diff --git a/modules/audio_filter/converter/mono.c b/modules/audio_filter/converter/mono.c index b3ae50dd7b..3bea3beee8 100644 --- a/modules/audio_filter/converter/mono.c +++ b/modules/audio_filter/converter/mono.c @@ -51,19 +51,15 @@ static int OpenFilter ( vlc_object_t * ); static void CloseFilter ( vlc_object_t * ); static block_t *Convert( filter_t *p_filter, block_t *p_block ); -static void stereo_mono_downmix( aout_instance_t *, aout_filter_t *, - aout_buffer_t *, aout_buffer_t * ); -static unsigned int stereo_to_mono( int16_t *, int16_t *, unsigned int ); -static void silence_channel( aout_instance_t *, aout_filter_t *, - aout_buffer_t *, aout_buffer_t * ); +static unsigned int stereo_to_mono( aout_instance_t *, aout_filter_t *, + aout_buffer_t *, aout_buffer_t * ); /***************************************************************************** * Local structures *****************************************************************************/ struct filter_sys_t { - vlc_bool_t b_block_channel; int i_nb_channels; /* number of float32 per sample */ unsigned int i_channel_selected; int i_bitspersample; @@ -118,7 +114,7 @@ static int OpenFilter( vlc_object_t *p_this ) if( (p_filter->fmt_in.i_codec != AOUT_FMT_S16_NE) || (p_filter->fmt_out.i_codec != AOUT_FMT_S16_NE) ) { - msg_Err( p_this, "invalid format" ); + msg_Err( p_this, "filter discarded (invalid format)" ); return -1; } @@ -144,23 +140,20 @@ static int OpenFilter( vlc_object_t *p_this ) p_sys->i_channel_selected = (unsigned int) var_GetInteger( p_this, MONO_CFG "mono-channel" ); - /* temporarily force channel silence */ - p_sys->b_block_channel = VLC_TRUE; - if( p_sys->b_block_channel ) - { - p_filter->fmt_out.audio.i_physical_channels = +#if 0 + p_filter->fmt_out.audio.i_physical_channels = AOUT_CHAN_CENTER; +#endif + p_filter->fmt_out.audio.i_physical_channels = (AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT); - } - else - p_filter->fmt_out.audio.i_physical_channels = AOUT_CHAN_CENTER; - p_filter->pf_audio_filter = Convert; p_filter->fmt_out.audio.i_rate = p_filter->fmt_in.audio.i_rate; p_filter->fmt_out.audio.i_format = p_filter->fmt_out.i_codec; - p_sys->i_nb_channels = aout_FormatNbChannels( &(p_filter->fmt_out.audio) ); + p_sys->i_nb_channels = aout_FormatNbChannels( &(p_filter->fmt_in.audio) ); p_sys->i_bitspersample = p_filter->fmt_out.audio.i_bitspersample; + p_filter->pf_audio_filter = Convert; + msg_Dbg( p_this, "%4.4s->%4.4s, channels %d->%d, bits per sample: %i->%i", (char *)&p_filter->fmt_in.i_codec, (char *)&p_filter->fmt_out.i_codec, @@ -193,6 +186,7 @@ static block_t *Convert( filter_t *p_filter, block_t *p_block ) aout_buffer_t in_buf, out_buf; block_t *p_out = NULL; int i_out_size; + unsigned int i_samples; if( !p_block || !p_block->i_samples ) { @@ -202,7 +196,7 @@ static block_t *Convert( filter_t *p_filter, block_t *p_block ) } i_out_size = p_block->i_samples * p_filter->p_sys->i_bitspersample/8 * - p_filter->p_sys->i_nb_channels; + aout_FormatNbChannels( &(p_filter->fmt_out.audio) ); p_out = p_filter->pf_audio_buffer_new( p_filter, i_out_size ); if( !p_out ) @@ -212,7 +206,8 @@ static block_t *Convert( filter_t *p_filter, block_t *p_block ) return NULL; } - p_out->i_samples = p_block->i_samples; + p_out->i_samples = (p_block->i_samples / p_filter->p_sys->i_nb_channels) * + aout_FormatNbChannels( &(p_filter->fmt_out.audio) ); p_out->i_dts = p_block->i_dts; p_out->i_pts = p_block->i_pts; p_out->i_length = p_block->i_length; @@ -228,13 +223,12 @@ static block_t *Convert( filter_t *p_filter, block_t *p_block ) in_buf.i_nb_samples = p_block->i_samples; #if 0 - if( in_buf.i_nb_bytes != (p_filter->p_sys->i_bitspersample/8) * in_buf.i_nb_samples ) + unsigned int i_in_size = in_buf.i_nb_samples * (p_filter->p_sys->i_bitspersample/8) * + aout_FormatNbChannels( &(p_filter->fmt_in.audio) ); + if( (in_buf.i_nb_bytes != i_in_size) && ((i_in_size % 32) != 0) ) /* is it word aligned?? */ { - msg_Err( p_filter, "input buffer is not alligned" ); -/* if( in_buf.i_nb_bytes > (p_filter->p_sys->i_bitspersample/8) * in_buf.i_nb_samples) - in_buf.i_nb_bytes = (p_filter->p_sys->i_bitspersample/8) * in_buf.i_nb_samples; - else - //in_buf*/ + msg_Err( p_filter, "input buffer is not word aligned" ); + /* Fix output buffer to be word aligned */ } #endif @@ -242,7 +236,8 @@ static block_t *Convert( filter_t *p_filter, block_t *p_block ) out_buf.i_nb_bytes = p_out->i_buffer; out_buf.i_nb_samples = p_out->i_samples; - stereo_mono_downmix( (aout_instance_t *)p_filter, &aout_filter, &in_buf, &out_buf ); + i_samples = stereo_to_mono( (aout_instance_t *)p_filter, &aout_filter, + &out_buf, &in_buf ); p_out->i_buffer = out_buf.i_nb_bytes; p_out->i_samples = out_buf.i_nb_samples; @@ -251,77 +246,29 @@ static block_t *Convert( filter_t *p_filter, block_t *p_block ) return p_out; } -static void stereo_mono_downmix( aout_instance_t * p_aout, aout_filter_t * p_filter, - aout_buffer_t * p_in_buf, aout_buffer_t * p_out_buf ) -{ - filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys; - - if( p_sys->b_block_channel ) - { - silence_channel( p_aout, p_filter, p_out_buf, p_in_buf ); - } - else - { - unsigned int i_samples; - - i_samples = stereo_to_mono( (int16_t *)p_out_buf->p_buffer, (int16_t *)p_in_buf->p_buffer, - p_out_buf->i_nb_samples ); - } - - p_out_buf->i_nb_samples = p_in_buf->i_nb_samples; -} - -/* silence_channel - play silence on all channels except the selected one. +/* stereo_to_mono - mix 2 channels (left,right) into one and play silence on + * all other channels. */ -static void silence_channel( aout_instance_t * p_aout, aout_filter_t * p_filter, - aout_buffer_t *p_out_buf, aout_buffer_t *p_in_buf ) +static unsigned int stereo_to_mono( aout_instance_t * p_aout, aout_filter_t *p_filter, + aout_buffer_t *p_output, aout_buffer_t *p_input ) { filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys; - unsigned int n = 0; int16_t *p_in, *p_out; + unsigned int n; - p_in = (int16_t *)p_in_buf->p_buffer; - p_out = (int16_t *)p_out_buf->p_buffer; + p_in = (int16_t *) p_input->p_buffer; + p_out = (int16_t *) p_output->p_buffer; - for( n = 0; n < p_in_buf->i_nb_samples * p_sys->i_nb_channels; n++ ) + for( n = 0; n < (p_input->i_nb_samples * p_sys->i_nb_channels); n++ ) { if( (n%p_sys->i_nb_channels) == p_sys->i_channel_selected ) { - p_out[n] = p_in[n]; + p_out[n] = (p_in[n] + p_in[n+1]) >> 1; } else { p_out[n] = 0x0; } } -} - -/* stereo_to_mono() function is from ffmpeg file libavcodec/resample.c - * Copyright (c) 2000 Fabrice Bellard. - */ -static unsigned int stereo_to_mono( int16_t *p_output, int16_t *p_input, - unsigned int i_samples ) -{ - int16_t *p, *q; - unsigned int n = i_samples; - - p = p_input; - q = p_output; - - while (n >= 4) { - q[0] = (p[0] + p[1]) >> 1; - q[1] = (p[2] + p[3]) >> 1; - q[2] = (p[4] + p[5]) >> 1; - q[3] = (p[6] + p[7]) >> 1; - q += 4; - p += 8; - n -= 4; - } - while (n > 0) { - q[0] = (p[0] + p[1]) >> 1; - q++; - p += 2; - n--; - } return n; }