[vlc-commits] aout: move input format to aout_owner
Rémi Denis-Courmont
git at videolan.org
Mon Aug 8 22:51:38 CEST 2011
vlc | branch: master | Rémi Denis-Courmont <remi at remlab.net> | Mon Aug 8 23:27:52 2011 +0300| [502b865b5ecb542ac693b14c3d95dff380d09ddb] | committer: Rémi Denis-Courmont
aout: move input format to aout_owner
> http://git.videolan.org/gitweb.cgi/vlc.git/?a=commit;h=502b865b5ecb542ac693b14c3d95dff380d09ddb
---
src/audio_output/aout_internal.h | 6 ++++--
src/audio_output/dec.c | 16 +++++++---------
src/audio_output/input.c | 31 ++++++++++++++++++-------------
3 files changed, 29 insertions(+), 24 deletions(-)
diff --git a/src/audio_output/aout_internal.h b/src/audio_output/aout_internal.h
index e03dd32..85caf0b 100644
--- a/src/audio_output/aout_internal.h
+++ b/src/audio_output/aout_internal.h
@@ -51,8 +51,8 @@ block_t *aout_FilterBufferNew( filter_t *, int );
/** an input stream for the audio output */
struct aout_input_t
{
- audio_sample_format_t input;
float multiplier; /**< Replay gain multiplier */
+ unsigned samplerate; /**< Input sample rate */
/* pre-filters */
filter_t * pp_filters[AOUT_MAX_FILTERS];
@@ -106,6 +106,7 @@ typedef struct
} volume; /**< Volume and gain management (FIXME: input manager?) */
audio_sample_format_t mixer_format;
+ audio_sample_format_t input_format;
/* Filters between mixer and output */
filter_t *filters[AOUT_MAX_FILTERS];
@@ -130,7 +131,8 @@ static inline aout_owner_t *aout_owner (audio_output_t *aout)
*****************************************************************************/
/* From input.c : */
-int aout_InputNew( audio_output_t * p_aout, aout_input_t * p_input, const aout_request_vout_t * );
+int aout_InputNew(audio_output_t *, const audio_sample_format_t *,
+ aout_input_t *, const aout_request_vout_t *);
int aout_InputDelete( audio_output_t * p_aout, aout_input_t * p_input );
block_t *aout_InputPlay( audio_output_t *p_aout, aout_input_t *p_input,
block_t *p_buffer, int i_input_rate, date_t * );
diff --git a/src/audio_output/dec.c b/src/audio_output/dec.c
index f60fa4d..50c3534 100644
--- a/src/audio_output/dec.c
+++ b/src/audio_output/dec.c
@@ -83,8 +83,6 @@ int aout_DecNew( audio_output_t *p_aout,
p_input->b_error = true;
- memcpy( &p_input->input, p_format,
- sizeof(audio_sample_format_t) );
if( p_replay_gain )
p_input->replay_gain = *p_replay_gain;
@@ -98,6 +96,7 @@ int aout_DecNew( audio_output_t *p_aout,
var_Destroy( p_aout, "audio-channels" );
/* Recreate the output using the new format. */
+ owner->input_format = *p_format;
if( aout_OutputNew( p_aout, p_format ) < 0 )
goto error;
@@ -108,7 +107,7 @@ int aout_DecNew( audio_output_t *p_aout,
date_Set (&owner->sync.date, VLC_TS_INVALID);
owner->input = p_input;
- aout_InputNew( p_aout, p_input, p_request_vout );
+ aout_InputNew( p_aout, p_format, p_input, p_request_vout );
aout_unlock( p_aout );
return 0;
error:
@@ -161,7 +160,7 @@ static void aout_CheckRestart (audio_output_t *aout)
owner->volume.mixer = NULL;
aout_OutputDelete (aout);
- if (aout_OutputNew (aout, &input->input))
+ if (aout_OutputNew (aout, &owner->input_format))
{
input->b_error = true;
return; /* we are officially screwed */
@@ -169,7 +168,7 @@ static void aout_CheckRestart (audio_output_t *aout)
owner->volume.mixer = aout_MixerNew (aout, owner->mixer_format.i_format);
- if (aout_InputNew (aout, input, &input->request_vout))
+ if (aout_InputNew (aout, &owner->input_format, input, &input->request_vout))
assert (input->b_error);
else
assert (!input->b_error);
@@ -187,10 +186,9 @@ block_t *aout_DecNewBuffer (audio_output_t *aout, size_t samples)
{
/* NOTE: the caller is responsible for serializing input change */
aout_owner_t *owner = aout_owner (aout);
- aout_input_t *input = owner->input;
- size_t length = samples * input->input.i_bytes_per_frame
- / input->input.i_frame_length;
+ size_t length = samples * owner->input_format.i_bytes_per_frame
+ / owner->input_format.i_frame_length;
block_t *block = block_Alloc( length );
if( likely(block != NULL) )
{
@@ -222,7 +220,7 @@ int aout_DecPlay (audio_output_t *p_aout, block_t *p_buffer, int i_input_rate)
assert( p_buffer->i_pts > 0 );
p_buffer->i_length = (mtime_t)p_buffer->i_nb_samples * 1000000
- / p_input->input.i_rate;
+ / owner->input_format.i_rate;
aout_lock( p_aout );
if( p_input->b_error )
diff --git a/src/audio_output/input.c b/src/audio_output/input.c
index 72fcba4..d51f8b6 100644
--- a/src/audio_output/input.c
+++ b/src/audio_output/input.c
@@ -66,7 +66,10 @@ static vout_thread_t *RequestVout( void *,
/*****************************************************************************
* aout_InputNew : allocate a new input and rework the filter pipeline
*****************************************************************************/
-int aout_InputNew( audio_output_t * p_aout, aout_input_t * p_input, const aout_request_vout_t *p_request_vout )
+int aout_InputNew( audio_output_t * p_aout,
+ const audio_sample_format_t *restrict format,
+ aout_input_t * p_input,
+ const aout_request_vout_t *p_request_vout )
{
aout_owner_t *owner = aout_owner (p_aout);
audio_sample_format_t chain_input_format;
@@ -75,7 +78,8 @@ int aout_InputNew( audio_output_t * p_aout, aout_input_t * p_input, const aout_r
char *psz_filters, *psz_visual, *psz_scaletempo;
int i_visual;
- aout_FormatPrint( p_aout, "input", &p_input->input );
+ aout_FormatPrint( p_aout, "input", format );
+ p_input->samplerate = format->i_rate;
p_input->i_nb_resamplers = p_input->i_nb_filters = 0;
@@ -91,9 +95,9 @@ int aout_InputNew( audio_output_t * p_aout, aout_input_t * p_input, const aout_r
}
/* Prepare format structure */
- chain_input_format = p_input->input;
+ chain_input_format = *format;
chain_output_format = owner->mixer_format;
- chain_output_format.i_rate = p_input->input.i_rate;
+ chain_output_format.i_rate = format->i_rate;
aout_FormatPrepare( &chain_output_format );
/* Now add user filters */
@@ -382,7 +386,7 @@ int aout_InputNew( audio_output_t * p_aout, aout_input_t * p_input, const aout_r
/* Create resamplers. */
if (AOUT_FMT_LINEAR(&owner->mixer_format))
{
- chain_output_format.i_rate = (__MAX(p_input->input.i_rate,
+ chain_output_format.i_rate = (__MAX(p_input->samplerate,
owner->mixer_format.i_rate)
* (100 + AOUT_MAX_RESAMPLING)) / 100;
if ( chain_output_format.i_rate == owner->mixer_format.i_rate )
@@ -400,7 +404,7 @@ int aout_InputNew( audio_output_t * p_aout, aout_input_t * p_input, const aout_r
}
/* Setup the initial rate of the resampler */
- p_input->pp_resamplers[0]->fmt_in.audio.i_rate = p_input->input.i_rate;
+ p_input->pp_resamplers[0]->fmt_in.audio.i_rate = p_input->samplerate;
}
p_input->i_resampling_type = AOUT_RESAMPLING_NONE;
@@ -453,13 +457,14 @@ int aout_InputDelete( audio_output_t * p_aout, aout_input_t * p_input )
*****************************************************************************/
void aout_InputCheckAndRestart( audio_output_t * p_aout, aout_input_t * p_input )
{
+ aout_owner_t *owner = aout_owner(p_aout);
aout_assert_locked( p_aout );
if( !p_input->b_restart )
return;
aout_InputDelete( p_aout, p_input );
- aout_InputNew( p_aout, p_input, &p_input->request_vout );
+ aout_InputNew( p_aout, &owner->input_format, p_input, &p_input->request_vout );
p_input->b_restart = false;
}
@@ -513,8 +518,8 @@ block_t *aout_InputPlay(audio_output_t *p_aout, aout_input_t *p_input,
{
unsigned int * const pi_rate = &p_input->p_playback_rate_filter->fmt_in.audio.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);
+ const int i_delta = *pi_rate - F(p_input->samplerate,p_input->i_last_input_rate);
+ *pi_rate = F(p_input->samplerate + i_delta, i_input_rate);
#undef F
p_input->i_last_input_rate = i_input_rate;
}
@@ -628,8 +633,8 @@ block_t *aout_InputPlay(audio_output_t *p_aout, aout_input_t *p_input,
* resampling */
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;
+ ? INPUT_RATE_DEFAULT * p_input->samplerate / i_input_rate
+ : p_input->samplerate;
if( p_input->pp_resamplers[0]->fmt_in.audio.i_rate == i_nominal_rate )
{
p_input->i_resampling_type = AOUT_RESAMPLING_NONE;
@@ -724,8 +729,8 @@ static void inputResamplingStop( aout_input_t *p_input )
{
p_input->pp_resamplers[0]->fmt_in.audio.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;
+ ? INPUT_RATE_DEFAULT * p_input->samplerate / p_input->i_last_input_rate
+ : p_input->samplerate;
}
}
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