Refactoring of audio_coding_module_impl
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
parent
a6ecd1ebb5
commit
bc1b43b297
File diff suppressed because it is too large
Load Diff
@ -24,383 +24,323 @@ class ACMGenericCodec;
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class CriticalSectionWrapper;
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class RWLockWrapper;
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//#define TIMED_LOGGING
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#ifdef TIMED_LOGGING
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#include "../test/timedtrace.h"
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#endif
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#ifdef ACM_QA_TEST
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# include <stdio.h>
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#endif
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class AudioCodingModuleImpl : public AudioCodingModule
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{
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public:
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// constructor
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AudioCodingModuleImpl(
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const WebRtc_Word32 id);
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class AudioCodingModuleImpl : public AudioCodingModule {
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public:
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// Constructor
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AudioCodingModuleImpl(const WebRtc_Word32 id);
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// destructor
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~AudioCodingModuleImpl();
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// Destructor
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~AudioCodingModuleImpl();
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// get version information for ACM and all components
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WebRtc_Word32 Version(
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char* version,
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WebRtc_UWord32& remainingBufferInBytes,
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WebRtc_UWord32& position) const;
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// change the unique identifier of this object
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virtual WebRtc_Word32 ChangeUniqueId(
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const WebRtc_Word32 id);
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// returns the number of milliseconds until the module want
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// a worker thread to call Process
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WebRtc_Word32 TimeUntilNextProcess();
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// Process any pending tasks such as timeouts
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WebRtc_Word32 Process();
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// Change the unique identifier of this object.
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virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
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// used in conference to go to and from active encoding, hence
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// in and out of mix
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WebRtc_Word32 SetMode(
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const bool passive);
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// Returns the number of milliseconds until the module want a worker thread
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// to call Process.
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WebRtc_Word32 TimeUntilNextProcess();
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// Process any pending tasks such as timeouts.
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WebRtc_Word32 Process();
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/////////////////////////////////////////
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// Sender
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//
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/////////////////////////////////////////
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// Sender
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//
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// Initialize send codec.
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WebRtc_Word32 InitializeSender();
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// initialize send codec
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WebRtc_Word32 InitializeSender();
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// Reset send codec.
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WebRtc_Word32 ResetEncoder();
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// reset send codec
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WebRtc_Word32 ResetEncoder();
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// Can be called multiple times for Codec, CNG, RED.
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WebRtc_Word32 RegisterSendCodec(const CodecInst& send_codec);
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// can be called multiple times for Codec, CNG, RED
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WebRtc_Word32 RegisterSendCodec(
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const CodecInst& sendCodec);
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// Get current send codec.
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WebRtc_Word32 SendCodec(CodecInst& current_codec) const;
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// get current send codec
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WebRtc_Word32 SendCodec(
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CodecInst& currentSendCodec) const;
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// Get current send frequency.
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WebRtc_Word32 SendFrequency() const;
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// get current send freq
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WebRtc_Word32 SendFrequency() const;
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// Get encode bitrate.
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// Adaptive rate codecs return their current encode target rate, while other
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// codecs return there longterm avarage or their fixed rate.
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WebRtc_Word32 SendBitrate() const;
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// Get encode bitrate
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// Adaptive rate codecs return their current encode target rate, while other codecs
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// return there longterm avarage or their fixed rate.
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WebRtc_Word32 SendBitrate() const;
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// Set available bandwidth, inform the encoder about the
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// estimated bandwidth received from the remote party.
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virtual WebRtc_Word32 SetReceivedEstimatedBandwidth(const WebRtc_Word32 bw);
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// set available bandwidth, inform the encoder about the
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// estimated bandwidth received from the remote party
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virtual WebRtc_Word32 SetReceivedEstimatedBandwidth(
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const WebRtc_Word32 bw);
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// Register a transport callback which will be
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// called to deliver the encoded buffers.
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WebRtc_Word32 RegisterTransportCallback(
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AudioPacketizationCallback* transport);
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// register a transport callback which will be
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// called to deliver the encoded buffers
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WebRtc_Word32 RegisterTransportCallback(
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AudioPacketizationCallback* transport);
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// Used by the module to deliver messages to the codec module/application
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// AVT(DTMF).
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WebRtc_Word32 RegisterIncomingMessagesCallback(
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AudioCodingFeedback* incoming_message, const ACMCountries cpt);
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// Used by the module to deliver messages to the codec module/application
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// AVT(DTMF)
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WebRtc_Word32 RegisterIncomingMessagesCallback(
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AudioCodingFeedback* incomingMessagesCallback,
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const ACMCountries cpt);
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// Add 10MS of raw (PCM) audio data to the encoder.
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WebRtc_Word32 Add10MsData(const AudioFrame& audio_frame);
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// Add 10MS of raw (PCM) audio data to the encoder
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WebRtc_Word32 Add10MsData(
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const AudioFrame& audioFrame);
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// set background noise mode for NetEQ, on, off or fade
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WebRtc_Word32 SetBackgroundNoiseMode(
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const ACMBackgroundNoiseMode mode);
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// get current background noise mode
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WebRtc_Word32 BackgroundNoiseMode(
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ACMBackgroundNoiseMode& mode);
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// Set background noise mode for NetEQ, on, off or fade.
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WebRtc_Word32 SetBackgroundNoiseMode(const ACMBackgroundNoiseMode mode);
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/////////////////////////////////////////
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// (FEC) Forward Error Correction
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//
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// Get current background noise mode.
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WebRtc_Word32 BackgroundNoiseMode(ACMBackgroundNoiseMode& mode);
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// configure FEC status i.e on/off
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WebRtc_Word32 SetFECStatus(
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const bool enable);
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// Get FEC status
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bool FECStatus() const;
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/////////////////////////////////////////
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// (FEC) Forward Error Correction
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//
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/////////////////////////////////////////
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// (VAD) Voice Activity Detection
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// and
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// (CNG) Comfort Noise Generation
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//
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// Configure FEC status i.e on/off.
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WebRtc_Word32 SetFECStatus(const bool enable_fec);
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WebRtc_Word32 SetVAD(
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const bool enableDTX = true,
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const bool enableVAD = false,
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const ACMVADMode vadMode = VADNormal);
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// Get FEC status.
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bool FECStatus() const;
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WebRtc_Word32 VAD(
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bool& dtxEnabled,
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bool& vadEnabled,
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ACMVADMode& vadMode) const;
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/////////////////////////////////////////
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// (VAD) Voice Activity Detection
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// and
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// (CNG) Comfort Noise Generation
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//
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WebRtc_Word32 RegisterVADCallback(
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ACMVADCallback* vadCallback);
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WebRtc_Word32 SetVAD(const bool enable_dtx = true,
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const bool enable_vad = false,
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const ACMVADMode mode = VADNormal);
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// Get VAD aggressiveness on the incoming stream
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ACMVADMode ReceiveVADMode() const;
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WebRtc_Word32 VAD(bool& dtx_enabled, bool& vad_enabled,
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ACMVADMode& mode) const;
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// Configure VAD aggressiveness on the incoming stream
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WebRtc_Word16 SetReceiveVADMode(
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const ACMVADMode mode);
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WebRtc_Word32 RegisterVADCallback(ACMVADCallback* vadCallback);
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// Get VAD aggressiveness on the incoming stream.
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ACMVADMode ReceiveVADMode() const;
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/////////////////////////////////////////
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// Receiver
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//
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// Configure VAD aggressiveness on the incoming stream.
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WebRtc_Word16 SetReceiveVADMode(const ACMVADMode mode);
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// initialize receiver, resets codec database etc
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WebRtc_Word32 InitializeReceiver();
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/////////////////////////////////////////
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// Receiver
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//
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// reset the decoder state
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WebRtc_Word32 ResetDecoder();
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// Initialize receiver, resets codec database etc.
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WebRtc_Word32 InitializeReceiver();
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// get current receive freq
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WebRtc_Word32 ReceiveFrequency() const;
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// Reset the decoder state.
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WebRtc_Word32 ResetDecoder();
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// get current playout freq
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WebRtc_Word32 PlayoutFrequency() const;
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// Get current receive frequency.
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WebRtc_Word32 ReceiveFrequency() const;
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// register possible reveive codecs, can be called multiple times,
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// for codecs, CNG, DTMF, RED
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WebRtc_Word32 RegisterReceiveCodec(
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const CodecInst& receiveCodec);
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// Get current playout frequency.
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WebRtc_Word32 PlayoutFrequency() const;
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// get current received codec
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WebRtc_Word32 ReceiveCodec(
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CodecInst& currentReceiveCodec) const;
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// Register possible reveive codecs, can be called multiple times,
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// for codecs, CNG, DTMF, RED.
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WebRtc_Word32 RegisterReceiveCodec(const CodecInst& receive_codec);
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// incoming packet from network parsed and ready for decode
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WebRtc_Word32 IncomingPacket(
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const WebRtc_UWord8* incomingPayload,
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const WebRtc_Word32 payloadLength,
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const WebRtcRTPHeader& rtpInfo);
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// Get current received codec.
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WebRtc_Word32 ReceiveCodec(CodecInst& current_codec) const;
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// Incoming payloads, without rtp-info, the rtp-info will be created in ACM.
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// One usage for this API is when pre-encoded files are pushed in ACM.
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WebRtc_Word32 IncomingPayload(
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const WebRtc_UWord8* incomingPayload,
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const WebRtc_Word32 payloadLength,
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const WebRtc_UWord8 payloadType,
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const WebRtc_UWord32 timestamp = 0);
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// Incoming packet from network parsed and ready for decode.
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WebRtc_Word32 IncomingPacket(const WebRtc_UWord8* incoming_payload,
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const WebRtc_Word32 payload_length,
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const WebRtcRTPHeader& rtp_info);
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// Minimum playout dealy (Used for lip-sync)
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WebRtc_Word32 SetMinimumPlayoutDelay(
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const WebRtc_Word32 timeMs);
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// Incoming payloads, without rtp-info, the rtp-info will be created in ACM.
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// One usage for this API is when pre-encoded files are pushed in ACM.
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WebRtc_Word32 IncomingPayload(const WebRtc_UWord8* incoming_payload,
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const WebRtc_Word32 payload_length,
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const WebRtc_UWord8 payload_type,
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const WebRtc_UWord32 timestamp = 0);
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// configure Dtmf playout status i.e on/off playout the incoming outband Dtmf tone
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WebRtc_Word32 SetDtmfPlayoutStatus(
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const bool enable);
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// Minimum playout dealy (used for lip-sync).
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WebRtc_Word32 SetMinimumPlayoutDelay(const WebRtc_Word32 time_ms);
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// Get Dtmf playout status
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bool DtmfPlayoutStatus() const;
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// Configure Dtmf playout status i.e on/off playout the incoming outband Dtmf
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// tone.
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WebRtc_Word32 SetDtmfPlayoutStatus(const bool enable);
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// Estimate the Bandwidth based on the incoming stream
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// This is also done in the RTP module
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// need this for one way audio where the RTCP send the BW estimate
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WebRtc_Word32 DecoderEstimatedBandwidth() const;
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// Get Dtmf playout status.
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bool DtmfPlayoutStatus() const;
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// Set playout mode voice, fax
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WebRtc_Word32 SetPlayoutMode(
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const AudioPlayoutMode mode);
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// Estimate the Bandwidth based on the incoming stream, needed
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// for one way audio where the RTCP send the BW estimate.
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// This is also done in the RTP module .
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WebRtc_Word32 DecoderEstimatedBandwidth() const;
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// Get playout mode voice, fax
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AudioPlayoutMode PlayoutMode() const;
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// Set playout mode voice, fax.
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WebRtc_Word32 SetPlayoutMode(const AudioPlayoutMode mode);
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// Get playout timestamp
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WebRtc_Word32 PlayoutTimestamp(
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WebRtc_UWord32& timestamp);
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// Get playout mode voice, fax.
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AudioPlayoutMode PlayoutMode() const;
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// Get 10 milliseconds of raw audio data to play out
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// automatic resample to the requested frequency if > 0
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WebRtc_Word32 PlayoutData10Ms(
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const WebRtc_Word32 desiredFreqHz,
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AudioFrame &audioFrame);
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// Get playout timestamp.
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WebRtc_Word32 PlayoutTimestamp(WebRtc_UWord32& timestamp);
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// Get 10 milliseconds of raw audio data to play out, and
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// automatic resample to the requested frequency if > 0.
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WebRtc_Word32 PlayoutData10Ms(const WebRtc_Word32 desired_freq_hz,
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AudioFrame &audio_frame);
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/////////////////////////////////////////
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// Statistics
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//
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/////////////////////////////////////////
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// Statistics
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//
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WebRtc_Word32 NetworkStatistics(
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ACMNetworkStatistics& statistics) const;
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WebRtc_Word32 NetworkStatistics(ACMNetworkStatistics& statistics) const;
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void DestructEncoderInst(void* ptrInst);
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void DestructEncoderInst(void* inst);
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WebRtc_Word16 AudioBuffer(WebRtcACMAudioBuff& audioBuff);
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WebRtc_Word16 AudioBuffer(WebRtcACMAudioBuff& buffer);
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// GET RED payload for iSAC. The method id called
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// when 'this' ACM is default ACM.
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WebRtc_Word32 REDPayloadISAC(
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const WebRtc_Word32 isacRate,
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const WebRtc_Word16 isacBwEstimate,
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WebRtc_UWord8* payload,
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WebRtc_Word16* payloadLenByte);
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WebRtc_Word16 SetAudioBuffer(WebRtcACMAudioBuff& audioBuff);
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WebRtc_UWord32 EarliestTimestamp() const;
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WebRtc_Word32 LastEncodedTimestamp(WebRtc_UWord32& timestamp) const;
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WebRtc_Word32 ReplaceInternalDTXWithWebRtc(
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const bool useWebRtcDTX);
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WebRtc_Word32 IsInternalDTXReplacedWithWebRtc(
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bool& usesWebRtcDTX);
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WebRtc_Word32 SetISACMaxRate(
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const WebRtc_UWord32 rateBitPerSec);
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WebRtc_Word32 SetISACMaxPayloadSize(
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const WebRtc_UWord16 payloadLenBytes);
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WebRtc_Word32 ConfigISACBandwidthEstimator(
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const WebRtc_UWord8 initFrameSizeMsec,
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const WebRtc_UWord16 initRateBitPerSec,
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const bool enforceFrameSize = false);
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WebRtc_Word32 UnregisterReceiveCodec(
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const WebRtc_Word16 payloadType);
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protected:
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void UnregisterSendCodec();
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WebRtc_Word32 UnregisterReceiveCodecSafe(
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const WebRtc_Word16 codecID);
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ACMGenericCodec* CreateCodec(
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const CodecInst& codec);
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WebRtc_Word16 DecoderParamByPlType(
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const WebRtc_UWord8 payloadType,
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WebRtcACMCodecParams& codecParams) const;
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WebRtc_Word16 DecoderListIDByPlName(
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const char* payloadName,
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const WebRtc_UWord16 sampFreqHz = 0) const;
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WebRtc_Word32 InitializeReceiverSafe();
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bool HaveValidEncoder(const char* callerName) const;
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WebRtc_Word32 RegisterRecCodecMSSafe(
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const CodecInst& receiveCodec,
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WebRtc_Word16 codecId,
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WebRtc_Word16 mirrorId,
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ACMNetEQ::JB jitterBuffer);
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private:
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// Change required states after starting to receive the codec corresponding
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// to |index|.
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int UpdateUponReceivingCodec(int index);
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// Remove all slaves and initialize a stereo slave with required codecs
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// from the master.
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int InitStereoSlave();
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// Returns true if the codec's |index| is registered with the master and
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// is a stereo codec, RED or CN.
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bool IsCodecForSlave(int index);
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// Returns true if the |codec| is RED.
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bool IsCodecRED(CodecInst codec);
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// ...or if its |index| is RED.
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bool IsCodecRED(int index);
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// Returns true if the |codec| is CN.
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bool IsCodecCN(int index);
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// ...or if its |index| is CN.
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bool IsCodecCN(CodecInst codec);
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AudioPacketizationCallback* _packetizationCallback;
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WebRtc_Word32 _id;
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WebRtc_UWord32 _lastTimestamp;
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WebRtc_UWord32 _lastInTimestamp;
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CodecInst _sendCodecInst;
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uint8_t _cng_nb_pltype;
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uint8_t _cng_wb_pltype;
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uint8_t _cng_swb_pltype;
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uint8_t _red_pltype;
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bool _vadEnabled;
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bool _dtxEnabled;
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ACMVADMode _vadMode;
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ACMGenericCodec* _codecs[ACMCodecDB::kMaxNumCodecs];
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ACMGenericCodec* _slaveCodecs[ACMCodecDB::kMaxNumCodecs];
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WebRtc_Word16 _mirrorCodecIdx[ACMCodecDB::kMaxNumCodecs];
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bool _stereoReceive[ACMCodecDB::kMaxNumCodecs];
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bool _stereoReceiveRegistered;
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bool _stereoSend;
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int _prev_received_channel;
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int _expected_channels;
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WebRtc_Word32 _currentSendCodecIdx;
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int _current_receive_codec_idx;
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bool _sendCodecRegistered;
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ACMResampler _inputResampler;
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ACMResampler _outputResampler;
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ACMNetEQ _netEq;
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CriticalSectionWrapper* _acmCritSect;
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ACMVADCallback* _vadCallback;
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WebRtc_UWord8 _lastRecvAudioCodecPlType;
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// RED/FEC
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bool _isFirstRED;
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bool _fecEnabled;
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WebRtc_UWord8* _redBuffer;
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RTPFragmentationHeader* _fragmentation;
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WebRtc_UWord32 _lastFECTimestamp;
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// if no RED is registered as receive codec this
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// will have an invalid value.
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WebRtc_UWord8 _receiveREDPayloadType;
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// 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[].
|
||||
// We define it as signed variable and initialize with -1 to indicate
|
||||
// unused elements.
|
||||
WebRtc_Word16 _registeredPlTypes[ACMCodecDB::kMaxNumCodecs];
|
||||
|
||||
// Used when payloads are pushed into ACM without any RTP info
|
||||
// One example is when pre-encoded bit-stream is pushed from
|
||||
// a file.
|
||||
WebRtcRTPHeader* _dummyRTPHeader;
|
||||
WebRtc_UWord16 _recvPlFrameSizeSmpls;
|
||||
|
||||
bool _receiverInitialized;
|
||||
ACMDTMFDetection* _dtmfDetector;
|
||||
|
||||
AudioCodingFeedback* _dtmfCallback;
|
||||
WebRtc_Word16 _lastDetectedTone;
|
||||
CriticalSectionWrapper* _callbackCritSect;
|
||||
#ifdef TIMED_LOGGING
|
||||
TimedTrace _trace;
|
||||
#endif
|
||||
|
||||
AudioFrame _audioFrame;
|
||||
// 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* length_bytes);
|
||||
|
||||
WebRtc_Word16 SetAudioBuffer(WebRtcACMAudioBuff& buffer);
|
||||
|
||||
WebRtc_UWord32 EarliestTimestamp() const;
|
||||
|
||||
WebRtc_Word32 LastEncodedTimestamp(WebRtc_UWord32& timestamp) const;
|
||||
|
||||
WebRtc_Word32 ReplaceInternalDTXWithWebRtc(const bool use_webrtc_dtx);
|
||||
|
||||
WebRtc_Word32 IsInternalDTXReplacedWithWebRtc(bool& uses_webrtc_dtx);
|
||||
|
||||
WebRtc_Word32 SetISACMaxRate(const WebRtc_UWord32 max_bit_per_sec);
|
||||
|
||||
WebRtc_Word32 SetISACMaxPayloadSize(const WebRtc_UWord16 max_size_bytes);
|
||||
|
||||
WebRtc_Word32 ConfigISACBandwidthEstimator(
|
||||
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 payload_type);
|
||||
|
||||
protected:
|
||||
void UnregisterSendCodec();
|
||||
|
||||
WebRtc_Word32 UnregisterReceiveCodecSafe(const WebRtc_Word16 id);
|
||||
|
||||
ACMGenericCodec* CreateCodec(const CodecInst& codec);
|
||||
|
||||
WebRtc_Word16 DecoderParamByPlType(const WebRtc_UWord8 payload_type,
|
||||
WebRtcACMCodecParams& codec_params) const;
|
||||
|
||||
WebRtc_Word16 DecoderListIDByPlName(
|
||||
const char* name, const WebRtc_UWord16 frequency = 0) const;
|
||||
|
||||
WebRtc_Word32 InitializeReceiverSafe();
|
||||
|
||||
bool HaveValidEncoder(const char* caller_name) const;
|
||||
|
||||
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
|
||||
// to |index|.
|
||||
int UpdateUponReceivingCodec(int index);
|
||||
|
||||
// Remove all slaves and initialize a stereo slave with required codecs
|
||||
// from the master.
|
||||
int InitStereoSlave();
|
||||
|
||||
// Returns true if the codec's |index| is registered with the master and
|
||||
// is a stereo codec, RED or CN.
|
||||
bool IsCodecForSlave(int index);
|
||||
|
||||
// Returns true if the |codec| is RED.
|
||||
bool IsCodecRED(CodecInst codec);
|
||||
// ...or if its |index| is RED.
|
||||
bool IsCodecRED(int index);
|
||||
|
||||
// Returns true if the |codec| is CN.
|
||||
bool IsCodecCN(int index);
|
||||
// ...or if its |index| is CN.
|
||||
bool IsCodecCN(CodecInst codec);
|
||||
|
||||
AudioPacketizationCallback* _packetizationCallback;
|
||||
WebRtc_Word32 _id;
|
||||
WebRtc_UWord32 _lastTimestamp;
|
||||
WebRtc_UWord32 _lastInTimestamp;
|
||||
CodecInst _sendCodecInst;
|
||||
uint8_t _cng_nb_pltype;
|
||||
uint8_t _cng_wb_pltype;
|
||||
uint8_t _cng_swb_pltype;
|
||||
uint8_t _red_pltype;
|
||||
bool _vadEnabled;
|
||||
bool _dtxEnabled;
|
||||
ACMVADMode _vadMode;
|
||||
ACMGenericCodec* _codecs[ACMCodecDB::kMaxNumCodecs];
|
||||
ACMGenericCodec* _slaveCodecs[ACMCodecDB::kMaxNumCodecs];
|
||||
WebRtc_Word16 _mirrorCodecIdx[ACMCodecDB::kMaxNumCodecs];
|
||||
bool _stereoReceive[ACMCodecDB::kMaxNumCodecs];
|
||||
bool _stereoReceiveRegistered;
|
||||
bool _stereoSend;
|
||||
int _prev_received_channel;
|
||||
int _expected_channels;
|
||||
WebRtc_Word32 _currentSendCodecIdx;
|
||||
int _current_receive_codec_idx;
|
||||
bool _sendCodecRegistered;
|
||||
ACMResampler _inputResampler;
|
||||
ACMResampler _outputResampler;
|
||||
ACMNetEQ _netEq;
|
||||
CriticalSectionWrapper* _acmCritSect;
|
||||
ACMVADCallback* _vadCallback;
|
||||
WebRtc_UWord8 _lastRecvAudioCodecPlType;
|
||||
|
||||
// RED/FEC.
|
||||
bool _isFirstRED;
|
||||
bool _fecEnabled;
|
||||
WebRtc_UWord8* _redBuffer;
|
||||
RTPFragmentationHeader* _fragmentation;
|
||||
WebRtc_UWord32 _lastFECTimestamp;
|
||||
// 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.
|
||||
WebRtc_UWord8 _previousPayloadType;
|
||||
|
||||
// This keeps track of payload types associated with _codecs[].
|
||||
// We define it as signed variable and initialize with -1 to indicate
|
||||
// unused elements.
|
||||
WebRtc_Word16 _registeredPlTypes[ACMCodecDB::kMaxNumCodecs];
|
||||
|
||||
// Used when payloads are pushed into ACM without any RTP info
|
||||
// One example is when pre-encoded bit-stream is pushed from
|
||||
// a file.
|
||||
WebRtcRTPHeader* _dummyRTPHeader;
|
||||
WebRtc_UWord16 _recvPlFrameSizeSmpls;
|
||||
|
||||
bool _receiverInitialized;
|
||||
ACMDTMFDetection* _dtmfDetector;
|
||||
|
||||
AudioCodingFeedback* _dtmfCallback;
|
||||
WebRtc_Word16 _lastDetectedTone;
|
||||
CriticalSectionWrapper* _callbackCritSect;
|
||||
|
||||
AudioFrame _audioFrame;
|
||||
|
||||
#ifdef ACM_QA_TEST
|
||||
FILE* _outgoingPL;
|
||||
FILE* _incomingPL;
|
||||
FILE* _outgoingPL;
|
||||
FILE* _incomingPL;
|
||||
#endif
|
||||
|
||||
};
|
||||
|
||||
} // namespace webrtc
|
||||
} // namespace webrtc
|
||||
|
||||
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_AUDIO_CODING_MODULE_IMPL_H_
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user