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Reformatting ACM. All changes are bit-exact in this CL.

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TEST=VoE auto-test, audio_coding_module_test; 

only 15 ms of teststereo_out_1.pcm is not bit-exact with output file of the head revision
Review URL: https://webrtc-codereview.appspot.com/937035

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3287 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
turaj@webrtc.org
2012-12-13 22:46:43 +00:00
parent ddebc17bee
commit c454fab03b
57 changed files with 5276 additions and 5241 deletions
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210
webrtc/modules/audio_coding/main/interface/audio_coding_module.h
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@@ -11,18 +11,20 @@
#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_INTERFACE_AUDIO_CODING_MODULE_H
#define WEBRTC_MODULES_AUDIO_CODING_MAIN_INTERFACE_AUDIO_CODING_MODULE_H
#include "audio_coding_module_typedefs.h"
#include "module.h"
// TODO(turajs): If possible, forward declare and remove the following include.
#include "module_common_types.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
#include "webrtc/modules/interface/module.h"
#include "webrtc/typedefs.h"
namespace webrtc {
// forward declarations
struct CodecInst;
struct WebRtcRTPHeader;
class AudioFrame;
class RTPFragmentationHeader;
#define WEBRTC_10MS_PCM_AUDIO 960 // 16 bits super wideband 48 Khz
#define WEBRTC_10MS_PCM_AUDIO 960 // 16 bits super wideband 48 kHz
// Callback class used for sending data ready to be packetized
class AudioPacketizationCallback {
@@ -30,8 +32,11 @@ class AudioPacketizationCallback {
virtual ~AudioPacketizationCallback() {}
virtual WebRtc_Word32 SendData(
FrameType frameType, WebRtc_UWord8 payloadType, WebRtc_UWord32 timeStamp,
const WebRtc_UWord8* payloadData, WebRtc_UWord16 payloadSize,
FrameType frame_type,
WebRtc_UWord8 payload_type,
WebRtc_UWord32 timestamp,
const WebRtc_UWord8* payload_data,
WebRtc_UWord16 payload_len_bytes,
const RTPFragmentationHeader* fragmentation) = 0;
};
@@ -40,7 +45,7 @@ class AudioCodingFeedback {
public:
virtual ~AudioCodingFeedback() {}
virtual WebRtc_Word32 IncomingDtmf(const WebRtc_UWord8 digitDtmf,
virtual WebRtc_Word32 IncomingDtmf(const WebRtc_UWord8 digit_dtmf,
const bool end) = 0;
};
@@ -96,17 +101,17 @@ class AudioCodingModule: public Module {
// Get supported codec with list number.
//
// Input:
// -listId : list number.
// -list_id : list number.
//
// Output:
// -codec : a structure where the parameters of the codec,
// given by list number is written to.
//
// Return value:
// -1 if the list number (listId) is invalid.
// -1 if the list number (list_id) is invalid.
// 0 if succeeded.
//
static WebRtc_Word32 Codec(const WebRtc_UWord8 listId, CodecInst& codec);
static WebRtc_Word32 Codec(const WebRtc_UWord8 list_id, CodecInst& codec);
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word32 Codec()
@@ -124,7 +129,7 @@ class AudioCodingModule: public Module {
// default parameters of the codec.
//
// Return value:
// -1 if the list number (listId) is invalid.
// -1 if no codec matches the given parameters.
// 0 if succeeded.
//
static WebRtc_Word32 Codec(const char* payload_name, CodecInst& codec,
@@ -154,10 +159,10 @@ class AudioCodingModule: public Module {
// Checks the validity of the parameters of the given codec.
//
// Input:
// -codec : the structur which keeps the parameters of the
// -codec : the structure which keeps the parameters of the
// codec.
//
// Reurn value:
// Return value:
// true if the parameters are valid,
// false if any parameter is not valid.
//
@@ -194,7 +199,7 @@ class AudioCodingModule: public Module {
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word32 RegisterSendCodec()
// Registers a codec, specified by "sendCodec," as sending codec.
// Registers a codec, specified by |send_codec|, as sending codec.
// This API can be called multiple of times to register Codec. The last codec
// registered overwrites the previous ones.
// The API can also be used to change payload type for CNG and RED, which are
@@ -211,7 +216,7 @@ class AudioCodingModule: public Module {
// secondary encoder will be unregistered.
//
// Input:
// -sendCodec : Parameters of the codec to be registered, c.f.
// -send_codec : Parameters of the codec to be registered, c.f.
// common_types.h for the definition of
// CodecInst.
//
@@ -219,7 +224,7 @@ class AudioCodingModule: public Module {
// -1 if failed to initialize,
// 0 if succeeded.
//
virtual WebRtc_Word32 RegisterSendCodec(const CodecInst& sendCodec) = 0;
virtual WebRtc_Word32 RegisterSendCodec(const CodecInst& send_codec) = 0;
///////////////////////////////////////////////////////////////////////////
// int RegisterSecondarySendCodec()
@@ -253,13 +258,13 @@ class AudioCodingModule: public Module {
// Get parameters for the codec currently registered as send codec.
//
// Output:
// -currentSendCodec : parameters of the send codec.
// -current_send_codec : parameters of the send codec.
//
// Return value:
// -1 if failed to get send codec,
// 0 if succeeded.
//
virtual WebRtc_Word32 SendCodec(CodecInst& currentSendCodec) const = 0;
virtual WebRtc_Word32 SendCodec(CodecInst& current_send_codec) const = 0;
///////////////////////////////////////////////////////////////////////////
// int SecondarySendCodec()
@@ -336,7 +341,7 @@ class AudioCodingModule: public Module {
// current encoder ACM will resample the audio.
//
// Input:
// -audioFrame : the input audio frame, containing raw audio
// -audio_frame : the input audio frame, containing raw audio
// sampling frequency etc.,
// c.f. module_common_types.h for definition of
// AudioFrame.
@@ -347,7 +352,7 @@ class AudioCodingModule: public Module {
// < -1 to add the frame to the buffer n samples had to be
// overwritten, -n is the return value in this case.
//
virtual WebRtc_Word32 Add10MsData(const AudioFrame& audioFrame) = 0;
virtual WebRtc_Word32 Add10MsData(const AudioFrame& audio_frame) = 0;
///////////////////////////////////////////////////////////////////////////
// (FEC) Forward Error Correction
@@ -368,14 +373,14 @@ class AudioCodingModule: public Module {
// This means that FEC is actually a RED scheme.
//
// Input:
// -enableFEC : if true FEC is enabled, otherwise FEC is
// -enable_fec : if true FEC is enabled, otherwise FEC is
// disabled.
//
// Return value:
// -1 if failed to set FEC status,
// 0 if succeeded.
//
virtual WebRtc_Word32 SetFECStatus(const bool enableFEC) = 0;
virtual WebRtc_Word32 SetFECStatus(const bool enable_fec) = 0;
///////////////////////////////////////////////////////////////////////////
// bool FECStatus()
@@ -394,7 +399,7 @@ class AudioCodingModule: public Module {
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word32 SetVAD()
// If DTX is enabled & the codec does not have internal DTX/VAD
// WebRtc VAD will be automatically enabled and 'enableVAD' is ignored.
// WebRtc VAD will be automatically enabled and |enable_vad| is ignored.
//
// If DTX is disabled but VAD is enabled no DTX packets are send,
// regardless of whether the codec has internal DTX/VAD or not. In this
@@ -403,11 +408,11 @@ class AudioCodingModule: public Module {
// NOTE! VAD/DTX is not supported when sending stereo.
//
// Inputs:
// -enableDTX : if true DTX is enabled,
// -enable_dtx : if true DTX is enabled,
// otherwise DTX is disabled.
// -enableVAD : if true VAD is enabled,
// -enable_vad : if true VAD is enabled,
// otherwise VAD is disabled.
// -vadMode : determines the aggressiveness of VAD. A more
// -vad_mode : determines the aggressiveness of VAD. A more
// aggressive mode results in more frames labeled
// as in-active, c.f. definition of
// ACMVADMode in audio_coding_module_typedefs.h
@@ -417,27 +422,27 @@ class AudioCodingModule: public Module {
// -1 if failed to set up VAD/DTX,
// 0 if succeeded.
//
virtual WebRtc_Word32 SetVAD(const bool enableDTX = true,
const bool enableVAD = false,
const ACMVADMode vadMode = VADNormal) = 0;
virtual WebRtc_Word32 SetVAD(const bool enable_dtx = true,
const bool enable_vad = false,
const ACMVADMode vad_mode = VADNormal) = 0;
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word32 VAD()
// Get VAD status.
//
// Outputs:
// -dtxEnabled : is set to true if DTX is enabled, otherwise
// -dtx_enabled : is set to true if DTX is enabled, otherwise
// is set to false.
// -vadEnabled : is set to true if VAD is enabled, otherwise
// -vad_enabled : is set to true if VAD is enabled, otherwise
// is set to false.
// -vadMode : is set to the current aggressiveness of VAD.
// -vad_mode : is set to the current aggressiveness of VAD.
//
// Return value:
// -1 if fails to retrieve the setting of DTX/VAD,
// 0 if succeeeded.
// 0 if succeeded.
//
virtual WebRtc_Word32 VAD(bool& dtxEnabled, bool& vadEnabled,
ACMVADMode& vadMode) const = 0;
virtual WebRtc_Word32 VAD(bool& dtx_enabled, bool& vad_enabled,
ACMVADMode& vad_mode) const = 0;
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word32 ReplaceInternalDTXWithWebRtc()
@@ -445,7 +450,7 @@ class AudioCodingModule: public Module {
// supported for G729, where this call replaces AnnexB with WebRtc DTX.
//
// Input:
// -useWebRtcDTX : if false (default) the codec built-in DTX/VAD
// -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.
//
@@ -454,7 +459,7 @@ class AudioCodingModule: public Module {
// 0 if succeeded.
//
virtual WebRtc_Word32 ReplaceInternalDTXWithWebRtc(
const bool useWebRtcDTX = false) = 0;
const bool use_webrtc_dtx = false) = 0;
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word32 IsInternalDTXReplacedWithWebRtc()
@@ -462,7 +467,7 @@ class AudioCodingModule: public Module {
// WebRtc DTX. This is only supported for G729.
//
// Output:
// -usesWebRtcDTX : is set to true if the codec internal DTX is
// -uses_webrtc_dtx : is set to true if the codec internal DTX is
// replaced with WebRtc DTX/VAD, otherwise it is set
// to false.
//
@@ -471,7 +476,7 @@ class AudioCodingModule: public Module {
// 0 if succeeded.
//
virtual WebRtc_Word32 IsInternalDTXReplacedWithWebRtc(
bool& usesWebRtcDTX) = 0;
bool& uses_webrtc_dtx) = 0;
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word32 RegisterVADCallback()
@@ -481,13 +486,13 @@ class AudioCodingModule: public Module {
// VAD is employed to identify a frame as active/inactive.
//
// Input:
// -vadCallback : pointer to a callback function.
// -vad_callback : pointer to a callback function.
//
// Return value:
// -1 if failed to register the callback function.
// 0 if the callback function is registered successfully.
//
virtual WebRtc_Word32 RegisterVADCallback(ACMVADCallback* vadCallback) = 0;
virtual WebRtc_Word32 RegisterVADCallback(ACMVADCallback* vad_callback) = 0;
///////////////////////////////////////////////////////////////////////////
// Receiver
@@ -544,7 +549,7 @@ class AudioCodingModule: public Module {
// codecs, CNG-NB, CNG-WB, CNG-SWB, AVT and RED.
//
// Input:
// -receiveCodec : parameters of the codec to be registered, c.f.
// -receive_codec : parameters of the codec to be registered, c.f.
// common_types.h for the definition of
// CodecInst.
//
@@ -552,7 +557,8 @@ class AudioCodingModule: public Module {
// -1 if failed to register the codec
// 0 if the codec registered successfully.
//
virtual WebRtc_Word32 RegisterReceiveCodec(const CodecInst& receiveCodec) = 0;
virtual WebRtc_Word32 RegisterReceiveCodec(
const CodecInst& receive_codec) = 0;
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word32 UnregisterReceiveCodec()
@@ -560,22 +566,22 @@ class AudioCodingModule: public Module {
// from the list of possible receive codecs.
//
// Input:
// -payloadType : The number representing the payload type to
// -payload_type : The number representing the payload type to
// unregister.
//
// Output:
// -1 if the unregistration fails.
// -1 if fails to unregister.
// 0 if the given codec is successfully unregistered.
//
virtual WebRtc_Word32 UnregisterReceiveCodec(
const WebRtc_Word16 receiveCodec) = 0;
const WebRtc_Word16 payload_type) = 0;
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word32 ReceiveCodec()
// Get the codec associated with last received payload.
//
// Output:
// -currRcvCodec : parameters of the codec associated with the last
// -curr_receive_codec : parameters of the codec associated with the last
// received payload, c.f. common_types.h for
// the definition of CodecInst.
//
@@ -583,25 +589,25 @@ class AudioCodingModule: public Module {
// -1 if failed to retrieve the codec,
// 0 if the codec is successfully retrieved.
//
virtual WebRtc_Word32 ReceiveCodec(CodecInst& currRcvCodec) const = 0;
virtual WebRtc_Word32 ReceiveCodec(CodecInst& curr_receive_codec) const = 0;
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word32 IncomingPacket()
// Call this function to insert a parsed RTP packet into ACM.
//
// Inputs:
// -incomingPayload : received payload.
// -payloadLengthByte : the length of payload in bytes.
// -rtpInfo : the relevant information retrieved from RTP
// -incoming_payload : received payload.
// -payload_len_bytes : the length of payload in bytes.
// -rtp_info : the relevant information retrieved from RTP
// header.
//
// Return value:
// -1 if failed to push in the payload
// 0 if payload is successfully pushed in.
//
virtual WebRtc_Word32 IncomingPacket(const WebRtc_UWord8* incomingPayload,
const WebRtc_Word32 payloadLengthByte,
const WebRtcRTPHeader& rtpInfo) = 0;
virtual WebRtc_Word32 IncomingPacket(const WebRtc_UWord8* incoming_payload,
const WebRtc_Word32 payload_len_bytes,
const WebRtcRTPHeader& rtp_info) = 0;
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word32 IncomingPayload()
@@ -610,9 +616,9 @@ class AudioCodingModule: public Module {
// pre-encoded files are pushed in ACM
//
// Inputs:
// -incomingPayload : received payload.
// -payloadLenghtByte : the length, in bytes, of the received payload.
// -payloadType : the payload-type. This specifies which codec has
// -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
@@ -627,9 +633,9 @@ class AudioCodingModule: public Module {
// -1 if failed to push in the payload
// 0 if payload is successfully pushed in.
//
virtual WebRtc_Word32 IncomingPayload(const WebRtc_UWord8* incomingPayload,
const WebRtc_Word32 payloadLengthByte,
const WebRtc_UWord8 payloadType,
virtual WebRtc_Word32 IncomingPayload(const WebRtc_UWord8* incoming_payload,
const WebRtc_Word32 payload_len_byte,
const WebRtc_UWord8 payload_type,
const WebRtc_UWord32 timestamp = 0) = 0;
///////////////////////////////////////////////////////////////////////////
@@ -637,22 +643,22 @@ class AudioCodingModule: public Module {
// Set Minimum playout delay, used for lip-sync.
//
// Input:
// -timeMs : minimum delay in milliseconds.
// -time_ms : minimum delay in milliseconds.
//
// Return value:
// -1 if failed to set the delay,
// 0 if the minimum delay is set.
//
virtual WebRtc_Word32 SetMinimumPlayoutDelay(const WebRtc_Word32 timeMs) = 0;
virtual WebRtc_Word32 SetMinimumPlayoutDelay(const WebRtc_Word32 time_ms) = 0;
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word32 RegisterIncomingMessagesCallback()
// Used by the module to deliver messages to the codec module/application
// when a Dtmf tone is detected, as well as when it stopped.
// when a DTMF tone is detected, as well as when it stopped.
//
// Inputs:
// -inMsgCallback : pointer to callback function which will be called
// if Dtmf is detected.
// -in_message_callback: pointer to callback function which will be called
// if DTMF is detected.
// -cpt : enables CPT (Call Progress Tone) detection for the
// specified country. c.f. definition of ACMCountries
// in audio_coding_module_typedefs.h for valid
@@ -665,20 +671,20 @@ class AudioCodingModule: public Module {
//
virtual WebRtc_Word32
RegisterIncomingMessagesCallback(
AudioCodingFeedback* inMsgCallback,
AudioCodingFeedback* in_message_callback,
const ACMCountries cpt = ACMDisableCountryDetection) = 0;
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word32 SetDtmfPlayoutStatus()
// Configure Dtmf playout, i.e. whether out-of-band
// Dtmf tones are played or not.
// Configure DTMF playout, i.e. whether out-of-band
// DTMF tones are played or not.
//
// Input:
// -enable : if true to enable playout out-of-band Dtmf tones,
// -enable : if true to enable playout out-of-band DTMF tones,
// false to disable.
//
// Return value:
// -1 if the method fails, e.g. Dtmf playout is not supported.
// -1 if the method fails, e.g. DTMF playout is not supported.
// 0 if the status is set successfully.
//
virtual WebRtc_Word32 SetDtmfPlayoutStatus(const bool enable) = 0;
@@ -696,7 +702,7 @@ class AudioCodingModule: public Module {
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word32 SetBackgroundNoiseMode()
// Sets the mode of the background noise playout in an event of long
// packetloss burst. For the valid modes see the declaration of
// packet loss burst. For the valid modes see the declaration of
// ACMBackgroundNoiseMode in audio_coding_module_typedefs.h.
//
// Input:
@@ -712,7 +718,7 @@ class AudioCodingModule: public Module {
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word32 BackgroundNoiseMode()
// Call this method to get the mode of the background noise playout.
// Playout of background noise is a result of a long packetloss burst.
// Playout of background noise is a result of a long packet loss burst.
// See ACMBackgroundNoiseMode in audio_coding_module_typedefs.h for
// possible modes.
//
@@ -746,7 +752,7 @@ class AudioCodingModule: public Module {
// 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 reciever.
// Currently only supported if iSAC is registered as a receiver.
//
// Return value:
// >0 bandwidth in bits/second.
@@ -762,7 +768,7 @@ class AudioCodingModule: public Module {
// delay. In FAX mode, NetEQ is optimized to have few delay changes as
// possible and maintain a constant delay, perhaps large relative to voice
// mode, to avoid PLC. In streaming mode, we tolerate a little more delay
// to acheive better jitter robustness.
// to achieve better jitter robustness.
//
// Input:
// -mode : playout mode. Possible inputs are:
@@ -786,7 +792,7 @@ class AudioCodingModule: public Module {
// fax: a mode that is optimized for receiving FAX signals.
// In this mode NetEq tries to maintain a constant high
// delay to avoid PLC if possible.
// streaming: a mode that is suitable for streaminq. In this mode we
// streaming: a mode that is suitable for streaming. In this mode we
// accept longer delay to improve jitter robustness.
//
virtual AudioPlayoutMode PlayoutMode() const = 0;
@@ -797,12 +803,12 @@ class AudioCodingModule: public Module {
// frequency. ACM will perform a resampling if required.
//
// Input:
// -desiredFreqHz : the desired sampling frequency, in Hertz, of the
// -desired_freq_hz : the desired sampling frequency, in Hertz, of the
// output audio. If set to -1, the function returns the
// audio at the current sampling frequency.
//
// Output:
// -audioFrame : output audio frame which contains raw audio data
// -audio_frame : output audio frame which contains raw audio data
// and other relevant parameters, c.f.
// module_common_types.h for the definition of
// AudioFrame.
@@ -812,8 +818,8 @@ class AudioCodingModule: public Module {
// 0 if the function succeeds.
//
virtual WebRtc_Word32
PlayoutData10Ms(const WebRtc_Word32 desiredFreqHz,
AudioFrame &audioFrame) = 0;
PlayoutData10Ms(const WebRtc_Word32 desired_freq_hz,
AudioFrame &audio_frame) = 0;
///////////////////////////////////////////////////////////////////////////
// (CNG) Comfort Noise Generation
@@ -855,43 +861,43 @@ class AudioCodingModule: public Module {
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word32 SetISACMaxRate()
// Set the maximum instantaneous rate of iSAC. For a payload of B bits
// with a frame-size of T sec the instantaneous rate is B/T bist per
// second. Therefore, (B/T < maxRateBitPerSec) and
// (B < maxPayloadLenBytes * 8) are always satisfied for iSAC payloads,
// with a frame-size of T sec the instantaneous rate is B/T bits per
// second. Therefore, (B/T < |max_rate_bps|) and
// (B < |max_payload_len_bytes| * 8) are always satisfied for iSAC payloads,
// c.f SetISACMaxPayloadSize().
//
// Input:
// -maxRateBitPerSec : maximum instantaneous bit-rate given in bits/sec.
// -max_rate_bps : maximum instantaneous bit-rate given in bits/sec.
//
// Return value:
// -1 if failed to set the maximum rate.
// 0 if the maximum rate is set successfully.
//
virtual WebRtc_Word32 SetISACMaxRate(
const WebRtc_UWord32 maxRateBitPerSec) = 0;
const WebRtc_UWord32 max_rate_bps) = 0;
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word32 SetISACMaxPayloadSize()
// Set the maximum payload size of iSAC packets. No iSAC payload,
// regardless of its frame-size, may exceed the given limit. For
// an iSAC payload of size B bits and frame-size T sec we have;
// (B < maxPayloadLenBytes * 8) and (B/T < maxRateBitPerSec), c.f.
// an iSAC payload of size B bits and frame-size T seconds we have;
// (B < |max_payload_len_bytes| * 8) and (B/T < |max_rate_bps|), c.f.
// SetISACMaxRate().
//
// Input:
// -maxPayloadLenBytes : maximum payload size in bytes.
// -max_payload_len_bytes : maximum payload size in bytes.
//
// Return value:
// -1 if failed to set the maximm payload-size.
// 0 if the given linit is seet successfully.
// -1 if failed to set the maximum payload-size.
// 0 if the given length is set successfully.
//
virtual WebRtc_Word32 SetISACMaxPayloadSize(
const WebRtc_UWord16 maxPayloadLenBytes) = 0;
const WebRtc_UWord16 max_payload_len_bytes) = 0;
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word32 ConfigISACBandwidthEstimator()
// Call this function to configure the bandwidth estimator of ISAC.
// During the adaptation of bit-rate, iSAC atomatically adjusts the
// 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
@@ -900,22 +906,22 @@ class AudioCodingModule: public Module {
// adaptation of frame-size. This is specified by the last parameter.
//
// Input:
// -initFrameSizeMsec : initial frame-size in milisecods. For iSAC-wb
// -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 indiates default value.
// -initRateBitPerSec : initial estimate of the bandwidth. Values
// value. Zero indicates default value.
// -init_rate_bps : initial estimate of the bandwidth. Values
// between 10000 and 58000 are acceptable.
// -enforceFrameSize : if true, the frame-size will not be adapted.
// -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 WebRtc_Word32 ConfigISACBandwidthEstimator(
const WebRtc_UWord8 initFrameSizeMsec,
const WebRtc_UWord16 initRateBitPerSec,
const bool enforceFrameSize = false) = 0;
const WebRtc_UWord8 init_frame_size_ms,
const WebRtc_UWord16 init_rate_bps,
const bool enforce_frame_size = false) = 0;
///////////////////////////////////////////////////////////////////////////
// statistics
@@ -926,14 +932,14 @@ class AudioCodingModule: public Module {
// Get network statistics.
//
// Input:
// -networkStatistics : a structure that contains network statistics.
// -network_statistics : a structure that contains network statistics.
//
// Return value:
// -1 if failed to set the network statistics,
// 0 if statistics are set successfully.
//
virtual WebRtc_Word32 NetworkStatistics(
ACMNetworkStatistics& networkStatistics) const = 0;
ACMNetworkStatistics& network_statistics) const = 0;
};
} // namespace webrtc