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Implement a few more audio options/functions

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Implement a few audio options in to the user interface as well as a few
inline audio functions in audio-io.h.

Make it so ffmpeg plugin automatically converts to the desired format.

Use regular interleaved float internally for audio instead of planar
float.
This commit is contained in:
jp9000
2014-02-23 16:27:19 -07:00
parent 0ff0d32731
commit c232ebde15
10 changed files with 311 additions and 54 deletions
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50
plugins/obs-ffmpeg/obs-ffmpeg-output.c
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@@ -21,6 +21,9 @@
#include <libavformat/avformat.h>
#include <libswscale/swscale.h>
/* NOTE: much of this stuff is test stuff that was more or less copied from
* the muxing.c ffmpeg example */
struct ffmpeg_data {
AVStream *video;
AVStream *audio;
@@ -34,6 +37,9 @@ struct ffmpeg_data {
int frame_size;
int total_frames;
enum audio_format audio_format;
size_t audio_planes;
size_t audio_size;
struct circlebuf excess_frames[MAX_AV_PLANES];
uint8_t *samples[MAX_AV_PLANES];
AVFrame *aframe;
@@ -52,12 +58,6 @@ struct ffmpeg_output {
/* ------------------------------------------------------------------------- */
/* TODO: remove these later */
#define SPS_TODO 44100
/* NOTE: much of this stuff is test stuff that was more or less copied from
* the muxing.c ffmpeg example */
static inline enum AVPixelFormat obs_to_ffmpeg_video_format(
enum video_format format)
{
@@ -76,6 +76,24 @@ static inline enum AVPixelFormat obs_to_ffmpeg_video_format(
return AV_PIX_FMT_NONE;
}
static inline enum audio_format convert_ffmpeg_sample_format(
enum AVSampleFormat format)
{
switch ((uint32_t)format) {
case AV_SAMPLE_FMT_U8: return AUDIO_FORMAT_U8BIT;
case AV_SAMPLE_FMT_S16: return AUDIO_FORMAT_16BIT;
case AV_SAMPLE_FMT_S32: return AUDIO_FORMAT_32BIT;
case AV_SAMPLE_FMT_FLT: return AUDIO_FORMAT_FLOAT;
case AV_SAMPLE_FMT_U8P: return AUDIO_FORMAT_U8BIT_PLANAR;
case AV_SAMPLE_FMT_S16P: return AUDIO_FORMAT_16BIT_PLANAR;
case AV_SAMPLE_FMT_S32P: return AUDIO_FORMAT_32BIT_PLANAR;
case AV_SAMPLE_FMT_FLTP: return AUDIO_FORMAT_FLOAT_PLANAR;
}
/* shouldn't get here */
return AUDIO_FORMAT_16BIT;
}
static bool new_stream(struct ffmpeg_data *data, AVStream **stream,
AVCodec **codec, enum AVCodecID id)
{
@@ -237,6 +255,10 @@ static bool create_audio_stream(struct ffmpeg_data *data)
context->sample_fmt = data->acodec->sample_fmts ?
data->acodec->sample_fmts[0] : AV_SAMPLE_FMT_FLTP;
data->audio_format = convert_ffmpeg_sample_format(context->sample_fmt);
data->audio_planes = get_audio_planes(data->audio_format, aoi.speakers);
data->audio_size = get_audio_size(data->audio_format, aoi.speakers, 1);
if (data->output->oformat->flags & AVFMT_GLOBALHEADER)
context->flags |= CODEC_FLAG_GLOBAL_HEADER;
@@ -532,21 +554,19 @@ static void receive_audio(void *param, const struct audio_data *frame)
struct ffmpeg_data *data = &output->ff_data;
AVCodecContext *context = data->audio->codec;
size_t planes = audio_output_planes(obs_audio());
size_t block_size = audio_output_blocksize(obs_audio());
size_t frame_size_bytes = (size_t)data->frame_size * block_size;
size_t frame_size_bytes = (size_t)data->frame_size * data->audio_size;
for (size_t i = 0; i < planes; i++)
for (size_t i = 0; i < data->audio_planes; i++)
circlebuf_push_back(&data->excess_frames[i], frame->data[0],
frame->frames * block_size);
frame->frames * data->audio_size);
while (data->excess_frames[0].size >= frame_size_bytes) {
for (size_t i = 0; i < planes; i++)
for (size_t i = 0; i < data->audio_planes; i++)
circlebuf_pop_front(&data->excess_frames[i],
data->samples[i], frame_size_bytes);
encode_audio(data, context, block_size);
encode_audio(data, context, data->audio_size);
}
}
@@ -575,9 +595,7 @@ static bool ffmpeg_output_start(void *data)
return false;
struct audio_convert_info aci = {
.samples_per_sec = SPS_TODO,
.format = AUDIO_FORMAT_FLOAT_PLANAR,
.speakers = SPEAKERS_STEREO
.format = output->ff_data.audio_format
};
struct video_scale_info vsi = {