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  • SDL Audio - Plays only Static Noise

    30 avril 2016, par bcpermafrost

    Im having an issue with playing audio.

    Im new to the SDL World of things so im learning from a tutorial.

    http://dranger.com/ffmpeg/tutorial03.html

    As far as audio goes, i have exactly what he put down and didnt get the result he says I should get. In the end of the lesson he specifies that the audio should play normally. However all i get is excessively loud static noise. This leads me to believe that the packets arent being read correctly. However I have no idea how to debug or look for the issue.

    Here is my main loop for parsing the packets :

     while (av_read_frame(pFormatCtx, &packet) >= 0) {
    

    
         if (packet.stream_index == videoStream) {
    
             avcodec_decode_video2(pCodecCtx, pFrame, &frameFinished, &packet);
    

    
             if (frameFinished){
    

    
                 AVPicture pict;
    

    
                 pict.data[0] = yPlane;
    
                 pict.data[1] = uPlane;
    
                 pict.data[2] = vPlane;
    
                 pict.linesize[0] = pCodecCtx->width;
    
                 pict.linesize[1] = uvPitch;
    
                 pict.linesize[2] = uvPitch;
    

    
                 sws_scale(sws_ctx,
    
                     pFrame->data, pFrame->linesize,
    
                     0, pCodecCtx->height,
    
                     pict.data, pict.linesize);
    

    
                 //SDL_UnlockTexture(bmp);
    

    
                 SDL_UpdateYUVTexture(bmp, 0, 
    
                     yPlane, pCodecCtx->width, 
    
                     uPlane, uvPitch, 
    
                     vPlane, uvPitch);
    

    

    
                 SDL_RenderClear(renderer);
    
                 SDL_RenderCopy(renderer, bmp, NULL, NULL);
    
                 SDL_RenderPresent(renderer);
    

    

    
                 av_free_packet(&packet);
    

    

    
             }
    

    
         }
    
         else if (packet.stream_index == audioStream) { 
    
             packet_queue_put(&audioq, &packet);
    

    
         }
    
         else
    
             av_free_packet(&packet);
    

    

    

    
         SDL_PollEvent(&event);
    

    
         switch (event.type) {
    
         case SDL_QUIT:
    
             quit = 1;
    
             SDL_DestroyTexture(bmp);
    
             SDL_DestroyRenderer(renderer);
    
             SDL_DestroyWindow(screen);
    
             SDL_Quit();
    
             exit(0);
    
             break;
    
         default:
    
             break;
    

    
         }
    

    
     }

    this is my initialization of the audio device :

    aCodecCtxOrig = pFormatCtx->streams[audioStream]->codec;
    
    aCodec = avcodec_find_decoder(aCodecCtxOrig->codec_id);
    
    if (!aCodec) {
    
        fprintf(stderr, "Unsupported codec!\n");
    
        return -1;
    
    }
    

    
    // Copy context
    
    aCodecCtx = avcodec_alloc_context3(aCodec);
    
    if (avcodec_copy_context(aCodecCtx, aCodecCtxOrig) != 0) {
    
        fprintf(stderr, "Couldn't copy codec context");
    
        return -1; // Error copying codec context
    
    }
    

    

    
    wanted_spec.freq = aCodecCtx->sample_rate;
    
    wanted_spec.format = AUDIO_U16SYS;
    
    wanted_spec.channels = aCodecCtx->channels;
    
    wanted_spec.silence = 0;
    
    wanted_spec.samples = SDL_AUDIO_BUFFER_SIZE;
    
    wanted_spec.callback = audio_callback;
    
    wanted_spec.userdata = aCodecCtx;
    

    

    
    if (SDL_OpenAudio( &wanted_spec, &spec) < 0) {
    
        fprintf(stderr, "SDL_OpenAudio: %s\n", SDL_GetError());
    
        return -1;
    
    }
    

    
    avcodec_open2(aCodecCtx, aCodec, NULL);
    

    
    // audio_st = pFormatCtx->streams[index]
    
    packet_queue_init(&audioq);
    
    SDL_PauseAudio(0);

    The Call back (same as the tutorial) :|

    void audio_callback(void *userdata, Uint8 *stream, int len) {
    

    
    AVCodecContext *aCodecCtx = (AVCodecContext *)userdata;
    
    int len1, audio_size;
    

    
    static uint8_t audio_buf[(MAX_AUDIO_FRAME_SIZE * 3) / 2];
    
    static unsigned int audio_buf_size = 0;
    
    static unsigned int audio_buf_index = 0;
    

    
    while (len > 0) {
    
        if (audio_buf_index >= audio_buf_size) {
    
            /* We have already sent all our data; get more */
    
            audio_size = audio_decode_frame(aCodecCtx, audio_buf, sizeof(audio_buf));
    
            if (audio_size < 0) {
    
                /* If error, output silence */
    
                audio_buf_size = 1024; // arbitrary?
    
                memset(audio_buf, 0, audio_buf_size);
    
            }
    
            else {
    
                audio_buf_size = audio_size;
    
            }
    
            audio_buf_index = 0;
    
        }
    
        len1 = audio_buf_size - audio_buf_index;
    
        if (len1 > len)
    
            len1 = len;
    
        memcpy(stream, (uint8_t *)audio_buf + audio_buf_index, len1);
    
        len -= len1;
    
        stream += len1;
    
        audio_buf_index += len1;
    
    }
    
}

  • SDL Audio - Plays only Static Noise

    19 août 2019, par bcpermafrost

    Im having an issue with playing audio.

    Im new to the SDL World of things so im learning from a tutorial.

    http://dranger.com/ffmpeg/tutorial03.html

    As far as audio goes, i have exactly what he put down and didnt get the result he says I should get. In the end of the lesson he specifies that the audio should play normally. However all i get is excessively loud static noise. This leads me to believe that the packets arent being read correctly. However I have no idea how to debug or look for the issue.

    Here is my main loop for parsing the packets :

     while (av_read_frame(pFormatCtx, &packet) >= 0) {
    

    
         if (packet.stream_index == videoStream) {
    
             avcodec_decode_video2(pCodecCtx, pFrame, &frameFinished, &packet);
    

    
             if (frameFinished){
    

    
                 AVPicture pict;
    

    
                 pict.data[0] = yPlane;
    
                 pict.data[1] = uPlane;
    
                 pict.data[2] = vPlane;
    
                 pict.linesize[0] = pCodecCtx->width;
    
                 pict.linesize[1] = uvPitch;
    
                 pict.linesize[2] = uvPitch;
    

    
                 sws_scale(sws_ctx,
    
                     pFrame->data, pFrame->linesize,
    
                     0, pCodecCtx->height,
    
                     pict.data, pict.linesize);
    

    
                 //SDL_UnlockTexture(bmp);
    

    
                 SDL_UpdateYUVTexture(bmp, 0, 
    
                     yPlane, pCodecCtx->width, 
    
                     uPlane, uvPitch, 
    
                     vPlane, uvPitch);
    

    

    
                 SDL_RenderClear(renderer);
    
                 SDL_RenderCopy(renderer, bmp, NULL, NULL);
    
                 SDL_RenderPresent(renderer);
    

    

    
                 av_free_packet(&packet);
    

    

    
             }
    

    
         }
    
         else if (packet.stream_index == audioStream) { 
    
             packet_queue_put(&audioq, &packet);
    

    
         }
    
         else
    
             av_free_packet(&packet);
    

    

    

    
         SDL_PollEvent(&event);
    

    
         switch (event.type) {
    
         case SDL_QUIT:
    
             quit = 1;
    
             SDL_DestroyTexture(bmp);
    
             SDL_DestroyRenderer(renderer);
    
             SDL_DestroyWindow(screen);
    
             SDL_Quit();
    
             exit(0);
    
             break;
    
         default:
    
             break;
    

    
         }
    

    
     }

    this is my initialization of the audio device :

    aCodecCtxOrig = pFormatCtx->streams[audioStream]->codec;
    
    aCodec = avcodec_find_decoder(aCodecCtxOrig->codec_id);
    
    if (!aCodec) {
    
        fprintf(stderr, "Unsupported codec!\n");
    
        return -1;
    
    }
    

    
    // Copy context
    
    aCodecCtx = avcodec_alloc_context3(aCodec);
    
    if (avcodec_copy_context(aCodecCtx, aCodecCtxOrig) != 0) {
    
        fprintf(stderr, "Couldn't copy codec context");
    
        return -1; // Error copying codec context
    
    }
    

    

    
    wanted_spec.freq = aCodecCtx->sample_rate;
    
    wanted_spec.format = AUDIO_U16SYS;
    
    wanted_spec.channels = aCodecCtx->channels;
    
    wanted_spec.silence = 0;
    
    wanted_spec.samples = SDL_AUDIO_BUFFER_SIZE;
    
    wanted_spec.callback = audio_callback;
    
    wanted_spec.userdata = aCodecCtx;
    

    

    
    if (SDL_OpenAudio( &wanted_spec, &spec) < 0) {
    
        fprintf(stderr, "SDL_OpenAudio: %s\n", SDL_GetError());
    
        return -1;
    
    }
    

    
    avcodec_open2(aCodecCtx, aCodec, NULL);
    

    
    // audio_st = pFormatCtx->streams[index]
    
    packet_queue_init(&audioq);
    
    SDL_PauseAudio(0);

    The Call back (same as the tutorial) :|

    void audio_callback(void *userdata, Uint8 *stream, int len) {
    

    
    AVCodecContext *aCodecCtx = (AVCodecContext *)userdata;
    
    int len1, audio_size;
    

    
    static uint8_t audio_buf[(MAX_AUDIO_FRAME_SIZE * 3) / 2];
    
    static unsigned int audio_buf_size = 0;
    
    static unsigned int audio_buf_index = 0;
    

    
    while (len > 0) {
    
        if (audio_buf_index >= audio_buf_size) {
    
            /* We have already sent all our data; get more */
    
            audio_size = audio_decode_frame(aCodecCtx, audio_buf, sizeof(audio_buf));
    
            if (audio_size < 0) {
    
                /* If error, output silence */
    
                audio_buf_size = 1024; // arbitrary?
    
                memset(audio_buf, 0, audio_buf_size);
    
            }
    
            else {
    
                audio_buf_size = audio_size;
    
            }
    
            audio_buf_index = 0;
    
        }
    
        len1 = audio_buf_size - audio_buf_index;
    
        if (len1 > len)
    
            len1 = len;
    
        memcpy(stream, (uint8_t *)audio_buf + audio_buf_index, len1);
    
        len -= len1;
    
        stream += len1;
    
        audio_buf_index += len1;
    
    }
    
}

  • lavu/tx : implement aarch64 NEON SIMD FFT

    3 février 2022, par Lynne
    lavu/tx : implement aarch64 NEON SIMD FFT

    The fastest fast Fourier transform in not just the west, but the world,
    
now for the most popular toy ISA.
    

    On a high level, it follows the design of the AVX2 version closely,
    
with the exception that the input is slightly less permuted as we don't have
    
to do lane switching with the input on double 4pt and 8pt.
    

    On a low level, the lack of subadd/addsub instructions REALLY penalizes
    
any attempt at writing an FFT. That single register matters a lot,
    
and reloading it simply takes unacceptably long.
    
In x86 land, vendors would've noticed developers need this.
    
In ARM land, you get a badly designed complex multiplication instruction
    
we cannot use, that's not present on 95% of devices. Because only
    
compilers matter, right ?
    

    Future optimization options are very few, perhaps better register
    
management to use more ld1/st1s.
    

    All timings below are in cycles :
    
A53 :
    
Length | C           | New (lavu)  | Old (lavc)  | FFTW
    

    

     |-------------|-------------|-------------|-----
    
4      |         842 | 420         | 1210        | 1460
    
8      |        1538 | 1020        | 1850        | 2520
    
16     |        3717 | 1900        | 3700        | 3990
    
32     |        9156 | 4070        | 8289        | 8860
    
64     |       21160 | 9931        | 18600       | 19625
    
128    |       49180 | 23278       | 41922       | 41922
    
256    |      112073 | 53876       | 93202       | 101092
    
512    |      252864 | 122884      | 205897      | 207868
    
1024   |      560512 | 278322      | 458071      | 453053
    
2048   |     1295402 | 775835      | 1038205     | 1020265
    
4096   |     3281263 | 2021221     | 2409718     | 2577554
    
8192   |     8577845 | 4780526     | 5673041     | 6802722
    

    Apple M1
    
New  - Total for len 512 reps 2097152 = 1.459141 s
    
Old  - Total for len 512 reps 2097152 = 2.251344 s
    
FFTW - Total for len 512 reps 2097152 = 1.868429 s
    

    New  - Total for len 1024 reps 4194304 = 6.490080 s
    
Old  - Total for len 1024 reps 4194304 = 9.604949 s
    
FFTW - Total for len 1024 reps 4194304 = 7.889281 s
    

    New  - Total for len 16384 reps 262144 = 10.374001 s
    
Old  - Total for len 16384 reps 262144 = 15.266713 s
    
FFTW - Total for len 16384 reps 262144 = 12.341745 s
    

    New  - Total for len 65536 reps 8192 = 1.769812 s
    
Old  - Total for len 65536 reps 8192 = 4.209413 s
    
FFTW - Total for len 65536 reps 8192 = 3.012365 s
    

    New  - Total for len 131072 reps 4096 = 1.942836 s
    
Old  - Segfaults
    
FFTW - Total for len 131072 reps 4096 = 3.713713 s
    

    Thanks to wbs for some simplifications, assembler fixes and a review
    
and to jannau for giving it a look.
    • [DH] libavutil/aarch64/Makefile
    • [DH] libavutil/aarch64/tx_float_init.c
    • [DH] libavutil/aarch64/tx_float_neon.S
    • [DH] libavutil/tx.c
    • [DH] libavutil/tx_priv.h

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