jp9000 84e1f47ced (API Change) Add support for multiple audio mixers
API changed:
--------------------------

void obs_output_set_audio_encoder(
		obs_output_t *output,
		obs_encoder_t *encoder);

obs_encoder_t *obs_output_get_audio_encoder(
		const obs_output_t *output);

obs_encoder_t *obs_audio_encoder_create(
		const char *id,
		const char *name,
		obs_data_t *settings);

Changed to:
--------------------------

/* 'idx' specifies the track index of the output */
void obs_output_set_audio_encoder(
		obs_output_t *output,
		obs_encoder_t *encoder,
		size_t idx);

/* 'idx' specifies the track index of the output */
obs_encoder_t *obs_output_get_audio_encoder(
		const obs_output_t *output,
		size_t idx);

/* 'mixer_idx' specifies the mixer index to capture audio from */
obs_encoder_t *obs_audio_encoder_create(
		const char *id,
		const char *name,
		obs_data_t *settings,
		size_t mixer_idx);

Overview
--------------------------
This feature allows multiple audio mixers to be used at a time.  This
capability was able to be added with surprisingly very little extra
overhead.  Audio will not be mixed unless it's assigned to a specific
mixer, and mixers will not mix unless they have an active mix
connection.

Mostly this will be useful for being able to separate out specific audio
for recording versus streaming, but will also be useful for certain
streaming services that support multiple audio streams via RTMP.

I didn't want to use a variable amount of mixers due to the desire to
reduce heap allocations, so currently I set the limit to 4 simultaneous
mixers; this number can be increased later if needed, but honestly I
feel like it's just the right number to use.

Sources:

Sources can now specify which audio mixers their audio is mixed to; this
can be a single mixer or multiple mixers at a time.  The
obs_source_set_audio_mixers function sets the audio mixer which an audio
source applies to.  For example, 0xF would mean that the source applies
to all four mixers.

Audio Encoders:

Audio encoders now must specify which specific audio mixer they use when
they encode audio data.

Outputs:

Outputs that use encoders can now support multiple audio tracks at once
if they have the OBS_OUTPUT_MULTI_TRACK capability flag set.  This is
mostly only useful for certain types of RTMP transmissions, though may
be useful for file formats that support multiple audio tracks as well
later on.
2015-02-04 16:51:29 -08:00

615 lines
16 KiB
C

/******************************************************************************
Copyright (C) 2014 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 <obs-module.h>
#include <obs-avc.h>
#include <util/platform.h>
#include <util/circlebuf.h>
#include <util/dstr.h>
#include <util/threading.h>
#include <inttypes.h>
#include "librtmp/rtmp.h"
#include "librtmp/log.h"
#include "flv-mux.h"
#define do_log(level, format, ...) \
blog(level, "[rtmp stream: '%s'] " format, \
obs_output_get_name(stream->output), ##__VA_ARGS__)
#define warn(format, ...) do_log(LOG_WARNING, format, ##__VA_ARGS__)
#define info(format, ...) do_log(LOG_INFO, format, ##__VA_ARGS__)
#define debug(format, ...) do_log(LOG_DEBUG, format, ##__VA_ARGS__)
#define OPT_DROP_THRESHOLD "drop_threshold_ms"
//#define TEST_FRAMEDROPS
struct rtmp_stream {
obs_output_t *output;
pthread_mutex_t packets_mutex;
struct circlebuf packets;
bool sent_headers;
bool connecting;
pthread_t connect_thread;
bool active;
pthread_t send_thread;
os_sem_t *send_sem;
os_event_t *stop_event;
struct dstr path, key;
struct dstr username, password;
/* frame drop variables */
int64_t drop_threshold_usec;
int64_t min_drop_dts_usec;
int min_priority;
int64_t last_dts_usec;
uint64_t total_bytes_sent;
int dropped_frames;
RTMP rtmp;
};
static const char *rtmp_stream_getname(void)
{
return obs_module_text("RTMPStream");
}
static void log_rtmp(int level, const char *format, va_list args)
{
if (level > RTMP_LOGWARNING)
return;
blogva(LOG_INFO, format, args);
}
static inline void free_packets(struct rtmp_stream *stream)
{
while (stream->packets.size) {
struct encoder_packet packet;
circlebuf_pop_front(&stream->packets, &packet, sizeof(packet));
obs_free_encoder_packet(&packet);
}
}
static void rtmp_stream_stop(void *data);
static void rtmp_stream_destroy(void *data)
{
struct rtmp_stream *stream = data;
if (stream->active)
rtmp_stream_stop(data);
if (stream) {
free_packets(stream);
dstr_free(&stream->path);
dstr_free(&stream->key);
dstr_free(&stream->username);
dstr_free(&stream->password);
os_event_destroy(stream->stop_event);
os_sem_destroy(stream->send_sem);
pthread_mutex_destroy(&stream->packets_mutex);
circlebuf_free(&stream->packets);
bfree(stream);
}
}
static void *rtmp_stream_create(obs_data_t *settings, obs_output_t *output)
{
struct rtmp_stream *stream = bzalloc(sizeof(struct rtmp_stream));
stream->output = output;
pthread_mutex_init_value(&stream->packets_mutex);
RTMP_Init(&stream->rtmp);
RTMP_LogSetCallback(log_rtmp);
RTMP_LogSetLevel(RTMP_LOGWARNING);
if (pthread_mutex_init(&stream->packets_mutex, NULL) != 0)
goto fail;
if (os_event_init(&stream->stop_event, OS_EVENT_TYPE_MANUAL) != 0)
goto fail;
UNUSED_PARAMETER(settings);
return stream;
fail:
rtmp_stream_destroy(stream);
return NULL;
}
static void rtmp_stream_stop(void *data)
{
struct rtmp_stream *stream = data;
void *ret;
os_event_signal(stream->stop_event);
if (stream->connecting)
pthread_join(stream->connect_thread, &ret);
if (stream->active) {
obs_output_end_data_capture(stream->output);
os_sem_post(stream->send_sem);
pthread_join(stream->send_thread, &ret);
RTMP_Close(&stream->rtmp);
}
os_event_reset(stream->stop_event);
stream->sent_headers = false;
}
static inline void set_rtmp_str(AVal *val, const char *str)
{
bool valid = (str && *str);
val->av_val = valid ? (char*)str : NULL;
val->av_len = valid ? (int)strlen(str) : 0;
}
static inline void set_rtmp_dstr(AVal *val, struct dstr *str)
{
bool valid = !dstr_is_empty(str);
val->av_val = valid ? str->array : NULL;
val->av_len = valid ? (int)str->len : 0;
}
static inline bool get_next_packet(struct rtmp_stream *stream,
struct encoder_packet *packet)
{
bool new_packet = false;
pthread_mutex_lock(&stream->packets_mutex);
if (stream->packets.size) {
circlebuf_pop_front(&stream->packets, packet,
sizeof(struct encoder_packet));
new_packet = true;
}
pthread_mutex_unlock(&stream->packets_mutex);
return new_packet;
}
static int send_packet(struct rtmp_stream *stream,
struct encoder_packet *packet, bool is_header)
{
uint8_t *data;
size_t size;
int ret = 0;
flv_packet_mux(packet, &data, &size, is_header);
#ifdef TEST_FRAMEDROPS
os_sleep_ms(rand() % 40);
#endif
ret = RTMP_Write(&stream->rtmp, (char*)data, (int)size);
bfree(data);
obs_free_encoder_packet(packet);
stream->total_bytes_sent += size;
return ret;
}
static inline void send_headers(struct rtmp_stream *stream);
static bool send_remaining_packets(struct rtmp_stream *stream)
{
struct encoder_packet packet;
if (!stream->sent_headers)
send_headers(stream);
while (get_next_packet(stream, &packet))
if (send_packet(stream, &packet, false) < 0)
return false;
return true;
}
static void *send_thread(void *data)
{
struct rtmp_stream *stream = data;
bool disconnected = false;
while (os_sem_wait(stream->send_sem) == 0) {
struct encoder_packet packet;
if (os_event_try(stream->stop_event) != EAGAIN)
break;
if (!get_next_packet(stream, &packet))
continue;
if (!stream->sent_headers)
send_headers(stream);
if (send_packet(stream, &packet, false) < 0) {
disconnected = true;
break;
}
}
if (!disconnected && !send_remaining_packets(stream))
disconnected = true;
if (disconnected) {
info("Disconnected from %s", stream->path.array);
free_packets(stream);
} else {
info("User stopped the stream");
}
if (os_event_try(stream->stop_event) == EAGAIN) {
pthread_detach(stream->send_thread);
obs_output_signal_stop(stream->output, OBS_OUTPUT_DISCONNECTED);
}
stream->active = false;
return NULL;
}
static void send_meta_data(struct rtmp_stream *stream)
{
uint8_t *meta_data;
size_t meta_data_size;
flv_meta_data(stream->output, &meta_data, &meta_data_size, false, 0);
RTMP_Write(&stream->rtmp, (char*)meta_data, (int)meta_data_size);
bfree(meta_data);
}
static void send_audio_header(struct rtmp_stream *stream)
{
obs_output_t *context = stream->output;
obs_encoder_t *aencoder = obs_output_get_audio_encoder(context, 0);
uint8_t *header;
struct encoder_packet packet = {
.type = OBS_ENCODER_AUDIO,
.timebase_den = 1
};
obs_encoder_get_extra_data(aencoder, &header, &packet.size);
packet.data = bmemdup(header, packet.size);
send_packet(stream, &packet, true);
}
static void send_video_header(struct rtmp_stream *stream)
{
obs_output_t *context = stream->output;
obs_encoder_t *vencoder = obs_output_get_video_encoder(context);
uint8_t *header;
size_t size;
struct encoder_packet packet = {
.type = OBS_ENCODER_VIDEO,
.timebase_den = 1,
.keyframe = true
};
obs_encoder_get_extra_data(vencoder, &header, &size);
packet.size = obs_parse_avc_header(&packet.data, header, size);
send_packet(stream, &packet, true);
}
static inline void send_headers(struct rtmp_stream *stream)
{
stream->sent_headers = true;
send_audio_header(stream);
send_video_header(stream);
}
static inline bool reset_semaphore(struct rtmp_stream *stream)
{
os_sem_destroy(stream->send_sem);
return os_sem_init(&stream->send_sem, 0) == 0;
}
#ifdef _WIN32
#define socklen_t int
#endif
#define MIN_SENDBUF_SIZE 65535
static void adjust_sndbuf_size(struct rtmp_stream *stream, int new_size)
{
int cur_sendbuf_size = new_size;
socklen_t int_size = sizeof(int);
getsockopt(stream->rtmp.m_sb.sb_socket, SOL_SOCKET, SO_SNDBUF,
(char*)&cur_sendbuf_size, &int_size);
if (cur_sendbuf_size < new_size) {
cur_sendbuf_size = new_size;
setsockopt(stream->rtmp.m_sb.sb_socket, SOL_SOCKET, SO_SNDBUF,
(const char*)&cur_sendbuf_size, int_size);
}
}
static int init_send(struct rtmp_stream *stream)
{
int ret;
#if defined(_WIN32)
adjust_sndbuf_size(stream, MIN_SENDBUF_SIZE);
#endif
reset_semaphore(stream);
ret = pthread_create(&stream->send_thread, NULL, send_thread, stream);
if (ret != 0) {
RTMP_Close(&stream->rtmp);
warn("Failed to create send thread");
return OBS_OUTPUT_ERROR;
}
stream->active = true;
send_meta_data(stream);
obs_output_begin_data_capture(stream->output, 0);
return OBS_OUTPUT_SUCCESS;
}
static int try_connect(struct rtmp_stream *stream)
{
if (dstr_is_empty(&stream->path)) {
warn("URL is empty");
return OBS_OUTPUT_BAD_PATH;
}
info("Connecting to RTMP URL %s...", stream->path.array);
if (!RTMP_SetupURL2(&stream->rtmp, stream->path.array,
stream->key.array))
return OBS_OUTPUT_BAD_PATH;
RTMP_EnableWrite(&stream->rtmp);
set_rtmp_dstr(&stream->rtmp.Link.pubUser, &stream->username);
set_rtmp_dstr(&stream->rtmp.Link.pubPasswd, &stream->password);
stream->rtmp.Link.swfUrl = stream->rtmp.Link.tcUrl;
set_rtmp_str(&stream->rtmp.Link.flashVer,
"FMLE/3.0 (compatible; FMSc/1.0)");
stream->rtmp.m_outChunkSize = 4096;
stream->rtmp.m_bSendChunkSizeInfo = true;
stream->rtmp.m_bUseNagle = true;
if (!RTMP_Connect(&stream->rtmp, NULL))
return OBS_OUTPUT_CONNECT_FAILED;
if (!RTMP_ConnectStream(&stream->rtmp, 0))
return OBS_OUTPUT_INVALID_STREAM;
info("Connection to %s successful", stream->path.array);
return init_send(stream);
}
static void *connect_thread(void *data)
{
struct rtmp_stream *stream = data;
int ret = try_connect(stream);
if (ret != OBS_OUTPUT_SUCCESS) {
obs_output_signal_stop(stream->output, ret);
info("Connection to %s failed: %d", stream->path.array, ret);
}
if (os_event_try(stream->stop_event) == EAGAIN)
pthread_detach(stream->connect_thread);
stream->connecting = false;
return NULL;
}
static bool rtmp_stream_start(void *data)
{
struct rtmp_stream *stream = data;
obs_service_t *service = obs_output_get_service(stream->output);
obs_data_t *settings;
if (!obs_output_can_begin_data_capture(stream->output, 0))
return false;
if (!obs_output_initialize_encoders(stream->output, 0))
return false;
stream->total_bytes_sent = 0;
stream->dropped_frames = 0;
settings = obs_output_get_settings(stream->output);
dstr_copy(&stream->path, obs_service_get_url(service));
dstr_copy(&stream->key, obs_service_get_key(service));
dstr_copy(&stream->username, obs_service_get_username(service));
dstr_copy(&stream->password, obs_service_get_password(service));
stream->drop_threshold_usec =
(int64_t)obs_data_get_int(settings, OPT_DROP_THRESHOLD) * 1000;
obs_data_release(settings);
return pthread_create(&stream->connect_thread, NULL, connect_thread,
stream) == 0;
}
static inline bool add_packet(struct rtmp_stream *stream,
struct encoder_packet *packet)
{
circlebuf_push_back(&stream->packets, packet,
sizeof(struct encoder_packet));
stream->last_dts_usec = packet->dts_usec;
return true;
}
static inline size_t num_buffered_packets(struct rtmp_stream *stream)
{
return stream->packets.size / sizeof(struct encoder_packet);
}
static void drop_frames(struct rtmp_stream *stream)
{
struct circlebuf new_buf = {0};
int drop_priority = 0;
uint64_t last_drop_dts_usec = 0;
int num_frames_dropped = 0;
debug("Previous packet count: %d", (int)num_buffered_packets(stream));
circlebuf_reserve(&new_buf, sizeof(struct encoder_packet) * 8);
while (stream->packets.size) {
struct encoder_packet packet;
circlebuf_pop_front(&stream->packets, &packet, sizeof(packet));
last_drop_dts_usec = packet.dts_usec;
if (packet.type == OBS_ENCODER_AUDIO) {
circlebuf_push_back(&new_buf, &packet, sizeof(packet));
} else {
if (drop_priority < packet.drop_priority)
drop_priority = packet.drop_priority;
num_frames_dropped++;
obs_free_encoder_packet(&packet);
}
}
circlebuf_free(&stream->packets);
stream->packets = new_buf;
stream->min_priority = drop_priority;
stream->min_drop_dts_usec = last_drop_dts_usec;
stream->dropped_frames += num_frames_dropped;
debug("New packet count: %d", (int)num_buffered_packets(stream));
}
static void check_to_drop_frames(struct rtmp_stream *stream)
{
struct encoder_packet first;
int64_t buffer_duration_usec;
if (num_buffered_packets(stream) < 5)
return;
circlebuf_peek_front(&stream->packets, &first, sizeof(first));
/* do not drop frames if frames were just dropped within this time */
if (first.dts_usec < stream->min_drop_dts_usec)
return;
/* if the amount of time stored in the buffered packets waiting to be
* sent is higher than threshold, drop frames */
buffer_duration_usec = stream->last_dts_usec - first.dts_usec;
if (buffer_duration_usec > stream->drop_threshold_usec) {
drop_frames(stream);
debug("dropping %" PRId64 " worth of frames",
buffer_duration_usec);
}
}
static bool add_video_packet(struct rtmp_stream *stream,
struct encoder_packet *packet)
{
check_to_drop_frames(stream);
/* if currently dropping frames, drop packets until it reaches the
* desired priority */
if (packet->priority < stream->min_priority) {
stream->dropped_frames++;
return false;
} else {
stream->min_priority = 0;
}
return add_packet(stream, packet);
}
static void rtmp_stream_data(void *data, struct encoder_packet *packet)
{
struct rtmp_stream *stream = data;
struct encoder_packet new_packet;
bool added_packet;
if (packet->type == OBS_ENCODER_VIDEO)
obs_parse_avc_packet(&new_packet, packet);
else
obs_duplicate_encoder_packet(&new_packet, packet);
pthread_mutex_lock(&stream->packets_mutex);
added_packet = (packet->type == OBS_ENCODER_VIDEO) ?
add_video_packet(stream, &new_packet) :
add_packet(stream, &new_packet);
pthread_mutex_unlock(&stream->packets_mutex);
if (added_packet)
os_sem_post(stream->send_sem);
else
obs_free_encoder_packet(&new_packet);
}
static void rtmp_stream_defaults(obs_data_t *defaults)
{
obs_data_set_default_int(defaults, OPT_DROP_THRESHOLD, 600);
}
static obs_properties_t *rtmp_stream_properties(void *unused)
{
UNUSED_PARAMETER(unused);
obs_properties_t *props = obs_properties_create();
obs_properties_add_int(props, OPT_DROP_THRESHOLD,
obs_module_text("RTMPStream.DropThreshold"),
200, 10000, 100);
return props;
}
static uint64_t rtmp_stream_total_bytes_sent(void *data)
{
struct rtmp_stream *stream = data;
return stream->total_bytes_sent;
}
static int rtmp_stream_dropped_frames(void *data)
{
struct rtmp_stream *stream = data;
return stream->dropped_frames;
}
struct obs_output_info rtmp_output_info = {
.id = "rtmp_output",
.flags = OBS_OUTPUT_AV |
OBS_OUTPUT_ENCODED |
OBS_OUTPUT_SERVICE,
.get_name = rtmp_stream_getname,
.create = rtmp_stream_create,
.destroy = rtmp_stream_destroy,
.start = rtmp_stream_start,
.stop = rtmp_stream_stop,
.encoded_packet = rtmp_stream_data,
.get_defaults = rtmp_stream_defaults,
.get_properties = rtmp_stream_properties,
.get_total_bytes = rtmp_stream_total_bytes_sent,
.get_dropped_frames = rtmp_stream_dropped_frames
};