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Implement RTMP module (still needs drop code)

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- 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.
This commit is contained in:
jp9000
2014-04-07 22:00:10 -07:00
parent 7eec72245d
commit 92522d1886
33 changed files with 517 additions and 234 deletions
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57
plugins/obs-ffmpeg/obs-ffmpeg-aac.c
View File

@@ -17,6 +17,7 @@
#include <util/base.h>
#include <util/circlebuf.h>
#include <util/darray.h>
#include <obs.h>
#include <libavformat/avformat.h>
@@ -30,11 +31,12 @@ struct aac_encoder {
AVCodec *aac;
AVCodecContext *context;
struct circlebuf buffers[MAX_AV_PLANES];
uint8_t *samples[MAX_AV_PLANES];
AVFrame *aframe;
int total_samples;
DARRAY(uint8_t) packet_buffer;
size_t audio_planes;
size_t audio_size;
@@ -63,15 +65,14 @@ static void aac_destroy(void *data)
{
struct aac_encoder *enc = data;
for (size_t i = 0; i < MAX_AV_PLANES; i++)
circlebuf_free(&enc->buffers[i]);
if (enc->samples[0])
av_freep(&enc->samples[0]);
if (enc->context)
avcodec_close(enc->context);
if (enc->aframe)
av_frame_free(&enc->aframe);
da_free(enc->packet_buffer);
bfree(enc);
}
@@ -109,6 +110,18 @@ static bool initialize_codec(struct aac_encoder *enc)
return true;
}
static void init_sizes(struct aac_encoder *enc, audio_t audio)
{
const struct audio_output_info *aoi;
enum audio_format format;
aoi = audio_output_getinfo(audio);
format = convert_ffmpeg_sample_format(enc->context->sample_fmt);
enc->audio_planes = get_audio_planes(format, aoi->speakers);
enc->audio_size = get_audio_size(format, aoi->speakers, 1);
}
static void *aac_create(obs_data_t settings, obs_encoder_t encoder)
{
struct aac_encoder *enc;
@@ -120,6 +133,8 @@ static void *aac_create(obs_data_t settings, obs_encoder_t encoder)
return NULL;
}
avcodec_register_all();
enc = bzalloc(sizeof(struct aac_encoder));
enc->encoder = encoder;
enc->aac = avcodec_find_encoder(AV_CODEC_ID_AAC);
@@ -128,22 +143,19 @@ static void *aac_create(obs_data_t settings, obs_encoder_t encoder)
goto fail;
}
avcodec_register(enc->aac);
enc->context = avcodec_alloc_context3(enc->aac);
if (!enc->context) {
aac_warn("aac_create", "Failed to create codec context");
goto fail;
}
enc->context->bit_rate = bitrate;
enc->context->bit_rate = bitrate * 1000;
enc->context->channels = (int)audio_output_channels(audio);
enc->context->sample_rate = audio_output_samplerate(audio);
enc->context->sample_fmt = enc->aac->sample_fmts ?
enc->aac->sample_fmts[0] : AV_SAMPLE_FMT_FLTP;
enc->audio_planes = audio_output_planes(audio);
enc->audio_size = audio_output_blocksize(audio);
init_sizes(enc, audio);
/* enable experimental FFmpeg encoder if the only one available */
enc->context->strict_std_compliance = -2;
@@ -192,9 +204,12 @@ static bool do_aac_encode(struct aac_encoder *enc,
if (!got_packet)
return true;
da_resize(enc->packet_buffer, 0);
da_push_back_array(enc->packet_buffer, avpacket.data, avpacket.size);
packet->pts = rescale_ts(avpacket.pts, enc->context, time_base);
packet->dts = rescale_ts(avpacket.dts, enc->context, time_base);
packet->data = bmemdup(avpacket.data, avpacket.size);
packet->data = enc->packet_buffer.array;
packet->size = avpacket.size;
packet->type = OBS_ENCODER_AUDIO;
packet->timebase_num = 1;
@@ -207,19 +222,9 @@ static bool aac_encode(void *data, struct encoder_frame *frame,
struct encoder_packet *packet, bool *received_packet)
{
struct aac_encoder *enc = data;
size_t size = frame->frames * enc->audio_size;
for (size_t i = 0; i < enc->audio_planes; i++)
circlebuf_push_back(&enc->buffers[i], frame->data[i], size);
if (enc->buffers[0].size < (size_t)enc->frame_size_bytes) {
*received_packet = false;
return true;
}
for (size_t i = 0; i < enc->audio_planes; i++)
circlebuf_pop_front(&enc->buffers[i], enc->samples[i],
enc->frame_size_bytes);
memcpy(enc->samples[i], frame->data[i], enc->frame_size_bytes);
return do_aac_encode(enc, packet, received_packet);
}
@@ -234,8 +239,7 @@ static obs_properties_t aac_properties(const char *locale)
obs_properties_t props = obs_properties_create(locale);
/* TODO: locale */
obs_properties_add_int(props, "bitrate", "Bitrate",
32, 320, 32);
obs_properties_add_int(props, "bitrate", "Bitrate", 32, 320, 32);
return props;
}
@@ -257,6 +261,12 @@ static bool aac_audio_info(void *data, struct audio_convert_info *info)
return true;
}
static size_t aac_frame_size(void *data)
{
struct aac_encoder *enc =data;
return enc->frame_size;
}
struct obs_encoder_info aac_encoder_info = {
.id = "ffmpeg_aac",
.type = OBS_ENCODER_AUDIO,
@@ -265,6 +275,7 @@ struct obs_encoder_info aac_encoder_info = {
.create = aac_create,
.destroy = aac_destroy,
.encode = aac_encode,
.frame_size = aac_frame_size,
.defaults = aac_defaults,
.properties = aac_properties,
.extra_data = aac_extra_data,

2
plugins/obs-ffmpeg/obs-ffmpeg-output.c
View File

@@ -760,7 +760,7 @@ static void *start_thread(void *data)
struct ffmpeg_output *output = data;
if (!try_connect(output))
obs_output_signal_start_fail(output->output,
obs_output_signal_stop(output->output,
OBS_OUTPUT_CONNECT_FAILED);
output->connecting = false;

4
plugins/obs-ffmpeg/obs-ffmpeg.c
View File

@@ -2,11 +2,13 @@
OBS_DECLARE_MODULE()
extern struct obs_output_info ffmpeg_output;
extern struct obs_output_info ffmpeg_output;
extern struct obs_encoder_info aac_encoder_info;
bool obs_module_load(uint32_t obs_version)
{
obs_register_output(&ffmpeg_output);
obs_register_encoder(&aac_encoder_info);
UNUSED_PARAMETER(obs_version);
return true;