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/******************************************************************************
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 "rtmp-stream.h"
#ifdef _WIN32
#include <util/windows/win-version.h>
#endif
Implement encoder interface (still preliminary) - Implement OBS encoder interface. It was previously incomplete, but now is reaching some level of completion, though probably should still be considered preliminary. I had originally implemented it so that encoders only have a 'reset' function to reset their parameters, but I felt that having both a 'start' and 'stop' function would be useful. Encoders are now assigned to a specific video/audio media output each rather than implicitely assigned to the main obs video/audio contexts. This allows separate encoder contexts that aren't necessarily assigned to the main video/audio context (which is useful for things such as recording specific sources). Will probably have to do this for regular obs outputs as well. When creating an encoder, you must now explicitely state whether that encoder is an audio or video encoder. Audio and video can optionally be automatically converted depending on what the encoder specifies. When something 'attaches' to an encoder, the first attachment starts the encoder, and the encoder automatically attaches to the media output context associated with it. Subsequent attachments won't have the same effect, they will just start receiving the same encoder data when the next keyframe plays (along with SEI if any). When detaching from the encoder, the last detachment will fully stop the encoder and detach the encoder from the media output context associated with the encoder. SEI must actually be exported separately; because new encoder attachments may not always be at the beginning of the stream, the first keyframe they get must have that SEI data in it. If the encoder has SEI data, it needs only add one small function to simply query that SEI data, and then that data will be handled automatically by libobs for all subsequent encoder attachments. - Implement x264 encoder plugin, move x264 files to separate plugin to separate necessary dependencies. - Change video/audio frame output structures to not use const qualifiers to prevent issues with non-const function usage elsewhere. This was an issue when writing the x264 encoder, as the x264 encoder expects non-const frame data. Change stagesurf_map to return a non-const data type to prevent this as well. - Change full range parameter of video scaler to be an enum rather than boolean
2014-03-16 16:21:34 -07:00
#ifndef SEC_TO_NSEC
#define SEC_TO_NSEC 1000000000ULL
#endif
#ifndef MSEC_TO_USEC
#define MSEC_TO_USEC 1000ULL
#endif
#ifndef MSEC_TO_NSEC
#define MSEC_TO_NSEC 1000000ULL
#endif
/* dynamic bitrate coefficients */
#define DBR_INC_TIMER (30ULL * SEC_TO_NSEC)
#define DBR_TRIGGER_USEC (200ULL * MSEC_TO_USEC)
#define MIN_ESTIMATE_DURATION_MS 1000
#define MAX_ESTIMATE_DURATION_MS 2000
static const char *rtmp_stream_getname(void *unused)
{
UNUSED_PARAMETER(unused);
2014-07-09 22:12:57 -07:00
return obs_module_text("RTMPStream");
}
static void log_rtmp(int level, const char *format, va_list args)
{
if (level > RTMP_LOGWARNING)
RTMP output: Implement frame drop code A little bit of history about frame dropping: I did a large number of experiments with frame dropping in old versions of OBS1, and it's not an easy thing to deal with. I tried just about everything from standard i-frame delay, to large buffers, to dumping packets, to super-unnecessarily-complex things that just ended up causing more problems than they was worth. When I did my experiments, I found that the most ideal frame drop system (in terms of reducing the amount of total data that needed to be dropped) was in the 0.4xx days where I had a 3 second frame-drop buffer where I could calculate the actual buffer size in bytes, and then intellgently choose packets in that buffer to trim it down to a specific size while minimizing the number of p-frames and i-frames dropped, and preventing the actual impact of dropped frames on the stream. The downside of it was that it required too much extra latency, and far too many people complained about it, so it was removed in favor of the current system. The current system I just refer to just as 'packet dumping', which when combined with low keyframe intervals (like most services use these days), is the next-best method from my experience. Just dump the buffer when you reach a threshold of buffering (which I prefer to measure with time rather than in size), then wait for a new i-frame. Simple, effective, and reduces the risk of consecutive buffering, while still having fairly low impact on the stream output due to the low keyframe interval of services. By the way, audio will not (and should not ever) be dropped, lest you end up with syncing issues (among other nasty things) specific to server implementation.
2014-04-12 04:34:15 -07:00
return;
RTMP output: Implement frame drop code A little bit of history about frame dropping: I did a large number of experiments with frame dropping in old versions of OBS1, and it's not an easy thing to deal with. I tried just about everything from standard i-frame delay, to large buffers, to dumping packets, to super-unnecessarily-complex things that just ended up causing more problems than they was worth. When I did my experiments, I found that the most ideal frame drop system (in terms of reducing the amount of total data that needed to be dropped) was in the 0.4xx days where I had a 3 second frame-drop buffer where I could calculate the actual buffer size in bytes, and then intellgently choose packets in that buffer to trim it down to a specific size while minimizing the number of p-frames and i-frames dropped, and preventing the actual impact of dropped frames on the stream. The downside of it was that it required too much extra latency, and far too many people complained about it, so it was removed in favor of the current system. The current system I just refer to just as 'packet dumping', which when combined with low keyframe intervals (like most services use these days), is the next-best method from my experience. Just dump the buffer when you reach a threshold of buffering (which I prefer to measure with time rather than in size), then wait for a new i-frame. Simple, effective, and reduces the risk of consecutive buffering, while still having fairly low impact on the stream output due to the low keyframe interval of services. By the way, audio will not (and should not ever) be dropped, lest you end up with syncing issues (among other nasty things) specific to server implementation.
2014-04-12 04:34:15 -07:00
blogva(LOG_INFO, format, args);
}
2015-11-01 15:19:50 -08:00
static inline size_t num_buffered_packets(struct rtmp_stream *stream);
static inline void free_packets(struct rtmp_stream *stream)
{
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size_t num_packets;
pthread_mutex_lock(&stream->packets_mutex);
2015-11-01 15:19:50 -08:00
num_packets = num_buffered_packets(stream);
if (num_packets)
info("Freeing %d remaining packets", (int)num_packets);
2015-11-01 15:19:50 -08:00
while (stream->packets.size) {
struct encoder_packet packet;
circlebuf_pop_front(&stream->packets, &packet, sizeof(packet));
obs_encoder_packet_release(&packet);
}
pthread_mutex_unlock(&stream->packets_mutex);
}
static inline bool stopping(struct rtmp_stream *stream)
{
return os_event_try(stream->stop_event) != EAGAIN;
}
2015-11-18 10:30:13 -08:00
static inline bool connecting(struct rtmp_stream *stream)
{
return os_atomic_load_bool(&stream->connecting);
}
static inline bool active(struct rtmp_stream *stream)
{
return os_atomic_load_bool(&stream->active);
}
static inline bool disconnected(struct rtmp_stream *stream)
{
return os_atomic_load_bool(&stream->disconnected);
}
static inline bool silently_reconnecting(struct rtmp_stream *stream)
{
return os_atomic_load_bool(&stream->silent_reconnect);
}
static void rtmp_stream_destroy(void *data)
{
struct rtmp_stream *stream = data;
2015-11-18 10:30:13 -08:00
if (stopping(stream) && !connecting(stream)) {
pthread_join(stream->send_thread, NULL);
2015-11-18 10:30:13 -08:00
} else if (connecting(stream) || active(stream)) {
if (stream->connecting)
pthread_join(stream->connect_thread, NULL);
stream->stop_ts = 0;
os_event_signal(stream->stop_event);
2015-11-18 10:30:13 -08:00
if (active(stream)) {
os_sem_post(stream->send_sem);
obs_output_end_data_capture(stream->output);
pthread_join(stream->send_thread, NULL);
}
}
RTMP_TLS_Free(&stream->rtmp);
free_packets(stream);
dstr_free(&stream->path);
dstr_free(&stream->key);
dstr_free(&stream->username);
dstr_free(&stream->password);
dstr_free(&stream->encoder_name);
dstr_free(&stream->bind_ip);
os_event_destroy(stream->stop_event);
os_sem_destroy(stream->send_sem);
pthread_mutex_destroy(&stream->packets_mutex);
circlebuf_free(&stream->packets);
#ifdef TEST_FRAMEDROPS
circlebuf_free(&stream->droptest_info);
#endif
circlebuf_free(&stream->dbr_frames);
pthread_mutex_destroy(&stream->dbr_mutex);
os_event_destroy(stream->buffer_space_available_event);
os_event_destroy(stream->buffer_has_data_event);
os_event_destroy(stream->socket_available_event);
os_event_destroy(stream->send_thread_signaled_exit);
pthread_mutex_destroy(&stream->write_buf_mutex);
if (stream->write_buf)
bfree(stream->write_buf);
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_LogSetCallback(log_rtmp);
RTMP_Init(&stream->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;
if (pthread_mutex_init(&stream->write_buf_mutex, NULL) != 0) {
warn("Failed to initialize write buffer mutex");
goto fail;
}
if (pthread_mutex_init(&stream->dbr_mutex, NULL) != 0) {
warn("Failed to initialize dbr mutex");
goto fail;
}
if (os_event_init(&stream->buffer_space_available_event,
OS_EVENT_TYPE_AUTO) != 0) {
warn("Failed to initialize write buffer event");
goto fail;
}
if (os_event_init(&stream->buffer_has_data_event, OS_EVENT_TYPE_AUTO) !=
0) {
warn("Failed to initialize data buffer event");
goto fail;
}
if (os_event_init(&stream->socket_available_event,
OS_EVENT_TYPE_AUTO) != 0) {
warn("Failed to initialize socket buffer event");
goto fail;
}
if (os_event_init(&stream->send_thread_signaled_exit,
OS_EVENT_TYPE_MANUAL) != 0) {
warn("Failed to initialize socket exit event");
goto fail;
}
UNUSED_PARAMETER(settings);
return stream;
fail:
rtmp_stream_destroy(stream);
return NULL;
}
static void rtmp_stream_stop(void *data, uint64_t ts)
{
struct rtmp_stream *stream = data;
if (stopping(stream) && ts != 0)
return;
2015-11-18 10:30:13 -08:00
if (connecting(stream))
pthread_join(stream->connect_thread, NULL);
stream->stop_ts = ts / 1000ULL;
if (ts)
stream->shutdown_timeout_ts =
ts +
(uint64_t)stream->max_shutdown_time_sec * 1000000000ULL;
2015-11-18 10:30:13 -08:00
if (active(stream)) {
os_event_signal(stream->stop_event);
if (stream->stop_ts == 0)
os_sem_post(stream->send_sem);
} else {
obs_output_signal_stop(stream->output, OBS_OUTPUT_SUCCESS);
}
}
static inline void set_rtmp_str(AVal *val, const char *str)
{
Implement RTMP module (still needs drop code) - Implement the RTMP output module. This time around, we just use a simple FLV muxer, then just write to the stream with RTMP_Write. Easy and effective. - Fix the FLV muxer, the muxer now outputs proper FLV packets. - Output API: * When using encoders, automatically interleave encoded packets before sending it to the output. * Pair encoders and have them automatically wait for the other to start to ensure sync. * Change 'obs_output_signal_start_fail' to 'obs_output_signal_stop' because it was a bit confusing, and doing this makes a lot more sense for outputs that need to stop suddenly (disconnections/etc). - Encoder API: * Remove some unnecessary encoder functions from the actual API and make them internal. Most of the encoder functions are handled automatically by outputs anyway, so there's no real need to expose them and end up inadvertently confusing plugin writers. * Have audio encoders wait for the video encoder to get a frame, then start at the exact data point that the first video frame starts to ensure the most accrate sync of video/audio possible. * Add a required 'frame_size' callback for audio encoders that returns the expected number of frames desired to encode with. This way, the libobs encoder API can handle the circular buffering internally automatically for the encoder modules, so encoder writers don't have to do it themselves. - Fix a few bugs in the serializer interface. It was passing the wrong variable for the data in a few cases. - If a source has video, make obs_source_update defer the actual update callback until the tick function is called to prevent threading issues.
2014-04-07 22:00:10 -07:00
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 inline void peek_next_packet(struct rtmp_stream *stream,
struct encoder_packet *packet)
{
pthread_mutex_lock(&stream->packets_mutex);
circlebuf_peek_front(&stream->packets, packet,
sizeof(struct encoder_packet));
pthread_mutex_unlock(&stream->packets_mutex);
}
static void reinsert_packet_at_front(struct rtmp_stream *stream,
struct encoder_packet *packet)
{
pthread_mutex_lock(&stream->packets_mutex);
circlebuf_push_front(&stream->packets, packet,
sizeof(struct encoder_packet));
pthread_mutex_unlock(&stream->packets_mutex);
os_sem_post(stream->send_sem);
}
#define RTMP_PACKET_TYPE_RECONNECT 0x20
static bool process_recv_data(struct rtmp_stream *stream, size_t size)
{
2015-11-16 13:49:34 -08:00
RTMP *rtmp = &stream->rtmp;
RTMPPacket packet = {0};
2015-11-16 13:49:34 -08:00
if (!RTMP_ReadPacket(rtmp, &packet)) {
2015-11-16 13:49:34 -08:00
#ifdef _WIN32
int error = WSAGetLastError();
2015-11-16 13:49:34 -08:00
#else
int error = errno;
2015-11-16 13:49:34 -08:00
#endif
do_log(LOG_ERROR, "RTMP_ReadPacket error: %d", error);
return false;
}
2015-11-16 13:49:34 -08:00
if (packet.m_body) {
if (packet.m_packetType == RTMP_PACKET_TYPE_RECONNECT) {
os_atomic_set_bool(&stream->silent_reconnect, true);
}
RTMPPacket_Free(&packet);
}
2015-11-16 13:49:34 -08:00
return true;
}
#ifdef TEST_FRAMEDROPS
static void droptest_cap_data_rate(struct rtmp_stream *stream, size_t size)
{
uint64_t ts = os_gettime_ns();
struct droptest_info info;
#if defined(_WIN32) && defined(TEST_FRAMEDROPS_WITH_BITRATE_SHORTCUTS)
uint64_t check_elapsed = ts - stream->droptest_last_key_check;
if (check_elapsed > (200ULL * MSEC_TO_NSEC)) {
size_t bitrate = 0;
stream->droptest_last_key_check = ts;
if (GetAsyncKeyState(VK_NUMPAD0) & 0x8000) {
stream->droptest_max = 0;
} else if (GetAsyncKeyState(VK_NUMPAD1) & 0x8000) {
bitrate = 1000;
} else if (GetAsyncKeyState(VK_NUMPAD2) & 0x8000) {
bitrate = 2000;
} else if (GetAsyncKeyState(VK_NUMPAD3) & 0x8000) {
bitrate = 3000;
} else if (GetAsyncKeyState(VK_NUMPAD4) & 0x8000) {
bitrate = 4000;
} else if (GetAsyncKeyState(VK_NUMPAD5) & 0x8000) {
bitrate = 5000;
} else if (GetAsyncKeyState(VK_NUMPAD6) & 0x8000) {
bitrate = 6000;
} else if (GetAsyncKeyState(VK_NUMPAD7) & 0x8000) {
bitrate = 7000;
} else if (GetAsyncKeyState(VK_NUMPAD8) & 0x8000) {
bitrate = 8000;
} else if (GetAsyncKeyState(VK_NUMPAD9) & 0x8000) {
bitrate = 9000;
}
if (bitrate) {
stream->droptest_max = (bitrate * 1000 / 8);
}
}
if (!stream->droptest_max) {
return;
}
#else
if (!stream->droptest_max) {
stream->droptest_max = DROPTEST_MAX_BYTES;
}
#endif
info.ts = ts;
info.size = size;
circlebuf_push_back(&stream->droptest_info, &info, sizeof(info));
stream->droptest_size += size;
if (stream->droptest_info.size) {
circlebuf_peek_front(&stream->droptest_info, &info,
sizeof(info));
if (stream->droptest_size > stream->droptest_max) {
uint64_t elapsed = ts - info.ts;
if (elapsed < 1000000000ULL) {
elapsed = 1000000000ULL - elapsed;
os_sleepto_ns(ts + elapsed);
}
while (stream->droptest_size > stream->droptest_max) {
circlebuf_pop_front(&stream->droptest_info,
&info, sizeof(info));
stream->droptest_size -= info.size;
}
}
}
}
#endif
static int socket_queue_data(RTMPSockBuf *sb, const char *data, int len,
void *arg)
{
UNUSED_PARAMETER(sb);
struct rtmp_stream *stream = arg;
retry_send:
if (!RTMP_IsConnected(&stream->rtmp))
return 0;
pthread_mutex_lock(&stream->write_buf_mutex);
if (stream->write_buf_len + len > stream->write_buf_size) {
pthread_mutex_unlock(&stream->write_buf_mutex);
if (os_event_wait(stream->buffer_space_available_event)) {
return 0;
}
goto retry_send;
}
memcpy(stream->write_buf + stream->write_buf_len, data, len);
stream->write_buf_len += len;
pthread_mutex_unlock(&stream->write_buf_mutex);
os_event_signal(stream->buffer_has_data_event);
return len;
}
static int send_packet(struct rtmp_stream *stream,
struct encoder_packet *packet, bool is_header,
size_t idx)
{
uint8_t *data;
size_t size;
int recv_size = 0;
Implement RTMP module (still needs drop code) - Implement the RTMP output module. This time around, we just use a simple FLV muxer, then just write to the stream with RTMP_Write. Easy and effective. - Fix the FLV muxer, the muxer now outputs proper FLV packets. - Output API: * When using encoders, automatically interleave encoded packets before sending it to the output. * Pair encoders and have them automatically wait for the other to start to ensure sync. * Change 'obs_output_signal_start_fail' to 'obs_output_signal_stop' because it was a bit confusing, and doing this makes a lot more sense for outputs that need to stop suddenly (disconnections/etc). - Encoder API: * Remove some unnecessary encoder functions from the actual API and make them internal. Most of the encoder functions are handled automatically by outputs anyway, so there's no real need to expose them and end up inadvertently confusing plugin writers. * Have audio encoders wait for the video encoder to get a frame, then start at the exact data point that the first video frame starts to ensure the most accrate sync of video/audio possible. * Add a required 'frame_size' callback for audio encoders that returns the expected number of frames desired to encode with. This way, the libobs encoder API can handle the circular buffering internally automatically for the encoder modules, so encoder writers don't have to do it themselves. - Fix a few bugs in the serializer interface. It was passing the wrong variable for the data in a few cases. - If a source has video, make obs_source_update defer the actual update callback until the tick function is called to prevent threading issues.
2014-04-07 22:00:10 -07:00
int ret = 0;
assert(idx < RTMP_MAX_STREAMS);
if (!stream->new_socket_loop) {
#ifdef _WIN32
ret = ioctlsocket(stream->rtmp.m_sb.sb_socket, FIONREAD,
(u_long *)&recv_size);
#else
ret = ioctl(stream->rtmp.m_sb.sb_socket, FIONREAD, &recv_size);
#endif
if (ret >= 0 && recv_size > 0) {
if (!process_recv_data(stream, (size_t)recv_size))
return -1;
}
2015-11-16 13:49:34 -08:00
}
if (idx > 0) {
flv_additional_packet_mux(
packet, is_header ? 0 : stream->start_dts_offset, &data,
&size, is_header, idx);
} else {
flv_packet_mux(packet, is_header ? 0 : stream->start_dts_offset,
&data, &size, is_header);
}
#ifdef TEST_FRAMEDROPS
droptest_cap_data_rate(stream, size);
#endif
ret = RTMP_Write(&stream->rtmp, (char *)data, (int)size, 0);
bfree(data);
if (is_header)
bfree(packet->data);
else
obs_encoder_packet_release(packet);
stream->total_bytes_sent += size;
return ret;
}
static inline bool send_headers(struct rtmp_stream *stream);
static inline bool can_shutdown_stream(struct rtmp_stream *stream,
struct encoder_packet *packet)
{
uint64_t cur_time = os_gettime_ns();
bool timeout = cur_time >= stream->shutdown_timeout_ts;
if (timeout)
info("Stream shutdown timeout reached (%d second(s))",
stream->max_shutdown_time_sec);
return timeout || packet->sys_dts_usec >= (int64_t)stream->stop_ts;
}
static void set_output_error(struct rtmp_stream *stream)
{
const char *msg = NULL;
#ifdef _WIN32
switch (stream->rtmp.last_error_code) {
case WSAETIMEDOUT:
msg = obs_module_text("ConnectionTimedOut");
break;
case WSAEACCES:
msg = obs_module_text("PermissionDenied");
break;
case WSAECONNABORTED:
msg = obs_module_text("ConnectionAborted");
break;
case WSAECONNRESET:
msg = obs_module_text("ConnectionReset");
break;
case WSAHOST_NOT_FOUND:
msg = obs_module_text("HostNotFound");
break;
case WSANO_DATA:
msg = obs_module_text("NoData");
break;
case WSAEADDRNOTAVAIL:
msg = obs_module_text("AddressNotAvailable");
break;
}
#else
switch (stream->rtmp.last_error_code) {
case ETIMEDOUT:
msg = obs_module_text("ConnectionTimedOut");
break;
case EACCES:
msg = obs_module_text("PermissionDenied");
break;
case ECONNABORTED:
msg = obs_module_text("ConnectionAborted");
break;
case ECONNRESET:
msg = obs_module_text("ConnectionReset");
break;
case HOST_NOT_FOUND:
msg = obs_module_text("HostNotFound");
break;
case NO_DATA:
msg = obs_module_text("NoData");
break;
case EADDRNOTAVAIL:
msg = obs_module_text("AddressNotAvailable");
break;
}
#endif
// non platform-specific errors
if (!msg) {
switch (stream->rtmp.last_error_code) {
case -0x2700:
msg = obs_module_text("SSLCertVerifyFailed");
break;
case -0x7680:
msg = "Failed to load root certificates for a secure TLS connection."
#if defined(__linux__)
" Check you have an up to date root certificate bundle in /etc/ssl/certs."
#endif
;
break;
}
}
if (msg)
obs_output_set_last_error(stream->output, msg);
}
static void dbr_add_frame(struct rtmp_stream *stream, struct dbr_frame *back)
{
struct dbr_frame front;
uint64_t dur;
circlebuf_push_back(&stream->dbr_frames, back, sizeof(*back));
circlebuf_peek_front(&stream->dbr_frames, &front, sizeof(front));
stream->dbr_data_size += back->size;
dur = (back->send_end - front.send_beg) / 1000000;
if (dur >= MAX_ESTIMATE_DURATION_MS) {
stream->dbr_data_size -= front.size;
circlebuf_pop_front(&stream->dbr_frames, NULL, sizeof(front));
}
stream->dbr_est_bitrate =
(dur >= MIN_ESTIMATE_DURATION_MS)
? (long)(stream->dbr_data_size * 1000 / dur)
: 0;
stream->dbr_est_bitrate *= 8;
stream->dbr_est_bitrate /= 1000;
if (stream->dbr_est_bitrate) {
stream->dbr_est_bitrate -= stream->audio_bitrate;
if (stream->dbr_est_bitrate < 50)
stream->dbr_est_bitrate = 50;
}
}
static void dbr_set_bitrate(struct rtmp_stream *stream);
static bool rtmp_stream_start(void *data);
#ifdef _WIN32
#define socklen_t int
#endif
static void log_sndbuf_size(struct rtmp_stream *stream)
{
int cur_sendbuf_size;
socklen_t int_size = sizeof(int);
if (!getsockopt(stream->rtmp.m_sb.sb_socket, SOL_SOCKET, SO_SNDBUF,
(char *)&cur_sendbuf_size, &int_size)) {
info("Socket send buffer is %d bytes", cur_sendbuf_size);
}
}
static void *send_thread(void *data)
{
struct rtmp_stream *stream = data;
os_set_thread_name("rtmp-stream: send_thread");
#if defined(_WIN32)
// Despite MSDN claiming otherwise, send buffer auto tuning on
// Windows 7 doesn't seem to work very well.
if (get_win_ver_int() == 0x601) {
DWORD cur_sendbuf_size;
DWORD desired_sendbuf_size = 524288;
socklen_t int_size = sizeof(int);
if (!getsockopt(stream->rtmp.m_sb.sb_socket, SOL_SOCKET,
SO_SNDBUF, (char *)&cur_sendbuf_size,
&int_size) &&
cur_sendbuf_size < desired_sendbuf_size) {
setsockopt(stream->rtmp.m_sb.sb_socket, SOL_SOCKET,
SO_SNDBUF, (char *)&desired_sendbuf_size,
sizeof(desired_sendbuf_size));
}
}
log_sndbuf_size(stream);
#endif
while (os_sem_wait(stream->send_sem) == 0) {
struct encoder_packet packet;
struct dbr_frame dbr_frame;
if (stopping(stream) && stream->stop_ts == 0) {
break;
}
if (!get_next_packet(stream, &packet))
continue;
if (stopping(stream)) {
if (can_shutdown_stream(stream, &packet)) {
obs_encoder_packet_release(&packet);
break;
}
}
if (!stream->sent_headers) {
if (!send_headers(stream)) {
os_atomic_set_bool(&stream->disconnected, true);
break;
}
}
/* silent reconnect signal received from server, reconnect on
* next keyframe */
if (silently_reconnecting(stream) &&
packet.type == OBS_ENCODER_VIDEO && packet.keyframe) {
reinsert_packet_at_front(stream, &packet);
break;
}
if (stream->dbr_enabled) {
dbr_frame.send_beg = os_gettime_ns();
dbr_frame.size = packet.size;
}
if (send_packet(stream, &packet, false, packet.track_idx) < 0) {
2015-11-03 14:09:48 -08:00
os_atomic_set_bool(&stream->disconnected, true);
break;
}
if (stream->dbr_enabled) {
dbr_frame.send_end = os_gettime_ns();
pthread_mutex_lock(&stream->dbr_mutex);
dbr_add_frame(stream, &dbr_frame);
pthread_mutex_unlock(&stream->dbr_mutex);
}
}
bool encode_error = os_atomic_load_bool(&stream->encode_error);
2015-11-18 10:30:13 -08:00
if (disconnected(stream)) {
info("Disconnected from %s", stream->path.array);
} else if (encode_error) {
info("Encoder error, disconnecting");
} else if (silently_reconnecting(stream)) {
info("Silent reconnect signal received from server");
} else {
info("User stopped the stream");
}
#if defined(_WIN32)
log_sndbuf_size(stream);
#endif
if (stream->new_socket_loop) {
os_event_signal(stream->send_thread_signaled_exit);
os_event_signal(stream->buffer_has_data_event);
pthread_join(stream->socket_thread, NULL);
stream->socket_thread_active = false;
stream->rtmp.m_bCustomSend = false;
}
set_output_error(stream);
RTMP_Close(&stream->rtmp);
/* reset bitrate on stop */
if (stream->dbr_enabled) {
if (stream->dbr_cur_bitrate != stream->dbr_orig_bitrate) {
stream->dbr_cur_bitrate = stream->dbr_orig_bitrate;
dbr_set_bitrate(stream);
}
}
if (!stopping(stream)) {
pthread_detach(stream->send_thread);
if (!silently_reconnecting(stream))
obs_output_signal_stop(stream->output,
OBS_OUTPUT_DISCONNECTED);
} else if (encode_error) {
obs_output_signal_stop(stream->output, OBS_OUTPUT_ENCODE_ERROR);
} else {
obs_output_end_data_capture(stream->output);
Implement RTMP module (still needs drop code) - Implement the RTMP output module. This time around, we just use a simple FLV muxer, then just write to the stream with RTMP_Write. Easy and effective. - Fix the FLV muxer, the muxer now outputs proper FLV packets. - Output API: * When using encoders, automatically interleave encoded packets before sending it to the output. * Pair encoders and have them automatically wait for the other to start to ensure sync. * Change 'obs_output_signal_start_fail' to 'obs_output_signal_stop' because it was a bit confusing, and doing this makes a lot more sense for outputs that need to stop suddenly (disconnections/etc). - Encoder API: * Remove some unnecessary encoder functions from the actual API and make them internal. Most of the encoder functions are handled automatically by outputs anyway, so there's no real need to expose them and end up inadvertently confusing plugin writers. * Have audio encoders wait for the video encoder to get a frame, then start at the exact data point that the first video frame starts to ensure the most accrate sync of video/audio possible. * Add a required 'frame_size' callback for audio encoders that returns the expected number of frames desired to encode with. This way, the libobs encoder API can handle the circular buffering internally automatically for the encoder modules, so encoder writers don't have to do it themselves. - Fix a few bugs in the serializer interface. It was passing the wrong variable for the data in a few cases. - If a source has video, make obs_source_update defer the actual update callback until the tick function is called to prevent threading issues.
2014-04-07 22:00:10 -07:00
}
if (!silently_reconnecting(stream)) {
free_packets(stream);
os_event_reset(stream->stop_event);
os_atomic_set_bool(&stream->active, false);
}
stream->sent_headers = false;
/* reset bitrate on stop */
if (stream->dbr_enabled) {
if (stream->dbr_cur_bitrate != stream->dbr_orig_bitrate) {
stream->dbr_cur_bitrate = stream->dbr_orig_bitrate;
dbr_set_bitrate(stream);
}
}
if (silently_reconnecting(stream)) {
rtmp_stream_start(stream);
}
return NULL;
}
static bool send_additional_meta_data(struct rtmp_stream *stream)
{
uint8_t *meta_data;
size_t meta_data_size;
bool success = true;
flv_additional_meta_data(stream->output, &meta_data, &meta_data_size);
success = RTMP_Write(&stream->rtmp, (char *)meta_data,
(int)meta_data_size, 0) >= 0;
bfree(meta_data);
return success;
}
static bool send_meta_data(struct rtmp_stream *stream)
{
uint8_t *meta_data;
size_t meta_data_size;
bool success = true;
flv_meta_data(stream->output, &meta_data, &meta_data_size, false);
success = RTMP_Write(&stream->rtmp, (char *)meta_data,
(int)meta_data_size, 0) >= 0;
bfree(meta_data);
return success;
}
static bool send_audio_header(struct rtmp_stream *stream, size_t idx,
bool *next)
{
obs_output_t *context = stream->output;
obs_encoder_t *aencoder = obs_output_get_audio_encoder(context, idx);
Implement RTMP module (still needs drop code) - Implement the RTMP output module. This time around, we just use a simple FLV muxer, then just write to the stream with RTMP_Write. Easy and effective. - Fix the FLV muxer, the muxer now outputs proper FLV packets. - Output API: * When using encoders, automatically interleave encoded packets before sending it to the output. * Pair encoders and have them automatically wait for the other to start to ensure sync. * Change 'obs_output_signal_start_fail' to 'obs_output_signal_stop' because it was a bit confusing, and doing this makes a lot more sense for outputs that need to stop suddenly (disconnections/etc). - Encoder API: * Remove some unnecessary encoder functions from the actual API and make them internal. Most of the encoder functions are handled automatically by outputs anyway, so there's no real need to expose them and end up inadvertently confusing plugin writers. * Have audio encoders wait for the video encoder to get a frame, then start at the exact data point that the first video frame starts to ensure the most accrate sync of video/audio possible. * Add a required 'frame_size' callback for audio encoders that returns the expected number of frames desired to encode with. This way, the libobs encoder API can handle the circular buffering internally automatically for the encoder modules, so encoder writers don't have to do it themselves. - Fix a few bugs in the serializer interface. It was passing the wrong variable for the data in a few cases. - If a source has video, make obs_source_update defer the actual update callback until the tick function is called to prevent threading issues.
2014-04-07 22:00:10 -07:00
uint8_t *header;
struct encoder_packet packet = {.type = OBS_ENCODER_AUDIO,
.timebase_den = 1};
if (!aencoder) {
*next = false;
return true;
}
Implement RTMP module (still needs drop code) - Implement the RTMP output module. This time around, we just use a simple FLV muxer, then just write to the stream with RTMP_Write. Easy and effective. - Fix the FLV muxer, the muxer now outputs proper FLV packets. - Output API: * When using encoders, automatically interleave encoded packets before sending it to the output. * Pair encoders and have them automatically wait for the other to start to ensure sync. * Change 'obs_output_signal_start_fail' to 'obs_output_signal_stop' because it was a bit confusing, and doing this makes a lot more sense for outputs that need to stop suddenly (disconnections/etc). - Encoder API: * Remove some unnecessary encoder functions from the actual API and make them internal. Most of the encoder functions are handled automatically by outputs anyway, so there's no real need to expose them and end up inadvertently confusing plugin writers. * Have audio encoders wait for the video encoder to get a frame, then start at the exact data point that the first video frame starts to ensure the most accrate sync of video/audio possible. * Add a required 'frame_size' callback for audio encoders that returns the expected number of frames desired to encode with. This way, the libobs encoder API can handle the circular buffering internally automatically for the encoder modules, so encoder writers don't have to do it themselves. - Fix a few bugs in the serializer interface. It was passing the wrong variable for the data in a few cases. - If a source has video, make obs_source_update defer the actual update callback until the tick function is called to prevent threading issues.
2014-04-07 22:00:10 -07:00
obs_encoder_get_extra_data(aencoder, &header, &packet.size);
packet.data = bmemdup(header, packet.size);
return send_packet(stream, &packet, true, idx) >= 0;
}
static bool 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 = {
Implement RTMP module (still needs drop code) - Implement the RTMP output module. This time around, we just use a simple FLV muxer, then just write to the stream with RTMP_Write. Easy and effective. - Fix the FLV muxer, the muxer now outputs proper FLV packets. - Output API: * When using encoders, automatically interleave encoded packets before sending it to the output. * Pair encoders and have them automatically wait for the other to start to ensure sync. * Change 'obs_output_signal_start_fail' to 'obs_output_signal_stop' because it was a bit confusing, and doing this makes a lot more sense for outputs that need to stop suddenly (disconnections/etc). - Encoder API: * Remove some unnecessary encoder functions from the actual API and make them internal. Most of the encoder functions are handled automatically by outputs anyway, so there's no real need to expose them and end up inadvertently confusing plugin writers. * Have audio encoders wait for the video encoder to get a frame, then start at the exact data point that the first video frame starts to ensure the most accrate sync of video/audio possible. * Add a required 'frame_size' callback for audio encoders that returns the expected number of frames desired to encode with. This way, the libobs encoder API can handle the circular buffering internally automatically for the encoder modules, so encoder writers don't have to do it themselves. - Fix a few bugs in the serializer interface. It was passing the wrong variable for the data in a few cases. - If a source has video, make obs_source_update defer the actual update callback until the tick function is called to prevent threading issues.
2014-04-07 22:00:10 -07:00
.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);
return send_packet(stream, &packet, true, 0) >= 0;
}
static inline bool send_headers(struct rtmp_stream *stream)
{
stream->sent_headers = true;
size_t i = 0;
bool next = true;
if (!send_audio_header(stream, i++, &next))
return false;
if (!send_video_header(stream))
return false;
while (next) {
if (!send_audio_header(stream, i++, &next))
return false;
}
return true;
}
static inline bool reset_semaphore(struct rtmp_stream *stream)
{
os_sem_destroy(stream->send_sem);
return os_sem_init(&stream->send_sem, 0) == 0;
}
RTMP output: Implement frame drop code A little bit of history about frame dropping: I did a large number of experiments with frame dropping in old versions of OBS1, and it's not an easy thing to deal with. I tried just about everything from standard i-frame delay, to large buffers, to dumping packets, to super-unnecessarily-complex things that just ended up causing more problems than they was worth. When I did my experiments, I found that the most ideal frame drop system (in terms of reducing the amount of total data that needed to be dropped) was in the 0.4xx days where I had a 3 second frame-drop buffer where I could calculate the actual buffer size in bytes, and then intellgently choose packets in that buffer to trim it down to a specific size while minimizing the number of p-frames and i-frames dropped, and preventing the actual impact of dropped frames on the stream. The downside of it was that it required too much extra latency, and far too many people complained about it, so it was removed in favor of the current system. The current system I just refer to just as 'packet dumping', which when combined with low keyframe intervals (like most services use these days), is the next-best method from my experience. Just dump the buffer when you reach a threshold of buffering (which I prefer to measure with time rather than in size), then wait for a new i-frame. Simple, effective, and reduces the risk of consecutive buffering, while still having fairly low impact on the stream output due to the low keyframe interval of services. By the way, audio will not (and should not ever) be dropped, lest you end up with syncing issues (among other nasty things) specific to server implementation.
2014-04-12 04:34:15 -07:00
static int init_send(struct rtmp_stream *stream)
{
int ret;
obs_output_t *context = stream->output;
RTMP output: Implement frame drop code A little bit of history about frame dropping: I did a large number of experiments with frame dropping in old versions of OBS1, and it's not an easy thing to deal with. I tried just about everything from standard i-frame delay, to large buffers, to dumping packets, to super-unnecessarily-complex things that just ended up causing more problems than they was worth. When I did my experiments, I found that the most ideal frame drop system (in terms of reducing the amount of total data that needed to be dropped) was in the 0.4xx days where I had a 3 second frame-drop buffer where I could calculate the actual buffer size in bytes, and then intellgently choose packets in that buffer to trim it down to a specific size while minimizing the number of p-frames and i-frames dropped, and preventing the actual impact of dropped frames on the stream. The downside of it was that it required too much extra latency, and far too many people complained about it, so it was removed in favor of the current system. The current system I just refer to just as 'packet dumping', which when combined with low keyframe intervals (like most services use these days), is the next-best method from my experience. Just dump the buffer when you reach a threshold of buffering (which I prefer to measure with time rather than in size), then wait for a new i-frame. Simple, effective, and reduces the risk of consecutive buffering, while still having fairly low impact on the stream output due to the low keyframe interval of services. By the way, audio will not (and should not ever) be dropped, lest you end up with syncing issues (among other nasty things) specific to server implementation.
2014-04-12 04:34:15 -07:00
if (!silently_reconnecting(stream))
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;
}
if (stream->new_socket_loop) {
int one = 1;
#ifdef _WIN32
if (ioctlsocket(stream->rtmp.m_sb.sb_socket, FIONBIO, &one)) {
stream->rtmp.last_error_code = WSAGetLastError();
#else
if (ioctl(stream->rtmp.m_sb.sb_socket, FIONBIO, &one)) {
stream->rtmp.last_error_code = errno;
#endif
warn("Failed to set non-blocking socket");
return OBS_OUTPUT_ERROR;
}
os_event_reset(stream->send_thread_signaled_exit);
info("New socket loop enabled by user");
if (stream->low_latency_mode)
info("Low latency mode enabled by user");
if (stream->write_buf)
bfree(stream->write_buf);
int total_bitrate = 0;
obs_encoder_t *vencoder = obs_output_get_video_encoder(context);
if (vencoder) {
obs_data_t *params = obs_encoder_get_settings(vencoder);
if (params) {
int bitrate =
obs_data_get_int(params, "bitrate");
if (!bitrate) {
warn("Video encoder didn't return a "
"valid bitrate, new network "
"code may function poorly. "
"Low latency mode disabled.");
stream->low_latency_mode = false;
bitrate = 10000;
}
total_bitrate += bitrate;
obs_data_release(params);
}
}
obs_encoder_t *aencoder =
obs_output_get_audio_encoder(context, 0);
if (aencoder) {
obs_data_t *params = obs_encoder_get_settings(aencoder);
if (params) {
int bitrate =
obs_data_get_int(params, "bitrate");
if (!bitrate)
bitrate = 160;
total_bitrate += bitrate;
obs_data_release(params);
}
}
// to bytes/sec
int ideal_buffer_size = total_bitrate * 128;
if (ideal_buffer_size < 131072)
ideal_buffer_size = 131072;
stream->write_buf_size = ideal_buffer_size;
stream->write_buf = bmalloc(ideal_buffer_size);
#ifdef _WIN32
ret = pthread_create(&stream->socket_thread, NULL,
socket_thread_windows, stream);
#else
warn("New socket loop not supported on this platform");
return OBS_OUTPUT_ERROR;
#endif
if (ret != 0) {
RTMP_Close(&stream->rtmp);
warn("Failed to create socket thread");
return OBS_OUTPUT_ERROR;
}
stream->socket_thread_active = true;
stream->rtmp.m_bCustomSend = true;
stream->rtmp.m_customSendFunc = socket_queue_data;
stream->rtmp.m_customSendParam = stream;
}
os_atomic_set_bool(&stream->active, true);
if (!send_meta_data(stream)) {
warn("Disconnected while attempting to send metadata");
set_output_error(stream);
return OBS_OUTPUT_DISCONNECTED;
}
obs_encoder_t *aencoder = obs_output_get_audio_encoder(context, 1);
if (aencoder && !send_additional_meta_data(stream)) {
warn("Disconnected while attempting to send additional "
"metadata");
return OBS_OUTPUT_DISCONNECTED;
}
if (obs_output_get_audio_encoder(context, 2) != NULL) {
warn("Additional audio streams not supported");
return OBS_OUTPUT_DISCONNECTED;
}
if (!silently_reconnecting(stream))
obs_output_begin_data_capture(stream->output, 0);
Implement RTMP module (still needs drop code) - Implement the RTMP output module. This time around, we just use a simple FLV muxer, then just write to the stream with RTMP_Write. Easy and effective. - Fix the FLV muxer, the muxer now outputs proper FLV packets. - Output API: * When using encoders, automatically interleave encoded packets before sending it to the output. * Pair encoders and have them automatically wait for the other to start to ensure sync. * Change 'obs_output_signal_start_fail' to 'obs_output_signal_stop' because it was a bit confusing, and doing this makes a lot more sense for outputs that need to stop suddenly (disconnections/etc). - Encoder API: * Remove some unnecessary encoder functions from the actual API and make them internal. Most of the encoder functions are handled automatically by outputs anyway, so there's no real need to expose them and end up inadvertently confusing plugin writers. * Have audio encoders wait for the video encoder to get a frame, then start at the exact data point that the first video frame starts to ensure the most accrate sync of video/audio possible. * Add a required 'frame_size' callback for audio encoders that returns the expected number of frames desired to encode with. This way, the libobs encoder API can handle the circular buffering internally automatically for the encoder modules, so encoder writers don't have to do it themselves. - Fix a few bugs in the serializer interface. It was passing the wrong variable for the data in a few cases. - If a source has video, make obs_source_update defer the actual update callback until the tick function is called to prevent threading issues.
2014-04-07 22:00:10 -07:00
return OBS_OUTPUT_SUCCESS;
}
#ifdef _WIN32
static void win32_log_interface_type(struct rtmp_stream *stream)
{
RTMP *rtmp = &stream->rtmp;
MIB_IPFORWARDROW route;
uint32_t dest_addr, source_addr;
char hostname[256];
HOSTENT *h;
if (rtmp->Link.hostname.av_len >= sizeof(hostname) - 1)
return;
strncpy(hostname, rtmp->Link.hostname.av_val, sizeof(hostname));
hostname[rtmp->Link.hostname.av_len] = 0;
h = gethostbyname(hostname);
if (!h)
return;
dest_addr = *(uint32_t *)h->h_addr_list[0];
if (rtmp->m_bindIP.addrLen == 0)
source_addr = 0;
else if (rtmp->m_bindIP.addr.ss_family == AF_INET)
2019-06-29 01:10:12 +02:00
source_addr = (*(struct sockaddr_in *)&rtmp->m_bindIP.addr)
.sin_addr.S_un.S_addr;
else
return;
if (!GetBestRoute(dest_addr, source_addr, &route)) {
MIB_IFROW row;
memset(&row, 0, sizeof(row));
row.dwIndex = route.dwForwardIfIndex;
if (!GetIfEntry(&row)) {
uint32_t speed = row.dwSpeed / 1000000;
char *type;
struct dstr other = {0};
if (row.dwType == IF_TYPE_ETHERNET_CSMACD) {
type = "ethernet";
} else if (row.dwType == IF_TYPE_IEEE80211) {
type = "802.11";
} else {
dstr_printf(&other, "type %lu", row.dwType);
type = other.array;
}
info("Interface: %s (%s, %lu mbps)", row.bDescr, type,
speed);
dstr_free(&other);
}
}
}
#endif
static void add_connect_data(char **penc, char *pend)
{
const AVal val = AVC("supportsGoAway");
*penc = AMF_EncodeNamedBoolean(*penc, pend, &val, true);
}
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);
// on reconnect we need to reset the internal variables of librtmp
// otherwise the data sent/received will not parse correctly on the other end
RTMP_Reset(&stream->rtmp);
// since we don't call RTMP_Init above, there's no other good place
// to reset this as doing it in RTMP_Close breaks the ugly RTMP
// authentication system
memset(&stream->rtmp.Link, 0, sizeof(stream->rtmp.Link));
stream->rtmp.last_error_code = 0;
if (!RTMP_SetupURL(&stream->rtmp, stream->path.array))
return OBS_OUTPUT_BAD_PATH;
Implement RTMP module (still needs drop code) - Implement the RTMP output module. This time around, we just use a simple FLV muxer, then just write to the stream with RTMP_Write. Easy and effective. - Fix the FLV muxer, the muxer now outputs proper FLV packets. - Output API: * When using encoders, automatically interleave encoded packets before sending it to the output. * Pair encoders and have them automatically wait for the other to start to ensure sync. * Change 'obs_output_signal_start_fail' to 'obs_output_signal_stop' because it was a bit confusing, and doing this makes a lot more sense for outputs that need to stop suddenly (disconnections/etc). - Encoder API: * Remove some unnecessary encoder functions from the actual API and make them internal. Most of the encoder functions are handled automatically by outputs anyway, so there's no real need to expose them and end up inadvertently confusing plugin writers. * Have audio encoders wait for the video encoder to get a frame, then start at the exact data point that the first video frame starts to ensure the most accrate sync of video/audio possible. * Add a required 'frame_size' callback for audio encoders that returns the expected number of frames desired to encode with. This way, the libobs encoder API can handle the circular buffering internally automatically for the encoder modules, so encoder writers don't have to do it themselves. - Fix a few bugs in the serializer interface. It was passing the wrong variable for the data in a few cases. - If a source has video, make obs_source_update defer the actual update callback until the tick function is called to prevent threading issues.
2014-04-07 22:00:10 -07:00
RTMP_EnableWrite(&stream->rtmp);
dstr_copy(&stream->encoder_name, "FMLE/3.0 (compatible; FMSc/1.0)");
set_rtmp_dstr(&stream->rtmp.Link.pubUser, &stream->username);
set_rtmp_dstr(&stream->rtmp.Link.pubPasswd, &stream->password);
set_rtmp_dstr(&stream->rtmp.Link.flashVer, &stream->encoder_name);
stream->rtmp.Link.swfUrl = stream->rtmp.Link.tcUrl;
stream->rtmp.Link.customConnectEncode = add_connect_data;
if (dstr_is_empty(&stream->bind_ip) ||
dstr_cmp(&stream->bind_ip, "default") == 0) {
memset(&stream->rtmp.m_bindIP, 0,
sizeof(stream->rtmp.m_bindIP));
} else {
bool success = netif_str_to_addr(&stream->rtmp.m_bindIP.addr,
&stream->rtmp.m_bindIP.addrLen,
stream->bind_ip.array);
if (success) {
int len = stream->rtmp.m_bindIP.addrLen;
bool ipv6 = len == sizeof(struct sockaddr_in6);
info("Binding to IPv%d", ipv6 ? 6 : 4);
}
}
RTMP_AddStream(&stream->rtmp, stream->key.array);
stream->rtmp.m_outChunkSize = 4096;
stream->rtmp.m_bSendChunkSizeInfo = true;
stream->rtmp.m_bUseNagle = true;
#ifdef _WIN32
win32_log_interface_type(stream);
#endif
if (!RTMP_Connect(&stream->rtmp, NULL)) {
set_output_error(stream);
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 bool init_connect(struct rtmp_stream *stream)
{
obs_service_t *service;
obs_data_t *settings;
const char *bind_ip;
int64_t drop_p;
int64_t drop_b;
uint32_t caps;
if (stopping(stream)) {
pthread_join(stream->send_thread, NULL);
}
free_packets(stream);
service = obs_output_get_service(stream->output);
if (!service)
return false;
2015-11-03 14:09:48 -08:00
os_atomic_set_bool(&stream->disconnected, false);
os_atomic_set_bool(&stream->encode_error, false);
stream->total_bytes_sent = 0;
stream->dropped_frames = 0;
stream->min_priority = 0;
stream->got_first_video = false;
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));
dstr_depad(&stream->path);
dstr_depad(&stream->key);
drop_b = (int64_t)obs_data_get_int(settings, OPT_DROP_THRESHOLD);
drop_p = (int64_t)obs_data_get_int(settings, OPT_PFRAME_DROP_THRESHOLD);
stream->max_shutdown_time_sec =
(int)obs_data_get_int(settings, OPT_MAX_SHUTDOWN_TIME_SEC);
obs_encoder_t *venc = obs_output_get_video_encoder(stream->output);
obs_encoder_t *aenc = obs_output_get_audio_encoder(stream->output, 0);
obs_data_t *vsettings = obs_encoder_get_settings(venc);
obs_data_t *asettings = obs_encoder_get_settings(aenc);
circlebuf_free(&stream->dbr_frames);
stream->audio_bitrate = (long)obs_data_get_int(asettings, "bitrate");
stream->dbr_data_size = 0;
stream->dbr_orig_bitrate = (long)obs_data_get_int(vsettings, "bitrate");
stream->dbr_cur_bitrate = stream->dbr_orig_bitrate;
stream->dbr_est_bitrate = 0;
stream->dbr_inc_bitrate = stream->dbr_orig_bitrate / 10;
stream->dbr_inc_timeout = 0;
stream->dbr_enabled = obs_data_get_bool(settings, OPT_DYN_BITRATE);
caps = obs_encoder_get_caps(venc);
if ((caps & OBS_ENCODER_CAP_DYN_BITRATE) == 0) {
stream->dbr_enabled = false;
}
if (obs_output_get_delay(stream->output) != 0) {
stream->dbr_enabled = false;
}
if (stream->dbr_enabled) {
info("Dynamic bitrate enabled. Dropped frames begone!");
}
obs_data_release(vsettings);
obs_data_release(asettings);
if (drop_p < (drop_b + 200))
drop_p = drop_b + 200;
stream->drop_threshold_usec = 1000 * drop_b;
stream->pframe_drop_threshold_usec = 1000 * drop_p;
bind_ip = obs_data_get_string(settings, OPT_BIND_IP);
dstr_copy(&stream->bind_ip, bind_ip);
stream->new_socket_loop =
obs_data_get_bool(settings, OPT_NEWSOCKETLOOP_ENABLED);
stream->low_latency_mode =
obs_data_get_bool(settings, OPT_LOWLATENCY_ENABLED);
// ugly hack for now, can be removed once new loop is reworked
if (stream->new_socket_loop &&
!strncmp(stream->path.array, "rtmps://", 8)) {
warn("Disabling network optimizations, not compatible with RTMPS");
stream->new_socket_loop = false;
}
obs_data_release(settings);
return true;
}
static void *connect_thread(void *data)
{
struct rtmp_stream *stream = data;
int ret;
os_set_thread_name("rtmp-stream: connect_thread");
if (!silently_reconnecting(stream)) {
if (!init_connect(stream)) {
obs_output_signal_stop(stream->output,
OBS_OUTPUT_BAD_PATH);
os_atomic_set_bool(&stream->silent_reconnect, false);
return NULL;
}
} else {
struct encoder_packet packet;
peek_next_packet(stream, &packet);
stream->start_dts_offset = get_ms_time(&packet, packet.dts);
}
ret = try_connect(stream);
if (ret != OBS_OUTPUT_SUCCESS) {
Implement RTMP module (still needs drop code) - Implement the RTMP output module. This time around, we just use a simple FLV muxer, then just write to the stream with RTMP_Write. Easy and effective. - Fix the FLV muxer, the muxer now outputs proper FLV packets. - Output API: * When using encoders, automatically interleave encoded packets before sending it to the output. * Pair encoders and have them automatically wait for the other to start to ensure sync. * Change 'obs_output_signal_start_fail' to 'obs_output_signal_stop' because it was a bit confusing, and doing this makes a lot more sense for outputs that need to stop suddenly (disconnections/etc). - Encoder API: * Remove some unnecessary encoder functions from the actual API and make them internal. Most of the encoder functions are handled automatically by outputs anyway, so there's no real need to expose them and end up inadvertently confusing plugin writers. * Have audio encoders wait for the video encoder to get a frame, then start at the exact data point that the first video frame starts to ensure the most accrate sync of video/audio possible. * Add a required 'frame_size' callback for audio encoders that returns the expected number of frames desired to encode with. This way, the libobs encoder API can handle the circular buffering internally automatically for the encoder modules, so encoder writers don't have to do it themselves. - Fix a few bugs in the serializer interface. It was passing the wrong variable for the data in a few cases. - If a source has video, make obs_source_update defer the actual update callback until the tick function is called to prevent threading issues.
2014-04-07 22:00:10 -07:00
obs_output_signal_stop(stream->output, ret);
info("Connection to %s failed: %d", stream->path.array, ret);
}
if (!stopping(stream))
pthread_detach(stream->connect_thread);
os_atomic_set_bool(&stream->silent_reconnect, false);
2015-11-03 14:09:48 -08:00
os_atomic_set_bool(&stream->connecting, false);
return NULL;
}
static bool rtmp_stream_start(void *data)
{
struct rtmp_stream *stream = data;
if (!silently_reconnecting(stream)) {
if (!obs_output_can_begin_data_capture(stream->output, 0))
return false;
if (!obs_output_initialize_encoders(stream->output, 0))
return false;
}
2015-11-03 14:09:48 -08:00
os_atomic_set_bool(&stream->connecting, true);
return pthread_create(&stream->connect_thread, NULL, connect_thread,
Implement RTMP module (still needs drop code) - Implement the RTMP output module. This time around, we just use a simple FLV muxer, then just write to the stream with RTMP_Write. Easy and effective. - Fix the FLV muxer, the muxer now outputs proper FLV packets. - Output API: * When using encoders, automatically interleave encoded packets before sending it to the output. * Pair encoders and have them automatically wait for the other to start to ensure sync. * Change 'obs_output_signal_start_fail' to 'obs_output_signal_stop' because it was a bit confusing, and doing this makes a lot more sense for outputs that need to stop suddenly (disconnections/etc). - Encoder API: * Remove some unnecessary encoder functions from the actual API and make them internal. Most of the encoder functions are handled automatically by outputs anyway, so there's no real need to expose them and end up inadvertently confusing plugin writers. * Have audio encoders wait for the video encoder to get a frame, then start at the exact data point that the first video frame starts to ensure the most accrate sync of video/audio possible. * Add a required 'frame_size' callback for audio encoders that returns the expected number of frames desired to encode with. This way, the libobs encoder API can handle the circular buffering internally automatically for the encoder modules, so encoder writers don't have to do it themselves. - Fix a few bugs in the serializer interface. It was passing the wrong variable for the data in a few cases. - If a source has video, make obs_source_update defer the actual update callback until the tick function is called to prevent threading issues.
2014-04-07 22:00:10 -07:00
stream) == 0;
}
RTMP output: Implement frame drop code A little bit of history about frame dropping: I did a large number of experiments with frame dropping in old versions of OBS1, and it's not an easy thing to deal with. I tried just about everything from standard i-frame delay, to large buffers, to dumping packets, to super-unnecessarily-complex things that just ended up causing more problems than they was worth. When I did my experiments, I found that the most ideal frame drop system (in terms of reducing the amount of total data that needed to be dropped) was in the 0.4xx days where I had a 3 second frame-drop buffer where I could calculate the actual buffer size in bytes, and then intellgently choose packets in that buffer to trim it down to a specific size while minimizing the number of p-frames and i-frames dropped, and preventing the actual impact of dropped frames on the stream. The downside of it was that it required too much extra latency, and far too many people complained about it, so it was removed in favor of the current system. The current system I just refer to just as 'packet dumping', which when combined with low keyframe intervals (like most services use these days), is the next-best method from my experience. Just dump the buffer when you reach a threshold of buffering (which I prefer to measure with time rather than in size), then wait for a new i-frame. Simple, effective, and reduces the risk of consecutive buffering, while still having fairly low impact on the stream output due to the low keyframe interval of services. By the way, audio will not (and should not ever) be dropped, lest you end up with syncing issues (among other nasty things) specific to server implementation.
2014-04-12 04:34:15 -07:00
static inline bool add_packet(struct rtmp_stream *stream,
struct encoder_packet *packet)
{
circlebuf_push_back(&stream->packets, packet,
sizeof(struct encoder_packet));
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, const char *name,
int highest_priority, bool pframes)
RTMP output: Implement frame drop code A little bit of history about frame dropping: I did a large number of experiments with frame dropping in old versions of OBS1, and it's not an easy thing to deal with. I tried just about everything from standard i-frame delay, to large buffers, to dumping packets, to super-unnecessarily-complex things that just ended up causing more problems than they was worth. When I did my experiments, I found that the most ideal frame drop system (in terms of reducing the amount of total data that needed to be dropped) was in the 0.4xx days where I had a 3 second frame-drop buffer where I could calculate the actual buffer size in bytes, and then intellgently choose packets in that buffer to trim it down to a specific size while minimizing the number of p-frames and i-frames dropped, and preventing the actual impact of dropped frames on the stream. The downside of it was that it required too much extra latency, and far too many people complained about it, so it was removed in favor of the current system. The current system I just refer to just as 'packet dumping', which when combined with low keyframe intervals (like most services use these days), is the next-best method from my experience. Just dump the buffer when you reach a threshold of buffering (which I prefer to measure with time rather than in size), then wait for a new i-frame. Simple, effective, and reduces the risk of consecutive buffering, while still having fairly low impact on the stream output due to the low keyframe interval of services. By the way, audio will not (and should not ever) be dropped, lest you end up with syncing issues (among other nasty things) specific to server implementation.
2014-04-12 04:34:15 -07:00
{
UNUSED_PARAMETER(pframes);
RTMP output: Implement frame drop code A little bit of history about frame dropping: I did a large number of experiments with frame dropping in old versions of OBS1, and it's not an easy thing to deal with. I tried just about everything from standard i-frame delay, to large buffers, to dumping packets, to super-unnecessarily-complex things that just ended up causing more problems than they was worth. When I did my experiments, I found that the most ideal frame drop system (in terms of reducing the amount of total data that needed to be dropped) was in the 0.4xx days where I had a 3 second frame-drop buffer where I could calculate the actual buffer size in bytes, and then intellgently choose packets in that buffer to trim it down to a specific size while minimizing the number of p-frames and i-frames dropped, and preventing the actual impact of dropped frames on the stream. The downside of it was that it required too much extra latency, and far too many people complained about it, so it was removed in favor of the current system. The current system I just refer to just as 'packet dumping', which when combined with low keyframe intervals (like most services use these days), is the next-best method from my experience. Just dump the buffer when you reach a threshold of buffering (which I prefer to measure with time rather than in size), then wait for a new i-frame. Simple, effective, and reduces the risk of consecutive buffering, while still having fairly low impact on the stream output due to the low keyframe interval of services. By the way, audio will not (and should not ever) be dropped, lest you end up with syncing issues (among other nasty things) specific to server implementation.
2014-04-12 04:34:15 -07:00
struct circlebuf new_buf = {0};
int num_frames_dropped = 0;
RTMP output: Implement frame drop code A little bit of history about frame dropping: I did a large number of experiments with frame dropping in old versions of OBS1, and it's not an easy thing to deal with. I tried just about everything from standard i-frame delay, to large buffers, to dumping packets, to super-unnecessarily-complex things that just ended up causing more problems than they was worth. When I did my experiments, I found that the most ideal frame drop system (in terms of reducing the amount of total data that needed to be dropped) was in the 0.4xx days where I had a 3 second frame-drop buffer where I could calculate the actual buffer size in bytes, and then intellgently choose packets in that buffer to trim it down to a specific size while minimizing the number of p-frames and i-frames dropped, and preventing the actual impact of dropped frames on the stream. The downside of it was that it required too much extra latency, and far too many people complained about it, so it was removed in favor of the current system. The current system I just refer to just as 'packet dumping', which when combined with low keyframe intervals (like most services use these days), is the next-best method from my experience. Just dump the buffer when you reach a threshold of buffering (which I prefer to measure with time rather than in size), then wait for a new i-frame. Simple, effective, and reduces the risk of consecutive buffering, while still having fairly low impact on the stream output due to the low keyframe interval of services. By the way, audio will not (and should not ever) be dropped, lest you end up with syncing issues (among other nasty things) specific to server implementation.
2014-04-12 04:34:15 -07:00
#ifdef _DEBUG
int start_packets = (int)num_buffered_packets(stream);
#else
UNUSED_PARAMETER(name);
#endif
RTMP output: Implement frame drop code A little bit of history about frame dropping: I did a large number of experiments with frame dropping in old versions of OBS1, and it's not an easy thing to deal with. I tried just about everything from standard i-frame delay, to large buffers, to dumping packets, to super-unnecessarily-complex things that just ended up causing more problems than they was worth. When I did my experiments, I found that the most ideal frame drop system (in terms of reducing the amount of total data that needed to be dropped) was in the 0.4xx days where I had a 3 second frame-drop buffer where I could calculate the actual buffer size in bytes, and then intellgently choose packets in that buffer to trim it down to a specific size while minimizing the number of p-frames and i-frames dropped, and preventing the actual impact of dropped frames on the stream. The downside of it was that it required too much extra latency, and far too many people complained about it, so it was removed in favor of the current system. The current system I just refer to just as 'packet dumping', which when combined with low keyframe intervals (like most services use these days), is the next-best method from my experience. Just dump the buffer when you reach a threshold of buffering (which I prefer to measure with time rather than in size), then wait for a new i-frame. Simple, effective, and reduces the risk of consecutive buffering, while still having fairly low impact on the stream output due to the low keyframe interval of services. By the way, audio will not (and should not ever) be dropped, lest you end up with syncing issues (among other nasty things) specific to server implementation.
2014-04-12 04:34:15 -07:00
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));
/* do not drop audio data or video keyframes */
if (packet.type == OBS_ENCODER_AUDIO ||
packet.drop_priority >= highest_priority) {
RTMP output: Implement frame drop code A little bit of history about frame dropping: I did a large number of experiments with frame dropping in old versions of OBS1, and it's not an easy thing to deal with. I tried just about everything from standard i-frame delay, to large buffers, to dumping packets, to super-unnecessarily-complex things that just ended up causing more problems than they was worth. When I did my experiments, I found that the most ideal frame drop system (in terms of reducing the amount of total data that needed to be dropped) was in the 0.4xx days where I had a 3 second frame-drop buffer where I could calculate the actual buffer size in bytes, and then intellgently choose packets in that buffer to trim it down to a specific size while minimizing the number of p-frames and i-frames dropped, and preventing the actual impact of dropped frames on the stream. The downside of it was that it required too much extra latency, and far too many people complained about it, so it was removed in favor of the current system. The current system I just refer to just as 'packet dumping', which when combined with low keyframe intervals (like most services use these days), is the next-best method from my experience. Just dump the buffer when you reach a threshold of buffering (which I prefer to measure with time rather than in size), then wait for a new i-frame. Simple, effective, and reduces the risk of consecutive buffering, while still having fairly low impact on the stream output due to the low keyframe interval of services. By the way, audio will not (and should not ever) be dropped, lest you end up with syncing issues (among other nasty things) specific to server implementation.
2014-04-12 04:34:15 -07:00
circlebuf_push_back(&new_buf, &packet, sizeof(packet));
} else {
num_frames_dropped++;
obs_encoder_packet_release(&packet);
RTMP output: Implement frame drop code A little bit of history about frame dropping: I did a large number of experiments with frame dropping in old versions of OBS1, and it's not an easy thing to deal with. I tried just about everything from standard i-frame delay, to large buffers, to dumping packets, to super-unnecessarily-complex things that just ended up causing more problems than they was worth. When I did my experiments, I found that the most ideal frame drop system (in terms of reducing the amount of total data that needed to be dropped) was in the 0.4xx days where I had a 3 second frame-drop buffer where I could calculate the actual buffer size in bytes, and then intellgently choose packets in that buffer to trim it down to a specific size while minimizing the number of p-frames and i-frames dropped, and preventing the actual impact of dropped frames on the stream. The downside of it was that it required too much extra latency, and far too many people complained about it, so it was removed in favor of the current system. The current system I just refer to just as 'packet dumping', which when combined with low keyframe intervals (like most services use these days), is the next-best method from my experience. Just dump the buffer when you reach a threshold of buffering (which I prefer to measure with time rather than in size), then wait for a new i-frame. Simple, effective, and reduces the risk of consecutive buffering, while still having fairly low impact on the stream output due to the low keyframe interval of services. By the way, audio will not (and should not ever) be dropped, lest you end up with syncing issues (among other nasty things) specific to server implementation.
2014-04-12 04:34:15 -07:00
}
}
circlebuf_free(&stream->packets);
stream->packets = new_buf;
if (stream->min_priority < highest_priority)
stream->min_priority = highest_priority;
if (!num_frames_dropped)
return;
RTMP output: Implement frame drop code A little bit of history about frame dropping: I did a large number of experiments with frame dropping in old versions of OBS1, and it's not an easy thing to deal with. I tried just about everything from standard i-frame delay, to large buffers, to dumping packets, to super-unnecessarily-complex things that just ended up causing more problems than they was worth. When I did my experiments, I found that the most ideal frame drop system (in terms of reducing the amount of total data that needed to be dropped) was in the 0.4xx days where I had a 3 second frame-drop buffer where I could calculate the actual buffer size in bytes, and then intellgently choose packets in that buffer to trim it down to a specific size while minimizing the number of p-frames and i-frames dropped, and preventing the actual impact of dropped frames on the stream. The downside of it was that it required too much extra latency, and far too many people complained about it, so it was removed in favor of the current system. The current system I just refer to just as 'packet dumping', which when combined with low keyframe intervals (like most services use these days), is the next-best method from my experience. Just dump the buffer when you reach a threshold of buffering (which I prefer to measure with time rather than in size), then wait for a new i-frame. Simple, effective, and reduces the risk of consecutive buffering, while still having fairly low impact on the stream output due to the low keyframe interval of services. By the way, audio will not (and should not ever) be dropped, lest you end up with syncing issues (among other nasty things) specific to server implementation.
2014-04-12 04:34:15 -07:00
stream->dropped_frames += num_frames_dropped;
#ifdef _DEBUG
debug("Dropped %s, prev packet count: %d, new packet count: %d", name,
start_packets, (int)num_buffered_packets(stream));
#endif
RTMP output: Implement frame drop code A little bit of history about frame dropping: I did a large number of experiments with frame dropping in old versions of OBS1, and it's not an easy thing to deal with. I tried just about everything from standard i-frame delay, to large buffers, to dumping packets, to super-unnecessarily-complex things that just ended up causing more problems than they was worth. When I did my experiments, I found that the most ideal frame drop system (in terms of reducing the amount of total data that needed to be dropped) was in the 0.4xx days where I had a 3 second frame-drop buffer where I could calculate the actual buffer size in bytes, and then intellgently choose packets in that buffer to trim it down to a specific size while minimizing the number of p-frames and i-frames dropped, and preventing the actual impact of dropped frames on the stream. The downside of it was that it required too much extra latency, and far too many people complained about it, so it was removed in favor of the current system. The current system I just refer to just as 'packet dumping', which when combined with low keyframe intervals (like most services use these days), is the next-best method from my experience. Just dump the buffer when you reach a threshold of buffering (which I prefer to measure with time rather than in size), then wait for a new i-frame. Simple, effective, and reduces the risk of consecutive buffering, while still having fairly low impact on the stream output due to the low keyframe interval of services. By the way, audio will not (and should not ever) be dropped, lest you end up with syncing issues (among other nasty things) specific to server implementation.
2014-04-12 04:34:15 -07:00
}
static bool find_first_video_packet(struct rtmp_stream *stream,
struct encoder_packet *first)
{
size_t count = stream->packets.size / sizeof(*first);
for (size_t i = 0; i < count; i++) {
struct encoder_packet *cur =
circlebuf_data(&stream->packets, i * sizeof(*first));
if (cur->type == OBS_ENCODER_VIDEO && !cur->keyframe) {
*first = *cur;
return true;
}
}
return false;
}
static bool dbr_bitrate_lowered(struct rtmp_stream *stream)
{
long prev_bitrate = stream->dbr_prev_bitrate;
long est_bitrate = 0;
long new_bitrate;
if (stream->dbr_est_bitrate &&
stream->dbr_est_bitrate < stream->dbr_cur_bitrate) {
stream->dbr_data_size = 0;
circlebuf_pop_front(&stream->dbr_frames, NULL,
stream->dbr_frames.size);
est_bitrate = stream->dbr_est_bitrate / 100 * 100;
if (est_bitrate < 50) {
est_bitrate = 50;
}
}
#if 0
if (prev_bitrate && est_bitrate) {
if (prev_bitrate < est_bitrate) {
blog(LOG_INFO, "going back to prev bitrate: "
"prev_bitrate (%d) < est_bitrate (%d)",
prev_bitrate,
est_bitrate);
new_bitrate = prev_bitrate;
} else {
new_bitrate = est_bitrate;
}
new_bitrate = est_bitrate;
} else if (prev_bitrate) {
new_bitrate = prev_bitrate;
info("going back to prev bitrate");
} else if (est_bitrate) {
new_bitrate = est_bitrate;
} else {
return false;
}
#else
if (est_bitrate) {
new_bitrate = est_bitrate;
} else if (prev_bitrate) {
new_bitrate = prev_bitrate;
info("going back to prev bitrate");
} else {
return false;
}
if (new_bitrate == stream->dbr_cur_bitrate) {
return false;
}
#endif
stream->dbr_prev_bitrate = 0;
stream->dbr_cur_bitrate = new_bitrate;
stream->dbr_inc_timeout = os_gettime_ns() + DBR_INC_TIMER;
info("bitrate decreased to: %ld", stream->dbr_cur_bitrate);
return true;
}
static void dbr_set_bitrate(struct rtmp_stream *stream)
{
obs_encoder_t *vencoder = obs_output_get_video_encoder(stream->output);
obs_data_t *settings = obs_encoder_get_settings(vencoder);
obs_data_set_int(settings, "bitrate", stream->dbr_cur_bitrate);
obs_encoder_update(vencoder, settings);
obs_data_release(settings);
}
static void dbr_inc_bitrate(struct rtmp_stream *stream)
{
stream->dbr_prev_bitrate = stream->dbr_cur_bitrate;
stream->dbr_cur_bitrate += stream->dbr_inc_bitrate;
if (stream->dbr_cur_bitrate >= stream->dbr_orig_bitrate) {
stream->dbr_cur_bitrate = stream->dbr_orig_bitrate;
info("bitrate increased to: %ld, done",
stream->dbr_cur_bitrate);
} else if (stream->dbr_cur_bitrate < stream->dbr_orig_bitrate) {
stream->dbr_inc_timeout = os_gettime_ns() + DBR_INC_TIMER;
info("bitrate increased to: %ld, waiting",
stream->dbr_cur_bitrate);
}
}
static void check_to_drop_frames(struct rtmp_stream *stream, bool pframes)
RTMP output: Implement frame drop code A little bit of history about frame dropping: I did a large number of experiments with frame dropping in old versions of OBS1, and it's not an easy thing to deal with. I tried just about everything from standard i-frame delay, to large buffers, to dumping packets, to super-unnecessarily-complex things that just ended up causing more problems than they was worth. When I did my experiments, I found that the most ideal frame drop system (in terms of reducing the amount of total data that needed to be dropped) was in the 0.4xx days where I had a 3 second frame-drop buffer where I could calculate the actual buffer size in bytes, and then intellgently choose packets in that buffer to trim it down to a specific size while minimizing the number of p-frames and i-frames dropped, and preventing the actual impact of dropped frames on the stream. The downside of it was that it required too much extra latency, and far too many people complained about it, so it was removed in favor of the current system. The current system I just refer to just as 'packet dumping', which when combined with low keyframe intervals (like most services use these days), is the next-best method from my experience. Just dump the buffer when you reach a threshold of buffering (which I prefer to measure with time rather than in size), then wait for a new i-frame. Simple, effective, and reduces the risk of consecutive buffering, while still having fairly low impact on the stream output due to the low keyframe interval of services. By the way, audio will not (and should not ever) be dropped, lest you end up with syncing issues (among other nasty things) specific to server implementation.
2014-04-12 04:34:15 -07:00
{
struct encoder_packet first;
int64_t buffer_duration_usec;
size_t num_packets = num_buffered_packets(stream);
const char *name = pframes ? "p-frames" : "b-frames";
int priority = pframes ? OBS_NAL_PRIORITY_HIGHEST
: OBS_NAL_PRIORITY_HIGH;
int64_t drop_threshold = pframes ? stream->pframe_drop_threshold_usec
: stream->drop_threshold_usec;
if (!pframes && stream->dbr_enabled) {
if (stream->dbr_inc_timeout) {
uint64_t t = os_gettime_ns();
if (t >= stream->dbr_inc_timeout) {
stream->dbr_inc_timeout = 0;
dbr_inc_bitrate(stream);
dbr_set_bitrate(stream);
}
}
}
if (num_packets < 5) {
if (!pframes)
stream->congestion = 0.0f;
RTMP output: Implement frame drop code A little bit of history about frame dropping: I did a large number of experiments with frame dropping in old versions of OBS1, and it's not an easy thing to deal with. I tried just about everything from standard i-frame delay, to large buffers, to dumping packets, to super-unnecessarily-complex things that just ended up causing more problems than they was worth. When I did my experiments, I found that the most ideal frame drop system (in terms of reducing the amount of total data that needed to be dropped) was in the 0.4xx days where I had a 3 second frame-drop buffer where I could calculate the actual buffer size in bytes, and then intellgently choose packets in that buffer to trim it down to a specific size while minimizing the number of p-frames and i-frames dropped, and preventing the actual impact of dropped frames on the stream. The downside of it was that it required too much extra latency, and far too many people complained about it, so it was removed in favor of the current system. The current system I just refer to just as 'packet dumping', which when combined with low keyframe intervals (like most services use these days), is the next-best method from my experience. Just dump the buffer when you reach a threshold of buffering (which I prefer to measure with time rather than in size), then wait for a new i-frame. Simple, effective, and reduces the risk of consecutive buffering, while still having fairly low impact on the stream output due to the low keyframe interval of services. By the way, audio will not (and should not ever) be dropped, lest you end up with syncing issues (among other nasty things) specific to server implementation.
2014-04-12 04:34:15 -07:00
return;
}
RTMP output: Implement frame drop code A little bit of history about frame dropping: I did a large number of experiments with frame dropping in old versions of OBS1, and it's not an easy thing to deal with. I tried just about everything from standard i-frame delay, to large buffers, to dumping packets, to super-unnecessarily-complex things that just ended up causing more problems than they was worth. When I did my experiments, I found that the most ideal frame drop system (in terms of reducing the amount of total data that needed to be dropped) was in the 0.4xx days where I had a 3 second frame-drop buffer where I could calculate the actual buffer size in bytes, and then intellgently choose packets in that buffer to trim it down to a specific size while minimizing the number of p-frames and i-frames dropped, and preventing the actual impact of dropped frames on the stream. The downside of it was that it required too much extra latency, and far too many people complained about it, so it was removed in favor of the current system. The current system I just refer to just as 'packet dumping', which when combined with low keyframe intervals (like most services use these days), is the next-best method from my experience. Just dump the buffer when you reach a threshold of buffering (which I prefer to measure with time rather than in size), then wait for a new i-frame. Simple, effective, and reduces the risk of consecutive buffering, while still having fairly low impact on the stream output due to the low keyframe interval of services. By the way, audio will not (and should not ever) be dropped, lest you end up with syncing issues (among other nasty things) specific to server implementation.
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if (!find_first_video_packet(stream, &first))
RTMP output: Implement frame drop code A little bit of history about frame dropping: I did a large number of experiments with frame dropping in old versions of OBS1, and it's not an easy thing to deal with. I tried just about everything from standard i-frame delay, to large buffers, to dumping packets, to super-unnecessarily-complex things that just ended up causing more problems than they was worth. When I did my experiments, I found that the most ideal frame drop system (in terms of reducing the amount of total data that needed to be dropped) was in the 0.4xx days where I had a 3 second frame-drop buffer where I could calculate the actual buffer size in bytes, and then intellgently choose packets in that buffer to trim it down to a specific size while minimizing the number of p-frames and i-frames dropped, and preventing the actual impact of dropped frames on the stream. The downside of it was that it required too much extra latency, and far too many people complained about it, so it was removed in favor of the current system. The current system I just refer to just as 'packet dumping', which when combined with low keyframe intervals (like most services use these days), is the next-best method from my experience. Just dump the buffer when you reach a threshold of buffering (which I prefer to measure with time rather than in size), then wait for a new i-frame. Simple, effective, and reduces the risk of consecutive buffering, while still having fairly low impact on the stream output due to the low keyframe interval of services. By the way, audio will not (and should not ever) be dropped, lest you end up with syncing issues (among other nasty things) specific to server implementation.
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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 (!pframes) {
stream->congestion =
(float)buffer_duration_usec / (float)drop_threshold;
}
/* alternatively, drop only pframes:
* (!pframes && stream->dbr_enabled)
* but let's test without dropping frames
* at all first */
if (stream->dbr_enabled) {
bool bitrate_changed = false;
if (pframes) {
return;
}
if ((uint64_t)buffer_duration_usec >= DBR_TRIGGER_USEC) {
pthread_mutex_lock(&stream->dbr_mutex);
bitrate_changed = dbr_bitrate_lowered(stream);
pthread_mutex_unlock(&stream->dbr_mutex);
}
if (bitrate_changed) {
debug("buffer_duration_msec: %" PRId64,
buffer_duration_usec / 1000);
dbr_set_bitrate(stream);
}
return;
}
if (buffer_duration_usec > drop_threshold) {
debug("buffer_duration_usec: %" PRId64, buffer_duration_usec);
drop_frames(stream, name, priority, pframes);
RTMP output: Implement frame drop code A little bit of history about frame dropping: I did a large number of experiments with frame dropping in old versions of OBS1, and it's not an easy thing to deal with. I tried just about everything from standard i-frame delay, to large buffers, to dumping packets, to super-unnecessarily-complex things that just ended up causing more problems than they was worth. When I did my experiments, I found that the most ideal frame drop system (in terms of reducing the amount of total data that needed to be dropped) was in the 0.4xx days where I had a 3 second frame-drop buffer where I could calculate the actual buffer size in bytes, and then intellgently choose packets in that buffer to trim it down to a specific size while minimizing the number of p-frames and i-frames dropped, and preventing the actual impact of dropped frames on the stream. The downside of it was that it required too much extra latency, and far too many people complained about it, so it was removed in favor of the current system. The current system I just refer to just as 'packet dumping', which when combined with low keyframe intervals (like most services use these days), is the next-best method from my experience. Just dump the buffer when you reach a threshold of buffering (which I prefer to measure with time rather than in size), then wait for a new i-frame. Simple, effective, and reduces the risk of consecutive buffering, while still having fairly low impact on the stream output due to the low keyframe interval of services. By the way, audio will not (and should not ever) be dropped, lest you end up with syncing issues (among other nasty things) specific to server implementation.
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}
}
static bool add_video_packet(struct rtmp_stream *stream,
struct encoder_packet *packet)
{
if (!silently_reconnecting(stream)) {
check_to_drop_frames(stream, false);
check_to_drop_frames(stream, true);
}
RTMP output: Implement frame drop code A little bit of history about frame dropping: I did a large number of experiments with frame dropping in old versions of OBS1, and it's not an easy thing to deal with. I tried just about everything from standard i-frame delay, to large buffers, to dumping packets, to super-unnecessarily-complex things that just ended up causing more problems than they was worth. When I did my experiments, I found that the most ideal frame drop system (in terms of reducing the amount of total data that needed to be dropped) was in the 0.4xx days where I had a 3 second frame-drop buffer where I could calculate the actual buffer size in bytes, and then intellgently choose packets in that buffer to trim it down to a specific size while minimizing the number of p-frames and i-frames dropped, and preventing the actual impact of dropped frames on the stream. The downside of it was that it required too much extra latency, and far too many people complained about it, so it was removed in favor of the current system. The current system I just refer to just as 'packet dumping', which when combined with low keyframe intervals (like most services use these days), is the next-best method from my experience. Just dump the buffer when you reach a threshold of buffering (which I prefer to measure with time rather than in size), then wait for a new i-frame. Simple, effective, and reduces the risk of consecutive buffering, while still having fairly low impact on the stream output due to the low keyframe interval of services. By the way, audio will not (and should not ever) be dropped, lest you end up with syncing issues (among other nasty things) specific to server implementation.
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/* if currently dropping frames, drop packets until it reaches the
* desired priority */
if (packet->drop_priority < stream->min_priority) {
stream->dropped_frames++;
RTMP output: Implement frame drop code A little bit of history about frame dropping: I did a large number of experiments with frame dropping in old versions of OBS1, and it's not an easy thing to deal with. I tried just about everything from standard i-frame delay, to large buffers, to dumping packets, to super-unnecessarily-complex things that just ended up causing more problems than they was worth. When I did my experiments, I found that the most ideal frame drop system (in terms of reducing the amount of total data that needed to be dropped) was in the 0.4xx days where I had a 3 second frame-drop buffer where I could calculate the actual buffer size in bytes, and then intellgently choose packets in that buffer to trim it down to a specific size while minimizing the number of p-frames and i-frames dropped, and preventing the actual impact of dropped frames on the stream. The downside of it was that it required too much extra latency, and far too many people complained about it, so it was removed in favor of the current system. The current system I just refer to just as 'packet dumping', which when combined with low keyframe intervals (like most services use these days), is the next-best method from my experience. Just dump the buffer when you reach a threshold of buffering (which I prefer to measure with time rather than in size), then wait for a new i-frame. Simple, effective, and reduces the risk of consecutive buffering, while still having fairly low impact on the stream output due to the low keyframe interval of services. By the way, audio will not (and should not ever) be dropped, lest you end up with syncing issues (among other nasty things) specific to server implementation.
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return false;
} else {
RTMP output: Implement frame drop code A little bit of history about frame dropping: I did a large number of experiments with frame dropping in old versions of OBS1, and it's not an easy thing to deal with. I tried just about everything from standard i-frame delay, to large buffers, to dumping packets, to super-unnecessarily-complex things that just ended up causing more problems than they was worth. When I did my experiments, I found that the most ideal frame drop system (in terms of reducing the amount of total data that needed to be dropped) was in the 0.4xx days where I had a 3 second frame-drop buffer where I could calculate the actual buffer size in bytes, and then intellgently choose packets in that buffer to trim it down to a specific size while minimizing the number of p-frames and i-frames dropped, and preventing the actual impact of dropped frames on the stream. The downside of it was that it required too much extra latency, and far too many people complained about it, so it was removed in favor of the current system. The current system I just refer to just as 'packet dumping', which when combined with low keyframe intervals (like most services use these days), is the next-best method from my experience. Just dump the buffer when you reach a threshold of buffering (which I prefer to measure with time rather than in size), then wait for a new i-frame. Simple, effective, and reduces the risk of consecutive buffering, while still having fairly low impact on the stream output due to the low keyframe interval of services. By the way, audio will not (and should not ever) be dropped, lest you end up with syncing issues (among other nasty things) specific to server implementation.
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stream->min_priority = 0;
}
RTMP output: Implement frame drop code A little bit of history about frame dropping: I did a large number of experiments with frame dropping in old versions of OBS1, and it's not an easy thing to deal with. I tried just about everything from standard i-frame delay, to large buffers, to dumping packets, to super-unnecessarily-complex things that just ended up causing more problems than they was worth. When I did my experiments, I found that the most ideal frame drop system (in terms of reducing the amount of total data that needed to be dropped) was in the 0.4xx days where I had a 3 second frame-drop buffer where I could calculate the actual buffer size in bytes, and then intellgently choose packets in that buffer to trim it down to a specific size while minimizing the number of p-frames and i-frames dropped, and preventing the actual impact of dropped frames on the stream. The downside of it was that it required too much extra latency, and far too many people complained about it, so it was removed in favor of the current system. The current system I just refer to just as 'packet dumping', which when combined with low keyframe intervals (like most services use these days), is the next-best method from my experience. Just dump the buffer when you reach a threshold of buffering (which I prefer to measure with time rather than in size), then wait for a new i-frame. Simple, effective, and reduces the risk of consecutive buffering, while still having fairly low impact on the stream output due to the low keyframe interval of services. By the way, audio will not (and should not ever) be dropped, lest you end up with syncing issues (among other nasty things) specific to server implementation.
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stream->last_dts_usec = packet->dts_usec;
RTMP output: Implement frame drop code A little bit of history about frame dropping: I did a large number of experiments with frame dropping in old versions of OBS1, and it's not an easy thing to deal with. I tried just about everything from standard i-frame delay, to large buffers, to dumping packets, to super-unnecessarily-complex things that just ended up causing more problems than they was worth. When I did my experiments, I found that the most ideal frame drop system (in terms of reducing the amount of total data that needed to be dropped) was in the 0.4xx days where I had a 3 second frame-drop buffer where I could calculate the actual buffer size in bytes, and then intellgently choose packets in that buffer to trim it down to a specific size while minimizing the number of p-frames and i-frames dropped, and preventing the actual impact of dropped frames on the stream. The downside of it was that it required too much extra latency, and far too many people complained about it, so it was removed in favor of the current system. The current system I just refer to just as 'packet dumping', which when combined with low keyframe intervals (like most services use these days), is the next-best method from my experience. Just dump the buffer when you reach a threshold of buffering (which I prefer to measure with time rather than in size), then wait for a new i-frame. Simple, effective, and reduces the risk of consecutive buffering, while still having fairly low impact on the stream output due to the low keyframe interval of services. By the way, audio will not (and should not ever) be dropped, lest you end up with syncing issues (among other nasty things) specific to server implementation.
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return add_packet(stream, packet);
}
static void rtmp_stream_data(void *data, struct encoder_packet *packet)
{
RTMP output: Implement frame drop code A little bit of history about frame dropping: I did a large number of experiments with frame dropping in old versions of OBS1, and it's not an easy thing to deal with. I tried just about everything from standard i-frame delay, to large buffers, to dumping packets, to super-unnecessarily-complex things that just ended up causing more problems than they was worth. When I did my experiments, I found that the most ideal frame drop system (in terms of reducing the amount of total data that needed to be dropped) was in the 0.4xx days where I had a 3 second frame-drop buffer where I could calculate the actual buffer size in bytes, and then intellgently choose packets in that buffer to trim it down to a specific size while minimizing the number of p-frames and i-frames dropped, and preventing the actual impact of dropped frames on the stream. The downside of it was that it required too much extra latency, and far too many people complained about it, so it was removed in favor of the current system. The current system I just refer to just as 'packet dumping', which when combined with low keyframe intervals (like most services use these days), is the next-best method from my experience. Just dump the buffer when you reach a threshold of buffering (which I prefer to measure with time rather than in size), then wait for a new i-frame. Simple, effective, and reduces the risk of consecutive buffering, while still having fairly low impact on the stream output due to the low keyframe interval of services. By the way, audio will not (and should not ever) be dropped, lest you end up with syncing issues (among other nasty things) specific to server implementation.
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struct rtmp_stream *stream = data;
struct encoder_packet new_packet;
bool added_packet = false;
if (disconnected(stream) || !active(stream))
return;
/* encoder fail */
if (!packet) {
os_atomic_set_bool(&stream->encode_error, true);
os_sem_post(stream->send_sem);
return;
}
if (packet->type == OBS_ENCODER_VIDEO) {
if (!stream->got_first_video) {
stream->start_dts_offset =
get_ms_time(packet, packet->dts);
stream->got_first_video = true;
}
obs_parse_avc_packet(&new_packet, packet);
} else {
obs_encoder_packet_ref(&new_packet, packet);
}
pthread_mutex_lock(&stream->packets_mutex);
RTMP output: Implement frame drop code A little bit of history about frame dropping: I did a large number of experiments with frame dropping in old versions of OBS1, and it's not an easy thing to deal with. I tried just about everything from standard i-frame delay, to large buffers, to dumping packets, to super-unnecessarily-complex things that just ended up causing more problems than they was worth. When I did my experiments, I found that the most ideal frame drop system (in terms of reducing the amount of total data that needed to be dropped) was in the 0.4xx days where I had a 3 second frame-drop buffer where I could calculate the actual buffer size in bytes, and then intellgently choose packets in that buffer to trim it down to a specific size while minimizing the number of p-frames and i-frames dropped, and preventing the actual impact of dropped frames on the stream. The downside of it was that it required too much extra latency, and far too many people complained about it, so it was removed in favor of the current system. The current system I just refer to just as 'packet dumping', which when combined with low keyframe intervals (like most services use these days), is the next-best method from my experience. Just dump the buffer when you reach a threshold of buffering (which I prefer to measure with time rather than in size), then wait for a new i-frame. Simple, effective, and reduces the risk of consecutive buffering, while still having fairly low impact on the stream output due to the low keyframe interval of services. By the way, audio will not (and should not ever) be dropped, lest you end up with syncing issues (among other nasty things) specific to server implementation.
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if (!disconnected(stream)) {
added_packet = (packet->type == OBS_ENCODER_VIDEO)
? add_video_packet(stream, &new_packet)
: add_packet(stream, &new_packet);
}
RTMP output: Implement frame drop code A little bit of history about frame dropping: I did a large number of experiments with frame dropping in old versions of OBS1, and it's not an easy thing to deal with. I tried just about everything from standard i-frame delay, to large buffers, to dumping packets, to super-unnecessarily-complex things that just ended up causing more problems than they was worth. When I did my experiments, I found that the most ideal frame drop system (in terms of reducing the amount of total data that needed to be dropped) was in the 0.4xx days where I had a 3 second frame-drop buffer where I could calculate the actual buffer size in bytes, and then intellgently choose packets in that buffer to trim it down to a specific size while minimizing the number of p-frames and i-frames dropped, and preventing the actual impact of dropped frames on the stream. The downside of it was that it required too much extra latency, and far too many people complained about it, so it was removed in favor of the current system. The current system I just refer to just as 'packet dumping', which when combined with low keyframe intervals (like most services use these days), is the next-best method from my experience. Just dump the buffer when you reach a threshold of buffering (which I prefer to measure with time rather than in size), then wait for a new i-frame. Simple, effective, and reduces the risk of consecutive buffering, while still having fairly low impact on the stream output due to the low keyframe interval of services. By the way, audio will not (and should not ever) be dropped, lest you end up with syncing issues (among other nasty things) specific to server implementation.
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pthread_mutex_unlock(&stream->packets_mutex);
RTMP output: Implement frame drop code A little bit of history about frame dropping: I did a large number of experiments with frame dropping in old versions of OBS1, and it's not an easy thing to deal with. I tried just about everything from standard i-frame delay, to large buffers, to dumping packets, to super-unnecessarily-complex things that just ended up causing more problems than they was worth. When I did my experiments, I found that the most ideal frame drop system (in terms of reducing the amount of total data that needed to be dropped) was in the 0.4xx days where I had a 3 second frame-drop buffer where I could calculate the actual buffer size in bytes, and then intellgently choose packets in that buffer to trim it down to a specific size while minimizing the number of p-frames and i-frames dropped, and preventing the actual impact of dropped frames on the stream. The downside of it was that it required too much extra latency, and far too many people complained about it, so it was removed in favor of the current system. The current system I just refer to just as 'packet dumping', which when combined with low keyframe intervals (like most services use these days), is the next-best method from my experience. Just dump the buffer when you reach a threshold of buffering (which I prefer to measure with time rather than in size), then wait for a new i-frame. Simple, effective, and reduces the risk of consecutive buffering, while still having fairly low impact on the stream output due to the low keyframe interval of services. By the way, audio will not (and should not ever) be dropped, lest you end up with syncing issues (among other nasty things) specific to server implementation.
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if (added_packet)
os_sem_post(stream->send_sem);
else
obs_encoder_packet_release(&new_packet);
RTMP output: Implement frame drop code A little bit of history about frame dropping: I did a large number of experiments with frame dropping in old versions of OBS1, and it's not an easy thing to deal with. I tried just about everything from standard i-frame delay, to large buffers, to dumping packets, to super-unnecessarily-complex things that just ended up causing more problems than they was worth. When I did my experiments, I found that the most ideal frame drop system (in terms of reducing the amount of total data that needed to be dropped) was in the 0.4xx days where I had a 3 second frame-drop buffer where I could calculate the actual buffer size in bytes, and then intellgently choose packets in that buffer to trim it down to a specific size while minimizing the number of p-frames and i-frames dropped, and preventing the actual impact of dropped frames on the stream. The downside of it was that it required too much extra latency, and far too many people complained about it, so it was removed in favor of the current system. The current system I just refer to just as 'packet dumping', which when combined with low keyframe intervals (like most services use these days), is the next-best method from my experience. Just dump the buffer when you reach a threshold of buffering (which I prefer to measure with time rather than in size), then wait for a new i-frame. Simple, effective, and reduces the risk of consecutive buffering, while still having fairly low impact on the stream output due to the low keyframe interval of services. By the way, audio will not (and should not ever) be dropped, lest you end up with syncing issues (among other nasty things) specific to server implementation.
2014-04-12 04:34:15 -07:00
}
static void rtmp_stream_defaults(obs_data_t *defaults)
RTMP output: Implement frame drop code A little bit of history about frame dropping: I did a large number of experiments with frame dropping in old versions of OBS1, and it's not an easy thing to deal with. I tried just about everything from standard i-frame delay, to large buffers, to dumping packets, to super-unnecessarily-complex things that just ended up causing more problems than they was worth. When I did my experiments, I found that the most ideal frame drop system (in terms of reducing the amount of total data that needed to be dropped) was in the 0.4xx days where I had a 3 second frame-drop buffer where I could calculate the actual buffer size in bytes, and then intellgently choose packets in that buffer to trim it down to a specific size while minimizing the number of p-frames and i-frames dropped, and preventing the actual impact of dropped frames on the stream. The downside of it was that it required too much extra latency, and far too many people complained about it, so it was removed in favor of the current system. The current system I just refer to just as 'packet dumping', which when combined with low keyframe intervals (like most services use these days), is the next-best method from my experience. Just dump the buffer when you reach a threshold of buffering (which I prefer to measure with time rather than in size), then wait for a new i-frame. Simple, effective, and reduces the risk of consecutive buffering, while still having fairly low impact on the stream output due to the low keyframe interval of services. By the way, audio will not (and should not ever) be dropped, lest you end up with syncing issues (among other nasty things) specific to server implementation.
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{
obs_data_set_default_int(defaults, OPT_DROP_THRESHOLD, 700);
obs_data_set_default_int(defaults, OPT_PFRAME_DROP_THRESHOLD, 900);
obs_data_set_default_int(defaults, OPT_MAX_SHUTDOWN_TIME_SEC, 30);
obs_data_set_default_string(defaults, OPT_BIND_IP, "default");
obs_data_set_default_bool(defaults, OPT_NEWSOCKETLOOP_ENABLED, false);
obs_data_set_default_bool(defaults, OPT_LOWLATENCY_ENABLED, false);
}
static obs_properties_t *rtmp_stream_properties(void *unused)
{
UNUSED_PARAMETER(unused);
obs_properties_t *props = obs_properties_create();
struct netif_saddr_data addrs = {0};
obs_property_t *p;
obs_properties_add_int(props, OPT_DROP_THRESHOLD,
2014-07-09 22:12:57 -07:00
obs_module_text("RTMPStream.DropThreshold"), 200,
10000, 100);
p = obs_properties_add_list(props, OPT_BIND_IP,
obs_module_text("RTMPStream.BindIP"),
OBS_COMBO_TYPE_LIST,
OBS_COMBO_FORMAT_STRING);
obs_property_list_add_string(p, obs_module_text("Default"), "default");
netif_get_addrs(&addrs);
for (size_t i = 0; i < addrs.addrs.num; i++) {
struct netif_saddr_item item = addrs.addrs.array[i];
obs_property_list_add_string(p, item.name, item.addr);
}
netif_saddr_data_free(&addrs);
obs_properties_add_bool(props, OPT_NEWSOCKETLOOP_ENABLED,
obs_module_text("RTMPStream.NewSocketLoop"));
obs_properties_add_bool(props, OPT_LOWLATENCY_ENABLED,
obs_module_text("RTMPStream.LowLatencyMode"));
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;
}
static float rtmp_stream_congestion(void *data)
{
struct rtmp_stream *stream = data;
if (stream->new_socket_loop)
return (float)stream->write_buf_len /
(float)stream->write_buf_size;
else
return stream->min_priority > 0 ? 1.0f : stream->congestion;
}
static int rtmp_stream_connect_time(void *data)
{
struct rtmp_stream *stream = data;
return stream->rtmp.connect_time_ms;
}
struct obs_output_info rtmp_output_info = {
.id = "rtmp_output",
.flags = OBS_OUTPUT_AV | OBS_OUTPUT_ENCODED | OBS_OUTPUT_SERVICE |
OBS_OUTPUT_MULTI_TRACK,
.encoded_video_codecs = "h264",
.encoded_audio_codecs = "aac",
.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_congestion = rtmp_stream_congestion,
.get_connect_time_ms = rtmp_stream_connect_time,
.get_dropped_frames = rtmp_stream_dropped_frames,
};