input.c

uclinux 下的vlc播放器源代码

    aout_FifoDestroy( p_aout, &p_input->fifo );

    return 0;
}

/*****************************************************************************
 * aout_InputPlay : play a buffer
 *****************************************************************************
 * This function must be entered with the input lock.
 *****************************************************************************/
int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input,
                    aout_buffer_t * p_buffer )
{
    mtime_t start_date;

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

        vlc_mutex_lock( &p_aout->mixer_lock );

        /* 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;

        vlc_mutex_unlock( &p_aout->mixer_lock );
    }

    /* 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. */
    vlc_mutex_lock( &p_aout->input_fifos_lock );
    start_date = aout_FifoNextStart( p_aout, &p_input->fifo );
    vlc_mutex_unlock( &p_aout->input_fifos_lock );

    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 ("I64Fd"), "
                  "clearing out", mdate() - start_date );
        vlc_mutex_lock( &p_aout->input_fifos_lock );
        aout_FifoSet( p_aout, &p_input->fifo, 0 );
        p_input->p_first_byte_to_mix = NULL;
        vlc_mutex_unlock( &p_aout->input_fifos_lock );
        if ( p_input->i_resampling_type != AOUT_RESAMPLING_NONE )
            msg_Warn( p_aout, "timing screwed, stopping resampling" );
        p_input->i_resampling_type = AOUT_RESAMPLING_NONE;
        if ( p_input->i_nb_resamplers != 0 )
        {
            p_input->pp_resamplers[0]->input.i_rate = p_input->input.i_rate;
            p_input->pp_resamplers[0]->b_continuity = VLC_FALSE;
        }
        start_date = 0;
        if( p_input->p_input_thread )
        {
            stats_UpdateInteger( p_input->p_input_thread, STATS_LOST_ABUFFERS, 1,
                                 NULL );
        }
    }

    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 ("I64Fd"), dropping buffer",
                  mdate() - p_buffer->start_date );
        if( p_input->p_input_thread )
        {
            stats_UpdateInteger( p_input->p_input_thread, STATS_LOST_ABUFFERS,
                                 1, NULL );
        }
        aout_BufferFree( p_buffer );
        p_input->i_resampling_type = AOUT_RESAMPLING_NONE;
        if ( p_input->i_nb_resamplers != 0 )
        {
            p_input->pp_resamplers[0]->input.i_rate = p_input->input.i_rate;
            p_input->pp_resamplers[0]->b_continuity = VLC_FALSE;
        }
        return 0;
    }

    /* If the audio drift is too big then it's not worth trying to resample
     * the audio. */
    if ( start_date != 0 &&
         ( start_date < p_buffer->start_date - 3 * AOUT_PTS_TOLERANCE ) )
    {
        msg_Warn( p_aout, "audio drift is too big ("I64Fd"), clearing out",
                  start_date - p_buffer->start_date );
        vlc_mutex_lock( &p_aout->input_fifos_lock );
        aout_FifoSet( p_aout, &p_input->fifo, 0 );
        p_input->p_first_byte_to_mix = NULL;
        vlc_mutex_unlock( &p_aout->input_fifos_lock );
        if ( p_input->i_resampling_type != AOUT_RESAMPLING_NONE )
            msg_Warn( p_aout, "timing screwed, stopping resampling" );
        p_input->i_resampling_type = AOUT_RESAMPLING_NONE;
        if ( p_input->i_nb_resamplers != 0 )
        {
            p_input->pp_resamplers[0]->input.i_rate = p_input->input.i_rate;
            p_input->pp_resamplers[0]->b_continuity = VLC_FALSE;
        }
        start_date = 0;
    }
    else if ( start_date != 0 &&
              ( start_date > p_buffer->start_date + 3 * AOUT_PTS_TOLERANCE ) )
    {
        msg_Warn( p_aout, "audio drift is too big ("I64Fd"), dropping buffer",
                  start_date - p_buffer->start_date );
        aout_BufferFree( p_buffer );
        if( p_input->p_input_thread )
        {
            stats_UpdateInteger( p_input->p_input_thread, STATS_LOST_ABUFFERS,
                                 1, NULL );
        }
        return 0;
    }

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

    /* Run pre-filters. */

    aout_FiltersPlay( p_aout, p_input->pp_filters, p_input->i_nb_filters,
                      &p_buffer );

    /* 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
         * 3. Some delay in the output stage, causing a loss of lip
         *    synchronization
         * Solution : resample the buffer to avoid a scratch.
         */
        mtime_t drift = p_buffer->start_date - start_date;

        p_input->i_resamp_start_date = mdate();
        p_input->i_resamp_start_drift = (int)drift;

        if ( drift > 0 )
            p_input->i_resampling_type = AOUT_RESAMPLING_DOWN;
        else
            p_input->i_resampling_type = AOUT_RESAMPLING_UP;

        msg_Warn( p_aout, "buffer is "I64Fd" %s, triggering %ssampling",
                          drift > 0 ? drift : -drift,
                          drift > 0 ? "in advance" : "late",
                          drift > 0 ? "down" : "up");
    }

    if ( p_input->i_resampling_type != AOUT_RESAMPLING_NONE )
    {
        /* Resampling has been triggered previously (because of dates
         * mismatch). We want the resampling to happen progressively so
         * it isn't too audible to the listener. */

        if( p_input->i_resampling_type == AOUT_RESAMPLING_UP )
        {
            p_input->pp_resamplers[0]->input.i_rate += 2; /* Hz */
        }
        else
        {
            p_input->pp_resamplers[0]->input.i_rate -= 2; /* Hz */
        }

        /* Check if everything is back to normal, in which case we can stop the
         * resampling */
        if( p_input->pp_resamplers[0]->input.i_rate ==
              p_input->input.i_rate )
        {
            p_input->i_resampling_type = AOUT_RESAMPLING_NONE;
            msg_Warn( p_aout, "resampling stopped after "I64Fi" usec "
                      "(drift: "I64Fi")",
                      mdate() - p_input->i_resamp_start_date,
                      p_buffer->start_date - start_date);
        }
        else if( abs( (int)(p_buffer->start_date - start_date) ) <
                 abs( p_input->i_resamp_start_drift ) / 2 )
        {
            /* if we reduced the drift from half, then it is time to switch
             * back the resampling direction. */
            if( p_input->i_resampling_type == AOUT_RESAMPLING_UP )
                p_input->i_resampling_type = AOUT_RESAMPLING_DOWN;
            else
                p_input->i_resampling_type = AOUT_RESAMPLING_UP;
            p_input->i_resamp_start_drift = 0;
        }
        else if( p_input->i_resamp_start_drift &&
                 ( abs( (int)(p_buffer->start_date - start_date) ) >
                   abs( p_input->i_resamp_start_drift ) * 3 / 2 ) )
        {
            /* If the drift is increasing and not decreasing, than something
             * is bad. We'd better stop the resampling right now. */
            msg_Warn( p_aout, "timing screwed, stopping resampling" );
            p_input->i_resampling_type = AOUT_RESAMPLING_NONE;
            p_input->pp_resamplers[0]->input.i_rate = p_input->input.i_rate;
        }
    }

    /* Adding the start date will be managed by aout_FifoPush(). */
    p_buffer->end_date = start_date +
        (p_buffer->end_date - p_buffer->start_date);
    p_buffer->start_date = start_date;

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

    vlc_mutex_lock( &p_aout->input_fifos_lock );
    aout_FifoPush( p_aout, &p_input->fifo, p_buffer );
    vlc_mutex_unlock( &p_aout->input_fifos_lock );

    return 0;
}

/*****************************************************************************
 * static functions
 *****************************************************************************/

static void inputFailure( aout_instance_t * p_aout, aout_input_t * p_input,
                          char * psz_error_message )
{
    /* error message */
    msg_Err( p_aout, "couldn't set an input pipeline" );

    /* clean up */
    aout_FiltersDestroyPipeline( p_aout, p_input->pp_filters,
                                 p_input->i_nb_filters );
    aout_FiltersDestroyPipeline( p_aout, p_input->pp_resamplers,
                                 p_input->i_nb_resamplers );
    aout_FifoDestroy( p_aout, &p_input->fifo );
    var_Destroy( p_aout, "visual" );
    var_Destroy( p_aout, "equalizer" );
    var_Destroy( p_aout, "audio-filter" );
    var_Destroy( p_aout, "audio-visual" );

    /* error flag */
    p_input->b_error = 1;
}

static int ChangeFiltersString( aout_instance_t * p_aout, char* psz_variable,
                                 char *psz_name, vlc_bool_t b_add )
{
    vlc_value_t val;
    char *psz_parser;

    var_Get( p_aout, psz_variable, &val );

    if( !val.psz_string ) val.psz_string = strdup("");

    psz_parser = strstr( val.psz_string, psz_name );

    if( b_add )
    {
        if( !psz_parser )
        {
            psz_parser = val.psz_string;
            asprintf( &val.psz_string, (*val.psz_string) ? "%s:%s" : "%s%s",
                      val.psz_string, psz_name );
            free( psz_parser );
        }
        else
        {
            return 0;
        }
    }
    else
    {
        if( psz_parser )
        {
            memmove( psz_parser, psz_parser + strlen(psz_name) +
                     (*(psz_parser + strlen(psz_name)) == ':' ? 1 : 0 ),
                     strlen(psz_parser + strlen(psz_name)) + 1 );
        }
        else
        {
            free( val.psz_string );
            return 0;
        }
    }

    var_Set( p_aout, psz_variable, val );
    free( val.psz_string );
    return 1;
}

static int VisualizationCallback( vlc_object_t *p_this, char const *psz_cmd,
                       vlc_value_t oldval, vlc_value_t newval, void *p_data )
{
    aout_instance_t *p_aout = (aout_instance_t *)p_this;
    char *psz_mode = newval.psz_string;
    vlc_value_t val;
    int i;

    if( !psz_mode || !*psz_mode )
    {
        ChangeFiltersString( p_aout, "audio-visual", "goom", VLC_FALSE );
        ChangeFiltersString( p_aout, "audio-visual", "visual", VLC_FALSE );
        ChangeFiltersString( p_aout, "audio-visual", "galaktos", VLC_FALSE );
    }
    else
    {
        if( !strcmp( "goom", psz_mode ) )
        {
            ChangeFiltersString( p_aout, "audio-visual", "visual", VLC_FALSE );
            ChangeFiltersString( p_aout, "audio-visual", "goom", VLC_TRUE );
            ChangeFiltersString( p_aout, "audio-visual", "galaktos", VLC_FALSE);
        }
        else if( !strcmp( "galaktos", psz_mode ) )
        {
            ChangeFiltersString( p_aout, "audio-visual", "visual", VLC_FALSE );
            ChangeFiltersString( p_aout, "audio-visual", "goom", VLC_FALSE );
            ChangeFiltersString( p_aout, "audio-visual", "galaktos", VLC_TRUE );
        }
        else
        {
            val.psz_string = psz_mode;
            var_Create( p_aout, "effect-list", VLC_VAR_STRING );
            var_Set( p_aout, "effect-list", val );

            ChangeFiltersString( p_aout, "audio-visual", "goom", VLC_FALSE );
            ChangeFiltersString( p_aout, "audio-visual", "visual", VLC_TRUE );
            ChangeFiltersString( p_aout, "audio-visual", "galaktos", VLC_FALSE);
        }
    }

    /* That sucks */
    for( i = 0; i < p_aout->i_nb_inputs; i++ )
    {
        p_aout->pp_inputs[i]->b_restart = VLC_TRUE;
    }

    return VLC_SUCCESS;
}

static int EqualizerCallback( vlc_object_t *p_this, char const *psz_cmd,
                       vlc_value_t oldval, vlc_value_t newval, void *p_data )
{
    aout_instance_t *p_aout = (aout_instance_t *)p_this;
    char *psz_mode = newval.psz_string;
    vlc_value_t val;
    int i;
    int i_ret;

    if( !psz_mode || !*psz_mode )
    {
        i_ret = ChangeFiltersString( p_aout, "audio-filter", "equalizer",
                                     VLC_FALSE );
    }
    else
    {
        val.psz_string = psz_mode;
        var_Create( p_aout, "equalizer-preset", VLC_VAR_STRING );
        var_Set( p_aout, "equalizer-preset", val );
        i_ret = ChangeFiltersString( p_aout, "audio-filter", "equalizer",
                                     VLC_TRUE );

    }

    /* That sucks */
    if( i_ret == 1 )
    {
        for( i = 0; i < p_aout->i_nb_inputs; i++ )
        {
            p_aout->pp_inputs[i]->b_restart = VLC_TRUE;
        }
    }

    return VLC_SUCCESS;
}