FFmpeg and types of samples

2019-04-13 00:01发布

问题:

Decoded audio using FFmpeg (function avcodec_decode_audio3) and try to reproduce it through Audiotrack in Android. When playing I hear some growling. No music. On the forums advised that the problem with the samples. Question: how to solve this problem, if it is associated with a mismatch of types of samples? Code decoding:



        while (av_read_frame(pFormatCtx, &packet)>= 0) {
            if (aCodecCtx->codec_type == AVMEDIA_TYPE_AUDIO) {
                        int data_size = AVCODEC_MAX_AUDIO_FRAME_SIZE * 2;
                        int size=packet.size;
 int decoded = 0;
                        while(size > 0) {
                                int len = avcodec_decode_audio3(aCodecCtx, (uint16_t *) pAudioBuffer, &data_size, &packet);


                                jbyte *bytes = (*env)->GetByteArrayElements(env, array, NULL);
                                memcpy(bytes + decoded, (uint16_t *) pAudioBuffer, len);
                                (*env)->ReleaseByteArrayElements(env, array, bytes, 0);

                                (*env)->CallStaticVoidMethod(env, cls, mid, array);


                                size -= len;
                                decoded += len;
                                }
            }

     }

Java code:


    protected void onCreate(Bundle savedInstanceState) {
        super.onCreate(savedInstanceState);
        setContentView(R.layout.main);

         int bufSize = AudioTrack.getMinBufferSize(44100,                                AudioFormat.CHANNEL_CONFIGURATION_MONO, 
                        AudioFormat.ENCODING_PCM_16BIT);
        track = new AudioTrack(AudioManager.STREAM_MUSIC, 44100, AudioFormat.CHANNEL_CONFIGURATION_MONO, 
                    AudioFormat.ENCODING_PCM_16BIT, bufSize, AudioTrack.MODE_STREAM);
        track.play();

            bytes = new byte[bufSize];
            int res = main(2, "/sdcard/muzika_iz_reklami_bmw_5_series_-_bmw_5_series.mp3", bytes);


            System.out.println(res);
    }
    private static void play(byte[] play) {

        track.write(play, 0, play.length);
    }

How to solve this problem? P.S. The file itself is checked - the standard player plays. Format: mp3.

回答1:

Keep a counter of decoded bytes that is accumulated with len after each call to avcodec_decode_audio for a single AVPacket. Then use it as offset at the Java byte array:

int decoded = 0;
  ...
int len = avcodec_decode_audio3(aCodecCtx, (uint8_t *) pAudioBuffer, &data_size, &packet);
  ...
memcpy(bytes + decoded, (uint8_t *) pAudioBuffer, len);
  ...
size -= len;
decoded += len;

You're overwriting the beginning of the buffer over and over instead of appending.



回答2:

here is my audio decode code, hope it would help you.

int16_t* samples = (int16_t*)av_malloc(AVCODEC_MAX_AUDIO_FRAME_SIZE);
int sample_size;

while(!m_quit && av_read_frame_wrapper(m_pAVFormatContext, &packet) >= 0) {
    if(packet.stream_index == m_audioStream) {
        org_data = packet.data;
        org_size = packet.size;
        LOGV("audio packet size = %d", packet.size);
        while (!m_quit && packet.size > 0) {
            sample_size = AVCODEC_MAX_AUDIO_FRAME_SIZE;
            len = avcodec_decode_audio3(m_pAVCodecContext, samples, &sample_size, &packet);
            LOGV("sample_size = %d, len = %d", sample_size, len);
            if (len < 0) {
                LOGE("Error when decoding...");
                break;
            }

            if (sample_size > 0) {
                //begin = now_ms();

                //audio_output_native(samples, sample_size);

                /* use audio queue */
                SampleNode* node = new SampleNode(samples, sample_size);
                m_pSampleQueue->enqueue(node);

                //end = now_ms();

                //LOGV("sample_size > 0, output audio samples cost %llu ms\n", end - begin);
            }
            packet.size -= len;
            packet.data += len;
        }

        packet.size = org_size;
        packet.data = org_data;

        av_free_packet(&packet);
    } else {
        av_free_packet(&packet);
    }
}