AudioStream類

AudioStream類是音頻流的基類，重寫了MediaStream的encodeWithMediaRecorder方法，實(shí)現(xiàn)了MediaRecorder錄制音頻的操作芥驳。同時(shí)作為抽象類申屹，它并沒有重寫encodeWithMediaCodec方法溉苛，而是留給子類去具體實(shí)現(xiàn)。

    @Override
    protected void encodeWithMediaRecorder() throws IOException {
        
        // We need a local socket to forward data output by the camera to the packetizer
        createSockets();

        Log.v(TAG,"Requested audio with "+mQuality.bitRate/1000+"kbps"+" at "+mQuality.samplingRate/1000+"kHz");
        
        mMediaRecorder = new MediaRecorder();
        mMediaRecorder.setAudioSource(mAudioSource);
        mMediaRecorder.setOutputFormat(mOutputFormat);
        mMediaRecorder.setAudioEncoder(mAudioEncoder);
        mMediaRecorder.setAudioChannels(1);
        mMediaRecorder.setAudioSamplingRate(mQuality.samplingRate);
        mMediaRecorder.setAudioEncodingBitRate(mQuality.bitRate);
        
        // We write the ouput of the camera in a local socket instead of a file !           
        // This one little trick makes streaming feasible quiet simply: data from the camera
        // can then be manipulated at the other end of the socket
        mMediaRecorder.setOutputFile(mSender.getFileDescriptor());

        mMediaRecorder.prepare();
        mMediaRecorder.start();

        try {
            // mReceiver.getInputStream contains the data from the camera
            // the mPacketizer encapsulates this stream in an RTP stream and send it over the network
            mPacketizer.setDestination(mDestination, mRtpPort, mRtcpPort);
            mPacketizer.setInputStream(mReceiver.getInputStream());
            mPacketizer.start();
            mStreaming = true;
        } catch (IOException e) {
            stop();
            throw new IOException("Something happened with the local sockets :/ Start failed !");
        }
        
    }

重寫encodeWithMediaRecorder方法贺喝，主要是MediaRecorder錄制音頻的操作菱鸥。首先啟動(dòng)本地Socket，再對(duì)MediaRecorder對(duì)象設(shè)置音頻來源躏鱼、音頻編碼氮采、音頻質(zhì)量等參數(shù)，然后開始錄制染苛。這里值得一提的是鹊漠，MediaRecorder輸出的音頻數(shù)據(jù)是寫入Socket中而不是文件中，這樣就可以在Socket的另一端進(jìn)行操作茶行。最后一步是使用Packetizer對(duì)象將數(shù)據(jù)打包并發(fā)送到網(wǎng)絡(luò)傳輸躯概。

AACStream類

AACStream類是一個(gè)關(guān)于AAC音頻格式的AudioStream的子類，內(nèi)部封裝了對(duì)AAC音頻格式的配置和編碼操作拢军。

    private static boolean AACStreamingSupported() {
        if (Build.VERSION.SDK_INT<14) return false;
        try {
            MediaRecorder.OutputFormat.class.getField("AAC_ADTS");
            return true;
        } catch (Exception e) {
            return false;
        }
    }

構(gòu)造函數(shù)中會(huì)判斷是否支持AAC編碼格式楞陷。

        // Checks if the user has supplied an exotic sampling rate
        int i=0;
        for (;i<AUDIO_SAMPLING_RATES.length;i++) {
            if (AUDIO_SAMPLING_RATES[i] == mQuality.samplingRate) {
                mSamplingRateIndex = i;
                break;
            }
        }
        // If he did, we force a reasonable one: 16 kHz
        if (i>12) mQuality.samplingRate = 16000;

configure()方法中的部分代碼。在start()開始時(shí)需要先檢查一下采樣率配置信息茉唉，不符合規(guī)范則強(qiáng)行設(shè)置默認(rèn)采樣率固蛾。

    @Override
    protected void encodeWithMediaRecorder() throws IOException {
        testADTS();
        ((AACADTSPacketizer)mPacketizer).setSamplingRate(mQuality.samplingRate);
        super.encodeWithMediaRecorder();
    }

重寫encodeWithMediaRecorder方法，testADTS()方法是先從麥克風(fēng)記錄AAC ADTS的簡短樣本度陆，以了解該設(shè)備支持的真實(shí)的采樣率艾凯，便于設(shè)置配置信息和防止報(bào)錯(cuò)。篇幅原因懂傀，這里就不貼出testADTS()的代碼了趾诗。

下面我們開始分析重寫encodeWithMediaCodec()方法里面的內(nèi)容，我把逐句分析寫在注釋里面蹬蚁。

        //計(jì)算出緩沖區(qū)的大小
        final int bufferSize = AudioRecord.getMinBufferSize(mQuality.samplingRate, AudioFormat.CHANNEL_IN_MONO, AudioFormat.ENCODING_PCM_16BIT)*2;
        //設(shè)置打包器的采樣率
        ((AACLATMPacketizer)mPacketizer).setSamplingRate(mQuality.samplingRate);
        //實(shí)例化AudioRecord恃泪，參數(shù)依次為：聲音來源、采樣率犀斋、聲道數(shù)贝乎、編碼方式、緩沖區(qū)
        mAudioRecord = new AudioRecord(MediaRecorder.AudioSource.MIC, mQuality.samplingRate, AudioFormat.CHANNEL_IN_MONO, AudioFormat.ENCODING_PCM_16BIT, bufferSize);
        //實(shí)例化MediaCodec編碼器
        mMediaCodec = MediaCodec.createEncoderByType("audio/mp4a-latm");
        //配置信息依次為：格式叽粹、位速率览效、頻道數(shù)、采樣率虫几、AAC文件锤灿、最大輸入緩沖區(qū)
        MediaFormat format = new MediaFormat();
        format.setString(MediaFormat.KEY_MIME, "audio/mp4a-latm");
        format.setInteger(MediaFormat.KEY_BIT_RATE, mQuality.bitRate);
        format.setInteger(MediaFormat.KEY_CHANNEL_COUNT, 1);
        format.setInteger(MediaFormat.KEY_SAMPLE_RATE, mQuality.samplingRate);
        format.setInteger(MediaFormat.KEY_AAC_PROFILE, MediaCodecInfo.CodecProfileLevel.AACObjectLC);
        format.setInteger(MediaFormat.KEY_MAX_INPUT_SIZE, bufferSize);
        mMediaCodec.configure(format, null, null, MediaCodec.CONFIGURE_FLAG_ENCODE);
        //開始錄制音頻
        mAudioRecord.startRecording();
        mMediaCodec.start();
        
        //設(shè)置編碼流，在后面的文章會(huì)詳細(xì)講到
        final MediaCodecInputStream inputStream = new MediaCodecInputStream(mMediaCodec);
        final ByteBuffer[] inputBuffers = mMediaCodec.getInputBuffers();

        //開啟一個(gè)線程辆脸，讀取和處理
        mThread = new Thread(new Runnable() {
            @Override
            public void run() {
                int len = 0, bufferIndex = 0;
                try {
                    //無限循環(huán)讀取
                    while (!Thread.interrupted()) {
                        //從輸入流隊(duì)列中取數(shù)據(jù)進(jìn)行編碼操作(出隊(duì)列)但校。
                        bufferIndex = mMediaCodec.dequeueInputBuffer(10000);
                        if (bufferIndex>=0) {
                            inputBuffers[bufferIndex].clear();
                            //從mAudioRecord讀取數(shù)據(jù)到inputBuffers[bufferIndex]中
                            len = mAudioRecord.read(inputBuffers[bufferIndex], bufferSize);
                            if (len ==  AudioRecord.ERROR_INVALID_OPERATION || len == AudioRecord.ERROR_BAD_VALUE) {
                                Log.e(TAG,"An error occured with the AudioRecord API !");
                            } else {
                                //Log.v(TAG,"Pushing raw audio to the decoder: len="+len+" bs: "+inputBuffers[bufferIndex].capacity());
                                //輸入流入隊(duì)列(往編碼器中添加數(shù)據(jù)做編碼處理)
                                mMediaCodec.queueInputBuffer(bufferIndex, 0, len, System.nanoTime()/1000, 0);
                            }
                        }
                    }
                } catch (RuntimeException e) {
                    e.printStackTrace();
                }
            }
        });

        mThread.start();
        //把編碼完成的數(shù)據(jù)流封裝打包并進(jìn)行網(wǎng)絡(luò)傳輸
        // The packetizer encapsulates this stream in an RTP stream and send it over the network
        mPacketizer.setDestination(mDestination, mRtpPort, mRtcpPort);
        mPacketizer.setInputStream(inputStream);
        mPacketizer.start();

        mStreaming = true;

以上可以看到，重寫encodeWithMediaCodec()方法主要流程就是：實(shí)例化和配置AudioRecord用來錄取音頻數(shù)據(jù)啡氢，實(shí)例化和配置MediaCodec用于對(duì)數(shù)據(jù)編碼状囱，開啟一個(gè)線程循環(huán)讀取AudioRecord錄取的音頻數(shù)據(jù)流州刽，并將原始音頻數(shù)據(jù)流添加到MediaCodec編碼器中進(jìn)行編碼，然后將編碼完成的數(shù)據(jù)流通過Packetizer打包器打包并發(fā)送出去浪箭。

AMRNBStream類

AMRNBStream類是一個(gè)關(guān)于ANR音頻格式的AudioStream的子類穗椅。

@Override
    protected void encodeWithMediaCodec() throws IOException {
        super.encodeWithMediaRecorder();
    }

AMRNBStream可操作的動(dòng)作不多，這里重寫encodeWithMediaCodec()方法直接指向了super.encodeWithMediaRecorder()奶栖。關(guān)于AAC和AMR的比較可參考：AAC和AMR音頻編碼標(biāo)準(zhǔn)介紹

至此我們完整的了解了音頻數(shù)據(jù)流的配置匹表、采集、編碼的整個(gè)過程宣鄙，下一篇我們將分析VideoStream類和它的子類袍镀，詳細(xì)了解視頻流的配置、采集和編碼的流程冻晤。