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Merge pull request #37 from OPENSPHERE-Inc/multitrackaudio
0.9.14
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| Original file line number | Diff line number | Diff line change |
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@@ -38,7 +38,7 @@ | |
| }, | ||
| "name": "osi-branch-output", | ||
| "displayName": "Branch Output Plugin", | ||
| "version": "0.9.13", | ||
| "version": "0.9.14", | ||
| "author": "OPENSPHERE Inc.", | ||
| "website": "https://opensphere.co.jp/", | ||
| "email": "[email protected]", | ||
|
|
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| Original file line number | Diff line number | Diff line change |
|---|---|---|
| @@ -0,0 +1,255 @@ | ||
| /* | ||
| Branch Output Plugin | ||
| Copyright (C) 2024 OPENSPHERE Inc. [email protected] | ||
| 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 <https://www.gnu.org/licenses/> | ||
| */ | ||
|
|
||
| #include <obs-module.h> | ||
|
|
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| #include "audio-capture.hpp" | ||
| #include "../plugin-support.h" | ||
|
|
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| #define MAX_AUDIO_BUFFER_FRAMES 131071 | ||
|
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| //--- AudioCapture class ---// | ||
|
|
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| AudioCapture::AudioCapture( | ||
| const char *_name, uint32_t _samplesPerSec, speaker_layout _speakers, audio_input_callback_t _captureCallback, | ||
| QObject *parent | ||
| ) | ||
| : QObject(parent), | ||
| name(_name), | ||
| samplesPerSec(_samplesPerSec), | ||
| speakers(_speakers), | ||
| audio(nullptr), | ||
| audioBuffer({0}), | ||
| audioBufferFrames(0), | ||
| audioConvBuffer(nullptr), | ||
| audioConvBufferSize(0) | ||
| { | ||
| audio_output_info aoi = {0}; | ||
| aoi.name = _name; | ||
| aoi.samples_per_sec = _samplesPerSec; | ||
| aoi.speakers = _speakers; | ||
| aoi.format = AUDIO_FORMAT_FLOAT_PLANAR; | ||
| aoi.input_param = this; | ||
| aoi.input_callback = _captureCallback; | ||
|
|
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| if (audio_output_open(&audio, &aoi) < 0) { | ||
| audio = nullptr; | ||
| return; | ||
| } | ||
|
|
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| active = true; | ||
| } | ||
|
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| AudioCapture::~AudioCapture() | ||
| { | ||
| active = false; | ||
|
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| if (audio) { | ||
| audio_output_close(audio); | ||
| } | ||
|
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| deque_free(&audioBuffer); | ||
| bfree(audioConvBuffer); | ||
| } | ||
|
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| uint64_t AudioCapture::popAudio(uint64_t startTsIn, uint32_t mixers, audio_output_data *audioData) | ||
| { | ||
| if (!active) { | ||
| return startTsIn; | ||
| } | ||
|
|
||
| QMutexLocker locker(&audioBufferMutex); | ||
| { | ||
| if (audioBufferFrames < AUDIO_OUTPUT_FRAMES) { | ||
| // Wait until enough frames are receved. | ||
| // DO NOT stall audio output pipeline | ||
| return startTsIn; | ||
| } | ||
|
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| size_t maxFrames = AUDIO_OUTPUT_FRAMES; | ||
|
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| while (maxFrames > 0 && audioBufferFrames) { | ||
| // Peek header of first chunk | ||
| deque_peek_front(&audioBuffer, audioConvBuffer, sizeof(AudioBufferHeader)); | ||
| auto header = (AudioBufferHeader *)audioConvBuffer; | ||
| auto dataSize = sizeof(AudioBufferHeader) + header->speakers * header->frames * 4; | ||
|
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| // Read chunk data | ||
| deque_peek_front(&audioBuffer, audioConvBuffer, dataSize); | ||
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| auto chunkFrames = header->frames - header->offset; | ||
| auto frames = (chunkFrames > maxFrames) ? maxFrames : chunkFrames; | ||
| auto outOffset = AUDIO_OUTPUT_FRAMES - maxFrames; | ||
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| for (auto tr = 0; tr < MAX_AUDIO_MIXES; tr++) { | ||
| if ((mixers & (1 << tr)) == 0) { | ||
| continue; | ||
| } | ||
| for (auto ch = 0; ch < header->speakers; ch++) { | ||
| auto out = audioData[tr].data[ch] + outOffset; | ||
| if (!header->data_idx[ch]) { | ||
| continue; | ||
| } | ||
| auto in = (float *)(audioConvBuffer + header->data_idx[ch]) + header->offset; | ||
|
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| for (size_t i = 0; i < frames; i++) { | ||
| *out += *(in++); | ||
| if (*out > 1.0f) { | ||
| *out = 1.0f; | ||
| } else if (*out < -1.0f) { | ||
| *out = -1.0f; | ||
| } | ||
| out++; | ||
| } | ||
| } | ||
| } | ||
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| if (frames == chunkFrames) { | ||
| // Remove fulfilled chunk from buffer | ||
| deque_pop_front(&audioBuffer, NULL, dataSize); | ||
| } else { | ||
| // Chunk frames are remaining -> modify chunk header | ||
| header->offset += frames; | ||
| deque_place(&audioBuffer, 0, header, sizeof(AudioBufferHeader)); | ||
| } | ||
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| maxFrames -= frames; | ||
|
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| // Decrement buffer usage | ||
| audioBufferFrames -= frames; | ||
| } | ||
| } | ||
| locker.unlock(); | ||
|
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| return startTsIn; | ||
| } | ||
|
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| void AudioCapture::pushAudio(const audio_data *audioData) | ||
| { | ||
| if (!active) { | ||
| return; | ||
| } | ||
|
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| QMutexLocker locker(&audioBufferMutex); | ||
| { | ||
| // Push audio data to buffer | ||
| if (audioBufferFrames + audioData->frames > MAX_AUDIO_BUFFER_FRAMES) { | ||
| obs_log(LOG_WARNING, "%s: The audio buffer is full", qUtf8Printable(name)); | ||
| deque_free(&audioBuffer); | ||
| deque_init(&audioBuffer); | ||
| audioBufferFrames = 0; | ||
| } | ||
|
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| // Compute header | ||
| AudioBufferHeader header = {0}; | ||
| header.frames = audioData->frames; | ||
| header.timestamp = audioData->timestamp; | ||
| header.samples_per_sec = samplesPerSec; | ||
| header.speakers = speakers; | ||
| header.format = AUDIO_FORMAT_FLOAT_PLANAR; | ||
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| for (auto i = 0, channels = 0; i < header.speakers; i++) { | ||
| if (!audioData->data[i]) { | ||
| continue; | ||
| } | ||
| header.data_idx[i] = sizeof(AudioBufferHeader) + channels * audioData->frames * 4; | ||
| channels++; | ||
| } | ||
|
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| // Push audio data to buffer | ||
| deque_push_back(&audioBuffer, &header, sizeof(AudioBufferHeader)); | ||
| for (auto i = 0; i < header.speakers; i++) { | ||
| if (!audioData->data[i]) { | ||
| continue; | ||
| } | ||
| deque_push_back(&audioBuffer, audioData->data[i], audioData->frames * 4); | ||
| } | ||
|
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| auto dataSize = sizeof(AudioBufferHeader) + header.speakers * header.frames * 4; | ||
| if (dataSize > audioConvBufferSize) { | ||
| obs_log( | ||
| LOG_DEBUG, "%s: Expand audioConvBuffer from %zu to %zu bytes", qUtf8Printable(name), | ||
| audioConvBufferSize, dataSize | ||
| ); | ||
| audioConvBuffer = (uint8_t *)brealloc(audioConvBuffer, dataSize); | ||
| audioConvBufferSize = dataSize; | ||
| } | ||
|
|
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| audioBufferFrames += audioData->frames; | ||
| } | ||
| locker.unlock(); | ||
| } | ||
|
|
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| // Convert obs_audio_data to audio_data | ||
| void AudioCapture::pushAudio(const obs_audio_data *audioData) | ||
| { | ||
| audio_data ad = {0}; | ||
| ad.frames = audioData->frames; | ||
| ad.timestamp = audioData->timestamp; | ||
| memcpy(ad.data, audioData->data, sizeof(audio_data::data)); | ||
|
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| pushAudio(&ad); | ||
| } | ||
|
|
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| // Callback from audio_output_open | ||
| bool AudioCapture::audioCapture( | ||
| void *param, uint64_t start_ts_in, uint64_t, uint64_t *out_ts, uint32_t mixers, audio_output_data *mixes | ||
| ) | ||
| { | ||
| auto *audioSource = static_cast<AudioCapture *>(param); | ||
| *out_ts = audioSource->popAudio(start_ts_in, mixers, mixes); | ||
| return true; | ||
| } | ||
|
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| // Callback from audio_output_open | ||
| bool AudioCapture::silenceCapture(void *, uint64_t startTsIn, uint64_t, uint64_t *outTs, uint32_t, audio_output_data *) | ||
| { | ||
| *outTs = startTsIn; | ||
| return true; | ||
| } | ||
|
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| //--- SourceAudioCapture class ---// | ||
|
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| SourceAudioCapture::SourceAudioCapture( | ||
| obs_source_t *source, uint32_t _samplesPerSec, speaker_layout _speakers, QObject *parent | ||
| ) | ||
| : AudioCapture(obs_source_get_name(source), _samplesPerSec, _speakers, AudioCapture::audioCapture, parent), | ||
| weakSource(obs_source_get_weak_source(source)) | ||
| { | ||
| obs_source_add_audio_capture_callback(source, sourceAudioCallback, this); | ||
| obs_log(LOG_DEBUG, "%s: Source audio capture created.", obs_source_get_name(source)); | ||
| } | ||
|
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| SourceAudioCapture::~SourceAudioCapture() | ||
| { | ||
| OBSSourceAutoRelease source = obs_weak_source_get_source(weakSource); | ||
| obs_source_remove_audio_capture_callback(source, sourceAudioCallback, this); | ||
|
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| obs_log(LOG_DEBUG, "%s: Source audio capture destroyed.", obs_source_get_name(source)); | ||
| } | ||
|
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| // Callback from obs_source_add_audio_capture_callback | ||
| void SourceAudioCapture::sourceAudioCallback(void *param, obs_source_t *, const audio_data *audioData, bool muted) | ||
| { | ||
| if (muted) { | ||
| return; | ||
| } | ||
|
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| auto sourceCapture = (SourceAudioCapture *)param; | ||
| sourceCapture->pushAudio(audioData); | ||
| } |
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