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[WIP] iOS 向けに AudioTrackSink を追加する #128

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[WIP] iOS 向けに AudioTrackSink を追加する #128

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b2dff31
Xcode 26.0 になったので修正する
zztkm Nov 18, 2025
fc792ae
iOS 向けに AudioTrackSink を追加する
zztkm Nov 18, 2025
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2 changes: 1 addition & 1 deletion .github/workflows/build.yml
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Original file line number Diff line number Diff line change
Expand Up @@ -74,7 +74,7 @@ jobs:
runs-on: macos-26
steps:
- uses: actions/checkout@v5
- name: Select Xcode 16.1
- name: Select Xcode 26.0
run: sudo xcode-select --switch /Applications/Xcode_26.0.app/Contents/Developer
- name: Setup Git User
run: |
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276 changes: 276 additions & 0 deletions patches/ios_audio_track_sink.patch
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Original file line number Diff line number Diff line change
@@ -0,0 +1,276 @@
diff --git a/sdk/BUILD.gn b/sdk/BUILD.gn
index 7b8a02462c..3ddacc9364 100644
--- a/sdk/BUILD.gn
+++ b/sdk/BUILD.gn
@@ -997,6 +997,9 @@ if (is_ios || is_mac) {
"objc/api/peerconnection/RTCAudioTrack+Private.h",
"objc/api/peerconnection/RTCAudioTrack.h",
"objc/api/peerconnection/RTCAudioTrack.mm",
+ "objc/api/peerconnection/RTCAudioTrack+Sink.h",
+ "objc/api/peerconnection/RTCAudioTrack+Sink.mm",
+ "objc/api/peerconnection/RTCAudioTrackSink.h",
"objc/api/peerconnection/RTCCertificate.h",
"objc/api/peerconnection/RTCCertificate.mm",
"objc/api/peerconnection/RTCConfiguration+Native.h",
diff --git a/sdk/objc/api/peerconnection/RTCAudioTrack+Sink.h b/sdk/objc/api/peerconnection/RTCAudioTrack+Sink.h
new file mode 100644
index 0000000000..902eb54341
--- /dev/null
+++ b/sdk/objc/api/peerconnection/RTCAudioTrack+Sink.h
@@ -0,0 +1,29 @@
+#import "RTCAudioTrack.h"
+#import "RTCAudioTrackSink.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/**
+ * RTCAudioTrack の Sink 管理カテゴリ
+ */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE(RTCAudioTrack) (Sink)
+
+/**
+ * AudioTrackSink を関連付ける
+ *
+ * @param sink 関連付ける AudioTrackSink
+ */
+- (void)addSink:(id<RTC_OBJC_TYPE(RTCAudioTrackSink)>)sink;
+
+/**
+ * AudioTrackSink の関連付けを解除する
+ *
+ * @param sink 関連付けを解除する AudioTrackSink
+ */
+- (void)removeSink:(id<RTC_OBJC_TYPE(RTCAudioTrackSink)>)sink;
+
+@end
+
+NS_ASSUME_NONNULL_END
+
diff --git a/sdk/objc/api/peerconnection/RTCAudioTrack+Sink.mm b/sdk/objc/api/peerconnection/RTCAudioTrack+Sink.mm
new file mode 100644
index 0000000000..002cfd78b0
--- /dev/null
+++ b/sdk/objc/api/peerconnection/RTCAudioTrack+Sink.mm
@@ -0,0 +1,171 @@
+#import "RTCAudioTrack+Sink.h"
+#import "RTCAudioTrack+Private.h"
+
+#include <memory>
+#include <map>
+#include <mutex>
+#include "api/media_stream_interface.h"
+#include "rtc_base/thread.h"
+
+namespace webrtc {
+namespace objc {
+
+/**
+ * C++ wrapper for Objective-C audio sink
+ * スレッドセーフな実装でObjective-CのAudioSinkをブリッジ
+ */
+class AudioSinkAdapter : public webrtc::AudioTrackSinkInterface {
+ public:
+ AudioSinkAdapter(id<RTC_OBJC_TYPE(RTCAudioTrackSink)> sink,
+ RTC_OBJC_TYPE(RTCAudioTrack)* track)
+ : sink_(sink), track_(track) {}
+
+ void OnData(const void* audio_data,
+ int bits_per_sample,
+ int sample_rate,
+ size_t number_of_channels,
+ size_t number_of_frames) override {
+ // 注意: このメソッドはWebRTCのオーディオスレッドから呼ばれる
+ // 高頻度（10-20ms間隔）で呼ばれるため、処理は最小限に
+
+ // OnData の呼び出しごとに生成される ObjC オブジェクトを即時解放し、
+ // オーディオスレッド上の長寿命ループでのメモリ膨張を防ぐ。
+ // 特に __weak から strong を取り出す際には ARC が retain→autorelease を
+ // 内部的に挿入するため、明示的な pool でその一時オブジェクトを毎回 drain する。
+ @autoreleasepool {
+ // weak参照のチェック（スレッドセーフ）
+ id<RTC_OBJC_TYPE(RTCAudioTrackSink)> strongSink = sink_;
+ RTC_OBJC_TYPE(RTCAudioTrack)* strongTrack = track_;
+
+ if (!strongSink || !strongTrack) {
+ return;
+ }
+
+ // データをNSDataに変換（コピーが発生）
+ size_t bytes_per_sample = bits_per_sample / 8;
+ size_t data_size = number_of_frames * number_of_channels * bytes_per_sample;
+ // Data/NSData は毎回新規インスタンスを生成してコピーし、バッファの再利用は行わない。
+ NSData* audioData = [[NSData alloc] initWithBytes:audio_data length:data_size];
+
+ // Objective-Cデリゲートの呼び出し
+ // 実装側で適切なキューへのディスパッチを行うことを期待
+ [strongSink didReceiveData:audioData
+ bitsPerSample:bits_per_sample
+ sampleRate:sample_rate
+ numberOfChannels:number_of_channels
+ numberOfFrames:number_of_frames];
+ }
+ }
+
+ int NumPreferredChannels() const override {
+ // weak参照のチェック
+ id<RTC_OBJC_TYPE(RTCAudioTrackSink)> strongSink = sink_;
+ if (strongSink && [strongSink respondsToSelector:@selector(preferredNumberOfChannels)]) {
+ return static_cast<int>([strongSink preferredNumberOfChannels]);
+ }
+ return -1;
+ }
+
+ private:
+ // weak参照でメモリリークを防ぐ
+ __weak id<RTC_OBJC_TYPE(RTCAudioTrackSink)> sink_;
+ __weak RTC_OBJC_TYPE(RTCAudioTrack)* track_;
+};
+
+} // namespace objc
+} // namespace webrtc
+
+// スレッドセーフなSinkアダプター管理
+namespace {
+
+class SinkAdapterManager {
+ public:
+ static SinkAdapterManager& GetInstance() {
+ static SinkAdapterManager instance;
+ return instance;
+ }
+
+ void AddAdapter(id<RTC_OBJC_TYPE(RTCAudioTrackSink)> sink,
+ std::unique_ptr<webrtc::objc::AudioSinkAdapter> adapter) {
+ std::lock_guard<std::mutex> lock(mutex_);
+ adapters_[sink] = std::move(adapter);
+ }
+
+ webrtc::objc::AudioSinkAdapter* GetAdapter(id<RTC_OBJC_TYPE(RTCAudioTrackSink)> sink) {
+ std::lock_guard<std::mutex> lock(mutex_);
+ auto it = adapters_.find(sink);
+ if (it != adapters_.end()) {
+ return it->second.get();
+ }
+ return nullptr;
+ }
+
+ std::unique_ptr<webrtc::objc::AudioSinkAdapter> RemoveAdapter(
+ id<RTC_OBJC_TYPE(RTCAudioTrackSink)> sink) {
+ std::lock_guard<std::mutex> lock(mutex_);
+ auto it = adapters_.find(sink);
+ if (it != adapters_.end()) {
+ auto adapter = std::move(it->second);
+ adapters_.erase(it);
+ return adapter;
+ }
+ return nullptr;
+ }
+
+ bool HasAdapter(id<RTC_OBJC_TYPE(RTCAudioTrackSink)> sink) {
+ std::lock_guard<std::mutex> lock(mutex_);
+ return adapters_.find(sink) != adapters_.end();
+ }
+
+ private:
+ std::mutex mutex_;
+ std::map<id<RTC_OBJC_TYPE(RTCAudioTrackSink)>,
+ std::unique_ptr<webrtc::objc::AudioSinkAdapter>> adapters_;
+};
+
+} // namespace
+
+@implementation RTC_OBJC_TYPE(RTCAudioTrack) (Sink)
+
+- (void)addSink:(id<RTC_OBJC_TYPE(RTCAudioTrackSink)>)sink {
+ if (!sink) {
+ return;
+ }
+
+ auto& manager = SinkAdapterManager::GetInstance();
+
+ // 既に存在するかチェック
+ if (manager.HasAdapter(sink)) {
+ return;
+ }
+
+ // 新しいアダプターを作成
+ auto adapter = std::make_unique<webrtc::objc::AudioSinkAdapter>(sink, self);
+ webrtc::objc::AudioSinkAdapter* adapterPtr = adapter.get();
+
+ // マネージャーに登録
+ manager.AddAdapter(sink, std::move(adapter));
+
+ // ネイティブトラックに追加（WebRTC側でスレッドセーフ）
+ self.nativeAudioTrack->AddSink(adapterPtr);
+}
+
+- (void)removeSink:(id<RTC_OBJC_TYPE(RTCAudioTrackSink)>)sink {
+ if (!sink) {
+ return;
+ }
+
+ auto& manager = SinkAdapterManager::GetInstance();
+
+ // アダプターを取得
+ auto* adapterPtr = manager.GetAdapter(sink);
+ if (adapterPtr) {
+ // ネイティブトラックから削除（WebRTC側でスレッドセーフ）
+ self.nativeAudioTrack->RemoveSink(adapterPtr);
+
+ // マネージャーから削除
+ manager.RemoveAdapter(sink);
+ }
+}
+
+@end
diff --git a/sdk/objc/api/peerconnection/RTCAudioTrackSink.h b/sdk/objc/api/peerconnection/RTCAudioTrackSink.h
new file mode 100644
index 0000000000..1999a5169f
--- /dev/null
+++ b/sdk/objc/api/peerconnection/RTCAudioTrackSink.h
@@ -0,0 +1,44 @@
+#import <Foundation/Foundation.h>
+#import "sdk/objc/base/RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@class RTC_OBJC_TYPE(RTCAudioTrack);
+
+/**
+ * RTCAudioTrackSink プロトコル
+ * 音声データを受信するためのインターフェース
+ */
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE(RTCAudioTrackSink) <NSObject>
+
+/**
+ * 音声データ受信コールバック
+ *
+ * @param audioData PCM 形式の音声データ。
+ * @param bitsPerSample 1 サンプルあたりのビット数。
+ * libwebrtc では PCM 形式の音声データは 16 bit 固定のため、常に 16 が渡されます。
+ * @param sampleRate サンプルレート (単位: Hz)
+ * @param numberOfChannels 音声データのチャンネル数。
+ * モノラルなら 1、ステレオなら 2 が渡されます。
+ * @param numberOfFrames audioData に含まれるフレーム数。
+ */
+- (void) didReceiveData:(NSData *)audioData
+ bitsPerSample:(NSInteger)bitsPerSample
+ sampleRate:(NSInteger)sampleRate
+ numberOfChannels:(NSInteger)numberOfChannels
+ numberOfFrames:(NSInteger)numberOfFrames;
+
+@optional
+/**
+ * onData で受け取る音声データのチャンネル数を指定するためのメソッドです。
+ * `-1` を指定した場合は音声データ規定のチャンネル数になります。
+ *
+ * @return チャンネル数（-1の場合は指定なし）
+ */
+- (NSInteger)preferredNumberOfChannels;
+
+@end
+
+NS_ASSUME_NONNULL_END
+
1 change: 1 addition & 0 deletions run.py
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Original file line number Diff line number Diff line change
Expand Up @@ -259,6 +259,7 @@ def get_depot_tools(source_dir, fetch=False):
"remove_crel.patch",
"revert_siso.patch",
"ios_revive_copy_framework_header.patch",
"ios_audio_track_sink.patch",
],
"android": [
"add_deps.patch",
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