jp9000 6c92cf5841 Implement output, improve video/audio subsystems
- Fill in the rest of the FFmpeg test output code for testing so it
   actually properly outputs data.

 - Improve the main video subsystem to be a bit more optimal and
   automatically output I420 or NV12 if needed.

 - Fix audio subsystem insertation and byte calculation.  Now it will
   seamlessly insert new audio data in to the audio stream based upon
   its timestamp value.  (Be extremely cautious when using floating
   point calculations for important things like this, and always round
   your values and check your values)

 - Use 32 byte alignment in case of future optimizations and export a
   function to get the current alignment.

 - Make os_sleepto_ns return true if slept, false if the time has
   already been passed before the call.

 - Fix sinewave output so that it actually properly calculates a middle
   C sinewave.

 - Change the use of row_bytes to linesize (also makes it a bit more
   consistent with FFmpeg's naming as well)
2014-02-09 05:51:06 -07:00

643 lines
17 KiB
C

/******************************************************************************
Copyright (C) 2013 by Hugh Bailey <obs.jim@gmail.com>
This program is free software: you can redistribute it and/or modify
it under the terms of the GNU General Public License as published by
the Free Software Foundation, either version 2 of the License, or
(at your option) any later version.
This program is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
GNU General Public License for more details.
You should have received a copy of the GNU General Public License
along with this program. If not, see <http://www.gnu.org/licenses/>.
******************************************************************************/
#include <math.h>
#include "../util/threading.h"
#include "../util/darray.h"
#include "../util/circlebuf.h"
#include "../util/platform.h"
#include "audio-io.h"
/* #define DEBUG_AUDIO */
#define nop() do {int invalid = 0;} while(0)
struct audio_input {
struct audio_convert_info conversion;
void (*callback)(void *param, const struct audio_data *data);
void *param;
};
struct audio_line {
char *name;
struct audio_output *audio;
struct circlebuf buffers[MAX_AUDIO_PLANES];
pthread_mutex_t mutex;
DARRAY(uint8_t) volume_buffers[MAX_AUDIO_PLANES];
uint64_t base_timestamp;
uint64_t last_timestamp;
/* states whether this line is still being used. if not, then when the
* buffer is depleted, it's destroyed */
bool alive;
struct audio_line **prev_next;
struct audio_line *next;
};
static inline void audio_line_destroy_data(struct audio_line *line)
{
for (size_t i = 0; i < MAX_AUDIO_PLANES; i++) {
circlebuf_free(&line->buffers[i]);
da_free(line->volume_buffers[i]);
}
pthread_mutex_destroy(&line->mutex);
bfree(line->name);
bfree(line);
}
struct audio_output {
struct audio_output_info info;
size_t block_size;
size_t channels;
size_t planes;
pthread_t thread;
event_t stop_event;
DARRAY(uint8_t) mix_buffers[MAX_AUDIO_PLANES];
bool initialized;
pthread_mutex_t line_mutex;
struct audio_line *first_line;
pthread_mutex_t input_mutex;
DARRAY(struct audio_input) inputs;
};
static inline void audio_output_removeline(struct audio_output *audio,
struct audio_line *line)
{
pthread_mutex_lock(&audio->line_mutex);
*line->prev_next = line->next;
if (line->next)
line->next->prev_next = line->prev_next;
pthread_mutex_unlock(&audio->line_mutex);
audio_line_destroy_data(line);
}
/* ------------------------------------------------------------------------- */
/* the following functions are used to calculate frame offsets based upon
* timestamps. this will actually work accurately as long as you handle the
* values correctly */
static inline double ts_to_frames(audio_t audio, uint64_t ts)
{
double audio_offset_d = (double)ts;
audio_offset_d /= 1000000000.0;
audio_offset_d *= (double)audio->info.samples_per_sec;
return audio_offset_d;
}
static inline double positive_round(double val)
{
return floor(val+0.5);
}
static size_t ts_diff_frames(audio_t audio, uint64_t ts1, uint64_t ts2)
{
double diff = ts_to_frames(audio, ts1) - ts_to_frames(audio, ts2);
return (size_t)positive_round(diff);
}
static size_t ts_diff_bytes(audio_t audio, uint64_t ts1, uint64_t ts2)
{
return ts_diff_frames(audio, ts1, ts2) * audio->block_size;
}
/* unless the value is 3+ hours worth of frames, this won't overflow */
static inline uint64_t conv_frames_to_time(audio_t audio, uint32_t frames)
{
return (uint64_t)frames * 1000000000ULL /
(uint64_t)audio->info.samples_per_sec;
}
/* ------------------------------------------------------------------------- */
static inline void clear_excess_audio_data(struct audio_line *line,
uint64_t prev_time)
{
size_t size = ts_diff_bytes(line->audio, prev_time,
line->base_timestamp);
blog(LOG_WARNING, "Excess audio data for audio line '%s', somehow "
"audio data went back in time by %lu bytes. "
"prev_time: %llu, line->base_timestamp: %llu",
line->name, (uint32_t)size,
prev_time, line->base_timestamp);
for (size_t i = 0; i < line->audio->planes; i++) {
size_t clear_size = (size > line->buffers[i].size) ?
(size_t)size : line->buffers[i].size;
circlebuf_pop_front(&line->buffers[i], NULL, clear_size);
}
}
static inline uint64_t min_uint64(uint64_t a, uint64_t b)
{
return a < b ? a : b;
}
static inline size_t min_size(size_t a, size_t b)
{
return a < b ? a : b;
}
/* TODO: this just overwrites. handle actual mixing */
static inline bool mix_audio_line(struct audio_output *audio,
struct audio_line *line, size_t size, uint64_t timestamp)
{
size_t time_offset = ts_diff_bytes(audio,
line->base_timestamp, timestamp);
if (time_offset > size)
return false;
size -= time_offset;
#ifdef DEBUG_AUDIO
blog(LOG_DEBUG, "shaved off %lu bytes", size);
#endif
for (size_t i = 0; i < audio->planes; i++) {
size_t pop_size = min_size(size, line->buffers[i].size);
circlebuf_pop_front(&line->buffers[i],
audio->mix_buffers[i].array + time_offset,
pop_size);
}
return true;
}
static inline void do_audio_output(struct audio_output *audio,
uint64_t timestamp, uint32_t frames)
{
struct audio_data data;
for (size_t i = 0; i < MAX_AUDIO_PLANES; i++)
data.data[i] = audio->mix_buffers[i].array;
data.frames = frames;
data.timestamp = timestamp;
data.volume = 1.0f;
/* TODO: conversion */
pthread_mutex_lock(&audio->input_mutex);
for (size_t i = 0; i < audio->inputs.num; i++) {
struct audio_input *input = audio->inputs.array+i;
input->callback(input->param, &data);
}
pthread_mutex_unlock(&audio->input_mutex);
}
static uint64_t mix_and_output(struct audio_output *audio, uint64_t audio_time,
uint64_t prev_time)
{
struct audio_line *line = audio->first_line;
uint32_t frames = (uint32_t)ts_diff_frames(audio, audio_time,
prev_time);
size_t bytes = frames * audio->block_size;
#ifdef DEBUG_AUDIO
blog(LOG_DEBUG, "audio_time: %llu, prev_time: %llu, bytes: %lu",
audio_time, prev_time, bytes);
#endif
/* return an adjusted audio_time according to the amount
* of data that was sampled to ensure seamless transmission */
audio_time = prev_time + conv_frames_to_time(audio, frames);
/* resize and clear mix buffers */
for (size_t i = 0; i < audio->planes; i++) {
da_resize(audio->mix_buffers[i], bytes);
memset(audio->mix_buffers[i].array, 0, bytes);
}
/* mix audio lines */
while (line) {
struct audio_line *next = line->next;
/* if line marked for removal, destroy and move to the next */
if (!line->buffers[0].size) {
if (!line->alive) {
audio_output_removeline(audio, line);
line = next;
continue;
}
}
pthread_mutex_lock(&line->mutex);
if (line->buffers[0].size && line->base_timestamp < prev_time) {
clear_excess_audio_data(line, prev_time);
line->base_timestamp = prev_time;
}
if (mix_audio_line(audio, line, bytes, prev_time))
line->base_timestamp = audio_time;
pthread_mutex_unlock(&line->mutex);
line = next;
}
/* output */
do_audio_output(audio, prev_time, frames);
return audio_time;
}
/* sample audio 40 times a second */
#define AUDIO_WAIT_TIME (1000/40)
static void *audio_thread(void *param)
{
struct audio_output *audio = param;
uint64_t buffer_time = audio->info.buffer_ms * 1000000;
uint64_t prev_time = os_gettime_ns() - buffer_time;
uint64_t audio_time;
while (event_try(&audio->stop_event) == EAGAIN) {
os_sleep_ms(AUDIO_WAIT_TIME);
pthread_mutex_lock(&audio->line_mutex);
audio_time = os_gettime_ns() - buffer_time;
audio_time = mix_and_output(audio, audio_time, prev_time);
prev_time = audio_time;
pthread_mutex_unlock(&audio->line_mutex);
}
return NULL;
}
/* ------------------------------------------------------------------------- */
static size_t audio_get_input_idx(audio_t video,
void (*callback)(void *param, const struct audio_data *data),
void *param)
{
for (size_t i = 0; i < video->inputs.num; i++) {
struct audio_input *input = video->inputs.array+i;
if (input->callback == callback && input->param == param)
return i;
}
return DARRAY_INVALID;
}
void audio_output_connect(audio_t audio,
struct audio_convert_info *conversion,
void (*callback)(void *param, const struct audio_data *data),
void *param)
{
pthread_mutex_lock(&audio->input_mutex);
if (audio_get_input_idx(audio, callback, param) == DARRAY_INVALID) {
struct audio_input input;
input.callback = callback;
input.param = param;
/* TODO: conversion */
if (conversion) {
input.conversion = *conversion;
} else {
input.conversion.format = audio->info.format;
input.conversion.speakers = audio->info.speakers;
input.conversion.samples_per_sec =
audio->info.samples_per_sec;
}
da_push_back(audio->inputs, &input);
}
pthread_mutex_unlock(&audio->input_mutex);
}
void audio_output_disconnect(audio_t audio,
void (*callback)(void *param, const struct audio_data *data),
void *param)
{
pthread_mutex_lock(&audio->input_mutex);
size_t idx = audio_get_input_idx(audio, callback, param);
if (idx != DARRAY_INVALID)
da_erase(audio->inputs, idx);
pthread_mutex_unlock(&audio->input_mutex);
}
static inline bool valid_audio_params(struct audio_output_info *info)
{
return info->format && info->name && info->samples_per_sec > 0 &&
info->speakers > 0;
}
int audio_output_open(audio_t *audio, struct audio_output_info *info)
{
struct audio_output *out;
pthread_mutexattr_t attr;
bool planar = is_audio_planar(info->format);
if (!valid_audio_params(info))
return AUDIO_OUTPUT_INVALIDPARAM;
out = bmalloc(sizeof(struct audio_output));
memset(out, 0, sizeof(struct audio_output));
memcpy(&out->info, info, sizeof(struct audio_output_info));
pthread_mutex_init_value(&out->line_mutex);
out->channels = get_audio_channels(info->speakers);
out->planes = planar ? out->channels : 1;
out->block_size = (planar ? 1 : out->channels) *
get_audio_bytes_per_channel(info->format);
if (pthread_mutexattr_init(&attr) != 0)
goto fail;
if (pthread_mutexattr_settype(&attr, PTHREAD_MUTEX_RECURSIVE) != 0)
goto fail;
if (pthread_mutex_init(&out->line_mutex, &attr) != 0)
goto fail;
if (pthread_mutex_init(&out->input_mutex, NULL) != 0)
goto fail;
if (event_init(&out->stop_event, EVENT_TYPE_MANUAL) != 0)
goto fail;
if (pthread_create(&out->thread, NULL, audio_thread, out) != 0)
goto fail;
out->initialized = true;
*audio = out;
return AUDIO_OUTPUT_SUCCESS;
fail:
audio_output_close(out);
return AUDIO_OUTPUT_FAIL;
}
void audio_output_close(audio_t audio)
{
void *thread_ret;
struct audio_line *line;
if (!audio)
return;
if (audio->initialized) {
event_signal(&audio->stop_event);
pthread_join(audio->thread, &thread_ret);
}
line = audio->first_line;
while (line) {
struct audio_line *next = line->next;
audio_line_destroy_data(line);
line = next;
}
for (size_t i = 0; i < MAX_AUDIO_PLANES; i++)
da_free(audio->mix_buffers[i]);
event_destroy(&audio->stop_event);
pthread_mutex_destroy(&audio->line_mutex);
bfree(audio);
}
audio_line_t audio_output_createline(audio_t audio, const char *name)
{
struct audio_line *line = bmalloc(sizeof(struct audio_line));
memset(line, 0, sizeof(struct audio_line));
line->alive = true;
line->audio = audio;
if (pthread_mutex_init(&line->mutex, NULL) != 0) {
blog(LOG_ERROR, "audio_output_createline: Failed to create "
"mutex");
bfree(line);
return NULL;
}
pthread_mutex_lock(&audio->line_mutex);
if (audio->first_line) {
audio->first_line->prev_next = &line->next;
line->next = audio->first_line;
}
line->prev_next = &audio->first_line;
audio->first_line = line;
pthread_mutex_unlock(&audio->line_mutex);
line->name = bstrdup(name ? name : "(unnamed audio line)");
return line;
}
const struct audio_output_info *audio_output_getinfo(audio_t audio)
{
return &audio->info;
}
void audio_line_destroy(struct audio_line *line)
{
if (line) {
if (!line->buffers[0].size)
audio_output_removeline(line->audio, line);
else
line->alive = false;
}
}
size_t audio_output_blocksize(audio_t audio)
{
return audio->block_size;
}
size_t audio_output_planes(audio_t audio)
{
return audio->planes;
}
size_t audio_output_channels(audio_t audio)
{
return audio->channels;
}
/* TODO: Optimization of volume multiplication functions */
static inline void mul_vol_u8bit(void *array, float volume, size_t total_num)
{
uint8_t *vals = array;
int16_t vol = (int16_t)(volume * 127.0f);
for (size_t i = 0; i < total_num; i++) {
int16_t val = (int16_t)(vals[i] ^ 0x80) << 8;
vals[i] = (uint8_t)((val * vol / 127) + 128);
}
}
static inline void mul_vol_16bit(void *array, float volume, size_t total_num)
{
uint16_t *vals = array;
int32_t vol = (int32_t)(volume * 32767.0f);
for (size_t i = 0; i < total_num; i++)
vals[i] = (int32_t)((int32_t)vals[i] * vol / 32767);
}
static inline float conv_24bit_to_float(uint8_t *vals)
{
int32_t val = ((int32_t)vals[0]) |
((int32_t)vals[1] << 8) |
((int32_t)vals[2] << 16);
if ((val & 0x800000) != 0)
val |= 0xFF000000;
return (float)val / 8388607.0f;
}
static inline void conv_float_to_24bit(float fval, uint8_t *vals)
{
int32_t val = (int32_t)(fval * 8388607.0f);
vals[0] = (val) & 0xFF;
vals[1] = (val >> 8) & 0xFF;
vals[2] = (val >> 16) & 0xFF;
}
static inline void mul_vol_24bit(void *array, float volume, size_t total_num)
{
uint8_t *vals = array;
for (size_t i = 0; i < total_num; i++) {
float val = conv_24bit_to_float(vals) * volume;
conv_float_to_24bit(val, vals);
vals += 3;
}
}
static inline void mul_vol_32bit(void *array, float volume, size_t total_num)
{
int32_t *vals = array;
for (size_t i = 0; i < total_num; i++) {
float val = (float)vals[i] / 2147483647.0f;
vals[i] = (int32_t)(val * volume / 2147483647.0f);
}
}
static inline void mul_vol_float(void *array, float volume, size_t total_num)
{
float *vals = array;
for (size_t i = 0; i < total_num; i++)
vals[i] *= volume;
}
static void audio_line_place_data_pos(struct audio_line *line,
const struct audio_data *data, size_t position)
{
bool planar = line->audio->planes > 1;
size_t total_num = data->frames * planar ? 1 : line->audio->channels;
size_t total_size = data->frames * line->audio->block_size;
for (size_t i = 0; i < line->audio->planes; i++) {
da_copy_array(line->volume_buffers[i], data->data[i],
total_size);
uint8_t *array = line->volume_buffers[i].array;
switch (line->audio->info.format) {
case AUDIO_FORMAT_U8BIT:
case AUDIO_FORMAT_U8BIT_PLANAR:
mul_vol_u8bit(array, data->volume, total_num);
break;
case AUDIO_FORMAT_16BIT:
case AUDIO_FORMAT_16BIT_PLANAR:
mul_vol_16bit(array, data->volume, total_num);
break;
case AUDIO_FORMAT_32BIT:
case AUDIO_FORMAT_32BIT_PLANAR:
mul_vol_32bit(array, data->volume, total_num);
break;
case AUDIO_FORMAT_FLOAT:
case AUDIO_FORMAT_FLOAT_PLANAR:
mul_vol_float(array, data->volume, total_num);
break;
case AUDIO_FORMAT_UNKNOWN:
blog(LOG_ERROR, "audio_line_place_data_pos: "
"Unknown format");
break;
}
circlebuf_place(&line->buffers[i], position,
line->volume_buffers[i].array, total_size);
}
}
void audio_line_place_data(struct audio_line *line,
const struct audio_data *data)
{
size_t pos = ts_diff_bytes(line->audio, data->timestamp,
line->base_timestamp);
#ifdef DEBUG_AUDIO
blog(LOG_DEBUG, "data->timestamp: %llu, line->base_timestamp: %llu, "
"pos: %lu, bytes: %lu, buf size: %lu",
data->timestamp, line->base_timestamp, pos,
data->frames * line->audio->block_size,
line->buffers[0].size);
#endif
audio_line_place_data_pos(line, data, pos);
}
void audio_line_output(audio_line_t line, const struct audio_data *data)
{
/* TODO: prevent insertation of data too far away from expected
* audio timing */
pthread_mutex_lock(&line->mutex);
if (!line->buffers[0].size) {
/* XXX: not entirely sure if this is the wisest course of
* action in all circumstances */
line->base_timestamp = data->timestamp -
line->audio->info.buffer_ms * 1000000;
audio_line_place_data(line, data);
} else if (line->base_timestamp <= data->timestamp) {
audio_line_place_data(line, data);
} else {
blog(LOG_DEBUG, "Bad timestamp for audio line '%s', "
"data->timestamp: %llu, "
"line->base_timestamp: %llu. This can "
"sometimes happen when there's a pause in "
"the threads.", line->name, data->timestamp,
line->base_timestamp);
}
pthread_mutex_unlock(&line->mutex);
}