input.c

VLC Player Source Code

                                                          "visualization",
                                                          psz_parser, true );
                    }
                }
            }

            /* failure */
            if ( p_filter->p_module == NULL )
            {
                msg_Err( p_aout, "cannot add user filter %s (skipped)",
                         psz_parser );

                vlc_object_detach( p_filter );
                vlc_object_release( p_filter );

                psz_parser = psz_next;
                continue;
            }

            /* complete the filter chain if necessary */
            if ( !AOUT_FMTS_IDENTICAL( &chain_input_format, &p_filter->input ) )
            {
                if ( aout_FiltersCreatePipeline( p_aout, p_input->pp_filters,
                                                 &p_input->i_nb_filters,
                                                 &chain_input_format,
                                                 &p_filter->input ) < 0 )
                {
                    msg_Err( p_aout, "cannot add user filter %s (skipped)",
                             psz_parser );

                    module_Unneed( p_filter, p_filter->p_module );
                    vlc_object_detach( p_filter );
                    vlc_object_release( p_filter );

                    psz_parser = psz_next;
                    continue;
                }
            }

            /* success */
            p_filter->b_continuity = false;
            p_input->pp_filters[p_input->i_nb_filters++] = p_filter;
            memcpy( &chain_input_format, &p_filter->output,
                    sizeof( audio_sample_format_t ) );

            /* next filter if any */
            psz_parser = psz_next;
        }
    }
    free( psz_filters );
    free( psz_visual );

    /* complete the filter chain if necessary */
    if ( !AOUT_FMTS_IDENTICAL( &chain_input_format, &chain_output_format ) )
    {
        if ( aout_FiltersCreatePipeline( p_aout, p_input->pp_filters,
                                         &p_input->i_nb_filters,
                                         &chain_input_format,
                                         &chain_output_format ) < 0 )
        {
            inputFailure( p_aout, p_input, "couldn't set an input pipeline" );
            return -1;
        }
    }

    /* Prepare hints for the buffer allocator. */
    p_input->input_alloc.i_alloc_type = AOUT_ALLOC_HEAP;
    p_input->input_alloc.i_bytes_per_sec = -1;

    /* Create resamplers. */
    if ( !AOUT_FMT_NON_LINEAR( &p_aout->mixer.mixer ) )
    {
        chain_output_format.i_rate = (__MAX(p_input->input.i_rate,
                                            p_aout->mixer.mixer.i_rate)
                                 * (100 + AOUT_MAX_RESAMPLING)) / 100;
        if ( chain_output_format.i_rate == p_aout->mixer.mixer.i_rate )
        {
            /* Just in case... */
            chain_output_format.i_rate++;
        }
        if ( aout_FiltersCreatePipeline( p_aout, p_input->pp_resamplers,
                                         &p_input->i_nb_resamplers,
                                         &chain_output_format,
                                         &p_aout->mixer.mixer ) < 0 )
        {
            inputFailure( p_aout, p_input, "couldn't set a resampler pipeline");
            return -1;
        }

        aout_FiltersHintBuffers( p_aout, p_input->pp_resamplers,
                                 p_input->i_nb_resamplers,
                                 &p_input->input_alloc );
        p_input->input_alloc.i_alloc_type = AOUT_ALLOC_HEAP;

        /* Setup the initial rate of the resampler */
        p_input->pp_resamplers[0]->input.i_rate = p_input->input.i_rate;
    }
    p_input->i_resampling_type = AOUT_RESAMPLING_NONE;

    p_input->p_playback_rate_filter = NULL;
    for( int i = 0; i < p_input->i_nb_filters; i++ )
    {
        aout_filter_t *p_filter = p_input->pp_filters[i];
        if( strcmp( "scaletempo", p_filter->psz_object_name ) == 0 )
        {
          p_input->p_playback_rate_filter = p_filter;
          break;
        }
    }
    if( ! p_input->p_playback_rate_filter && p_input->i_nb_resamplers > 0 )
    {
        p_input->p_playback_rate_filter = p_input->pp_resamplers[0];
    }

    aout_FiltersHintBuffers( p_aout, p_input->pp_filters,
                             p_input->i_nb_filters,
                             &p_input->input_alloc );
    p_input->input_alloc.i_alloc_type = AOUT_ALLOC_HEAP;

    /* i_bytes_per_sec is still == -1 if no filters */
    p_input->input_alloc.i_bytes_per_sec = __MAX(
                                    p_input->input_alloc.i_bytes_per_sec,
                                    (int)(p_input->input.i_bytes_per_frame
                                     * p_input->input.i_rate
                                     / p_input->input.i_frame_length) );

    ReplayGainSelect( p_aout, p_input );

    /* Success */
    p_input->b_error = false;
    p_input->b_restart = false;
    p_input->i_last_input_rate = INPUT_RATE_DEFAULT;

    return 0;
}

/*****************************************************************************
 * aout_InputDelete : delete an input
 *****************************************************************************
 * This function must be entered with the mixer lock.
 *****************************************************************************/
int aout_InputDelete( aout_instance_t * p_aout, aout_input_t * p_input )
{
    AOUT_ASSERT_MIXER_LOCKED;
    if ( p_input->b_error ) return 0;

    aout_FiltersDestroyPipeline( p_aout, p_input->pp_filters,
                                 p_input->i_nb_filters );
    p_input->i_nb_filters = 0;
    aout_FiltersDestroyPipeline( p_aout, p_input->pp_resamplers,
                                 p_input->i_nb_resamplers );
    p_input->i_nb_resamplers = 0;
    aout_FifoDestroy( p_aout, &p_input->fifo );

    return 0;
}

/*****************************************************************************
 * aout_InputPlay : play a buffer
 *****************************************************************************
 * This function must be entered with the input lock.
 *****************************************************************************/
/* XXX Do not activate it !! */
//#define AOUT_PROCESS_BEFORE_CHEKS
int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input,
                    aout_buffer_t * p_buffer, int i_input_rate )
{
    mtime_t start_date;
    AOUT_ASSERT_INPUT_LOCKED;

    if( p_input->b_restart )
    {
        aout_fifo_t fifo, dummy_fifo;
        uint8_t     *p_first_byte_to_mix;

        aout_lock_mixer( p_aout );
        aout_lock_input_fifos( p_aout );

        /* A little trick to avoid loosing our input fifo */
        aout_FifoInit( p_aout, &dummy_fifo, p_aout->mixer.mixer.i_rate );
        p_first_byte_to_mix = p_input->p_first_byte_to_mix;
        fifo = p_input->fifo;
        p_input->fifo = dummy_fifo;
        aout_InputDelete( p_aout, p_input );
        aout_InputNew( p_aout, p_input );
        p_input->p_first_byte_to_mix = p_first_byte_to_mix;
        p_input->fifo = fifo;

        aout_unlock_input_fifos( p_aout );
        aout_unlock_mixer( p_aout );
    }

    if( i_input_rate != INPUT_RATE_DEFAULT && p_input->p_playback_rate_filter == NULL )
    {
        inputDrop( p_aout, p_input, p_buffer );
        return 0;
    }

#ifdef AOUT_PROCESS_BEFORE_CHEKS
    /* Run pre-filters. */
    aout_FiltersPlay( p_aout, p_input->pp_filters, p_input->i_nb_filters,
                      &p_buffer );

    /* Actually run the resampler now. */
    if ( p_input->i_nb_resamplers > 0 )
    {
        const mtime_t i_date = p_buffer->start_date;
        aout_FiltersPlay( p_aout, p_input->pp_resamplers,
                          p_input->i_nb_resamplers,
                          &p_buffer );
    }

    if( p_buffer->i_nb_samples <= 0 )
    {
        aout_BufferFree( p_buffer );
        return 0;
    }
#endif

    /* Handle input rate change, but keep drift correction */
    if( i_input_rate != p_input->i_last_input_rate )
    {
        unsigned int * const pi_rate = &p_input->p_playback_rate_filter->input.i_rate;
#define F(r,ir) ( INPUT_RATE_DEFAULT * (r) / (ir) )
        const int i_delta = *pi_rate - F(p_input->input.i_rate,p_input->i_last_input_rate);
        *pi_rate = F(p_input->input.i_rate + i_delta, i_input_rate);
#undef F
        p_input->i_last_input_rate = i_input_rate;
    }

    /* We don't care if someone changes the start date behind our back after
     * this. We'll deal with that when pushing the buffer, and compensate
     * with the next incoming buffer. */
    aout_lock_input_fifos( p_aout );
    start_date = aout_FifoNextStart( p_aout, &p_input->fifo );
    aout_unlock_input_fifos( p_aout );

    if ( start_date != 0 && start_date < mdate() )
    {
        /* The decoder is _very_ late. This can only happen if the user
         * pauses the stream (or if the decoder is buggy, which cannot
         * happen :). */
        msg_Warn( p_aout, "computed PTS is out of range (%"PRId64"), "
                  "clearing out", mdate() - start_date );
        aout_lock_input_fifos( p_aout );
        aout_FifoSet( p_aout, &p_input->fifo, 0 );
        p_input->p_first_byte_to_mix = NULL;
        aout_unlock_input_fifos( p_aout );
        if ( p_input->i_resampling_type != AOUT_RESAMPLING_NONE )
            msg_Warn( p_aout, "timing screwed, stopping resampling" );
        inputResamplingStop( p_input );
        start_date = 0;
    }

    if ( p_buffer->start_date < mdate() + AOUT_MIN_PREPARE_TIME )
    {
        /* The decoder gives us f*cked up PTS. It's its business, but we
         * can't present it anyway, so drop the buffer. */
        msg_Warn( p_aout, "PTS is out of range (%"PRId64"), dropping buffer",
                  mdate() - p_buffer->start_date );

        inputDrop( p_aout, p_input, p_buffer );
        inputResamplingStop( p_input );
        return 0;
    }

    /* If the audio drift is too big then it's not worth trying to resample
     * the audio. */
    mtime_t i_pts_tolerance = 3 * AOUT_PTS_TOLERANCE * i_input_rate / INPUT_RATE_DEFAULT;
    if ( start_date != 0 &&
         ( start_date < p_buffer->start_date - i_pts_tolerance ) )
    {
        msg_Warn( p_aout, "audio drift is too big (%"PRId64"), clearing out",
                  start_date - p_buffer->start_date );
        aout_lock_input_fifos( p_aout );
        aout_FifoSet( p_aout, &p_input->fifo, 0 );
        p_input->p_first_byte_to_mix = NULL;
        aout_unlock_input_fifos( p_aout );
        if ( p_input->i_resampling_type != AOUT_RESAMPLING_NONE )
            msg_Warn( p_aout, "timing screwed, stopping resampling" );
        inputResamplingStop( p_input );
        start_date = 0;
    }
    else if ( start_date != 0 &&
              ( start_date > p_buffer->start_date + i_pts_tolerance) )
    {
        msg_Warn( p_aout, "audio drift is too big (%"PRId64"), dropping buffer",
                  start_date - p_buffer->start_date );
        inputDrop( p_aout, p_input, p_buffer );
        return 0;
    }

    if ( start_date == 0 ) start_date = p_buffer->start_date;

#ifndef AOUT_PROCESS_BEFORE_CHEKS
    /* Run pre-filters. */
    aout_FiltersPlay( p_aout, p_input->pp_filters, p_input->i_nb_filters,
                      &p_buffer );
#endif

    /* Run the resampler if needed.
     * We first need to calculate the output rate of this resampler. */
    if ( ( p_input->i_resampling_type == AOUT_RESAMPLING_NONE ) &&
         ( start_date < p_buffer->start_date - AOUT_PTS_TOLERANCE
           || start_date > p_buffer->start_date + AOUT_PTS_TOLERANCE ) &&
         p_input->i_nb_resamplers > 0 )
    {
        /* Can happen in several circumstances :
         * 1. A problem at the input (clock drift)
         * 2. A small pause triggered by the user