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submodule
ffmpeg
Commits
6304f78e
Commit
6304f78e
authored
Sep 30, 2012
by
Justin Ruggles
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Plain Diff
avplay: support mid-stream sample rate changes
Resample to the rate that was configured in SDL.
parent
f5962229
Hide whitespace changes
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Showing
1 changed file
with
21 additions
and
14 deletions
+21
-14
avplay.c
avplay.c
+21
-14
No files found.
avplay.c
View file @
6304f78e
...
...
@@ -164,8 +164,10 @@ typedef struct VideoState {
enum
AVSampleFormat
sdl_sample_fmt
;
uint64_t
sdl_channel_layout
;
int
sdl_channels
;
int
sdl_sample_rate
;
enum
AVSampleFormat
resample_sample_fmt
;
uint64_t
resample_channel_layout
;
int
resample_sample_rate
;
AVAudioResampleContext
*
avr
;
AVFrame
*
frame
;
...
...
@@ -758,7 +760,7 @@ static void video_audio_display(VideoState *s)
the last buffer computation */
if
(
audio_callback_time
)
{
time_diff
=
av_gettime
()
-
audio_callback_time
;
delay
-=
(
time_diff
*
s
->
audio_st
->
codec
->
sample_rate
)
/
1000000
;
delay
-=
(
time_diff
*
s
->
sdl_
sample_rate
)
/
1000000
;
}
delay
+=
2
*
data_used
;
...
...
@@ -960,7 +962,7 @@ static double get_audio_clock(VideoState *is)
hw_buf_size
=
audio_write_get_buf_size
(
is
);
bytes_per_sec
=
0
;
if
(
is
->
audio_st
)
{
bytes_per_sec
=
is
->
audio_st
->
codec
->
sample_rate
*
is
->
sdl_channels
*
bytes_per_sec
=
is
->
sdl_
sample_rate
*
is
->
sdl_channels
*
av_get_bytes_per_sample
(
is
->
sdl_sample_fmt
);
}
if
(
bytes_per_sec
)
...
...
@@ -1817,7 +1819,7 @@ static int synchronize_audio(VideoState *is, short *samples,
avg_diff
=
is
->
audio_diff_cum
*
(
1
.
0
-
is
->
audio_diff_avg_coef
);
if
(
fabs
(
avg_diff
)
>=
is
->
audio_diff_threshold
)
{
wanted_size
=
samples_size
+
((
int
)(
diff
*
is
->
audio_st
->
codec
->
sample_rate
)
*
n
);
wanted_size
=
samples_size
+
((
int
)(
diff
*
is
->
sdl_
sample_rate
)
*
n
);
nb_samples
=
samples_size
/
n
;
min_size
=
((
nb_samples
*
(
100
-
SAMPLE_CORRECTION_PERCENT_MAX
))
/
100
)
*
n
;
...
...
@@ -1907,11 +1909,13 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
is
->
frame
->
nb_samples
,
is
->
frame
->
format
,
1
);
audio_resample
=
is
->
frame
->
format
!=
is
->
sdl_sample_fmt
||
is
->
frame
->
channel_layout
!=
is
->
sdl_channel_layout
;
audio_resample
=
is
->
frame
->
format
!=
is
->
sdl_sample_fmt
||
is
->
frame
->
channel_layout
!=
is
->
sdl_channel_layout
||
is
->
frame
->
sample_rate
!=
is
->
sdl_sample_rate
;
resample_changed
=
is
->
frame
->
format
!=
is
->
resample_sample_fmt
||
is
->
frame
->
channel_layout
!=
is
->
resample_channel_layout
;
resample_changed
=
is
->
frame
->
format
!=
is
->
resample_sample_fmt
||
is
->
frame
->
channel_layout
!=
is
->
resample_channel_layout
||
is
->
frame
->
sample_rate
!=
is
->
resample_sample_rate
;
if
((
!
is
->
avr
&&
audio_resample
)
||
resample_changed
)
{
int
ret
;
...
...
@@ -1928,9 +1932,9 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
av_opt_set_int
(
is
->
avr
,
"in_channel_layout"
,
is
->
frame
->
channel_layout
,
0
);
av_opt_set_int
(
is
->
avr
,
"in_sample_fmt"
,
is
->
frame
->
format
,
0
);
av_opt_set_int
(
is
->
avr
,
"in_sample_rate"
,
is
->
frame
->
sample_rate
,
0
);
av_opt_set_int
(
is
->
avr
,
"out_channel_layout"
,
is
->
sdl_channel_layout
,
0
);
av_opt_set_int
(
is
->
avr
,
"out_sample_fmt"
,
is
->
sdl_sample_fmt
,
0
);
av_opt_set_int
(
is
->
avr
,
"out_sample_rate"
,
dec
->
sample_rate
,
0
);
av_opt_set_int
(
is
->
avr
,
"out_channel_layout"
,
is
->
sdl_channel_layout
,
0
);
av_opt_set_int
(
is
->
avr
,
"out_sample_fmt"
,
is
->
sdl_sample_fmt
,
0
);
av_opt_set_int
(
is
->
avr
,
"out_sample_rate"
,
is
->
sdl_
sample_rate
,
0
);
if
((
ret
=
avresample_open
(
is
->
avr
))
<
0
)
{
fprintf
(
stderr
,
"error initializing libavresample
\n
"
);
...
...
@@ -1939,6 +1943,7 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
}
is
->
resample_sample_fmt
=
is
->
frame
->
format
;
is
->
resample_channel_layout
=
is
->
frame
->
channel_layout
;
is
->
resample_sample_rate
=
is
->
frame
->
sample_rate
;
}
if
(
audio_resample
)
{
...
...
@@ -1977,7 +1982,7 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
*
pts_ptr
=
pts
;
n
=
is
->
sdl_channels
*
av_get_bytes_per_sample
(
is
->
sdl_sample_fmt
);
is
->
audio_clock
+=
(
double
)
data_size
/
(
double
)(
n
*
dec
->
sample_rate
);
(
double
)(
n
*
is
->
sdl_
sample_rate
);
#ifdef DEBUG
{
static
double
last_clock
;
...
...
@@ -2092,8 +2097,7 @@ static int stream_component_open(VideoState *is, int stream_index)
/* prepare audio output */
if
(
avctx
->
codec_type
==
AVMEDIA_TYPE_AUDIO
)
{
wanted_spec
.
freq
=
avctx
->
sample_rate
;
wanted_spec
.
format
=
AUDIO_S16SYS
;
is
->
sdl_sample_rate
=
avctx
->
sample_rate
;
if
(
!
avctx
->
channel_layout
)
avctx
->
channel_layout
=
av_get_default_channel_layout
(
avctx
->
channels
);
...
...
@@ -2107,6 +2111,8 @@ static int stream_component_open(VideoState *is, int stream_index)
is
->
sdl_channel_layout
=
AV_CH_LAYOUT_STEREO
;
is
->
sdl_channels
=
av_get_channel_layout_nb_channels
(
is
->
sdl_channel_layout
);
wanted_spec
.
format
=
AUDIO_S16SYS
;
wanted_spec
.
freq
=
is
->
sdl_sample_rate
;
wanted_spec
.
channels
=
is
->
sdl_channels
;
wanted_spec
.
silence
=
0
;
wanted_spec
.
samples
=
SDL_AUDIO_BUFFER_SIZE
;
...
...
@@ -2119,7 +2125,8 @@ static int stream_component_open(VideoState *is, int stream_index)
is
->
audio_hw_buf_size
=
spec
.
size
;
is
->
sdl_sample_fmt
=
AV_SAMPLE_FMT_S16
;
is
->
resample_sample_fmt
=
is
->
sdl_sample_fmt
;
is
->
resample_channel_layout
=
is
->
sdl_channel_layout
;
is
->
resample_channel_layout
=
avctx
->
channel_layout
;
is
->
resample_sample_rate
=
avctx
->
sample_rate
;
}
ic
->
streams
[
stream_index
]
->
discard
=
AVDISCARD_DEFAULT
;
...
...
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