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Commit 35d05dcb authored by Chia-chi Yeh's avatar Chia-chi Yeh
Browse files

RTP: support payloads with larger packetization interval. 

RFC 3551 section 4.2 said that a receiver should accept packets
representing between 0 and 200ms of audio data. Now we add the
ability to decode multiple frames in a payload as long as the
jitter buffer is not full. This change covers G711, GSM, and
GSM-EFR. AMR will be added later.

Bug: 3029736
Change-Id: Ifd194596766d14f02177925c58432cd620e44dd7
parent 1ed7a407

voip/jni/rtp/AmrCodec.cpp

+12 −8
Original line number Diff line number Diff line
@@ -52,7 +52,7 @@ public: 


    int set(int sampleRate, const char *fmtp);

    int encode(void *payload, int16_t *samples);

    int decode(int16_t *samples, void *payload, int length);

    int decode(int16_t *samples, int count, void *payload, int length);



private:

    void *mEncoder;
@@ -128,7 +128,7 @@ int AmrCodec::encode(void *payload, int16_t *samples) 
    return length;

}



int AmrCodec::decode(int16_t *samples, void *payload, int length)

int AmrCodec::decode(int16_t *samples, int count, void *payload, int length)

{

    unsigned char *bytes = (unsigned char *)payload;

    Frame_Type_3GPP type;
@@ -213,7 +213,7 @@ public: 
    }



    int encode(void *payload, int16_t *samples);

    int decode(int16_t *samples, void *payload, int length);

    int decode(int16_t *samples, int count, void *payload, int length);



private:

    void *mEncoder;
@@ -239,20 +239,24 @@ int GsmEfrCodec::encode(void *payload, int16_t *samples) 
    return -1;

}



int GsmEfrCodec::decode(int16_t *samples, void *payload, int length)

int GsmEfrCodec::decode(int16_t *samples, int count, void *payload, int length)

{

    unsigned char *bytes = (unsigned char *)payload;

    if (length == 31 && (bytes[0] >> 4) == 0x0C) {

    int n = 0;

    while (n + 160 <= count && length >= 31 && (bytes[0] >> 4) == 0x0C) {

        for (int i = 0; i < 30; ++i) {

            bytes[i] = (bytes[i] << 4) | (bytes[i + 1] >> 4);

        }

        bytes[30] <<= 4;



        if (AMRDecode(mDecoder, AMR_122, bytes, samples, MIME_IETF) == 31) {

            return 160;

        if (AMRDecode(mDecoder, AMR_122, bytes, &samples[n], MIME_IETF) != 31) {

            break;

        }

        n += 160;

        length -= 31;

        bytes += 31;

    }

    return -1;

    return n;

}



} // namespace

voip/jni/rtp/AudioCodec.h

+1 −1
Original line number Diff line number Diff line
@@ -30,7 +30,7 @@ public: 
    // Returns the length of payload in bytes.

    virtual int encode(void *payload, int16_t *samples) = 0;

    // Returns the number of decoded samples.

    virtual int decode(int16_t *samples, void *payload, int length) = 0;

    virtual int decode(int16_t *samples, int count, void *payload, int length) = 0;

};



AudioCodec *newAudioCodec(const char *codecName);

voip/jni/rtp/AudioGroup.cpp

+11 −10
Original line number Diff line number Diff line
@@ -395,7 +395,8 @@ void AudioStream::decode(int tick) 
        mLatencyTimer = tick;

    }



    if (mBufferTail - mBufferHead > BUFFER_SIZE - mInterval) {

    int count = (BUFFER_SIZE - (mBufferTail - mBufferHead)) * mSampleRate;

    if (count < mSampleCount) {

        // Buffer overflow. Drop the packet.

        LOGV("stream[%d] buffer overflow", mSocket);

        recv(mSocket, &c, 1, MSG_DONTWAIT);
@@ -403,19 +404,18 @@ void AudioStream::decode(int tick) 
    }



    // Receive the packet and decode it.

    int16_t samples[mSampleCount];

    int length = 0;

    int16_t samples[count];

    if (!mCodec) {

        // Special case for device stream.

        length = recv(mSocket, samples, sizeof(samples),

        count = recv(mSocket, samples, sizeof(samples),

            MSG_TRUNC | MSG_DONTWAIT) >> 1;

    } else {

        __attribute__((aligned(4))) uint8_t buffer[2048];

        sockaddr_storage remote;

        socklen_t len = sizeof(remote);

        socklen_t addrlen = sizeof(remote);



        length = recvfrom(mSocket, buffer, sizeof(buffer),

            MSG_TRUNC | MSG_DONTWAIT, (sockaddr *)&remote, &len);

        int length = recvfrom(mSocket, buffer, sizeof(buffer),

            MSG_TRUNC | MSG_DONTWAIT, (sockaddr *)&remote, &addrlen);



        // Do we need to check SSRC, sequence, and timestamp? They are not

        // reliable but at least they can be used to identify duplicates?
@@ -433,14 +433,15 @@ void AudioStream::decode(int tick) 
        }

        length -= offset;

        if (length >= 0) {

            length = mCodec->decode(samples, &buffer[offset], length);

            length = mCodec->decode(samples, count, &buffer[offset], length);

        }

        if (length > 0 && mFixRemote) {

            mRemote = remote;

            mFixRemote = false;

        }

        count = length;

    }

    if (length <= 0) {

    if (count <= 0) {

        LOGV("stream[%d] decoder error", mSocket);

        return;

    }
@@ -462,7 +463,7 @@ void AudioStream::decode(int tick) 


    // Append to the jitter buffer.

    int tail = mBufferTail * mSampleRate;

    for (int i = 0; i < mSampleCount; ++i) {

    for (int i = 0; i < count; ++i) {

        mBuffer[tail & mBufferMask] = samples[i];

        ++tail;

    }

voip/jni/rtp/G711Codec.cpp

+10 −4
Original line number Diff line number Diff line
@@ -39,7 +39,7 @@ public: 
        return mSampleCount;

    }

    int encode(void *payload, int16_t *samples);

    int decode(int16_t *samples, void *payload, int length);

    int decode(int16_t *samples, int count, void *payload, int length);

private:

    int mSampleCount;

};
@@ -64,9 +64,12 @@ int UlawCodec::encode(void *payload, int16_t *samples) 
    return mSampleCount;

}



int UlawCodec::decode(int16_t *samples, void *payload, int length)

int UlawCodec::decode(int16_t *samples, int count, void *payload, int length)

{

    int8_t *ulaws = (int8_t *)payload;

    if (length > count) {

        length = count;

    }

    for (int i = 0; i < length; ++i) {

        int ulaw = ~ulaws[i];

        int exponent = (ulaw >> 4) & 0x07;
@@ -87,7 +90,7 @@ public: 
        return mSampleCount;

    }

    int encode(void *payload, int16_t *samples);

    int decode(int16_t *samples, void *payload, int length);

    int decode(int16_t *samples, int count, void *payload, int length);

private:

    int mSampleCount;

};
@@ -111,9 +114,12 @@ int AlawCodec::encode(void *payload, int16_t *samples) 
    return mSampleCount;

}



int AlawCodec::decode(int16_t *samples, void *payload, int length)

int AlawCodec::decode(int16_t *samples, int count, void *payload, int length)

{

    int8_t *alaws = (int8_t *)payload;

    if (length > count) {

        length = count;

    }

    for (int i = 0; i < length; ++i) {

        int alaw = alaws[i] ^ 0x55;

        int exponent = (alaw >> 4) & 0x07;

voip/jni/rtp/GsmCodec.cpp

+10 −6
Original line number Diff line number Diff line
@@ -44,7 +44,7 @@ public: 
    }



    int encode(void *payload, int16_t *samples);

    int decode(int16_t *samples, void *payload, int length);

    int decode(int16_t *samples, int count, void *payload, int length);



private:

    gsm mEncode;
@@ -57,13 +57,17 @@ int GsmCodec::encode(void *payload, int16_t *samples) 
    return 33;

}



int GsmCodec::decode(int16_t *samples, void *payload, int length)

int GsmCodec::decode(int16_t *samples, int count, void *payload, int length)

{

    if (length == 33 &&

        gsm_decode(mDecode, (unsigned char *)payload, samples) == 0) {

        return 160;

    unsigned char *bytes = (unsigned char *)payload;

    int n = 0;

    while (n + 160 <= count && length >= 33 &&

        gsm_decode(mDecode, bytes, &samples[n]) == 0) {

        n += 160;

        length -= 33;

        bytes += 33;

    }

    return -1;

    return n;

}



} // namespace