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elphel-apps-camogm
Commit ecb78ce2 authored by Mikhail Karpenko's avatar Mikhail Karpenko
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WIP: write audio samples to file 

Audio samples are recorded after each frame which brakes the structure
of video chunks in case when frames_per_chunk > 1. This will be fixed
later.
parent c5b381e3
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Showing with 82 additions and 50 deletions
src/camogm_audio.c
View file @ ecb78ce2
...@@ -53,8 +53,8 @@ void audio_init(struct audio *audio, bool restart) ...@@ -53,8 +53,8 @@ void audio_init(struct audio *audio, bool restart)
int err = 0; int err = 0;
bool init_ok = false; bool init_ok = false;
unsigned int t = audio->audio_rate; unsigned int t = audio->audio_rate;
unsigned int period_time = 40 * 1000; unsigned int period_time = SAMPLE_TIME * 1000;
unsigned int buffer_time = 2000000; unsigned int buffer_time = BUFFER_TIME * 1000;
snd_pcm_hw_params_t *hw_params; // allocated on stack, do not free snd_pcm_hw_params_t *hw_params; // allocated on stack, do not free
snd_pcm_sw_params_t *sw_params; // allocated on stack, do not free snd_pcm_sw_params_t *sw_params; // allocated on stack, do not free
snd_pcm_status_t *status; // allocated on stack, do not free snd_pcm_status_t *status; // allocated on stack, do not free
...@@ -65,7 +65,7 @@ void audio_init(struct audio *audio, bool restart) ...@@ -65,7 +65,7 @@ void audio_init(struct audio *audio, bool restart)
audio->ctx_a.sbuffer_len -= audio->ctx_a.sbuffer_len % 2; audio->ctx_a.sbuffer_len -= audio->ctx_a.sbuffer_len % 2;
// 'while' loop here just to break initialization sequence after an error // 'while' loop here just to break initialization sequence after an error
while (true) { while (true) {
audio->ctx_a.sbuffer = (void *)malloc(audio->ctx_a.sbuffer_len * 8 + AUDIO_SBUFFER_PREFIX); audio->ctx_a.sbuffer = (void *)malloc(audio->ctx_a.sbuffer_len * audio->audio_channels * AUDIO_BPS);
if (audio->ctx_a.sbuffer == NULL) { if (audio->ctx_a.sbuffer == NULL) {
D0(fprintf(debug_file, "error: can not allocate buffer for audio samples: %s\n", strerror(errno))); D0(fprintf(debug_file, "error: can not allocate buffer for audio samples: %s\n", strerror(errno)));
break; break;
...@@ -170,14 +170,16 @@ void audio_start(struct audio *audio) ...@@ -170,14 +170,16 @@ void audio_start(struct audio *audio)
*/ */
void audio_process(struct audio *audio) void audio_process(struct audio *audio)
{ {
int slen; snd_pcm_sframes_t slen;
int counter = 0; int counter = 0;
void *_buf; void *_buf;
int _buf_len; long _buf_len;
struct timeval tv_sys; struct timeval tv_sys;
snd_timestamp_t ts; snd_timestamp_t ts;
snd_pcm_status_t *status; // allocated on stack, do not free snd_pcm_status_t *status; // allocated on stack, do not free
assert(audio->write_samples);
if (audio->audio_enable == 0) if (audio->audio_enable == 0)
return; return;
...@@ -193,7 +195,7 @@ void audio_process(struct audio *audio) ...@@ -193,7 +195,7 @@ void audio_process(struct audio *audio)
snd_pcm_status(audio->ctx_a.capture_hnd, status); snd_pcm_status(audio->ctx_a.capture_hnd, status);
snd_pcm_status_get_tstamp(status, &ts); snd_pcm_status_get_tstamp(status, &ts);
avail = snd_pcm_status_get_avail(status); avail = snd_pcm_status_get_avail(status);
int to_read = audio->ctx_a.sbuffer_len; // length in samples snd_pcm_uframes_t to_read = audio->ctx_a.sbuffer_len; // length in samples
if (audio->ctx_a.rem_samples < 0) if (audio->ctx_a.rem_samples < 0)
audio->ctx_a.rem_samples = 0; audio->ctx_a.rem_samples = 0;
if (avail >= audio->ctx_a.sbuffer_len && audio->ctx_a.rem_samples == 0) if (avail >= audio->ctx_a.sbuffer_len && audio->ctx_a.rem_samples == 0)
...@@ -214,7 +216,7 @@ void audio_process(struct audio *audio) ...@@ -214,7 +216,7 @@ void audio_process(struct audio *audio)
} }
} }
if (to_push_flag) { if (to_push_flag) {
slen = snd_pcm_readi(audio->ctx_a.capture_hnd, (void *)(audio->ctx_a.sbuffer + AUDIO_SBUFFER_PREFIX), to_read); slen = snd_pcm_readi(audio->ctx_a.capture_hnd, (void *)audio->ctx_a.sbuffer, to_read);
if (slen > 0) { if (slen > 0) {
int flag = 1; int flag = 1;
long offset = 0; long offset = 0;
...@@ -238,12 +240,12 @@ void audio_process(struct audio *audio) ...@@ -238,12 +240,12 @@ void audio_process(struct audio *audio)
break; break;
} }
} }
// we need to skip some samples in a new session, but if we just switch the frames than // we need to skip some samples in a new session, but if we just switch the frames then
// we need to split new samples in the buffer into two parts - for the previous file, // we need to split new samples in the buffer into two parts - for the previous file,
// and the next one... // and the next one...
// so we can just save in the first file new data, and in the next use "skip_samples" field // so we can just save in the first file new data, and in the next use "skip_samples" field
if (audio->ctx_a.audio_skip_samples != 0) { if (audio->ctx_a.audio_skip_samples != 0) {
D5(fprintf(debug_file, "skip_samples = %lld, available samples = %d\n", audio->ctx_a.audio_skip_samples, slen)); D5(fprintf(debug_file, "skip_samples = %lld, available samples = %ld\n", audio->ctx_a.audio_skip_samples, slen));
if (audio->ctx_a.audio_skip_samples >= slen) { if (audio->ctx_a.audio_skip_samples >= slen) {
audio->ctx_a.audio_skip_samples -= slen; audio->ctx_a.audio_skip_samples -= slen;
flag = 0; flag = 0;
...@@ -255,11 +257,10 @@ void audio_process(struct audio *audio) ...@@ -255,11 +257,10 @@ void audio_process(struct audio *audio)
if (flag) { if (flag) {
long samples = slen - offset; long samples = slen - offset;
audio->ctx_a.audio_count += samples; audio->ctx_a.audio_count += samples;
_buf = (void *) (audio->ctx_a.sbuffer + AUDIO_SBUFFER_PREFIX); _buf = (void *)audio->ctx_a.sbuffer;
_buf = (void *) ((char *) _buf + offset * 2 * audio->audio_channels); _buf = (void *)((char *) _buf + offset * AUDIO_BPS * audio->audio_channels);
_buf_len = samples * 2 * audio->audio_channels; _buf_len = samples * AUDIO_BPS * audio->audio_channels;
assert(audio->write_samples); audio->write_samples(audio, _buf, _buf_len, samples);
audio->write_samples(_buf, _buf_len, samples);
float tr = 1.0 / audio->audio_rate; float tr = 1.0 / audio->audio_rate;
float l = tr * audio->ctx_a.audio_count; float l = tr * audio->ctx_a.audio_count;
...@@ -274,6 +275,7 @@ void audio_process(struct audio *audio) ...@@ -274,6 +275,7 @@ void audio_process(struct audio *audio)
} }
} }
} else { } else {
D3(fprintf(debug_file, "error reading from ALSA buffer, error code %ld\n", slen));
break; break;
} }
} }
......
src/camogm_audio.h
View file @ ecb78ce2
...@@ -27,14 +27,15 @@ ...@@ -27,14 +27,15 @@
#define SAMPLE_RATE 44100 #define SAMPLE_RATE 44100
#define SAMPLE_CHANNELS 2 #define SAMPLE_CHANNELS 2
#define SAMPLE_TIME 200 #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" #define DEFAULT_SND_DEVICE "plughw:0,0"
#define AUDIO_SBUFFER_PREFIX 16
#define AUDIO_CHANNELS_MIN 1 #define AUDIO_CHANNELS_MIN 1
#define AUDIO_CHANNELS_MAX 2 #define AUDIO_CHANNELS_MAX 2
#define AUDIO_RATE_MIN 11025 #define AUDIO_RATE_MIN 11025
#define AUDIO_RATE_MAX 44100 #define AUDIO_RATE_MAX 44100
#define DEFAULT_AUDIO_VOLUME 0xffff #define DEFAULT_AUDIO_VOLUME 0xffff
#define AUDIO_BPS 2 ///< bytes per sample for a single channel (can be 1 or 2)
struct context_audio { struct context_audio {
char *sbuffer; ///< buffer for audio samples char *sbuffer; ///< buffer for audio samples
...@@ -78,7 +79,7 @@ struct audio { ...@@ -78,7 +79,7 @@ struct audio {
int frame_period; ///< video frame period, used to calculate time stamps for audio samples int frame_period; ///< video frame period, used to calculate time stamps for audio samples
void (*get_fpga_time)(const struct audio *audio, struct timeval *tv);//< callback function which can get FPGA time void (*get_fpga_time)(const struct audio *audio, struct timeval *tv);//< callback function which can get FPGA time
int (*write_samples)(void *buff, int len, int slen); ///< callback function which actually write data to file, this must be set 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 ///< in the camogm_init_* function when appropriate format is selected
}; };
......
src/camogm_mov.c
View file @ ecb78ce2
...@@ -32,7 +32,7 @@ ...@@ -32,7 +32,7 @@
#define QUICKTIME_MIN_HEADER 0x300 #define QUICKTIME_MIN_HEADER 0x300
/** @brief The length in bytes of sample-to-chunk table entry as defined in QuickTime format specification */ /** @brief The length in bytes of sample-to-chunk table entry as defined in QuickTime format specification */
#define S2C_ENTRY_LEN 12 #define S2C_ENTRY_LEN 12
/** @brief The number of entries in sample-to-chunk table */ /** @brief The number of entries in sample-to-chunk table. See camogm_start_mov for the reason why we need 3 entries. */
#define S2C_ENTRIES 3 #define S2C_ENTRIES 3
// for the parser // for the parser
...@@ -76,7 +76,7 @@ int quicktime_template_parser(camogm_state *state, ...@@ -76,7 +76,7 @@ int quicktime_template_parser(camogm_state *state,
void putBigEndian(unsigned long d, int l); void putBigEndian(unsigned long d, int l);
int parse_special(camogm_state *state); int parse_special(camogm_state *state);
int parse(camogm_state *state, int top); int parse(camogm_state *state, int top);
static int camogm_audio_mov(void *buff, int len, int slen); static int camogm_audio_mov(struct audio *audio, void *buff, long len, long slen);
static inline bool is_audio_frame(unsigned long len); static inline bool is_audio_frame(unsigned long len);
static inline void mark_audio(unsigned long *len); static inline void mark_audio(unsigned long *len);
static inline void unmark_audio(unsigned long *len); static inline void unmark_audio(unsigned long *len);
...@@ -146,6 +146,12 @@ int camogm_start_mov(camogm_state *state) ...@@ -146,6 +146,12 @@ int camogm_start_mov(camogm_state *state)
state->frame_index = 0; state->frame_index = 0;
if (audio->audio_enable) { if (audio->audio_enable) {
/* Allocate memory for sample-to-chunk buffers. For simplicity, all audio chunks must be the same size and
* we enforce this by reading from ALSA buffer (see camogm_audio.c ) only when it contains the appropriate
* number of samples. Such approach simplifies the building of sample-to-chunk atoms, although there are
* two corner cases: the first and the last chunks in file can contain different number of samples, thus we
* need 3 entries in total (first, last and all in between). That is why S2C_ENTRIES = 3.
*/
audio->audio_samples_to_chunk = malloc(S2C_ENTRY_LEN * S2C_ENTRIES); audio->audio_samples_to_chunk = malloc(S2C_ENTRY_LEN * S2C_ENTRIES);
if (!audio->audio_samples_to_chunk) { if (!audio->audio_samples_to_chunk) {
return -CAMOGM_FRAME_MALLOC; return -CAMOGM_FRAME_MALLOC;
...@@ -168,7 +174,7 @@ int camogm_start_mov(camogm_state *state) ...@@ -168,7 +174,7 @@ int camogm_start_mov(camogm_state *state)
*/ */
data_offset = QUICKTIME_MIN_HEADER + 16; data_offset = QUICKTIME_MIN_HEADER + 16;
data_offset += 4 * state->max_frames; // space for sample size atom - video data_offset += 4 * state->max_frames; // space for sample size atom - video
data_offset += 4 * state->max_frames; // space for chunk offsets atom - video data_offset += (4 * state->max_frames) / state->frames_per_chunk; // space for chunk offsets atom - video
if (audio->audio_enable) { if (audio->audio_enable) {
data_offset += 4 * state->max_frames; // space for chunk offsets atom - audio data_offset += 4 * state->max_frames; // space for chunk offsets atom - audio
data_offset += S2C_ENTRY_LEN * S2C_ENTRIES; // space for samples size atom - audio data_offset += S2C_ENTRY_LEN * S2C_ENTRIES; // space for samples size atom - audio
...@@ -188,7 +194,7 @@ int camogm_start_mov(camogm_state *state) ...@@ -188,7 +194,7 @@ int camogm_start_mov(camogm_state *state)
*/ */
int camogm_frame_mov(camogm_state *state) int camogm_frame_mov(camogm_state *state)
{ {
int ret; int ret = 0;
int i, j; int i, j;
ssize_t iovlen, l; ssize_t iovlen, l;
struct iovec chunks_iovec[7]; struct iovec chunks_iovec[7];
...@@ -218,17 +224,43 @@ int camogm_frame_mov(camogm_state *state) ...@@ -218,17 +224,43 @@ int camogm_frame_mov(camogm_state *state)
/** /**
* Write audio samples to file. * Write audio samples to file.
* @param buff * @param[in] buff pointer to buffer containing audio samples
* @param len * @param[in] len the size of buffer, in bytes
* @param slen * @param[in] slen the number of audio samples in buffer
* @return * @return 0 if data was recorded successfully and negative error code otherwise
*/ */
static int camogm_audio_mov(void *buff, int len, int slen) static int camogm_audio_mov(struct audio *audio, void *buff, long len, long slen)
{ {
int ret_val = 0; int ret_val = 0;
unsigned long k;
ssize_t wr_len;
camogm_state *state = container_of(audio, camogm_state, audio);
D6(fprintf(debug_file, "write audio sample, len = %d, slen = %d\n", len, slen)); D6(fprintf(debug_file, "write audio sample, len = %d, slen = %d\n", len, slen));
wr_len = write(state->ivf, buff, len);
if (wr_len < len) {
D0(fprintf(debug_file, "audio samples write error: %s; returned %d, expected %d\n", strerror(errno), wr_len, len));
close(state->ivf);
state->ivf = -1;
return CAMOGM_FRAME_FILE_ERR;
}
k = len;
mark_audio(&k);
state->frame_lengths[state->frame_index] = k;
state->frame_index++;
if (audio->audio_samples_to_chunk[0] == -1) {
// this slot contains the number of samples in first chunk in file
audio->audio_samples_to_chunk[0] = slen;
} else {
// these slots contain the number of samples in the last and in the one before last chunks
audio->audio_samples_to_chunk[1] = audio->audio_samples_to_chunk[2];
audio->audio_samples_to_chunk[2] = slen;
}
audio->audio_frameno++;
audio->audio_samples += slen;
return ret_val; return ret_val;
} }
...@@ -350,7 +382,7 @@ int parse_special(camogm_state *state) ...@@ -350,7 +382,7 @@ int parse_special(camogm_state *state)
if (strcmp(str, "frame_duration") == 0) { if (strcmp(str, "frame_duration") == 0) {
putBigEndian(sample_dur, 4); return 0; putBigEndian(sample_dur, 4); return 0;
} }
if (strcmp(str, "samples_chunk") == 0) { if (strcmp(str, "samples_chunk") == 0) { // 'stsc' video atom
putBigEndian(samplesPerChunk, 4); return 0; putBigEndian(samplesPerChunk, 4); return 0;
} // will put zeroes on pass 1 } // will put zeroes on pass 1
...@@ -383,15 +415,17 @@ int parse_special(camogm_state *state) ...@@ -383,15 +415,17 @@ int parse_special(camogm_state *state)
putBigEndian(state->audio.audio_channels * 2, 4); putBigEndian(state->audio.audio_channels * 2, 4);
return 0; return 0;
} }
if (strcmp(str, "audio_stsz") == 0) { if (strcmp(str, "audio_stsz") == 0) {
putBigEndian(state->audio.audio_channels * 2, 4); putBigEndian(state->audio.audio_channels * 2, 4);
putBigEndian(state->audio.audio_samples, 4); /* sample size table in 'stsz' atom contains entry for every sample, sound samples are
* all the same size thus this table is not needed - put 0 as the number of entries here
*/
putBigEndian(0, 4);
return 0; return 0;
} }
if (strcmp(str, "audio_stco") == 0) { // (4 + 4 * chunk_count) bytes if (strcmp(str, "audio_stco") == 0) {
long offset = 0; long offset = 0;
n = state->audio.audio_frameno; n = state->audio.audio_frameno;
//fprintf(stderr, "chunk_offsets; n == %d; nframes == %d; samplesPerChunk == %d\n", n, nframes, samplesPerChunk);
putBigEndian(n, 4); putBigEndian(n, 4);
j = 0; j = 0;
for (i = 0; i < state->frame_index; i++) { for (i = 0; i < state->frame_index; i++) {
...@@ -408,18 +442,19 @@ int parse_special(camogm_state *state) ...@@ -408,18 +442,19 @@ int parse_special(camogm_state *state)
D0(fprintf(debug_file, "Error MOV: wrong records for \"audio_stco\", have written %d, need to write %d\n", j, n)); D0(fprintf(debug_file, "Error MOV: wrong records for \"audio_stco\", have written %d, need to write %d\n", j, n));
return 0; return 0;
} }
// TODO!!!
if (strcmp(str, "audio_stsc") == 0) { if (strcmp(str, "audio_stsc") == 0) {
n = 0; n = 0;
for (i = 0; i < S2C_ENTRIES; i++) { for (int entry = 0; entry < S2C_ENTRIES; entry++) {
if (state->audio.audio_samples_to_chunk[i] != -1) { if (state->audio.audio_samples_to_chunk[entry] != -1) {
n++; n++;
} }
} }
putBigEndian(n, 4); putBigEndian(n, 4);
// first table entry refers to first audio chunk in file
putBigEndian(1, 4); putBigEndian(1, 4);
putBigEndian(state->audio.audio_samples_to_chunk[0], 4); putBigEndian(state->audio.audio_samples_to_chunk[0], 4);
putBigEndian(01, 4); // TODO: 02 ??? putBigEndian(01, 4);
// second table entry, most chunks in file refer here
n = 2; n = 2;
if (state->audio.audio_samples_to_chunk[1] != -1) { if (state->audio.audio_samples_to_chunk[1] != -1) {
putBigEndian(n, 4); putBigEndian(n, 4);
...@@ -427,6 +462,7 @@ int parse_special(camogm_state *state) ...@@ -427,6 +462,7 @@ int parse_special(camogm_state *state)
putBigEndian(01, 4); putBigEndian(01, 4);
n = state->audio.audio_frameno; n = state->audio.audio_frameno;
} }
// last table entry corresponds to the last audio chunk in file
if (state->audio.audio_samples_to_chunk[2] != -1) { if (state->audio.audio_samples_to_chunk[2] != -1) {
putBigEndian(n, 4); putBigEndian(n, 4);
putBigEndian(state->audio.audio_samples_to_chunk[2], 4); putBigEndian(state->audio.audio_samples_to_chunk[2], 4);
...@@ -434,10 +470,6 @@ int parse_special(camogm_state *state) ...@@ -434,10 +470,6 @@ int parse_special(camogm_state *state)
} }
return 0; return 0;
} }
if (strcmp(str, "audio_samples_chunk") == 0) {
putBigEndian(1, 4);
return 0;
} // will put zeroes on pass 1
if (strcmp(str, "sample_sizes") == 0) { // 'stsz' video atom if (strcmp(str, "sample_sizes") == 0) { // 'stsz' video atom
// index for video stream only, audio index is build separately // index for video stream only, audio index is build separately
j = 0; j = 0;
...@@ -449,7 +481,7 @@ int parse_special(camogm_state *state) ...@@ -449,7 +481,7 @@ int parse_special(camogm_state *state)
} }
} }
if (j != nframes) if (j != nframes)
D0(fprintf(debug_file, "Error MOV: wrong records for \"samples_sizes\": have write: %d, need to write: %d\n", j, n)); D0(fprintf(debug_file, "Error MOV: wrong records for \"samples_sizes\": have write: %d, need to write: %d\n", j, nframes));
return 0; return 0;
} }
if (strcmp(str, "chunk_offsets") == 0) { // 'stco' video atom if (strcmp(str, "chunk_offsets") == 0) { // 'stco' video atom
...@@ -602,7 +634,7 @@ int quicktime_template_parser( camogm_state *state, ...@@ -602,7 +634,7 @@ int quicktime_template_parser( camogm_state *state,
lseek(ofd, 0, SEEK_SET); lseek(ofd, 0, SEEK_SET);
audio_timescale = state->audio.audio_rate; audio_timescale = state->audio.audio_rate;
audio_rate = audio_timescale; audio_rate = audio_timescale; // QuickTime defines sample rate as unsigned 16.16 fixed-point number
audio_rate <<= 16; audio_rate <<= 16;
audio_duration = state->audio.audio_samples; audio_duration = state->audio.audio_samples;
audio_channels = state->audio.audio_channels; audio_channels = state->audio.audio_channels;
......
src/qt_audio
View file @ ecb78ce2
...@@ -177,7 +177,7 @@ ...@@ -177,7 +177,7 @@
!time # Creation time !time # Creation time
!time # Modification time !time # Modification time
!audio_timescale # Time Scale - 44100 // ??? !audio_timescale # Time Scale - 44100 // ???
!audio_duration # Duration // ??? !audio_duration # Duration in units of the time scale
0000 # Language 0000 # Language
0000 # Quality 0000 # Quality
} # 'mdhd } # 'mdhd
...@@ -187,7 +187,6 @@ ...@@ -187,7 +187,6 @@
'mhlr # Component type 'mhlr # Component type
'soun # Component subtype 'soun # Component subtype
00000000 # Component manufacturer 00000000 # Component manufacturer
# 'niko # Component manufacturer
00000000 # Component Flags 00000000 # Component Flags
00000000 # Component Flags Mask 00000000 # Component Flags Mask
00 # Component name 00 # Component name
...@@ -233,8 +232,6 @@ ...@@ -233,8 +232,6 @@
000000 # Flags 000000 # Flags
00000001 # Number of entries 00000001 # Number of entries
{'sowt # 16 bit LE, Data format {'sowt # 16 bit LE, Data format
# 0001 # Version
# 0000 # Revision
00000000 0000 # (reserved) 00000000 0000 # (reserved)
0001 # Data reference index 0001 # Data reference index
0001 # Version 0001 # Version
...@@ -268,15 +265,15 @@ ...@@ -268,15 +265,15 @@
000000 # Flags 000000 # Flags
!audio_stsc # !audio_stsc #
# 00000000 # Number of entries # 00000000 # Number of entries
#[ 00000000 # first chunk #[ 00000000 # the first chunk number using this table entry (it starts from 1 in video)
#... 00000000 # samples per chunk #... 00000000 # the number of samples per chunk
#] 00000000 # samples description ID - 01 or 02 ? #] 00000000 # sample description ID; defines the entry in stsd table which describes this chunk (it starts from 1 in video)
} }
{'stsz # Sample size atom {'stsz # Sample size atom
00 # version 00 # version
000000 # Flags 000000 # Flags
!audio_stsz # Sample Size - 2 for Mono, 4 for Stereo - ? !audio_stsz # Sample Size - 2 for Mono, 4 for Stereo
# 00000000 # Number of entries - empty # 00000000 # Number of entries; this table should be empty if all the samples are the same size
} }
{'stco # Chunk offset atom {'stco # Chunk offset atom
00 # version 00 # version
......
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