()
// Get status if the codec internal DTX is replaced with WebRtc DTX.
// This should always be true if codec does not have an internal DTX.
//
// Output:
// -uses_webrtc_dtx : is set to true if the codec internal DTX is
// replaced with WebRtc DTX/VAD, otherwise it is set
// to false.
//
// Return value:
// -1 if failed to determine if codec internal DTX is replaced with WebRtc,
// 0 if succeeded.
//
virtual int32_t IsInternalDTXReplacedWithWebRtc(
bool* uses_webrtc_dtx) = 0;
///////////////////////////////////////////////////////////////////////////
// int32_t RegisterVADCallback()
// Call this method to register a callback function which is called
@ -533,17 +461,6 @@ class AudioCodingModule {
//
virtual int32_t InitializeReceiver() = 0;
///////////////////////////////////////////////////////////////////////////
// int32_t ResetDecoder()
// This API resets the states of decoders. ACM will not lose any
// decoder-related settings, such as registered codecs.
//
// Return value:
// -1 if failed to initialize,
// 0 if succeeded.
//
virtual int32_t ResetDecoder() = 0;
///////////////////////////////////////////////////////////////////////////
// int32_t ReceiveFrequency()
// Get sampling frequency of the last received payload.
@ -738,19 +655,6 @@ class AudioCodingModule {
// TODO(tlegrand): Change function to return the timestamp.
virtual int32_t PlayoutTimestamp(uint32_t* timestamp) = 0;
///////////////////////////////////////////////////////////////////////////
// int32_t DecoderEstimatedBandwidth()
// Get the estimate of the Bandwidth, in bits/second, based on the incoming
// stream. This API is useful in one-way communication scenarios, where
// the bandwidth information is sent in an out-of-band fashion.
// Currently only supported if iSAC is registered as a receiver.
//
// Return value:
// >0 bandwidth in bits/second.
// -1 if failed to get a bandwidth estimate.
//
virtual int32_t DecoderEstimatedBandwidth() const = 0;
///////////////////////////////////////////////////////////////////////////
// int32_t SetPlayoutMode()
// Call this API to set the playout mode. Playout mode could be optimized
@ -849,35 +753,6 @@ class AudioCodingModule {
//
virtual int SetISACMaxPayloadSize(int max_payload_len_bytes) = 0;
///////////////////////////////////////////////////////////////////////////
// int32_t ConfigISACBandwidthEstimator()
// Call this function to configure the bandwidth estimator of ISAC.
// During the adaptation of bit-rate, iSAC automatically adjusts the
// frame-size (either 30 or 60 ms) to save on RTP header. The initial
// frame-size can be specified by the first argument. The configuration also
// regards the initial estimate of bandwidths. The estimator starts from
// this point and converges to the actual bottleneck. This is given by the
// second parameter. Furthermore, it is also possible to control the
// adaptation of frame-size. This is specified by the last parameter.
//
// Input:
// -init_frame_size_ms : initial frame-size in milliseconds. For iSAC-wb
// 30 ms and 60 ms (default) are acceptable values,
// and for iSAC-swb 30 ms is the only acceptable
// value. Zero indicates default value.
// -init_rate_bps : initial estimate of the bandwidth. Values
// between 10000 and 58000 are acceptable.
// -enforce_srame_size : if true, the frame-size will not be adapted.
//
// Return value:
// -1 if failed to configure the bandwidth estimator,
// 0 if the configuration was successfully applied.
//
virtual int32_t ConfigISACBandwidthEstimator(
int init_frame_size_ms,
int init_rate_bps,
bool enforce_frame_size = false) = 0;
///////////////////////////////////////////////////////////////////////////
// int SetOpusApplication()
// Sets the intended application if current send codec is Opus. Opus uses this