Add ExternalAudioSource for pushing PCM frames from the application

api/media_stream_interface.h

@@ -270,6 +270,11 @@ class RTC_EXPORT AudioSourceInterface : public MediaSourceInterface {
   // audio network adaptation on the source is the wrong layer of abstraction).
   virtual const AudioOptions options() const;
   virtual void SetOptions(const AudioOptions & /* options */) {}
+
+  // Returns true if this source delivers audio externally (via AddSink),
+  // bypassing the ADM/AudioState audio distribution path.
+  // When true, AudioSendStream should not register with AudioState.
+  virtual bool is_external_source() const { return false; }
 };
 
 // Interface of the audio processor used by the audio track to collect

audio/audio_send_stream.cc

@@ -373,8 +373,13 @@ void AudioSendStream::Start() {
   }
   channel_send_->StartSend();
   sending_ = true;
-  audio_state()->AddSendingStream(this, encoder_sample_rate_hz_,
-                                  encoder_num_channels_);
+  // Only register with AudioState if not using an external source.
+  // External sources deliver audio directly via AddSink, so AudioState
+  // must not also push device-captured audio into this stream.
+  if (!config_.external_source) {
+    audio_state()->AddSendingStream(this, encoder_sample_rate_hz_,
+                                    encoder_num_channels_);
+  }
 }
 
 void AudioSendStream::Stop() {
@@ -386,7 +391,10 @@ void AudioSendStream::Stop() {
   RemoveBitrateObserver();
   channel_send_->StopSend();
   sending_ = false;
-  audio_state()->RemoveSendingStream(this);
+  // Only unregister if we registered (when not using an external source).
+  if (!config_.external_source) {
+    audio_state()->RemoveSendingStream(this);
+  }
 }
 
 void AudioSendStream::SendAudioData(std::unique_ptr<AudioFrame> audio_frame) {
@@ -577,7 +585,7 @@ void AudioSendStream::StoreEncoderProperties(int sample_rate_hz,
                                              size_t num_channels) {
   encoder_sample_rate_hz_ = sample_rate_hz;
   encoder_num_channels_ = num_channels;
-  if (sending_) {
+  if (sending_ && !config_.external_source) {
     // Update AudioState's information about the stream.
     audio_state()->AddSendingStream(this, sample_rate_hz, num_channels);
   }

call/audio_send_stream.cc

@@ -38,6 +38,7 @@ std::string AudioSendStream::Config::ToString() const {
   ss << ", has audio_network_adaptor_config: "
      << (audio_network_adaptor_config ? "true" : "false");
   ss << ", has_dscp: " << (has_dscp ? "true" : "false");
+  ss << ", external_source: " << (external_source ? "true" : "false");
   ss << ", send_codec_spec: "
      << (send_codec_spec ? send_codec_spec->ToString() : "<unset>");
   ss << "}";

call/audio_send_stream.h

@@ -178,6 +178,12 @@ class AudioSendStream : public AudioSender {
     // An optional frame transformer used by insertable streams to transform
     // encoded frames.
     scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer;
+
+    // When true, this stream uses an external audio source (not ADM).
+    // AudioState will NOT send device-captured audio to this stream.
+    // Audio is delivered directly via the source's AddSink mechanism.
+    // This prevents mixing of device audio with externally-sourced audio.
+    bool external_source = false;
   };
 
   virtual ~AudioSendStream() = default;

media/base/audio_source.h

@@ -49,6 +49,10 @@ class AudioSource {
   // to the source at a time.
   virtual void SetSink(Sink* sink) = 0;
 
+  // Returns true if this source delivers audio externally (bypassing ADM).
+  // When true, AudioSendStream should not register with AudioState.
+  virtual bool is_external_source() const { return false; }
+
  protected:
   virtual ~AudioSource() {}
 };

media/engine/webrtc_voice_engine.cc

@@ -938,6 +938,19 @@ class WebRtcVoiceSendChannel::WebRtcAudioSendStream : public AudioSource::Sink {
   void SetSource(AudioSource* source) {
     RTC_DCHECK_RUN_ON(&worker_thread_checker_);
     RTC_DCHECK(source);
+    // If the source delivers audio externally (via its own AddSink path),
+    // mark the config so AudioState doesn't also push device audio into this
+    // stream. Evaluated before the early return below because on track
+    // replacement the adapter object stays the same while the underlying
+    // source type may change. The stream is stopped across the flip so that
+    // its AudioState registration is always added and removed under the same
+    // flag value, then restarted under the new one.
+    if (source->is_external_source() != config_.external_source) {
+      stream_->Stop();
+      config_.external_source = source->is_external_source();
+      ReconfigureAudioSendStream(nullptr);
+      UpdateSendState();
+    }
     if (source_) {
       RTC_DCHECK(source_ == source);
       return;

pc/BUILD.gn

@@ -1262,6 +1262,29 @@ rtc_library("local_audio_source") {
     "../api:scoped_refptr",
   ]
 }
+rtc_library("external_audio_source") {
+  visibility = [ "*" ]
+  sources = [
+    "external_audio_source.cc",
+    "external_audio_source.h",
+  ]
+  public_deps = [ ":local_audio_source" ]
+  deps = [
+    "../api:audio_options_api",
+    "../api:make_ref_counted",
+    "../api:media_stream_interface",
+    "../api:scoped_refptr",
+    "../api/task_queue",
+    "../api/units:time_delta",
+    "../rtc_base:checks",
+    "../rtc_base:logging",
+    "../rtc_base:macromagic",
+    "../rtc_base/synchronization:mutex",
+    "../rtc_base/task_utils:repeating_task",
+    "../system_wrappers",
+    "//third_party/abseil-cpp/absl/functional:any_invocable",
+  ]
+}
 rtc_library("peer_connection") {
   visibility = [ ":*" ]
   sources = [
@@ -2538,6 +2561,7 @@ if (rtc_include_tests && !build_with_chromium) {
       "data_channel_integrationtest.cc",
       "data_channel_unittest.cc",
       "dtmf_sender_unittest.cc",
+      "external_audio_source_unittest.cc",
       "ice_server_parsing_unittest.cc",
       "jitter_buffer_delay_unittest.cc",
       "jsep_session_description_unittest.cc",
@@ -2595,6 +2619,7 @@ if (rtc_include_tests && !build_with_chromium) {
       ":dtls_transport",
       ":dtmf_sender",
       ":enable_fake_media",
+      ":external_audio_source",
       ":fake_codec_lookup_helper",
       ":ice_server_parsing",
       ":integration_test_helpers",
@@ -2774,6 +2799,7 @@ if (rtc_include_tests && !build_with_chromium) {
       "../rtc_base:network",
       "../rtc_base:network_constants",
       "../rtc_base:null_socket_server",
+      "../rtc_base:platform_thread",
       "../rtc_base:random",
       "../rtc_base:refcount",
       "../rtc_base:rtc_base_tests_utils",
@@ -2803,6 +2829,7 @@ if (rtc_include_tests && !build_with_chromium) {
       "../test:test_support",
       "../test:wait_until",
       "../test/pc/sctp:fake_sctp_transport",
+      "../test/time_controller",
       "scenario_tests:pc_scenario_tests",
       "//third_party/abseil-cpp/absl/algorithm:container",
       "//third_party/abseil-cpp/absl/base:nullability",