如何使用 AudioUnit 播放来自服务器的音频流？

Question

- (void)openPlayThreadWithRtmpURL:(NSString *)rtmpURL {
spx_int16_t *input_buffer;

do {
    if (self.rtmpDelegate) {
        [self.rtmpDelegate evenCallbackWithEvent:2000];
    }

    //init speex decoder and config;
    speex_bits_init(&dbits);
    dec_state = speex_decoder_init(&speex_wb_mode);

    speex_decoder_ctl(dec_state, SPEEX_GET_FRAME_SIZE, &dec_frame_size);

    input_buffer = malloc(dec_frame_size * sizeof(short));

    NSLog(@"Init Speex decoder success frame_size = %d",dec_frame_size);

    //init rtmp
    pPlayRtmp = RTMP_Alloc();
    RTMP_Init(pPlayRtmp);
    NSLog(@"Play RTMP_Init %@\n", rtmpURL);

    if (!RTMP_SetupURL(pPlayRtmp, (char*)[rtmpURL UTF8String])) {
        NSLog(@"Play RTMP_SetupURL error\n");
        if(self.rtmpDelegate) {
            [self.rtmpDelegate evenCallbackWithEvent:2002];
        }
        break;
    }

    if (!RTMP_Connect(pPlayRtmp, NULL) || !RTMP_ConnectStream(pPlayRtmp, 0)) {
        NSLog(@"Play RTMP_Connect or RTMP_ConnectStream error\n");
        if(self.rtmpDelegate) {
            [self.rtmpDelegate evenCallbackWithEvent:2002];
        }
        break;
    }

    if(self.rtmpDelegate) {
        [self.rtmpDelegate evenCallbackWithEvent:2001];
    }
    NSLog(@"Player RTMP_Connected \n");

    RTMPPacket rtmp_pakt = {0};
    isStartPlay = YES;
    while (isStartPlay && RTMP_ReadPacket(pPlayRtmp, &rtmp_pakt)) {
        if (RTMPPacket_IsReady(&rtmp_pakt)) {
            if (!rtmp_pakt.m_nBodySize) {
                continue;
            }
            if (rtmp_pakt.m_packetType == RTMP_PACKET_TYPE_AUDIO) {
                NSLog(@"Audio size = %d head = %d time = %d", rtmp_pakt.m_nBodySize, rtmp_pakt.m_body[0], rtmp_pakt.m_nTimeStamp);
                speex_bits_read_from(&dbits, rtmp_pakt.m_body + 1, rtmp_pakt.m_nBodySize - 1);
                speex_decode_int(dec_state, &dbits, input_buffer);  //audioData in the input_buffer
                //do something...



            } else if (rtmp_pakt.m_packetType == RTMP_PACKET_TYPE_VIDEO) {
                // 处理视频数据包
            } else if (rtmp_pakt.m_packetType == RTMP_PACKET_TYPE_INVOKE) {
                // 处理invoke包
                NSLog(@"RTMP_PACKET_TYPE_INVOKE");
                RTMP_ClientPacket(pPlayRtmp,&rtmp_pakt);
            } else if (rtmp_pakt.m_packetType == RTMP_PACKET_TYPE_INFO) {
                // 处理信息包
                //NSLog(@"RTMP_PACKET_TYPE_INFO");
            } else if (rtmp_pakt.m_packetType == RTMP_PACKET_TYPE_FLASH_VIDEO) {
                // 其他数据
                int index = 0;
                while (1) {
                    int StreamType; //1-byte
                    int MediaSize; //3-byte
                    int TiMMER; //3-byte
                    int Reserve; //4-byte
                    char* MediaData; //MediaSize-byte
                    int TagLen; //4-byte

                    StreamType = rtmp_pakt.m_body[index];
                    index += 1;
                    MediaSize = bigThreeByteToInt(rtmp_pakt.m_body + index);
                    index += 3;
                    TiMMER = bigThreeByteToInt(rtmp_pakt.m_body + index);
                    index += 3;
                    Reserve = bigFourByteToInt(rtmp_pakt.m_body + index);
                    index += 4;
                    MediaData = rtmp_pakt.m_body + index;
                    index += MediaSize;
                    TagLen = bigFourByteToInt(rtmp_pakt.m_body + index);
                    index += 4;
                    //NSLog(@"bodySize:%d   index:%d",rtmp_pakt.m_nBodySize,index);
                    //LOGI("StreamType:%d MediaSize:%d  TiMMER:%d TagLen:%d\n", StreamType, MediaSize, TiMMER, TagLen);
                    if (StreamType == 0x08) {
                        //音频包
                        //int MediaSize = bigThreeByteToInt(rtmp_pakt.m_body+1);
                        //  LOGI("FLASH audio size:%d  head:%d time:%d\n", MediaSize, MediaData[0], TiMMER);
                        speex_bits_read_from(&dbits, MediaData + 1, MediaSize - 1);
                        speex_decode_int(dec_state, &dbits, input_buffer);

                        //[mAudioPlayer putAudioData:input_buffer];
                        //  putAudioQueue(output_buffer,dec_frame_size);
                    } else if (StreamType == 0x09) {
                        //视频包
                        //  LOGI( "video size:%d  head:%d\n", MediaSize, MediaData[0]);
                    }
                    if (rtmp_pakt.m_nBodySize == index) {
                        break;
                    }
                }
            }
            RTMPPacket_Free(&rtmp_pakt);
        }
    }
    if (isStartPlay) {
        if(self.rtmpDelegate) {
            [self.rtmpDelegate evenCallbackWithEvent:2005];
        }
        isStartPlay = NO;
    }
} while (0);
[mAudioPlayer stopPlay];
if (self.rtmpDelegate) {
    [self.rtmpDelegate evenCallbackWithEvent:2004];
}
if (RTMP_IsConnected(pPlayRtmp)) {
    RTMP_Close(pPlayRtmp);
}
RTMP_Free(pPlayRtmp);
free(input_buffer);
speex_bits_destroy(&dbits);
speex_decoder_destroy(dec_state);

}

这是我的自定义方法。 rtmpURL是一个NSString的对象，它是一个流服务器地址。使用这种方法，我可以从服务器获取音频流的编码，之后，我使用speex解码器解码我得到的数据，就像这样：

//init speex decoder and config;
    speex_bits_init(&dbits);
    dec_state = speex_decoder_init(&speex_wb_mode);

    speex_decoder_ctl(dec_state, SPEEX_GET_FRAME_SIZE, &dec_frame_size);

    input_buffer = malloc(dec_frame_size * sizeof(short));

    NSLog(@"Init Speex decoder success frame_size = %d",dec_frame_size);
 if (rtmp_pakt.m_packetType == RTMP_PACKET_TYPE_AUDIO) {
                NSLog(@"Audio size = %d head = %d time = %d", rtmp_pakt.m_nBodySize, rtmp_pakt.m_body[0], rtmp_pakt.m_nTimeStamp);
                speex_bits_read_from(&dbits, rtmp_pakt.m_body + 1, rtmp_pakt.m_nBodySize - 1);
                speex_decode_int(dec_state, &dbits, input_buffer);  //audioData in the input_buffer
                //do something...



            }

现在，解码后的音频数据存储在input_buffer中，这是我的困惑。如何使用AudioUnit播放音频data.And这是我的播放回调函数：

OSStatus playCallback(void                            *inRefCon,
                  AudioUnitRenderActionFlags      *ioActionFlags,
                  const AudioTimeStamp            *inTimeStamp,
                  UInt32                          inBusNumber,
                  UInt32                          inNumberFrames,
                  AudioBufferList                 *ioData){
AudioPlayer *THIS = (__bridge AudioPlayer *)inRefCon;
//How do I use the AudioUnit to play the audio stream from server?

return noErr;

}

希望有朋友解决我的困惑，如果你用过audioUnit，非常感谢！

Answer 1

这里有一些非常好的资源link

Answer 2

在您的 playCallback 中，您需要将音频复制到缓冲区 ioData。例如

memcpy (ioData->mBuffers[0].mData,  input_buffer + offset, numBytes );
// increase offset based on how many frames it requests.

输入变量inNumberFrames是准备好的帧数。这可能少于 input_buffer 中的帧数。所以你需要跟踪你的游戏位置。

我不知道你的音频流基本描述中指定的音频格式。您需要计算需要复制多少字节，考虑 mono/stereo、每个通道的字节数，当然还有 inNumberFrames。

如何使用 AudioUnit 播放来自服务器的音频流？

How do I use the AudioUnit to play the audio stream from server?

objective-c

stream

playback

audiounit

ios