react-native-audio-api 0.8.3-nightly-ea268f4-20251006 → 0.9.0-nightly-96a5bcd-20251007

package/common/cpp/audioapi/libs/ffmpeg/FFmpegDecoding.h

@@ -1,11 +1,11 @@
 /*
- * This file dynamically links to the FFmpeg library, which is licensed under the
- * GNU Lesser General Public License (LGPL) version 2.1 or later.
+ * This file dynamically 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 and dynamic linking
- * allows you to use this code without your entire project being subject to the
- * terms of the LGPL. However, note that if you link statically to FFmpeg, you must
- * comply with the terms of the LGPL for FFmpeg itself.
+ * Our own code in this file is licensed under the MIT License and dynamic
+ * linking allows you to use this code without your entire project being subject
+ * to the terms of the LGPL. However, note that if you link statically to
+ * FFmpeg, you must comply with the terms of the LGPL for FFmpeg itself.
  */
 
 #include <audioapi/utils/AudioBus.h>
@@ -14,24 +14,62 @@
 #include <vector>
 
 extern "C" {
-    #include <libavcodec/avcodec.h>
-    #include <libavformat/avformat.h>
-    #include <libavutil/opt.h>
-    #include <libswresample/swresample.h>
+#include <libavcodec/avcodec.h>
+#include <libavformat/avformat.h>
+#include <libavutil/opt.h>
+#include <libswresample/swresample.h>
 }
+class AudioBuffer;
 
-namespace audioapi::ffmpegdecoding {
+namespace audioapi::ffmpegdecoder {
 // Custom IO context for reading from memory
 struct MemoryIOContext {
-    const uint8_t *data;
-    size_t size;
-    size_t pos;
-  };
+  const uint8_t *data;
+  size_t size;
+  size_t pos;
+};
+
+struct AudioStreamContext {
+  AVFormatContext *fmt_ctx = nullptr;
+  AVCodecContext *codec_ctx = nullptr;
+  int audio_stream_index = -1;
+};
 
 int read_packet(void *opaque, uint8_t *buf, int buf_size);
 int64_t seek_packet(void *opaque, int64_t offset, int whence);
-std::vector<int16_t> readAllPcmFrames(AVFormatContext *fmt_ctx, AVCodecContext *codec_ctx, int out_sample_rate, int audio_stream_index, int channels, size_t &framesRead);
-std::vector<int16_t> decodeWithMemoryBlock(const void *data, size_t size, const int channel_count, int sample_rate);
-std::vector<int16_t> decodeWithFilePath(const std::string &path, const int channel_count, int sample_rate);
+inline int findAudioStreamIndex(AVFormatContext *fmt_ctx);
+std::vector<float> readAllPcmFrames(
+    AVFormatContext *fmt_ctx,
+    AVCodecContext *codec_ctx,
+    int out_sample_rate,
+    int output_channel_count,
+    int audio_stream_index,
+    size_t &framesRead);
+
+void convertFrameToBuffer(
+    SwrContext *swr,
+    AVFrame *frame,
+    int output_channel_count,
+    std::vector<float> &buffer,
+    size_t &framesRead,
+    uint8_t **&resampled_data,
+    int &max_resampled_samples);
+bool setupDecoderContext(
+    AVFormatContext *fmt_ctx,
+    int &audio_stream_index,
+    std::unique_ptr<AVCodecContext, decltype(&avcodec_free_context)>
+        &codec_ctx);
+std::shared_ptr<AudioBuffer> decodeAudioFrames(
+    AVFormatContext *fmt_ctx,
+    AVCodecContext *codec_ctx,
+    int audio_stream_index,
+    int sample_rate);
+
+std::shared_ptr<AudioBuffer>
+decodeWithMemoryBlock(const void *data, size_t size, int sample_rate);
+
+std::shared_ptr<AudioBuffer> decodeWithFilePath(
+    const std::string &path,
+    int sample_rate);
 
-} // namespace audioapi::ffmpegdecoder
+} // namespace audioapi::ffmpegdecoder

package/common/cpp/test/CMakeLists.txt

@@ -21,7 +21,7 @@ enable_testing()
 file(GLOB_RECURSE RNAUDIOAPI_SRC
   CONFIGURE_DEPENDS
   "${ROOT}/node_modules/react-native-audio-api/common/cpp/audioapi/*.cpp"
-  "${ROOT}/node_modules/react-native-audio-api/android/src/main/cpp/audioapi/android/core/AudioDecoder.cpp"
+  "${ROOT}/node_modules/react-native-audio-api/android/src/main/cpp/audioapi/android/core/utils/AudioDecoder.cpp"
 )
 
 # exclude HostObjects from tests
@@ -67,6 +67,7 @@ add_executable(
   OscillatorTest.cpp
   GainTest.cpp
   AudioParamTest.cpp
+  StereoPannerTest.cpp
 )
 
 add_compile_definitions(AUDIO_API_TEST_SUITE)

package/common/cpp/test/GainTest.cpp

@@ -28,10 +28,10 @@ class TestableGainNode : public audioapi::GainNode {
     getGainParam()->setValue(value);
   }
 
-  void processNode(
+  std::shared_ptr<audioapi::AudioBus> processNode(
       const std::shared_ptr<audioapi::AudioBus> &processingBus,
       int framesToProcess) override {
-    audioapi::GainNode::processNode(processingBus, framesToProcess);
+    return audioapi::GainNode::processNode(processingBus, framesToProcess);
   }
 };
 
@@ -52,9 +52,9 @@ TEST_F(GainTest, GainModulatesVolumeCorrectly) {
     bus->getChannel(0)->getData()[i] = i + 1;
   }
 
-  gainNode->processNode(bus, FRAMES_TO_PROCESS);
-  for (size_t i = 0; i < bus->getSize(); ++i) {
-    EXPECT_FLOAT_EQ((*bus->getChannel(0))[i], (i + 1) * GAIN_VALUE);
+  auto resultBus = gainNode->processNode(bus, FRAMES_TO_PROCESS);
+  for (size_t i = 0; i < FRAMES_TO_PROCESS; ++i) {
+    EXPECT_FLOAT_EQ((*resultBus->getChannel(0))[i], (i + 1) * GAIN_VALUE);
   }
 }
 
@@ -71,9 +71,9 @@ TEST_F(GainTest, GainModulatesVolumeCorrectlyMultiChannel) {
     bus->getChannel(1)->getData()[i] = -i - 1;
   }
 
-  gainNode->processNode(bus, FRAMES_TO_PROCESS);
-  for (size_t i = 0; i < bus->getSize(); ++i) {
-    EXPECT_FLOAT_EQ((*bus->getChannel(0))[i], (i + 1) * GAIN_VALUE);
-    EXPECT_FLOAT_EQ((*bus->getChannel(1))[i], (-i - 1) * GAIN_VALUE);
+  auto resultBus = gainNode->processNode(bus, FRAMES_TO_PROCESS);
+  for (size_t i = 0; i < FRAMES_TO_PROCESS; ++i) {
+    EXPECT_FLOAT_EQ((*resultBus->getChannel(0))[i], (i + 1) * GAIN_VALUE);
+    EXPECT_FLOAT_EQ((*resultBus->getChannel(1))[i], (-i - 1) * GAIN_VALUE);
   }
 }

package/common/cpp/test/StereoPannerTest.cpp

@@ -0,0 +1,129 @@
+#include <audioapi/core/OfflineAudioContext.h>
+#include <audioapi/core/effects/StereoPannerNode.h>
+#include <audioapi/core/utils/worklets/SafeIncludes.h>
+#include <audioapi/utils/AudioArray.h>
+#include <audioapi/utils/AudioBus.h>
+#include <gtest/gtest.h>
+#include "MockAudioEventHandlerRegistry.h"
+
+using namespace audioapi;
+
+class StereoPannerTest : public ::testing::Test {
+ protected:
+  std::shared_ptr<IAudioEventHandlerRegistry> eventRegistry;
+  std::unique_ptr<OfflineAudioContext> context;
+  static constexpr int sampleRate = 44100;
+
+  void SetUp() override {
+    eventRegistry = std::make_shared<MockAudioEventHandlerRegistry>();
+    context = std::make_unique<OfflineAudioContext>(
+        2, 5 * sampleRate, sampleRate, eventRegistry, RuntimeRegistry{});
+  }
+};
+
+class TestableStereoPannerNode : public StereoPannerNode {
+ public:
+  explicit TestableStereoPannerNode(BaseAudioContext *context)
+      : StereoPannerNode(context) {}
+
+  void setPanParam(float value) {
+    getPanParam()->setValue(value);
+  }
+
+  std::shared_ptr<AudioBus> processNode(
+      const std::shared_ptr<AudioBus> &processingBus,
+      int framesToProcess) override {
+    return StereoPannerNode::processNode(processingBus, framesToProcess);
+  }
+};
+
+TEST_F(StereoPannerTest, StereoPannerCanBeCreated) {
+  auto panner = context->createStereoPanner();
+  ASSERT_NE(panner, nullptr);
+}
+
+TEST_F(StereoPannerTest, PanModulatesInputMonoCorrectly) {
+  static constexpr float PAN_VALUE = 0.5;
+  static constexpr int FRAMES_TO_PROCESS = 4;
+  auto panNode = std::make_shared<TestableStereoPannerNode>(context.get());
+  panNode->setPanParam(PAN_VALUE);
+
+  auto bus =
+      std::make_shared<audioapi::AudioBus>(FRAMES_TO_PROCESS, 1, sampleRate);
+  for (size_t i = 0; i < bus->getSize(); ++i) {
+    (*bus->getChannelByType(AudioBus::ChannelLeft))[i] = i + 1;
+  }
+
+  auto resultBus = panNode->processNode(bus, FRAMES_TO_PROCESS);
+  // x = (0.5 + 1) / 2 = 0.75
+  // gainL = cos(x * (π / 2)) = cos(0.75 * (π / 2)) = 0.38268343236508984
+  // gainR = sin(x * (π / 2)) = sin(0.75 * (π / 2)) = 0.9238795325112867
+  for (size_t i = 0; i < FRAMES_TO_PROCESS; ++i) {
+    EXPECT_NEAR(
+        (*resultBus->getChannelByType(AudioBus::ChannelLeft))[i],
+        (i + 1) * 0.38268343236508984,
+        1e-4);
+    EXPECT_NEAR(
+        (*resultBus->getChannelByType(AudioBus::ChannelRight))[i],
+        (i + 1) * 0.9238795325112867,
+        1e-4);
+  }
+}
+
+TEST_F(StereoPannerTest, PanModulatesInputStereoCorrectlyWithNegativePan) {
+  static constexpr float PAN_VALUE = -0.5;
+  static constexpr int FRAMES_TO_PROCESS = 4;
+  auto panNode = std::make_shared<TestableStereoPannerNode>(context.get());
+  panNode->setPanParam(PAN_VALUE);
+
+  auto bus =
+      std::make_shared<audioapi::AudioBus>(FRAMES_TO_PROCESS, 2, sampleRate);
+  for (size_t i = 0; i < bus->getSize(); ++i) {
+    (*bus->getChannelByType(AudioBus::ChannelLeft))[i] = i + 1;
+    (*bus->getChannelByType(AudioBus::ChannelRight))[i] = i + 1;
+  }
+
+  auto resultBus = panNode->processNode(bus, FRAMES_TO_PROCESS);
+  // x = -0.5 + 1 = 0.5
+  // gainL = cos(x * (π / 2)) = cos(0.5 * (π / 2)) = 0.7071067811865476
+  // gainR = sin(x * (π / 2)) = sin(0.5 * (π / 2)) = 0.7071067811865476
+  for (size_t i = 0; i < FRAMES_TO_PROCESS; ++i) {
+    EXPECT_NEAR(
+        (*resultBus->getChannelByType(AudioBus::ChannelLeft))[i],
+        (i + 1) + (i + 1) * 0.7071067811865476,
+        1e-4);
+    EXPECT_NEAR(
+        (*resultBus->getChannelByType(AudioBus::ChannelRight))[i],
+        (i + 1) * 0.7071067811865476,
+        1e-4);
+  }
+}
+
+TEST_F(StereoPannerTest, PanModulatesInputStereoCorrectlyWithPositivePan) {
+  static constexpr float PAN_VALUE = 0.75;
+  static constexpr int FRAMES_TO_PROCESS = 4;
+  auto panNode = std::make_shared<TestableStereoPannerNode>(context.get());
+  panNode->setPanParam(PAN_VALUE);
+
+  auto bus =
+      std::make_shared<audioapi::AudioBus>(FRAMES_TO_PROCESS, 2, sampleRate);
+  for (size_t i = 0; i < bus->getSize(); ++i) {
+    (*bus->getChannelByType(AudioBus::ChannelLeft))[i] = i + 1;
+    (*bus->getChannelByType(AudioBus::ChannelRight))[i] = i + 1;
+  }
+
+  auto resultBus = panNode->processNode(bus, FRAMES_TO_PROCESS);
+  // x = 0.75
+  // gainL = cos(x * (π / 2)) = cos(0.75 * (π / 2)) = 0.38268343236508984
+  // gainR = sin(x * (π / 2)) = sin(0.75 * (π / 2)) = 0.9238795325112867
+  for (size_t i = 0; i < FRAMES_TO_PROCESS; ++i) {
+    EXPECT_NEAR(
+        (*resultBus->getChannelByType(AudioBus::ChannelLeft))[i],
+        (i + 1) * 0.38268343236508984,
+        1e-4);
+    EXPECT_NEAR(
+        (*resultBus->getChannelByType(AudioBus::ChannelRight))[i],
+        (i + 1) + (i + 1) * 0.9238795325112867,
+        1e-4);
+  }
+}

package/ios/audioapi/ios/core/IOSAudioRecorder.h

@@ -18,8 +18,7 @@ class IOSAudioRecorder : public AudioRecorder {
   IOSAudioRecorder(
       float sampleRate,
       int bufferLength,
-      const std::shared_ptr<AudioEventHandlerRegistry>
-          &audioEventHandlerRegistry);
+      const std::shared_ptr<AudioEventHandlerRegistry> &audioEventHandlerRegistry);
 
   ~IOSAudioRecorder() override;
 

package/ios/audioapi/ios/core/utils/AudioDecoder.mm

@@ -0,0 +1,160 @@
+#define MINIAUDIO_IMPLEMENTATION
+#import <audioapi/libs/miniaudio/miniaudio.h>
+
+#include <audioapi/libs/miniaudio/decoders/libopus/miniaudio_libopus.h>
+#include <audioapi/libs/miniaudio/decoders/libvorbis/miniaudio_libvorbis.h>
+
+#include <audioapi/core/sources/AudioBuffer.h>
+#include <audioapi/core/utils/AudioDecoder.h>
+#include <audioapi/dsp/VectorMath.h>
+#include <audioapi/libs/audio-stretch/stretch.h>
+#include <audioapi/libs/base64/base64.h>
+#include <audioapi/libs/ffmpeg/FFmpegDecoding.h>
+#include <audioapi/utils/AudioArray.h>
+#include <audioapi/utils/AudioBus.h>
+
+namespace audioapi {
+
+// Decoding audio in fixed-size chunks because total frame count can't be
+// determined in advance. Note: ma_decoder_get_length_in_pcm_frames() always
+// returns 0 for Vorbis decoders.
+std::vector<float> AudioDecoder::readAllPcmFrames(ma_decoder &decoder, int outputChannels)
+{
+  std::vector<float> buffer;
+  std::vector<float> temp(CHUNK_SIZE * outputChannels);
+  ma_uint64 outFramesRead = 0;
+
+  while (true) {
+    ma_uint64 tempFramesDecoded = 0;
+    ma_decoder_read_pcm_frames(&decoder, temp.data(), CHUNK_SIZE, &tempFramesDecoded);
+    if (tempFramesDecoded == 0) {
+      break;
+    }
+
+    buffer.insert(buffer.end(), temp.data(), temp.data() + tempFramesDecoded * outputChannels);
+    outFramesRead += tempFramesDecoded;
+  }
+
+  if (outFramesRead == 0) {
+    NSLog(@"Failed to decode");
+  }
+  return buffer;
+}
+
+std::shared_ptr<AudioBuffer> AudioDecoder::makeAudioBufferFromFloatBuffer(
+    const std::vector<float> &buffer,
+    float outputSampleRate,
+    int outputChannels)
+{
+  if (buffer.empty()) {
+    return nullptr;
+  }
+
+  auto outputFrames = buffer.size() / outputChannels;
+  auto audioBus = std::make_shared<AudioBus>(outputFrames, outputChannels, outputSampleRate);
+
+  for (int ch = 0; ch < outputChannels; ++ch) {
+    auto channelData = audioBus->getChannel(ch)->getData();
+    for (int i = 0; i < outputFrames; ++i) {
+      channelData[i] = buffer[i * outputChannels + ch];
+    }
+  }
+  return std::make_shared<AudioBuffer>(audioBus);
+}
+
+std::shared_ptr<AudioBuffer> AudioDecoder::decodeWithFilePath(const std::string &path, float sampleRate)
+{
+  if (AudioDecoder::pathHasExtension(path, {".mp4", ".m4a", ".aac"})) {
+    auto buffer = ffmpegdecoder::decodeWithFilePath(path, static_cast<int>(sampleRate));
+    if (buffer == nullptr) {
+      NSLog(@"Failed to decode with FFmpeg: %s", path.c_str());
+      return nullptr;
+    }
+    return buffer;
+  }
+  ma_decoder decoder;
+  ma_decoder_config config = ma_decoder_config_init(ma_format_f32, 0, static_cast<int>(sampleRate));
+  ma_decoding_backend_vtable *customBackends[] = {ma_decoding_backend_libvorbis, ma_decoding_backend_libopus};
+
+  config.ppCustomBackendVTables = customBackends;
+  config.customBackendCount = sizeof(customBackends) / sizeof(customBackends[0]);
+
+  if (ma_decoder_init_file(path.c_str(), &config, &decoder) != MA_SUCCESS) {
+    NSLog(@"Failed to initialize decoder for file: %s", path.c_str());
+    ma_decoder_uninit(&decoder);
+    return nullptr;
+  }
+
+  auto outputSampleRate = static_cast<float>(decoder.outputSampleRate);
+  auto outputChannels = static_cast<int>(decoder.outputChannels);
+
+  std::vector<float> buffer = readAllPcmFrames(decoder, outputChannels);
+  ma_decoder_uninit(&decoder);
+  return makeAudioBufferFromFloatBuffer(buffer, outputSampleRate, outputChannels);
+}
+
+std::shared_ptr<AudioBuffer> AudioDecoder::decodeWithMemoryBlock(const void *data, size_t size, float sampleRate)
+{
+  const AudioFormat format = AudioDecoder::detectAudioFormat(data, size);
+  if (format == AudioFormat::MP4 || format == AudioFormat::M4A || format == AudioFormat::AAC) {
+    auto buffer = ffmpegdecoder::decodeWithMemoryBlock(data, size, static_cast<int>(sampleRate));
+    if (buffer == nullptr) {
+      NSLog(@"Failed to decode with FFmpeg");
+      return nullptr;
+    }
+    return buffer;
+  }
+  ma_decoder decoder;
+  ma_decoder_config config = ma_decoder_config_init(ma_format_f32, 0, static_cast<int>(sampleRate));
+
+  ma_decoding_backend_vtable *customBackends[] = {ma_decoding_backend_libvorbis, ma_decoding_backend_libopus};
+
+  config.ppCustomBackendVTables = customBackends;
+  config.customBackendCount = sizeof(customBackends) / sizeof(customBackends[0]);
+
+  if (ma_decoder_init_memory(data, size, &config, &decoder) != MA_SUCCESS) {
+    NSLog(@"Failed to initialize decoder for memory block");
+    ma_decoder_uninit(&decoder);
+    return nullptr;
+  }
+
+  auto outputSampleRate = static_cast<float>(decoder.outputSampleRate);
+  auto outputChannels = static_cast<int>(decoder.outputChannels);
+
+  std::vector<float> buffer = readAllPcmFrames(decoder, outputChannels);
+  ma_decoder_uninit(&decoder);
+  return makeAudioBufferFromFloatBuffer(buffer, outputSampleRate, outputChannels);
+}
+
+std::shared_ptr<AudioBuffer> AudioDecoder::decodeWithPCMInBase64(
+    const std::string &data,
+    float inputSampleRate,
+    int inputChannelCount,
+    bool interleaved)
+{
+  auto decodedData = base64_decode(data, false);
+  const auto uint8Data = reinterpret_cast<uint8_t *>(decodedData.data());
+  size_t numFramesDecoded = decodedData.size() / (inputChannelCount * sizeof(int16_t));
+
+  auto audioBus = std::make_shared<AudioBus>(numFramesDecoded, inputChannelCount, inputSampleRate);
+
+  for (int ch = 0; ch < inputChannelCount; ++ch) {
+    auto channelData = audioBus->getChannel(ch)->getData();
+
+    for (size_t i = 0; i < numFramesDecoded; ++i) {
+      size_t offset;
+      if (interleaved) {
+        // Ch1, Ch2, Ch1, Ch2, ...
+        offset = (i * inputChannelCount + ch) * sizeof(int16_t);
+      } else {
+        // Ch1, Ch1, Ch1, ..., Ch2, Ch2, Ch2, ...
+        offset = (ch * numFramesDecoded + i) * sizeof(int16_t);
+      }
+
+      channelData[i] = uint8ToFloat(uint8Data[offset], uint8Data[offset + 1]);
+    }
+  }
+  return std::make_shared<AudioBuffer>(audioBus);
+}
+
+} // namespace audioapi

package/lib/commonjs/api.js

@@ -219,6 +219,18 @@ Object.defineProperty(exports, "WorkletSourceNode", {
     return _WorkletSourceNode.default;
   }
 });
+Object.defineProperty(exports, "decodeAudioData", {
+  enumerable: true,
+  get: function () {
+    return _AudioDecoder.decodeAudioData;
+  }
+});
+Object.defineProperty(exports, "decodePCMInBase64", {
+  enumerable: true,
+  get: function () {
+    return _AudioDecoder.decodePCMInBase64;
+  }
+});
 Object.defineProperty(exports, "useSystemVolume", {
   enumerable: true,
   get: function () {
@@ -250,6 +262,7 @@ var _StreamerNode = _interopRequireDefault(require("./core/StreamerNode"));
 var _ConstantSourceNode = _interopRequireDefault(require("./core/ConstantSourceNode"));
 var _system = _interopRequireDefault(require("./system"));
 var _useSystemVolume = _interopRequireDefault(require("./hooks/useSystemVolume"));
+var _AudioDecoder = require("./core/AudioDecoder");
 var _types = require("./types");
 var _errors = require("./errors");
 function _interopRequireDefault(e) { return e && e.__esModule ? e : { default: e }; }
@@ -257,7 +270,7 @@ function _interopRequireDefault(e) { return e && e.__esModule ? e : { default: e
 
 /* eslint-disable no-var */
 
-if (global.createAudioContext == null || global.createOfflineAudioContext == null || global.createAudioRecorder == null || global.AudioEventEmitter == null) {
+if (global.createAudioContext == null || global.createOfflineAudioContext == null || global.createAudioRecorder == null || global.createAudioDecoder == null || global.AudioEventEmitter == null) {
   if (!_specs.NativeAudioAPIModule) {
     throw new Error(`Failed to install react-native-audio-api: The native module could not be found.`);
   }

package/lib/commonjs/api.js.map

@@ -1 +1 @@
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package/lib/commonjs/core/AudioDecoder.js

@@ -0,0 +1,48 @@
+"use strict";
+
+Object.defineProperty(exports, "__esModule", {
+  value: true
+});
+exports.decodeAudioData = decodeAudioData;
+exports.decodePCMInBase64 = decodePCMInBase64;
+var _AudioBuffer = _interopRequireDefault(require("./AudioBuffer"));
+function _interopRequireDefault(e) { return e && e.__esModule ? e : { default: e }; }
+class AudioDecoder {
+  static instance = null;
+  constructor() {
+    this.decoder = global.createAudioDecoder();
+  }
+  static getInstance() {
+    if (!AudioDecoder.instance) {
+      AudioDecoder.instance = new AudioDecoder();
+    }
+    return AudioDecoder.instance;
+  }
+  async decodeAudioDataInstance(input, sampleRate) {
+    let buffer;
+    if (typeof input === 'string') {
+      // Remove the file:// prefix if it exists
+      if (input.startsWith('file://')) {
+        input = input.replace('file://', '');
+      }
+      buffer = await this.decoder.decodeWithFilePath(input, sampleRate ?? 0);
+    } else if (input instanceof ArrayBuffer) {
+      buffer = await this.decoder.decodeWithMemoryBlock(new Uint8Array(input), sampleRate ?? 0);
+    }
+    if (!buffer) {
+      throw new Error('Unsupported input type or failed to decode audio');
+    }
+    return new _AudioBuffer.default(buffer);
+  }
+  async decodePCMInBase64Instance(base64String, inputSampleRate, inputChannelCount, interleaved) {
+    const buffer = await this.decoder.decodeWithPCMInBase64(base64String, inputSampleRate, inputChannelCount, interleaved);
+    return new _AudioBuffer.default(buffer);
+  }
+}
+async function decodeAudioData(input, sampleRate) {
+  return AudioDecoder.getInstance().decodeAudioDataInstance(input, sampleRate);
+}
+async function decodePCMInBase64(base64String, inputSampleRate, inputChannelCount, isInterleaved = true) {
+  return AudioDecoder.getInstance().decodePCMInBase64Instance(base64String, inputSampleRate, inputChannelCount, isInterleaved);
+}
+//# sourceMappingURL=AudioDecoder.js.map

package/lib/commonjs/core/AudioDecoder.js.map

@@ -0,0 +1 @@
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package/lib/commonjs/core/BaseAudioContext.js

@@ -14,6 +14,7 @@ var _AudioBufferQueueSourceNode = _interopRequireDefault(require("./AudioBufferQ
 var _AudioBufferSourceNode = _interopRequireDefault(require("./AudioBufferSourceNode"));
 var _AudioDestinationNode = _interopRequireDefault(require("./AudioDestinationNode"));
 var _BiquadFilterNode = _interopRequireDefault(require("./BiquadFilterNode"));
+var _ConstantSourceNode = _interopRequireDefault(require("./ConstantSourceNode"));
 var _GainNode = _interopRequireDefault(require("./GainNode"));
 var _OscillatorNode = _interopRequireDefault(require("./OscillatorNode"));
 var _PeriodicWave = _interopRequireDefault(require("./PeriodicWave"));
@@ -21,7 +22,7 @@ var _RecorderAdapterNode = _interopRequireDefault(require("./RecorderAdapterNode
 var _StereoPannerNode = _interopRequireDefault(require("./StereoPannerNode"));
 var _StreamerNode = _interopRequireDefault(require("./StreamerNode"));
 var _WorkletNode = _interopRequireDefault(require("./WorkletNode"));
-var _ConstantSourceNode = _interopRequireDefault(require("./ConstantSourceNode"));
+var _AudioDecoder = require("./AudioDecoder");
 function _interopRequireDefault(e) { return e && e.__esModule ? e : { default: e }; }
 class BaseAudioContext {
   constructor(context) {
@@ -35,6 +36,15 @@ class BaseAudioContext {
   get state() {
     return this.context.state;
   }
+  async decodeAudioData(input, sampleRate) {
+    if (!(typeof input === 'string' || input instanceof ArrayBuffer)) {
+      throw new TypeError('Input must be a string or ArrayBuffer');
+    }
+    return await (0, _AudioDecoder.decodeAudioData)(input, sampleRate ?? this.sampleRate);
+  }
+  async decodePCMInBase64(base64String, inputSampleRate, inputChannelCount, isInterleaved = true) {
+    return await (0, _AudioDecoder.decodePCMInBase64)(base64String, inputSampleRate, inputChannelCount, isInterleaved);
+  }
   createWorkletNode(callback, bufferLength, inputChannelCount, workletRuntime = 'AudioRuntime') {
     if (inputChannelCount < 1 || inputChannelCount > 32) {
       throw new _errors.NotSupportedError(`The number of input channels provided (${inputChannelCount}) can not be less than 1 or greater than 32`);
@@ -132,23 +142,6 @@ class BaseAudioContext {
   createAnalyser() {
     return new _AnalyserNode.default(this, this.context.createAnalyser());
   }
-
-  /** Decodes audio data from a local file path. */
-  async decodeAudioDataSource(sourcePath) {
-    // Remove the file:// prefix if it exists
-    if (sourcePath.startsWith('file://')) {
-      sourcePath = sourcePath.replace('file://', '');
-    }
-    return new _AudioBuffer.default(await this.context.decodeAudioDataSource(sourcePath));
-  }
-
-  /** Decodes audio data from an ArrayBuffer. */
-  async decodeAudioData(data) {
-    return new _AudioBuffer.default(await this.context.decodeAudioData(new Uint8Array(data)));
-  }
-  async decodePCMInBase64Data(base64, playbackRate = 1.0) {
-    return new _AudioBuffer.default(await this.context.decodePCMAudioDataInBase64(base64, playbackRate));
-  }
 }
 exports.default = BaseAudioContext;
 //# sourceMappingURL=BaseAudioContext.js.map

package/lib/commonjs/core/BaseAudioContext.js.map

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