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- fixes + correct decode of DTX/CNG periods

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This commit is contained in:
Dmytro Bogovych
2026-01-11 17:48:42 +03:00
parent fba022c7f6
commit f650eaccb7
5 changed files with 300 additions and 261 deletions
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src/engine/media/MT_AudioReceiver.h
+56 -42
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@@ -6,6 +6,7 @@
#ifndef __MT_AUDIO_RECEIVER_H
#define __MT_AUDIO_RECEIVER_H
#include "../engine_config.h"
#include "MT_Stream.h"
#include "MT_CodecList.h"
#include "MT_CngHelper.h"
@@ -61,17 +62,17 @@ public:
RtpBuffer(Statistics& stat);
~RtpBuffer();
unsigned ssrc();
unsigned ssrc() const;
void setSsrc(unsigned ssrc);
void setHigh(int milliseconds);
int high();
int high() const;
void setLow(int milliseconds);
int low();
int low() const;
void setPrebuffer(int milliseconds);
int prebuffer();
int prebuffer() const;
int getNumberOfReturnedPackets() const;
int getNumberOfAddPackets() const;
@@ -88,12 +89,12 @@ public:
FetchResult fetch(ResultList& rl);
protected:
unsigned mSsrc = 0;
int mHigh = RTP_BUFFER_HIGH,
mLow = RTP_BUFFER_LOW,
mPrebuffer = RTP_BUFFER_PREBUFFER;
int mReturnedCounter = 0,
mAddCounter = 0;
unsigned mSsrc = 0;
int mHigh = RTP_BUFFER_HIGH,
mLow = RTP_BUFFER_LOW,
mPrebuffer = RTP_BUFFER_PREBUFFER;
int mReturnedCounter = 0,
mAddCounter = 0;
mutable Mutex mGuard;
typedef std::vector<std::shared_ptr<Packet>> PacketList;
@@ -105,7 +106,7 @@ protected:
std::optional<uint32_t> mLastSeqno;
// To calculate average interval between packet add. It is close to jitter but more useful in debugging.
float mLastAddTime = 0.0;
float mLastAddTime = 0.0f;
};
class Receiver
@@ -125,29 +126,37 @@ public:
~AudioReceiver();
// Update codec settings
void setCodecSettings(const CodecList::Settings& codecSettings);
CodecList::Settings& getCodecSettings();
void setCodecSettings(const CodecList::Settings& codecSettings);
CodecList::Settings& getCodecSettings();
// Returns false when packet is rejected as illegal. codec parameter will show codec which will be used for decoding.
// Lifetime of pointer to codec is limited by lifetime of AudioReceiver (it is container).
bool add(const std::shared_ptr<jrtplib::RTPPacket>& p, Codec** codec = nullptr);
// Returns false when there is no rtp data from jitter
enum DecodeOptions
/*enum DecodeOptions
{
DecodeOptions_ResampleToMainRate = 0,
DecodeOptions_DontResample = 1,
DecodeOptions_FillCngGap = 2,
DecodeOptions_SkipDecode = 4
};*/
struct DecodeOptions
{
bool mResampleToMainRate = true;
bool mFillGapByCNG = false;
bool mSkipDecode = false;
};
enum DecodeResult
{
DecodeResult_Ok,
DecodeResult_Skip,
DecodeResult_BadPacket
DecodeResult_Ok, // Decoded ok
DecodeResult_Skip, // Just no data - emit silence instead
DecodeResult_BadPacket // Error happened during the decode
};
DecodeResult getAudio(Audio::DataWindow& output, int options = DecodeOptions_ResampleToMainRate, int* rate = nullptr);
DecodeResult getAudio(Audio::DataWindow& output, DecodeOptions options = {.mResampleToMainRate = true, .mFillGapByCNG = false, .mSkipDecode = false}, int* rate = nullptr);
// Looks for codec by payload type
Codec* findCodec(int payloadType);
@@ -163,52 +172,57 @@ public:
int samplerateFor(jrtplib::RTPPacket& p);
protected:
RtpBuffer mBuffer;
CodecMap mCodecMap;
PCodec mCodec;
int mFrameCount = 0;
CodecList::Settings mCodecSettings;
CodecList mCodecList;
JitterStatistics mJitterStats;
RtpBuffer mBuffer; // Jitter buffer itself
CodecMap mCodecMap;
PCodec mCodec;
int mFrameCount = 0;
CodecList::Settings mCodecSettings;
CodecList mCodecList;
JitterStatistics mJitterStats;
std::shared_ptr<jrtplib::RTPPacket> mCngPacket;
CngDecoder mCngDecoder;
// Decode RTP early, do not wait for speaker callback
bool mEarlyDecode = false;
CngDecoder mCngDecoder;
size_t mDTXSamplesToEmit = 0; // How much silence (or CNG) should be emited before next RTP packet gets into the action
// Buffer to hold decoded data
char mDecodedFrame[65536];
int mDecodedLength = 0;
int16_t mDecodedFrame[MT_MAX_DECODEBUFFER];
size_t mDecodedLength = 0;
// Buffer to hold data converted to stereo/mono
char mConvertedFrame[32768];
int mConvertedLength = 0;
// Buffer to hold data converted to stereo/mono; there is multiplier 2 as it can be stereo audio
int16_t mConvertedFrame[MT_MAX_DECODEBUFFER * 2];
size_t mConvertedLength = 0;
// Buffer to hold data resampled to AUDIO_SAMPLERATE
char mResampledFrame[65536];
int mResampledLength = 0;
int16_t mResampledFrame[MT_MAX_DECODEBUFFER];
size_t mResampledLength = 0;
// Last packet time length
int mLastPacketTimeLength = 0;
std::optional<uint32_t> mLastPacketTimestamp;
int mFailedCount = 0;
Audio::Resampler mResampler8, mResampler16,
mResampler32, mResampler48;
Audio::Resampler mResampler8, mResampler16, mResampler32, mResampler48;
Audio::PWavFileWriter mDecodedDump;
float mLastDecodeTime = 0.0; // Time last call happened to codec->decode()
std::optional<std::chrono::steady_clock::time_point> mLastDecodeTimestamp; // Time last call happened to codec->decode()
float mIntervalSum = 0.0;
float mIntervalSum = 0.0f;
int mIntervalCount = 0;
// Zero rate will make audio mono but resampling will be skipped
void makeMonoAndResample(int rate, int channels);
// Resamples, sends to analysis, writes to dump and queues to output decoded frames from mDecodedFrame
void processDecoded(Audio::DataWindow& output, int options);
void processDecoded(Audio::DataWindow& output, DecodeOptions options);
void produceSilence(std::chrono::milliseconds length, Audio::DataWindow& output, DecodeOptions options);
void produceCNG(std::chrono::milliseconds length, Audio::DataWindow& output, DecodeOptions options);
void processStatisticsWithAmrCodec(Codec* c);
// Calculate bitrate switch statistics for AMR codecs
void updateAmrCodecStats(Codec* c);
DecodeResult decodeGap(Audio::DataWindow& output, DecodeOptions options);
DecodeResult decodePacket(const RtpBuffer::ResultList& rl, Audio::DataWindow& output, DecodeOptions options, int* rate = nullptr);
DecodeResult decodeNone(Audio::DataWindow& output, DecodeOptions options);
};
class DtmfReceiver: public Receiver