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libremedia
Tethys
FFmpeg
Commits
f199f385
Commit
f199f385
authored
13 years ago
by
Justin Ruggles
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avplay: use avcodec_decode_audio4()
parent
e2a2c49f
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1 changed file
avplay.c
+27
-16
27 additions, 16 deletions
avplay.c
with
27 additions
and
16 deletions
avplay.c
+
27
−
16
View file @
f199f385
...
...
@@ -153,18 +153,16 @@ typedef struct VideoState {
AVStream
*
audio_st
;
PacketQueue
audioq
;
int
audio_hw_buf_size
;
/* samples output by the codec. we reserve more space for avsync
compensation */
DECLARE_ALIGNED
(
16
,
uint8_t
,
audio_buf1
)[(
AVCODEC_MAX_AUDIO_FRAME_SIZE
*
3
)
/
2
];
DECLARE_ALIGNED
(
16
,
uint8_t
,
audio_buf2
)[(
AVCODEC_MAX_AUDIO_FRAME_SIZE
*
3
)
/
2
];
uint8_t
silence_buf
[
SDL_AUDIO_BUFFER_SIZE
];
uint8_t
*
audio_buf
;
uint8_t
*
audio_buf1
;
unsigned
int
audio_buf_size
;
/* in bytes */
int
audio_buf_index
;
/* in bytes */
AVPacket
audio_pkt_temp
;
AVPacket
audio_pkt
;
enum
AVSampleFormat
audio_src_fmt
;
AVAudioConvert
*
reformat_ctx
;
AVFrame
*
frame
;
int
show_audio
;
/* if true, display audio samples */
int16_t
sample_array
[
SAMPLE_ARRAY_SIZE
];
...
...
@@ -2010,7 +2008,7 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
AVPacket
*
pkt_temp
=
&
is
->
audio_pkt_temp
;
AVPacket
*
pkt
=
&
is
->
audio_pkt
;
AVCodecContext
*
dec
=
is
->
audio_st
->
codec
;
int
n
,
len1
,
data_size
;
int
n
,
len1
,
data_size
,
got_frame
;
double
pts
;
int
new_packet
=
0
;
int
flush_complete
=
0
;
...
...
@@ -2018,13 +2016,16 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
for
(;;)
{
/* NOTE: the audio packet can contain several frames */
while
(
pkt_temp
->
size
>
0
||
(
!
pkt_temp
->
data
&&
new_packet
))
{
if
(
!
is
->
frame
)
{
if
(
!
(
is
->
frame
=
avcodec_alloc_frame
()))
return
AVERROR
(
ENOMEM
);
}
else
avcodec_get_frame_defaults
(
is
->
frame
);
if
(
flush_complete
)
break
;
new_packet
=
0
;
data_size
=
sizeof
(
is
->
audio_buf1
);
len1
=
avcodec_decode_audio3
(
dec
,
(
int16_t
*
)
is
->
audio_buf1
,
&
data_size
,
pkt_temp
);
len1
=
avcodec_decode_audio4
(
dec
,
is
->
frame
,
&
got_frame
,
pkt_temp
);
if
(
len1
<
0
)
{
/* if error, we skip the frame */
pkt_temp
->
size
=
0
;
...
...
@@ -2034,12 +2035,15 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
pkt_temp
->
data
+=
len1
;
pkt_temp
->
size
-=
len1
;
if
(
data_size
<=
0
)
{
if
(
!
got_frame
)
{
/* stop sending empty packets if the decoder is finished */
if
(
!
pkt_temp
->
data
&&
dec
->
codec
->
capabilities
&
CODEC_CAP_DELAY
)
flush_complete
=
1
;
continue
;
}
data_size
=
av_samples_get_buffer_size
(
NULL
,
dec
->
channels
,
is
->
frame
->
nb_samples
,
dec
->
sample_fmt
,
1
);
if
(
dec
->
sample_fmt
!=
is
->
audio_src_fmt
)
{
if
(
is
->
reformat_ctx
)
...
...
@@ -2056,21 +2060,26 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
}
if
(
is
->
reformat_ctx
)
{
const
void
*
ibuf
[
6
]
=
{
is
->
audio_buf1
};
void
*
obuf
[
6
]
=
{
is
->
audio_buf2
}
;
const
void
*
ibuf
[
6
]
=
{
is
->
frame
->
data
[
0
]
};
void
*
obuf
[
6
];
int
istride
[
6
]
=
{
av_get_bytes_per_sample
(
dec
->
sample_fmt
)};
int
ostride
[
6
]
=
{
2
};
int
len
=
data_size
/
istride
[
0
];
obuf
[
0
]
=
av_realloc
(
is
->
audio_buf1
,
FFALIGN
(
len
*
ostride
[
0
],
32
));
if
(
!
obuf
[
0
])
{
return
AVERROR
(
ENOMEM
);
}
is
->
audio_buf1
=
obuf
[
0
];
if
(
av_audio_convert
(
is
->
reformat_ctx
,
obuf
,
ostride
,
ibuf
,
istride
,
len
)
<
0
)
{
printf
(
"av_audio_convert() failed
\n
"
);
break
;
}
is
->
audio_buf
=
is
->
audio_buf
2
;
is
->
audio_buf
=
is
->
audio_buf
1
;
/* FIXME: existing code assume that data_size equals framesize*channels*2
remove this legacy cruft */
data_size
=
len
*
2
;
}
else
{
is
->
audio_buf
=
is
->
audio_buf1
;
is
->
audio_buf
=
is
->
frame
->
data
[
0
]
;
}
/* if no pts, then compute it */
...
...
@@ -2106,8 +2115,7 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
if
(
pkt
->
data
==
flush_pkt
.
data
)
avcodec_flush_buffers
(
dec
);
pkt_temp
->
data
=
pkt
->
data
;
pkt_temp
->
size
=
pkt
->
size
;
*
pkt_temp
=
*
pkt
;
/* if update the audio clock with the pts */
if
(
pkt
->
pts
!=
AV_NOPTS_VALUE
)
{
...
...
@@ -2275,6 +2283,9 @@ static void stream_component_close(VideoState *is, int stream_index)
if
(
is
->
reformat_ctx
)
av_audio_convert_free
(
is
->
reformat_ctx
);
is
->
reformat_ctx
=
NULL
;
av_freep
(
&
is
->
audio_buf1
);
is
->
audio_buf
=
NULL
;
av_freep
(
&
is
->
frame
);
if
(
is
->
rdft
)
{
av_rdft_end
(
is
->
rdft
);
...
...
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