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Refactoring of audio_coding_module_impl

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First patch set: pure formatting.

Review URL: https://webrtc-codereview.appspot.com/522001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2125 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
tina.legrand@webrtc.org 2012-04-26 08:53:45 +00:00
parent a6ecd1ebb5
commit bc1b43b297
2 changed files with 2177 additions and 2654 deletions
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2247
src/modules/audio_coding/main/source/audio_coding_module_impl.cc
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276
src/modules/audio_coding/main/source/audio_coding_module_impl.h
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@ -24,113 +24,83 @@ class ACMGenericCodec;
class CriticalSectionWrapper;
class RWLockWrapper;
//#define TIMED_LOGGING
#ifdef TIMED_LOGGING
#include "../test/timedtrace.h"
#endif
#ifdef ACM_QA_TEST
# include <stdio.h>
#endif
class AudioCodingModuleImpl : public AudioCodingModule
{
class AudioCodingModuleImpl : public AudioCodingModule {
public:
// constructor
AudioCodingModuleImpl(
const WebRtc_Word32 id);
// Constructor
AudioCodingModuleImpl(const WebRtc_Word32 id);
// destructor
// Destructor
~AudioCodingModuleImpl();
// get version information for ACM and all components
WebRtc_Word32 Version(
char* version,
WebRtc_UWord32& remainingBufferInBytes,
WebRtc_UWord32& position) const;
// Change the unique identifier of this object.
virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
// change the unique identifier of this object
virtual WebRtc_Word32 ChangeUniqueId(
const WebRtc_Word32 id);
// returns the number of milliseconds until the module want
// a worker thread to call Process
// Returns the number of milliseconds until the module want a worker thread
// to call Process.
WebRtc_Word32 TimeUntilNextProcess();
// Process any pending tasks such as timeouts
// Process any pending tasks such as timeouts.
WebRtc_Word32 Process();
// used in conference to go to and from active encoding, hence
// in and out of mix
WebRtc_Word32 SetMode(
const bool passive);
/////////////////////////////////////////
// Sender
//
// initialize send codec
// Initialize send codec.
WebRtc_Word32 InitializeSender();
// reset send codec
// Reset send codec.
WebRtc_Word32 ResetEncoder();
// can be called multiple times for Codec, CNG, RED
WebRtc_Word32 RegisterSendCodec(
const CodecInst& sendCodec);
// Can be called multiple times for Codec, CNG, RED.
WebRtc_Word32 RegisterSendCodec(const CodecInst& send_codec);
// get current send codec
WebRtc_Word32 SendCodec(
CodecInst& currentSendCodec) const;
// Get current send codec.
WebRtc_Word32 SendCodec(CodecInst& current_codec) const;
// get current send freq
// Get current send frequency.
WebRtc_Word32 SendFrequency() const;
// Get encode bitrate
// Adaptive rate codecs return their current encode target rate, while other codecs
// return there longterm avarage or their fixed rate.
// Get encode bitrate.
// Adaptive rate codecs return their current encode target rate, while other
// codecs return there longterm avarage or their fixed rate.
WebRtc_Word32 SendBitrate() const;
// set available bandwidth, inform the encoder about the
// estimated bandwidth received from the remote party
virtual WebRtc_Word32 SetReceivedEstimatedBandwidth(
const WebRtc_Word32 bw);
// Set available bandwidth, inform the encoder about the
// estimated bandwidth received from the remote party.
virtual WebRtc_Word32 SetReceivedEstimatedBandwidth(const WebRtc_Word32 bw);
// register a transport callback which will be
// called to deliver the encoded buffers
// Register a transport callback which will be
// called to deliver the encoded buffers.
WebRtc_Word32 RegisterTransportCallback(
AudioPacketizationCallback* transport);
// Used by the module to deliver messages to the codec module/application
// AVT(DTMF)
// AVT(DTMF).
WebRtc_Word32 RegisterIncomingMessagesCallback(
AudioCodingFeedback* incomingMessagesCallback,
const ACMCountries cpt);
AudioCodingFeedback* incoming_message, const ACMCountries cpt);
// Add 10MS of raw (PCM) audio data to the encoder
WebRtc_Word32 Add10MsData(
const AudioFrame& audioFrame);
// Add 10MS of raw (PCM) audio data to the encoder.
WebRtc_Word32 Add10MsData(const AudioFrame& audio_frame);
// set background noise mode for NetEQ, on, off or fade
WebRtc_Word32 SetBackgroundNoiseMode(
const ACMBackgroundNoiseMode mode);
// Set background noise mode for NetEQ, on, off or fade.
WebRtc_Word32 SetBackgroundNoiseMode(const ACMBackgroundNoiseMode mode);
// get current background noise mode
WebRtc_Word32 BackgroundNoiseMode(
ACMBackgroundNoiseMode& mode);
// Get current background noise mode.
WebRtc_Word32 BackgroundNoiseMode(ACMBackgroundNoiseMode& mode);
/////////////////////////////////////////
// (FEC) Forward Error Correction
//
// configure FEC status i.e on/off
WebRtc_Word32 SetFECStatus(
const bool enable);
// Configure FEC status i.e on/off.
WebRtc_Word32 SetFECStatus(const bool enable_fec);
// Get FEC status
// Get FEC status.
bool FECStatus() const;
/////////////////////////////////////////
@ -139,171 +109,144 @@ public:
// (CNG) Comfort Noise Generation
//
WebRtc_Word32 SetVAD(
const bool enableDTX = true,
const bool enableVAD = false,
const ACMVADMode vadMode = VADNormal);
WebRtc_Word32 SetVAD(const bool enable_dtx = true,
const bool enable_vad = false,
const ACMVADMode mode = VADNormal);
WebRtc_Word32 VAD(
bool& dtxEnabled,
bool& vadEnabled,
ACMVADMode& vadMode) const;
WebRtc_Word32 VAD(bool& dtx_enabled, bool& vad_enabled,
ACMVADMode& mode) const;
WebRtc_Word32 RegisterVADCallback(
ACMVADCallback* vadCallback);
WebRtc_Word32 RegisterVADCallback(ACMVADCallback* vadCallback);
// Get VAD aggressiveness on the incoming stream
// Get VAD aggressiveness on the incoming stream.
ACMVADMode ReceiveVADMode() const;
// Configure VAD aggressiveness on the incoming stream
WebRtc_Word16 SetReceiveVADMode(
const ACMVADMode mode);
// Configure VAD aggressiveness on the incoming stream.
WebRtc_Word16 SetReceiveVADMode(const ACMVADMode mode);
/////////////////////////////////////////
// Receiver
//
// initialize receiver, resets codec database etc
// Initialize receiver, resets codec database etc.
WebRtc_Word32 InitializeReceiver();
// reset the decoder state
// Reset the decoder state.
WebRtc_Word32 ResetDecoder();
// get current receive freq
// Get current receive frequency.
WebRtc_Word32 ReceiveFrequency() const;
// get current playout freq
// Get current playout frequency.
WebRtc_Word32 PlayoutFrequency() const;
// register possible reveive codecs, can be called multiple times,
// for codecs, CNG, DTMF, RED
WebRtc_Word32 RegisterReceiveCodec(
const CodecInst& receiveCodec);
// Register possible reveive codecs, can be called multiple times,
// for codecs, CNG, DTMF, RED.
WebRtc_Word32 RegisterReceiveCodec(const CodecInst& receive_codec);
// get current received codec
WebRtc_Word32 ReceiveCodec(
CodecInst& currentReceiveCodec) const;
// Get current received codec.
WebRtc_Word32 ReceiveCodec(CodecInst& current_codec) const;
// incoming packet from network parsed and ready for decode
WebRtc_Word32 IncomingPacket(
const WebRtc_UWord8* incomingPayload,
const WebRtc_Word32 payloadLength,
const WebRtcRTPHeader& rtpInfo);
// Incoming packet from network parsed and ready for decode.
WebRtc_Word32 IncomingPacket(const WebRtc_UWord8* incoming_payload,
const WebRtc_Word32 payload_length,
const WebRtcRTPHeader& rtp_info);
// Incoming payloads, without rtp-info, the rtp-info will be created in ACM.
// One usage for this API is when pre-encoded files are pushed in ACM.
WebRtc_Word32 IncomingPayload(
const WebRtc_UWord8* incomingPayload,
const WebRtc_Word32 payloadLength,
const WebRtc_UWord8 payloadType,
WebRtc_Word32 IncomingPayload(const WebRtc_UWord8* incoming_payload,
const WebRtc_Word32 payload_length,
const WebRtc_UWord8 payload_type,
const WebRtc_UWord32 timestamp = 0);
// Minimum playout dealy (Used for lip-sync)
WebRtc_Word32 SetMinimumPlayoutDelay(
const WebRtc_Word32 timeMs);
// Minimum playout dealy (used for lip-sync).
WebRtc_Word32 SetMinimumPlayoutDelay(const WebRtc_Word32 time_ms);
// configure Dtmf playout status i.e on/off playout the incoming outband Dtmf tone
WebRtc_Word32 SetDtmfPlayoutStatus(
const bool enable);
// Configure Dtmf playout status i.e on/off playout the incoming outband Dtmf
// tone.
WebRtc_Word32 SetDtmfPlayoutStatus(const bool enable);
// Get Dtmf playout status
// Get Dtmf playout status.
bool DtmfPlayoutStatus() const;
// Estimate the Bandwidth based on the incoming stream
// This is also done in the RTP module
// need this for one way audio where the RTCP send the BW estimate
// Estimate the Bandwidth based on the incoming stream, needed
// for one way audio where the RTCP send the BW estimate.
// This is also done in the RTP module .
WebRtc_Word32 DecoderEstimatedBandwidth() const;
// Set playout mode voice, fax
WebRtc_Word32 SetPlayoutMode(
const AudioPlayoutMode mode);
// Set playout mode voice, fax.
WebRtc_Word32 SetPlayoutMode(const AudioPlayoutMode mode);
// Get playout mode voice, fax
// Get playout mode voice, fax.
AudioPlayoutMode PlayoutMode() const;
// Get playout timestamp
WebRtc_Word32 PlayoutTimestamp(
WebRtc_UWord32& timestamp);
// Get 10 milliseconds of raw audio data to play out
// automatic resample to the requested frequency if > 0
WebRtc_Word32 PlayoutData10Ms(
const WebRtc_Word32 desiredFreqHz,
AudioFrame &audioFrame);
// Get playout timestamp.
WebRtc_Word32 PlayoutTimestamp(WebRtc_UWord32& timestamp);
// Get 10 milliseconds of raw audio data to play out, and
// automatic resample to the requested frequency if > 0.
WebRtc_Word32 PlayoutData10Ms(const WebRtc_Word32 desired_freq_hz,
AudioFrame &audio_frame);
/////////////////////////////////////////
// Statistics
//
WebRtc_Word32 NetworkStatistics(
ACMNetworkStatistics& statistics) const;
WebRtc_Word32 NetworkStatistics(ACMNetworkStatistics& statistics) const;
void DestructEncoderInst(void* ptrInst);
void DestructEncoderInst(void* inst);
WebRtc_Word16 AudioBuffer(WebRtcACMAudioBuff& audioBuff);
WebRtc_Word16 AudioBuffer(WebRtcACMAudioBuff& buffer);
// GET RED payload for iSAC. The method id called
// when 'this' ACM is default ACM.
WebRtc_Word32 REDPayloadISAC(
const WebRtc_Word32 isacRate,
const WebRtc_Word16 isacBwEstimate,
// GET RED payload for iSAC. The method id called when 'this' ACM is
// the default ACM.
WebRtc_Word32 REDPayloadISAC(const WebRtc_Word32 isac_rate,
const WebRtc_Word16 isac_bw_estimate,
WebRtc_UWord8* payload,
WebRtc_Word16* payloadLenByte);
WebRtc_Word16* length_bytes);
WebRtc_Word16 SetAudioBuffer(WebRtcACMAudioBuff& audioBuff);
WebRtc_Word16 SetAudioBuffer(WebRtcACMAudioBuff& buffer);
WebRtc_UWord32 EarliestTimestamp() const;
WebRtc_Word32 LastEncodedTimestamp(WebRtc_UWord32& timestamp) const;
WebRtc_Word32 ReplaceInternalDTXWithWebRtc(
const bool useWebRtcDTX);
WebRtc_Word32 ReplaceInternalDTXWithWebRtc(const bool use_webrtc_dtx);
WebRtc_Word32 IsInternalDTXReplacedWithWebRtc(
bool& usesWebRtcDTX);
WebRtc_Word32 IsInternalDTXReplacedWithWebRtc(bool& uses_webrtc_dtx);
WebRtc_Word32 SetISACMaxRate(
const WebRtc_UWord32 rateBitPerSec);
WebRtc_Word32 SetISACMaxRate(const WebRtc_UWord32 max_bit_per_sec);
WebRtc_Word32 SetISACMaxPayloadSize(
const WebRtc_UWord16 payloadLenBytes);
WebRtc_Word32 SetISACMaxPayloadSize(const WebRtc_UWord16 max_size_bytes);
WebRtc_Word32 ConfigISACBandwidthEstimator(
const WebRtc_UWord8 initFrameSizeMsec,
const WebRtc_UWord16 initRateBitPerSec,
const bool enforceFrameSize = false);
const WebRtc_UWord8 frame_size_ms,
const WebRtc_UWord16 rate_bit_per_sec,
const bool enforce_frame_size = false);
WebRtc_Word32 UnregisterReceiveCodec(
const WebRtc_Word16 payloadType);
WebRtc_Word32 UnregisterReceiveCodec(const WebRtc_Word16 payload_type);
protected:
void UnregisterSendCodec();
WebRtc_Word32 UnregisterReceiveCodecSafe(
const WebRtc_Word16 codecID);
WebRtc_Word32 UnregisterReceiveCodecSafe(const WebRtc_Word16 id);
ACMGenericCodec* CreateCodec(
const CodecInst& codec);
ACMGenericCodec* CreateCodec(const CodecInst& codec);
WebRtc_Word16 DecoderParamByPlType(
const WebRtc_UWord8 payloadType,
WebRtcACMCodecParams& codecParams) const;
WebRtc_Word16 DecoderParamByPlType(const WebRtc_UWord8 payload_type,
WebRtcACMCodecParams& codec_params) const;
WebRtc_Word16 DecoderListIDByPlName(
const char* payloadName,
const WebRtc_UWord16 sampFreqHz = 0) const;
const char* name, const WebRtc_UWord16 frequency = 0) const;
WebRtc_Word32 InitializeReceiverSafe();
bool HaveValidEncoder(const char* callerName) const;
bool HaveValidEncoder(const char* caller_name) const;
WebRtc_Word32 RegisterRecCodecMSSafe(
const CodecInst& receiveCodec,
WebRtc_Word16 codecId,
WebRtc_Word16 mirrorId,
ACMNetEQ::JB jitterBuffer);
WebRtc_Word32 RegisterRecCodecMSSafe(const CodecInst& receive_codec,
WebRtc_Word16 codec_id,
WebRtc_Word16 mirror_id,
ACMNetEQ::JB jitter_buffer);
private:
// Change required states after starting to receive the codec corresponding
@ -358,17 +301,17 @@ private:
ACMVADCallback* _vadCallback;
WebRtc_UWord8 _lastRecvAudioCodecPlType;
// RED/FEC
// RED/FEC.
bool _isFirstRED;
bool _fecEnabled;
WebRtc_UWord8* _redBuffer;
RTPFragmentationHeader* _fragmentation;
WebRtc_UWord32 _lastFECTimestamp;
// if no RED is registered as receive codec this
// If no RED is registered as receive codec this
// will have an invalid value.
WebRtc_UWord8 _receiveREDPayloadType;
// This is to keep track of CN instances where we can send DTMFs
// This is to keep track of CN instances where we can send DTMFs.
WebRtc_UWord8 _previousPayloadType;
// This keeps track of payload types associated with _codecs[].
@ -388,9 +331,6 @@ private:
AudioCodingFeedback* _dtmfCallback;
WebRtc_Word16 _lastDetectedTone;
CriticalSectionWrapper* _callbackCritSect;
#ifdef TIMED_LOGGING
TimedTrace _trace;
#endif
AudioFrame _audioFrame;