[vlc-devel] [PATCH] Add Scaletempo audio filter
Rov Juvano
rovjuvano at users.sourceforge.net
Sun Jun 22 18:09:27 CEST 2008
Scaletempo maintains the audio pitch when playback rate != 1.0 (i.e.
no chipmunk effect). This fixes the pitch scaling caused by using the
resampler to handle playback rate.
Ported from GStreamer. Inspired by SoundTouch library by Olli Parviainen.
---
configure.ac | 1 +
include/vlc_aout.h | 2 +
modules/audio_filter/Modules.am | 1 +
modules/audio_filter/scaletempo.c | 512 +++++++++++++++++++++++++++++++++++++
src/audio_output/input.c | 38 +++-
5 files changed, 546 insertions(+), 8 deletions(-)
create mode 100644 modules/audio_filter/scaletempo.c
diff --git a/configure.ac b/configure.ac
index 0ee5d85..ddf9bfe 100644
--- a/configure.ac
+++ b/configure.ac
@@ -1216,6 +1216,7 @@ if test "${SYS}" != "mingwce"; then
VLC_ADD_PLUGIN([normvol])
VLC_ADD_PLUGIN([equalizer])
VLC_ADD_PLUGIN([param_eq])
+ VLC_ADD_PLUGIN([scaletempo])
VLC_ADD_PLUGIN([converter_float])
VLC_ADD_PLUGIN([a52tospdif])
VLC_ADD_PLUGIN([dtstospdif])
diff --git a/include/vlc_aout.h b/include/vlc_aout.h
index 1eddf89..cbc7b9b 100644
--- a/include/vlc_aout.h
+++ b/include/vlc_aout.h
@@ -255,6 +255,8 @@ struct aout_input_t
aout_filter_t * pp_filters[AOUT_MAX_FILTERS];
int i_nb_filters;
+ aout_filter_t * p_playback_rate_filter;
+
/* resamplers */
aout_filter_t * pp_resamplers[AOUT_MAX_FILTERS];
int i_nb_resamplers;
diff --git a/modules/audio_filter/Modules.am b/modules/audio_filter/Modules.am
index eade69d..5255a3b 100644
--- a/modules/audio_filter/Modules.am
+++ b/modules/audio_filter/Modules.am
@@ -3,3 +3,4 @@ SOURCES_equalizer = equalizer.c equalizer_presets.h
SOURCES_normvol = normvol.c
SOURCES_audio_format = format.c
SOURCES_param_eq = param_eq.c
+SOURCES_scaletempo = scaletempo.c
diff --git a/modules/audio_filter/scaletempo.c b/modules/audio_filter/scaletempo.c
new file mode 100644
index 0000000..a15bf8a
--- /dev/null
+++ b/modules/audio_filter/scaletempo.c
@@ -0,0 +1,512 @@
+/*****************************************************************************
+ * scaletempo.c: Scale audio tempo while maintaining pitch
+ *****************************************************************************
+ * Copyright © 2008 the VideoLAN team
+ * $Id$
+ *
+ * Authors: Rov Juvano <rovjuvano at users.sourceforge.net>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
+ *****************************************************************************/
+
+/*****************************************************************************
+ * Preamble
+ *****************************************************************************/
+#ifdef HAVE_CONFIG_H
+# include "config.h"
+#endif
+
+#include <vlc_common.h>
+#include <vlc_plugin.h>
+#include <vlc_aout.h>
+
+#include <string.h> /* for memset */
+#include <limits.h> /* form INT_MIN */
+
+/*****************************************************************************
+ * Module descriptor
+ *****************************************************************************/
+static int Open( vlc_object_t * );
+static void Close( vlc_object_t * );
+static void DoWork( aout_instance_t *, aout_filter_t *,
+ aout_buffer_t *, aout_buffer_t * );
+
+vlc_module_begin();
+ set_description( N_("Scale audio tempo in sync with playback rate") );
+ set_shortname( N_("Scaletempo") );
+ set_capability( "audio filter", 0 );
+ set_category( CAT_AUDIO );
+ set_subcategory( SUBCAT_AUDIO_AFILTER );
+
+ add_integer_with_range( "scaletempo-stride", 30, 1, 2000, NULL,
+ N_("Stride Length"), N_("Length in milliseconds to output each stride"), true );
+ add_float_with_range( "scaletempo-overlap", .20, 0.0, 1.0, NULL,
+ N_("Overlap Length"), N_("Percentage of stride to overlap"), true );
+ add_integer_with_range( "scaletempo-search", 14, 0, 200, NULL,
+ N_("Search Length"), N_("Length in milliseconds to search for best overlap position"), true );
+
+ set_callbacks( Open, Close );
+vlc_module_end();
+
+/*
+ * Scaletempo works by producing audio in constant sized chunks (a "stride") but
+ * consuming chunks proportional to the playback rate.
+ *
+ * Scaletempo then smooths the output by blending the end of one stride with
+ * the next ("overlap").
+ *
+ * Scaletempo smooths the overlap further by searching within the input buffer
+ * for the best overlap position. Scaletempo uses a statistical cross correlation
+ * (roughly a dot-product). Scaletempo consumes most of its CPU cycles here.
+ *
+ * NOTE:
+ * sample: a single audio sample for one channel
+ * frame: a single set of samples, one for each channel
+ * VLC uses these terms differently
+ */
+typedef struct aout_filter_sys_t
+{
+ /* Filter static config */
+ double scale;
+ /* parameters */
+ uint ms_stride;
+ double percent_overlap;
+ uint ms_search;
+ /* audio format */
+ uint samples_per_frame; /* AKA number of channels */
+ uint bytes_per_sample;
+ uint bytes_per_frame;
+ uint sample_rate;
+ /* stride */
+ double frames_stride_scaled;
+ double frames_stride_error;
+ uint bytes_stride;
+ double bytes_stride_scaled;
+ uint bytes_queue_max;
+ uint bytes_queued;
+ uint bytes_to_slide;
+ uint8_t *buf_queue;
+ /* overlap */
+ uint samples_overlap;
+ uint samples_standing;
+ uint bytes_overlap;
+ uint bytes_standing;
+ void *buf_overlap;
+ void *table_blend;
+ void (*output_overlap)( aout_filter_t *p_filter, void *p_out_buf, uint bytes_off );
+ /* best overlap */
+ uint frames_search;
+ void *buf_pre_corr;
+ void *table_window;
+ uint (*best_overlap_offset)( aout_filter_t *p_filter );
+ /* for "audio filter" only, manage own buffers */
+ int i_buf;
+ uint8_t *p_buffers[2];
+} aout_filter_sys_t;
+
+/*****************************************************************************
+ * best_overlap_offset: calculate best offset for overlap
+ *****************************************************************************/
+static uint best_overlap_offset_float( aout_filter_t *p_filter )
+{
+ aout_filter_sys_t *p = p_filter->p_sys;
+ float *pw, *po, *ppc, *search_start;
+ float best_corr = INT_MIN;
+ uint best_off = 0;
+ uint i, off;
+
+ pw = p->table_window;
+ po = p->buf_overlap;
+ po += p->samples_per_frame;
+ ppc = p->buf_pre_corr;
+ for( i = p->samples_per_frame; i < p->samples_overlap; i++ ) {
+ *ppc++ = *pw++ * *po++;
+ }
+
+ search_start = (float *)p->buf_queue + p->samples_per_frame;
+ for( off = 0; off < p->frames_search; off++ ) {
+ float corr = 0;
+ float *ps = search_start;
+ ppc = p->buf_pre_corr;
+ for( i = p->samples_per_frame; i < p->samples_overlap; i++ ) {
+ corr += *ppc++ * *ps++;
+ }
+ if( corr > best_corr ) {
+ best_corr = corr;
+ best_off = off;
+ }
+ search_start += p->samples_per_frame;
+ }
+
+ return best_off * p->bytes_per_frame;
+}
+
+/*****************************************************************************
+ * output_overlap: blend end of previous stride with beginning of current stride
+ *****************************************************************************/
+static void output_overlap_float( aout_filter_t *p_filter,
+ void *buf_out,
+ uint bytes_off )
+{
+ aout_filter_sys_t *p = p_filter->p_sys;
+ float *pout = buf_out;
+ float *pb = p->table_blend;
+ float *po = p->buf_overlap;
+ float *pin = (float *)( p->buf_queue + bytes_off );
+ uint i;
+ for( i = 0; i < p->samples_overlap; i++ ) {
+ *pout++ = *po - *pb++ * ( *po - *pin++ ); po++;
+ }
+}
+
+/*****************************************************************************
+ * fill_queue: fill p_sys->buf_queue as much possible, skipping samples as needed
+ *****************************************************************************/
+static size_t fill_queue( aout_filter_t *p_filter,
+ uint8_t *p_buffer,
+ size_t i_buffer,
+ size_t offset )
+{
+ aout_filter_sys_t *p = p_filter->p_sys;
+ uint bytes_in = i_buffer - offset;
+ size_t offset_unchanged = offset;
+
+ if( p->bytes_to_slide > 0 ) {
+ if( p->bytes_to_slide < p->bytes_queued ) {
+ uint bytes_in_move = p->bytes_queued - p->bytes_to_slide;
+ memmove( p->buf_queue,
+ p->buf_queue + p->bytes_to_slide,
+ bytes_in_move );
+ p->bytes_to_slide = 0;
+ p->bytes_queued = bytes_in_move;
+ } else {
+ uint bytes_in_skip;
+ p->bytes_to_slide -= p->bytes_queued;
+ bytes_in_skip = __MIN( p->bytes_to_slide, bytes_in );
+ p->bytes_queued = 0;
+ p->bytes_to_slide -= bytes_in_skip;
+ offset += bytes_in_skip;
+ bytes_in -= bytes_in_skip;
+ }
+ }
+
+ if( bytes_in > 0 ) {
+ uint bytes_in_copy = __MIN( p->bytes_queue_max - p->bytes_queued, bytes_in );
+ memcpy( p->buf_queue + p->bytes_queued,
+ p_buffer + offset,
+ bytes_in_copy );
+ p->bytes_queued += bytes_in_copy;
+ offset += bytes_in_copy;
+ }
+
+ return offset - offset_unchanged;
+}
+
+/*****************************************************************************
+ * transform_buffer: main filter loop
+ *****************************************************************************/
+static size_t transform_buffer( aout_filter_t *p_filter,
+ uint8_t *p_buffer,
+ size_t i_buffer,
+ uint8_t *pout )
+{
+ aout_filter_sys_t *p = p_filter->p_sys;
+
+ size_t offset_in = fill_queue( p_filter, p_buffer, i_buffer, 0 );
+ uint bytes_out = 0;
+ while( p->bytes_queued >= p->bytes_queue_max ) {
+ uint bytes_off = 0;
+
+ // output stride
+ if( p->output_overlap ) {
+ if( p->best_overlap_offset ) {
+ bytes_off = p->best_overlap_offset( p_filter );
+ }
+ p->output_overlap( p_filter, pout, bytes_off );
+ }
+ memcpy( pout + p->bytes_overlap,
+ p->buf_queue + bytes_off + p->bytes_overlap,
+ p->bytes_standing );
+ pout += p->bytes_stride;
+ bytes_out += p->bytes_stride;
+
+ // input stride
+ memcpy( p->buf_overlap,
+ p->buf_queue + bytes_off + p->bytes_stride,
+ p->bytes_overlap );
+ double frames_to_slide = p->frames_stride_scaled + p->frames_stride_error;
+ uint frames_to_stride_whole = (int)frames_to_slide;
+ p->bytes_to_slide = frames_to_stride_whole * p->bytes_per_frame;
+ p->frames_stride_error = frames_to_slide - frames_to_stride_whole;
+
+ offset_in += fill_queue( p_filter, p_buffer, i_buffer, offset_in );
+ }
+
+ return bytes_out;
+}
+
+/*****************************************************************************
+ * calculate_output_buffer_size
+ *****************************************************************************/
+static size_t calculate_output_buffer_size( aout_filter_t *p_filter,
+ size_t bytes_in )
+{
+ aout_filter_sys_t *p = p_filter->p_sys;
+ size_t bytes_out = 0;
+ int bytes_to_out = bytes_in + p->bytes_queued - p->bytes_to_slide;
+ if( bytes_to_out >= (int)p->bytes_queue_max ) {
+ /* while (total_buffered - stride_length * n >= queue_max) n++ */
+ bytes_out = p->bytes_stride * ( (uint)(
+ ( bytes_to_out - p->bytes_queue_max + /* rounding protection */ p->bytes_per_frame )
+ / p->bytes_stride_scaled ) + 1 );
+ }
+ return bytes_out;
+}
+
+/*****************************************************************************
+ * reinit_buffers: reinitializes buffers in p_filter->p_sys
+ *****************************************************************************/
+static int reinit_buffers( aout_filter_t *p_filter )
+{
+ aout_filter_sys_t *p = p_filter->p_sys;
+ uint i,j;
+
+ uint frames_stride = p->ms_stride * p->sample_rate / 1000.0;
+ p->bytes_stride = frames_stride * p->bytes_per_frame;
+
+ /* overlap */
+ uint frames_overlap = frames_stride * p->percent_overlap;
+ if( frames_overlap < 1 ) { /* if no overlap */
+ p->bytes_overlap = 0;
+ p->bytes_standing = p->bytes_stride;
+ p->samples_standing = p->bytes_standing / p->bytes_per_sample;
+ p->output_overlap = NULL;
+ } else {
+ uint prev_overlap = p->bytes_overlap;
+ p->bytes_overlap = frames_overlap * p->bytes_per_frame;
+ p->samples_overlap = frames_overlap * p->samples_per_frame;
+ p->bytes_standing = p->bytes_stride - p->bytes_overlap;
+ p->samples_standing = p->bytes_standing / p->bytes_per_sample;
+ p->buf_overlap = malloc( p->bytes_overlap );
+ p->table_blend = malloc( p->samples_overlap * 4 ); /* sizeof (int32|float) */
+ if( ! p->buf_overlap || ! p->table_blend ) {
+ return VLC_ENOMEM;
+ }
+ if( p->bytes_overlap > prev_overlap ) {
+ memset( (uint8_t *)p->buf_overlap + prev_overlap, 0, p->bytes_overlap - prev_overlap );
+ }
+ float *pb = p->table_blend;
+ float t = (float)frames_overlap;
+ for( i = 0; i<frames_overlap; i++ ) {
+ float v = i / t;
+ for( j = 0; j < p->samples_per_frame; j++ ) {
+ *pb++ = v;
+ }
+ }
+ p->output_overlap = output_overlap_float;
+ }
+
+ /* best overlap */
+ p->frames_search = ( frames_overlap <= 1 ) ? 0 : p->ms_search * p->sample_rate / 1000.0;
+ if( p->frames_search < 1 ) { /* if no search */
+ p->best_overlap_offset = NULL;
+ } else {
+ uint bytes_pre_corr = ( p->samples_overlap - p->samples_per_frame ) * 4; /* sizeof (int32|float) */
+ p->buf_pre_corr = malloc( bytes_pre_corr );
+ p->table_window = malloc( bytes_pre_corr );
+ if( ! p->buf_pre_corr || ! p->table_window ) {
+ return VLC_ENOMEM;
+ }
+ float *pw = p->table_window;
+ for( i = 1; i<frames_overlap; i++ ) {
+ float v = i * ( frames_overlap - i );
+ for( j = 0; j < p->samples_per_frame; j++ ) {
+ *pw++ = v;
+ }
+ }
+ p->best_overlap_offset = best_overlap_offset_float;
+ }
+
+ uint new_size = ( p->frames_search + frames_stride + frames_overlap ) * p->bytes_per_frame;
+ if( p->bytes_queued > new_size ) {
+ if( p->bytes_to_slide > p->bytes_queued ) {
+ p->bytes_to_slide -= p->bytes_queued;
+ p->bytes_queued = 0;
+ } else {
+ uint new_queued = __MIN( p->bytes_queued - p->bytes_to_slide, new_size );
+ memmove( p->buf_queue,
+ p->buf_queue + p->bytes_queued - new_queued,
+ new_queued );
+ p->bytes_to_slide = 0;
+ p->bytes_queued = new_queued;
+ }
+ }
+ p->bytes_queue_max = new_size;
+ p->buf_queue = malloc( p->bytes_queue_max );
+ if( ! p->buf_queue ) {
+ return VLC_ENOMEM;
+ }
+
+ p->bytes_stride_scaled = p->bytes_stride * p->scale;
+ p->frames_stride_scaled = p->bytes_stride_scaled / p->bytes_per_frame;
+
+ msg_Dbg( VLC_OBJECT(p_filter),
+ "%.3f scale, %.3f stride_in, %i stride_out, %i standing, %i overlap, %i search, %i queue, %s mode",
+ p->scale,
+ p->frames_stride_scaled,
+ (int)( p->bytes_stride / p->bytes_per_frame ),
+ (int)( p->bytes_standing / p->bytes_per_frame ),
+ (int)( p->bytes_overlap / p->bytes_per_frame ),
+ p->frames_search,
+ (int)( p->bytes_queue_max / p->bytes_per_frame ),
+ "fl32");
+
+ return VLC_SUCCESS;
+}
+
+/*****************************************************************************
+ * Open: initialize as "audio filter"
+ *****************************************************************************/
+static int Open( vlc_object_t *p_this )
+{
+ aout_filter_t *p_filter = (aout_filter_t *)p_this;
+ aout_filter_sys_t *p_sys;
+ bool b_fit = true;
+
+ if( p_filter->input.i_format != VLC_FOURCC('f','l','3','2' ) ||
+ p_filter->output.i_format != VLC_FOURCC('f','l','3','2') )
+ {
+ b_fit = false;
+ p_filter->input.i_format = p_filter->output.i_format = VLC_FOURCC('f','l','3','2');
+ msg_Warn( p_filter, "bad input or output format" );
+ }
+ if( ! AOUT_FMTS_SIMILAR( &p_filter->input, &p_filter->output ) )
+ {
+ b_fit = false;
+ memcpy( &p_filter->output, &p_filter->input, sizeof(audio_sample_format_t) );
+ msg_Warn( p_filter, "input and output formats are not similar" );
+ }
+
+ if( ! b_fit )
+ {
+ return VLC_EGENERIC;
+ }
+
+ p_filter->pf_do_work = DoWork;
+ p_filter->b_in_place = false;
+
+ /* Allocate structure */
+ p_sys = p_filter->p_sys = malloc( sizeof(aout_filter_sys_t) );
+ if( ! p_sys )
+ {
+ return VLC_ENOMEM;
+ }
+
+ p_sys->scale = 1.0;
+ p_sys->sample_rate = p_filter->input.i_rate;
+ p_sys->samples_per_frame = aout_FormatNbChannels( &p_filter->input );
+ p_sys->bytes_per_sample = 4;
+ p_sys->bytes_per_frame = p_sys->samples_per_frame * p_sys->bytes_per_sample;
+
+ msg_Dbg( p_this, "format: %5i rate, %i nch, %i bps, %s",
+ p_sys->sample_rate,
+ p_sys->samples_per_frame,
+ p_sys->bytes_per_sample,
+ "fl32" );
+
+ p_sys->ms_stride = config_GetInt( p_this, "scaletempo-stride" );
+ p_sys->percent_overlap = config_GetFloat( p_this, "scaletempo-overlap" );
+ p_sys->ms_search = config_GetInt( p_this, "scaletempo-search" );
+
+ msg_Dbg( p_this, "params: %i stride, %.3f overlap, %i search",
+ p_sys->ms_stride, p_sys->percent_overlap, p_sys->ms_search );
+
+ p_sys->i_buf = 0;
+ p_sys->p_buffers[0] = NULL;
+ p_sys->p_buffers[1] = NULL;
+
+ p_sys->buf_queue = NULL;
+ p_sys->buf_overlap = NULL;
+ p_sys->table_blend = NULL;
+ p_sys->buf_pre_corr = NULL;
+ p_sys->table_window = NULL;
+ p_sys->bytes_overlap = 0;
+ p_sys->bytes_queued = 0;
+ p_sys->bytes_to_slide = 0;
+ p_sys->frames_stride_error = 0;
+ return reinit_buffers( p_filter );
+}
+
+static void Close( vlc_object_t *p_this )
+{
+ aout_filter_t *p_filter = (aout_filter_t *)p_this;
+ aout_filter_sys_t *p_sys = p_filter->p_sys;
+ free( p_sys->buf_queue );
+ free( p_sys->buf_overlap );
+ free( p_sys->table_blend );
+ free( p_sys->buf_pre_corr );
+ free( p_sys->table_window );
+ free( p_sys->p_buffers[0] );
+ free( p_sys->p_buffers[1] );
+ free( p_filter->p_sys );
+}
+
+/*****************************************************************************
+ * DoWork: aout_filter wrapper for transform_buffer
+ *****************************************************************************/
+static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter,
+ aout_buffer_t * p_in_buf, aout_buffer_t * p_out_buf )
+{
+ VLC_UNUSED(p_aout);
+ aout_filter_sys_t *p = p_filter->p_sys;
+
+ if( p_filter->input.i_rate == p->sample_rate ) {
+ memcpy( p_out_buf->p_buffer, p_in_buf->p_buffer, p_in_buf->i_nb_bytes );
+ p_out_buf->i_nb_bytes = p_in_buf->i_nb_bytes;
+ p_out_buf->i_nb_samples = p_in_buf->i_nb_samples;
+ return;
+ }
+
+ double scale = p_filter->input.i_rate / (double)p->sample_rate;
+ if( scale != p->scale ) {
+ p->scale = scale;
+ p->bytes_stride_scaled = p->bytes_stride * p->scale;
+ p->frames_stride_scaled = p->bytes_stride_scaled / p->bytes_per_frame;
+ p->bytes_to_slide = 0;
+ msg_Dbg( p_filter, "%.3f scale, %.3f stride_in, %i stride_out",
+ p->scale,
+ p->frames_stride_scaled,
+ (int)( p->bytes_stride / p->bytes_per_frame ) );
+ }
+
+ size_t i_outsize = calculate_output_buffer_size ( p_filter, p_in_buf->i_nb_bytes );
+ if( i_outsize > p_out_buf->i_size ) {
+ void *temp = realloc( p->p_buffers[ p->i_buf ], i_outsize );
+ if( temp == NULL )
+ {
+ return;
+ }
+ p->p_buffers[ p->i_buf ] = temp;
+ p_out_buf->p_buffer = p->p_buffers[ p->i_buf ];
+ p->i_buf = ! p->i_buf;
+ }
+
+ size_t bytes_out = transform_buffer( p_filter,
+ p_in_buf->p_buffer, p_in_buf->i_nb_bytes,
+ p_out_buf->p_buffer );
+
+ p_out_buf->i_nb_bytes = bytes_out;
+ p_out_buf->i_nb_samples = bytes_out / p->bytes_per_frame;
+}
diff --git a/src/audio_output/input.c b/src/audio_output/input.c
index 184e8f7..4761706 100644
--- a/src/audio_output/input.c
+++ b/src/audio_output/input.c
@@ -401,6 +401,21 @@ int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input )
}
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 );
@@ -475,7 +490,7 @@ int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input,
vlc_mutex_unlock( &p_aout->mixer_lock );
}
- if( i_input_rate != INPUT_RATE_DEFAULT && p_input->i_nb_resamplers <= 0 )
+ if( i_input_rate != INPUT_RATE_DEFAULT && p_input->p_playback_rate_filter == NULL )
{
inputDrop( p_aout, p_input, p_buffer );
return 0;
@@ -502,10 +517,10 @@ int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input,
}
#endif
- /* Handle input rate change by modifying resampler input rate */
+ /* 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->pp_resamplers[0]->input.i_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);
@@ -551,8 +566,9 @@ int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input,
/* 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 - 3 * AOUT_PTS_TOLERANCE ) )
+ ( 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 );
@@ -566,7 +582,7 @@ int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input,
start_date = 0;
}
else if ( start_date != 0 &&
- ( start_date > p_buffer->start_date + 3 * AOUT_PTS_TOLERANCE ) )
+ ( 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 );
@@ -629,7 +645,11 @@ int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input,
/* Check if everything is back to normal, in which case we can stop the
* resampling */
- if( p_input->pp_resamplers[0]->input.i_rate == 1000 * p_input->input.i_rate / i_input_rate )
+ unsigned int i_nominal_rate =
+ (p_input->pp_resamplers[0] == p_input->p_playback_rate_filter)
+ ? INPUT_RATE_DEFAULT * p_input->input.i_rate / i_input_rate
+ : p_input->input.i_rate;
+ if( p_input->pp_resamplers[0]->input.i_rate == i_nominal_rate )
{
p_input->i_resampling_type = AOUT_RESAMPLING_NONE;
msg_Warn( p_aout, "resampling stopped after %"PRIi64" usec "
@@ -733,8 +753,10 @@ static void inputResamplingStop( aout_input_t *p_input )
p_input->i_resampling_type = AOUT_RESAMPLING_NONE;
if( p_input->i_nb_resamplers != 0 )
{
- p_input->pp_resamplers[0]->input.i_rate = INPUT_RATE_DEFAULT *
- p_input->input.i_rate / p_input->i_last_input_rate;
+ p_input->pp_resamplers[0]->input.i_rate =
+ ( p_input->pp_resamplers[0] == p_input->p_playback_rate_filter )
+ ? INPUT_RATE_DEFAULT * p_input->input.i_rate / p_input->i_last_input_rate
+ : p_input->input.i_rate;
p_input->pp_resamplers[0]->b_continuity = false;
}
}
--
1.5.4.3
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