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Elphel
elphel-apps-camogm
Commits
313f638e
Commit
313f638e
authored
Jun 15, 2017
by
Mikhail Karpenko
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Process audio between video frames
parent
b50eb743
Changes
7
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7 changed files
with
292 additions
and
330 deletions
+292
-330
camogm.c
src/camogm.c
+64
-93
camogm.h
src/camogm.h
+1
-0
camogm_audio.c
src/camogm_audio.c
+198
-164
camogm_audio.h
src/camogm_audio.h
+25
-21
camogm_mov.c
src/camogm_mov.c
+4
-17
thelper.c
src/thelper.c
+0
-33
thelper.h
src/thelper.h
+0
-2
No files found.
src/camogm.c
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313f638e
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src/camogm.h
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313f638e
...
...
@@ -223,6 +223,7 @@ typedef struct {
int
max_frames
;
int
set_max_frames
;
int
frames_per_chunk
;
///< QuickTime, the number of samples (images or audio samples) in a chunk
unsigned
int
chunk_frame_cntr
;
///< QuickTime, number of video frames recorded in current chunk
int
set_frames_per_chunk
;
///< QuickTime, index for fast forward (sample-to-chunk atom)
int
frameno
;
///< total image frame counter, does not include audio samples
unsigned
long
*
frame_lengths
;
///< QuickTime; pointer to an array of frame lengths wich includes both image frames
...
...
src/camogm_audio.c
View file @
313f638e
This diff is collapsed.
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src/camogm_audio.h
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313f638e
...
...
@@ -25,8 +25,8 @@
#include <sys/time.h>
#include <alsa/asoundlib.h>
#define SAMPLE_RATE 44100
#define SAMPLE_CHANNELS 2
#define SAMPLE_RATE 44100
///< default sampling rate
#define SAMPLE_CHANNELS 2
///< default number of audio channels
#define SAMPLE_TIME 200 ///< restrict ALSA to have this period, in milliseconds
#define BUFFER_TIME 1000 ///< approximate ALSA buffer duration, in milliseconds
#define DEFAULT_SND_DEVICE "plughw:0,0"
...
...
@@ -36,27 +36,34 @@
#define AUDIO_RATE_MAX 44100
#define DEFAULT_AUDIO_VOLUME 0xffff
/**
* @brief Audio recording context related to stream management.
* Members of this structure should not be used outside audio module.
*/
struct
context_audio
{
char
*
sbuffer
;
///< buffer for audio samples
long
sbuffer_len
;
///< the length of samples buffer in samples
long
sbuffer_len
;
///< total length of audio buffer, in audio frames
long
sbuffer_pos
;
///< pointer to current write position in audio buffer, in frames
long
read_frames
;
///< read granularity, in frames
long
sample_time
;
///< duration of one chunk of audio data, in ms
struct
timeval
time_start
;
///< start time, set only when stream starts and updated with each new file
struct
timeval
time_last
;
///< calculated time of last audio sample (this value is not taken from ALSA)
long
rem_samples
;
///< remaining samples
int
begin_of_stream_with_audio
;
///<
int
audio_trigger
;
///< indicates the beginning of audio recording to make some initial set ups
long
long
audio_skip_samples
;
///<
snd_pcm_format_t
audio_format
;
///< format of audio samples as defined in 'enum snd_pcm_format_t'
snd_pcm_t
*
capture_hnd
;
///< ALSA PCM handle
};
/**
* @brief Various parameters related to audio recording.
*/
struct
audio
{
int
audio_enable
;
///< flag indicating that audio is enabled
int
audio_rate
;
///< sample rate, in Hz
int
audio_channels
;
///< number of channels
int
audio_volume
;
///< volume set in range [0..0xFFFF]
int
sync_port
;
///< synch audio stream to this sensor port
int
set_audio_enable
;
///< temporary storage for new value
int
set_audio_rate
;
///< temporary storage for new value
...
...
@@ -74,27 +81,24 @@ struct audio {
struct
timeval
ts_audio
;
///< time stamp when audio stream started
struct
timeval
ts_video
;
///< time stamp of each new frame
struct
timeval
ts_video_start
;
///< time stamp of starting video frame
int
frame_period
;
///< video frame period, in microseconds
snd_pcm_format_t
audio_format
;
///< format of audio samples as defined in 'enum snd_pcm_format_t'
int
frame_period_us
;
///< video frame period measured for #sync_port, in microseconds
unsigned
long
audio_skip_samples
;
///< skip this number audio frames to sync to video
int
begin_of_stream_with_audio
;
///< flag indicating that A/V sync is in progress
int
audio_trigger
;
///< indicates the beginning of audio recording to make some initial set ups
bool
save_data
;
///< flag indicating that audio data should be recorded, otherwise audio frames should be
///< stored in buffer for delayed recording
unsigned
int
sleep_period_us
;
///< sleep period between frames while processing audio stream, in microseconds
void
(
*
get_fpga_time
)(
const
struct
audio
*
audio
,
struct
timeval
*
tv
);
//< callback function which can get FPGA time
int
(
*
write_samples
)(
struct
audio
*
audio
,
void
*
buff
,
long
len
,
long
slen
);
///< callback function which actually write data to file, this must be set
///< in the camogm_init_* function when appropriate format is selected
// === debug ===
struct
timeval
sf_timediff
;
// system to fpga time difference at the beginning of the stream
struct
timeval
m_len
;
struct
timeval
sys_fpga_timediff
;
int
avail_samples
;
long
calc_frames
;
// calculated number of frames by current video frame
struct
timeval
prev_ts_video
;
long
long
skip_samples
;
// === end of debug ===
};
void
audio_init
(
struct
audio
*
audio
,
bool
restart
);
void
audio_
start
(
struct
audio
*
audio
);
void
audio_init
_hw
(
struct
audio
*
audio
,
bool
restart
);
void
audio_
init_sw
(
struct
audio
*
audio
,
bool
restart
,
int
frames
);
void
audio_process
(
struct
audio
*
audio
);
void
audio_finish
(
struct
audio
*
audio
,
bool
reset
);
void
audio_set_volume
(
int
nvolume
);
unsigned
long
audio_get_hw_buffer_max
(
void
);
#endif
/* _CAMOGM_AUDIO_H */
src/camogm_mov.c
View file @
313f638e
...
...
@@ -26,11 +26,7 @@
#include <sys/types.h>
#include <assert.h>
// for debug only
#include <math.h>
#include "camogm_mov.h"
#include "thelper.h"
/** @brief QuickTime header length (w/o index tables) enough to accommodate static data */
#define QUICKTIME_MIN_HEADER 0x300
...
...
@@ -228,9 +224,9 @@ int camogm_frame_mov(camogm_state *state)
/**
* Write audio samples to file.
* @param[in] buff pointer to buffer containing audio
sampl
es
* @param[in] buff pointer to buffer containing audio
fram
es
* @param[in] len the size of buffer, in bytes
* @param[in] slen the number of audio
sampl
es in buffer
* @param[in] slen the number of audio
fram
es in buffer
* @return 0 if data was recorded successfully and negative error code otherwise
*/
static
int
camogm_audio_mov
(
struct
audio
*
audio
,
void
*
buff
,
long
len
,
long
slen
)
...
...
@@ -240,7 +236,7 @@ static int camogm_audio_mov(struct audio *audio, void *buff, long len, long slen
ssize_t
wr_len
;
camogm_state
*
state
=
container_of
(
audio
,
camogm_state
,
audio
);
D6
(
fprintf
(
debug_file
,
"write audio
sample
, len = %ld, slen = %ld
\n
"
,
len
,
slen
));
D6
(
fprintf
(
debug_file
,
"write audio
chunk
, len = %ld, slen = %ld
\n
"
,
len
,
slen
));
wr_len
=
write
(
state
->
ivf
,
buff
,
len
);
if
(
wr_len
<
len
)
{
...
...
@@ -289,19 +285,15 @@ int camogm_end_mov(camogm_state *state)
q_template
,
// string containing header template
state
->
ivf
,
// output file descriptor (opened)
state
->
width
,
// width in pixels
state
->
height
,
state
->
height
,
// height in pixels
state
->
frameno
,
// the number of image frames
state
->
frame_period
[
port
]
/
(
1000000
/
timescale
),
state
->
frames_per_chunk
,
0
,
// frame size - will look in the table
(
int
)((
float
)
timescale
/
(
state
->
timescale
)),
// state->frame_lengths, // array of frame lengths to build an index
NULL
,
// array of frame lengths to build an index
state
->
frame_data_start
);
// === debug code ===
fprintf
(
debug_file
,
"total # of video frames: %d, total # of audio samples: %ld
\n
"
,
state
->
frameno
,
state
->
audio
.
audio_samples
);
// === end of debug ===
close
(
state
->
ivf
);
state
->
ivf
=
-
1
;
// free memory used for index
...
...
@@ -640,11 +632,6 @@ int quicktime_template_parser( camogm_state *state,
iFileLen
=
strlen
(
iFile
);
lseek
(
ofd
,
0
,
SEEK_SET
);
// === debug ===
struct
timeval
m_len
=
state
->
audio
.
m_len
;
// duration of movie
fprintf
(
debug_file
,
"frameno: %d, duration: %ld:%06ld, audio_samples: %ld
\n
"
,
state
->
frameno
,
m_len
.
tv_sec
,
m_len
.
tv_usec
,
state
->
audio
.
audio_samples
);
// === ebd of debug ===
audio_timescale
=
state
->
audio
.
audio_rate
;
audio_rate
=
audio_timescale
;
// QuickTime defines sample rate as unsigned 16.16 fixed-point number
audio_rate
<<=
16
;
...
...
src/thelper.c
View file @
313f638e
...
...
@@ -51,36 +51,3 @@ int time_comp(struct timeval *t1, struct timeval *t2)
}
return
-
1
;
}
/**
* Subtract one time value from another and return the difference
* @param tv1 time value to subtract from
* @param tv2 time value to be subtracted
* @return tv1 - tv2
*/
struct
timeval
time_sub
(
const
struct
timeval
*
tv1
,
const
struct
timeval
*
tv2
)
{
struct
timeval
ret_val
=
*
tv1
;
ret_val
.
tv_sec
-=
1
;
ret_val
.
tv_usec
+=
1000000
;
ret_val
.
tv_sec
-=
tv2
->
tv_sec
;
ret_val
.
tv_usec
-=
tv2
->
tv_usec
;
time_normalize
(
&
ret_val
);
return
ret_val
;
}
/**
* Add one time value to another and return the sum
*/
struct
timeval
time_add
(
const
struct
timeval
*
tv1
,
const
struct
timeval
*
tv2
)
{
struct
timeval
ret_val
=
*
tv1
;
ret_val
.
tv_sec
+=
tv2
->
tv_sec
;
ret_val
.
tv_usec
+=
tv2
->
tv_usec
;
time_normalize
(
&
ret_val
);
return
ret_val
;
}
src/thelper.h
View file @
313f638e
...
...
@@ -26,8 +26,6 @@
void
time_normalize
(
struct
timeval
*
tv
);
int
time_comp
(
struct
timeval
*
t1
,
struct
timeval
*
t2
);
struct
timeval
time_sub
(
const
struct
timeval
*
tv1
,
const
struct
timeval
*
tv2
);
struct
timeval
time_add
(
const
struct
timeval
*
tv1
,
const
struct
timeval
*
tv2
);
/**
* Convert time represented by timeval structure to time in microseconds
...
...
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