obs-studio/libobs/media-io/audio-io.h

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2013-09-30 19:37:13 -07:00
/******************************************************************************
Copyright (C) 2013 by Hugh Bailey <obs.jim@gmail.com>
This program is free software: you can redistribute it and/or modify
it under the terms of the GNU General Public License as published by
the Free Software Foundation, either version 2 of the License, or
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(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/>.
******************************************************************************/
#pragma once
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#include "media-io-defs.h"
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#include "../util/c99defs.h"
#ifdef __cplusplus
extern "C" {
#endif
/*
* Base audio output component. Use this to create an audio output track
* for the media.
*/
struct audio_output;
struct audio_line;
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typedef struct audio_output *audio_t;
typedef struct audio_line *audio_line_t;
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enum audio_format {
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AUDIO_FORMAT_UNKNOWN,
AUDIO_FORMAT_U8BIT,
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AUDIO_FORMAT_16BIT,
AUDIO_FORMAT_32BIT,
AUDIO_FORMAT_FLOAT,
AUDIO_FORMAT_U8BIT_PLANAR,
AUDIO_FORMAT_16BIT_PLANAR,
AUDIO_FORMAT_32BIT_PLANAR,
AUDIO_FORMAT_FLOAT_PLANAR,
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};
enum speaker_layout {
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SPEAKERS_UNKNOWN,
SPEAKERS_MONO,
SPEAKERS_STEREO,
SPEAKERS_2POINT1,
SPEAKERS_QUAD,
SPEAKERS_4POINT1,
SPEAKERS_5POINT1,
SPEAKERS_5POINT1_SURROUND,
SPEAKERS_7POINT1,
SPEAKERS_7POINT1_SURROUND,
SPEAKERS_SURROUND,
};
struct audio_data {
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
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uint8_t *data[MAX_AV_PLANES];
uint32_t frames;
uint64_t timestamp;
float volume;
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};
struct audio_output_info {
const char *name;
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uint32_t samples_per_sec;
enum audio_format format;
enum speaker_layout speakers;
uint64_t buffer_ms;
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};
struct audio_convert_info {
uint32_t samples_per_sec;
enum audio_format format;
enum speaker_layout speakers;
};
static inline uint32_t get_audio_channels(enum speaker_layout speakers)
{
switch (speakers) {
case SPEAKERS_MONO: return 1;
case SPEAKERS_STEREO: return 2;
case SPEAKERS_2POINT1: return 3;
case SPEAKERS_SURROUND:
case SPEAKERS_QUAD: return 4;
case SPEAKERS_4POINT1: return 5;
case SPEAKERS_5POINT1:
case SPEAKERS_5POINT1_SURROUND: return 6;
case SPEAKERS_7POINT1: return 8;
case SPEAKERS_7POINT1_SURROUND: return 8;
case SPEAKERS_UNKNOWN: return 0;
}
return 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.
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static inline size_t get_audio_bytes_per_channel(enum audio_format format)
{
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.
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switch (format) {
case AUDIO_FORMAT_U8BIT:
case AUDIO_FORMAT_U8BIT_PLANAR:
return 1;
case AUDIO_FORMAT_16BIT:
case AUDIO_FORMAT_16BIT_PLANAR:
return 2;
case AUDIO_FORMAT_FLOAT:
case AUDIO_FORMAT_FLOAT_PLANAR:
case AUDIO_FORMAT_32BIT:
case AUDIO_FORMAT_32BIT_PLANAR:
return 4;
case AUDIO_FORMAT_UNKNOWN:
return 0;
}
return 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.
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static inline bool is_audio_planar(enum audio_format format)
{
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.
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switch (format) {
case AUDIO_FORMAT_U8BIT:
case AUDIO_FORMAT_16BIT:
case AUDIO_FORMAT_32BIT:
case AUDIO_FORMAT_FLOAT:
return false;
case AUDIO_FORMAT_U8BIT_PLANAR:
case AUDIO_FORMAT_FLOAT_PLANAR:
case AUDIO_FORMAT_16BIT_PLANAR:
case AUDIO_FORMAT_32BIT_PLANAR:
return true;
case AUDIO_FORMAT_UNKNOWN:
return false;
}
return false;
}
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.
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static inline size_t get_audio_planes(enum audio_format format,
enum speaker_layout speakers)
{
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.
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return (is_audio_planar(format) ? get_audio_channels(speakers) : 1);
}
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.
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static inline size_t get_audio_size(enum audio_format format,
enum speaker_layout speakers, uint32_t frames)
{
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 planar = is_audio_planar(format);
return (planar ? 1 : get_audio_channels(speakers)) *
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.
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get_audio_bytes_per_channel(format) *
frames;
}
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#define AUDIO_OUTPUT_SUCCESS 0
#define AUDIO_OUTPUT_INVALIDPARAM -1
#define AUDIO_OUTPUT_FAIL -2
EXPORT int audio_output_open(audio_t *audio, struct audio_output_info *info);
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EXPORT void audio_output_close(audio_t audio);
EXPORT bool audio_output_connect(audio_t video,
const struct audio_convert_info *conversion,
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
void (*callback)(void *param, struct audio_data *data),
void *param);
EXPORT void audio_output_disconnect(audio_t video,
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
void (*callback)(void *param, struct audio_data *data),
void *param);
EXPORT bool audio_output_active(audio_t audio);
EXPORT size_t audio_output_blocksize(audio_t audio);
EXPORT size_t audio_output_planes(audio_t audio);
EXPORT size_t audio_output_channels(audio_t audio);
EXPORT uint32_t audio_output_samplerate(audio_t audio);
EXPORT const struct audio_output_info *audio_output_getinfo(audio_t audio);
EXPORT audio_line_t audio_output_createline(audio_t audio, const char *name);
EXPORT void audio_line_destroy(audio_line_t line);
EXPORT void audio_line_output(audio_line_t line, const struct audio_data *data);
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#ifdef __cplusplus
}
#endif