react-native-audio-api 0.7.2 → 0.8.0-nightly-daaceff-20250902

package/common/cpp/audioapi/core/sources/StreamerNode.h

@@ -0,0 +1,123 @@
+/*
+ * This file links to the FFmpeg library, which is licensed under the
+ * GNU Lesser General Public License (LGPL) version 2.1 or later.
+ *
+ * Our own code in this file is licensed under the MIT License, but note
+ * that linking with FFmpeg means you must comply with the terms of the
+ * LGPL for FFmpeg itself.
+ */
+
+#pragma once
+
+#include <audioapi/core/sources/AudioScheduledSourceNode.h>
+
+#ifndef AUDIO_API_TEST_SUITE
+extern "C" {
+  #include <libavformat/avformat.h>
+  #include <libavcodec/avcodec.h>
+  #include <libavutil/samplefmt.h>
+  #include <libavutil/channel_layout.h>
+  #include <libavutil/opt.h>
+  #include <libswresample/swresample.h>
+}
+#endif
+
+#include <cmath>
+#include <memory>
+#include <string>
+#include <atomic>
+
+namespace audioapi {
+
+class AudioBus;
+
+class StreamerNode : public AudioScheduledSourceNode {
+ public:
+  explicit StreamerNode(BaseAudioContext *context);
+  ~StreamerNode() override;
+
+  /**
+   * @brief Initialize all necessary ffmpeg components for streaming audio
+  */
+  bool initialize(const std::string& inputUrl);
+  void stop(double when) override;
+
+ private:
+  static constexpr int SIZE = 4'000'000; // 4MB
+
+ public:
+  static constexpr int getEstimatedSize() {
+    return StreamerNode::SIZE;
+  } // in bytes
+
+ protected:
+  void processNode(const std::shared_ptr<AudioBus>& processingBus, int framesToProcess) override;
+
+ private:
+  #ifndef AUDIO_API_TEST_SUITE
+  AVFormatContext* fmtCtx_;
+  AVCodecContext* codecCtx_;
+  const AVCodec* decoder_;
+  AVCodecParameters* codecpar_;
+  AVPacket* pkt_;
+  AVFrame* frame_; // Frame that is currently being processed
+  AVFrame* pendingFrame_; // Frame that is saved if bufferedBus is full
+  std::shared_ptr<AudioBus> bufferedBus_; // audio bus for buffering hls frames
+  size_t bufferedBusIndex_; // index in the buffered bus where we write the next frame
+  size_t maxBufferSize_; // maximum size of the buffered bus
+  int audio_stream_index_; // index of the audio stream channel in the input
+  SwrContext* swrCtx_;
+  uint8_t** resampledData_; // weird ffmpeg way of using raw byte pointers for resampled data
+  int maxResampledSamples_;
+  std::mutex mutex_;
+  std::thread streamingThread_;
+  std::atomic<bool> streamFlag; // Flag to control the streaming thread
+  static constexpr float BUFFER_LENGTH_SECONDS = 5.0f; // Length of the buffer in seconds
+  static constexpr int INITIAL_MAX_RESAMPLED_SAMPLES = 8192; // Initial size for resampled data
+
+  /**
+   * @brief Setting up the resampler
+   * @return true if successful, false otherwise
+   */
+  bool setupResampler();
+
+  /**
+   * @brief Resample the audio frame, change its sample format and channel layout
+   * @param frame The AVFrame to resample
+   * @return true if successful, false otherwise
+   */
+  bool processFrameWithResampler(AVFrame* frame);
+
+  /**
+   * @brief Thread function to continuously read and process audio frames
+   * @details This function runs in a separate thread to avoid blocking the main audio processing thread
+   * @note It will read frames from the input stream, resample them, and store them in the buffered bus
+   * @note The thread will stop when streamFlag is set to false
+   */
+  void streamAudio();
+
+  /** @brief Clean up resources */
+  void cleanup();
+
+  /**
+   * @brief Open the input stream
+   * @param input_url The URL of the input stream
+   * @return true if successful, false otherwise
+   * @note This function initializes the FFmpeg libraries and opens the input stream
+   */
+  bool openInput(const std::string& inputUrl);
+
+  /**
+   * @brief Find the audio stream channel in the input
+   * @return true if audio stream was found, false otherwise
+   */
+  bool findAudioStream();
+
+  /**
+   * @brief Set up the decoder for the audio stream
+   * @return true if successful, false otherwise
+   */
+  bool setupDecoder();
+  #endif // AUDIO_API_TEST_SUITE
+};
+} // namespace audioapi

package/common/cpp/audioapi/external/ffmpeg_include/libavcodec/ac3_parser.h

@@ -0,0 +1,36 @@
+/*
+ * AC-3 parser prototypes
+ * Copyright (c) 2003 Fabrice Bellard
+ * Copyright (c) 2003 Michael Niedermayer
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg 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
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_AC3_PARSER_H
+#define AVCODEC_AC3_PARSER_H
+
+#include <stddef.h>
+#include <stdint.h>
+
+/**
+ * Extract the bitstream ID and the frame size from AC-3 data.
+ */
+int av_ac3_parse_header(const uint8_t *buf, size_t size,
+                        uint8_t *bitstream_id, uint16_t *frame_size);
+
+
+#endif /* AVCODEC_AC3_PARSER_H */

package/common/cpp/audioapi/external/ffmpeg_include/libavcodec/adts_parser.h

@@ -0,0 +1,37 @@
+/*
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg 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
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_ADTS_PARSER_H
+#define AVCODEC_ADTS_PARSER_H
+
+#include <stddef.h>
+#include <stdint.h>
+
+#define AV_AAC_ADTS_HEADER_SIZE 7
+
+/**
+ * Extract the number of samples and frames from AAC data.
+ * @param[in]  buf     pointer to AAC data buffer
+ * @param[out] samples Pointer to where number of samples is written
+ * @param[out] frames  Pointer to where number of frames is written
+ * @return Returns 0 on success, error code on failure.
+ */
+int av_adts_header_parse(const uint8_t *buf, uint32_t *samples,
+                         uint8_t *frames);
+
+#endif /* AVCODEC_ADTS_PARSER_H */