Surface the number of encoded channels

Two audio channels going into the AudioSource::Sink can either be
down-mixed to mono or encoded as stereo. This change enables WebRTC
users (such as Chromium) to query the number of audio channels actually
encoded. That information can in turn be used to tailor the audio
processing to the number of channels actually encoded.

This change fixes webrtc:8133 from a WebRTC perspective and will be
followed up with the necessary Chromium changes.

Bug: webrtc:8133
Change-Id: I8e8a08292002919784c05a5aacb21707918809c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/197426
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32836}

api/media_stream_interface.h

@@ -216,6 +216,11 @@ class AudioTrackSinkInterface {
                   number_of_frames);
   }
 
+  // Returns the number of channels encoded by the sink. This can be less than
+  // the number_of_channels if down-mixing occur. A value of -1 means an unknown
+  // number.
+  virtual int NumPreferredChannels() const { return -1; }
+
  protected:
   virtual ~AudioTrackSinkInterface() {}
 };

media/base/audio_source.h

@@ -36,6 +36,11 @@ class AudioSource {
     // Called when the AudioSource is going away.
     virtual void OnClose() = 0;
 
+    // Returns the number of channels encoded by the sink. This can be less than
+    // the number_of_channels if down-mixing occur. A value of -1 means an
+    // unknown number.
+    virtual int NumPreferredChannels() const = 0;
+
    protected:
     virtual ~Sink() {}
   };

media/base/fake_media_engine.h

@@ -371,6 +371,7 @@ class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> {
                 size_t number_of_frames,
                 absl::optional<int64_t> absolute_capture_timestamp_ms) override;
     void OnClose() override;
+    int NumPreferredChannels() const override { return -1; }
     AudioSource* source() const;
 
    private:

media/engine/webrtc_voice_engine.cc

@@ -1114,6 +1114,14 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
         *audio_codec_spec_);
 
     UpdateAllowedBitrateRange();
+
+    // Encoder will only use two channels if the stereo parameter is set.
+    const auto& it = send_codec_spec.format.parameters.find("stereo");
+    if (it != send_codec_spec.format.parameters.end() && it->second == "1") {
+      num_encoded_channels_ = 2;
+    } else {
+      num_encoded_channels_ = 1;
+    }
   }
 
   void UpdateAudioNetworkAdaptorConfig() {
@@ -1133,6 +1141,8 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
     stream_->Reconfigure(config_);
   }
 
+  int NumPreferredChannels() const override { return num_encoded_channels_; }
+
   const AdaptivePtimeConfig adaptive_ptime_config_;
   rtc::ThreadChecker worker_thread_checker_;
   rtc::RaceChecker audio_capture_race_checker_;
@@ -1154,6 +1164,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
   // TODO(webrtc:11717): Remove this once audio_network_adaptor in AudioOptions
   // has been removed.
   absl::optional<std::string> audio_network_adaptor_config_from_options_;
+  int num_encoded_channels_ = -1;
 };
 
 class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {

pc/rtp_sender.cc

@@ -405,6 +405,7 @@ void LocalAudioSinkAdapter::OnData(
   if (sink_) {
     sink_->OnData(audio_data, bits_per_sample, sample_rate, number_of_channels,
                   number_of_frames, absolute_capture_timestamp_ms);
+    num_preferred_channels_ = sink_->NumPreferredChannels();
   }
 }
 

pc/rtp_sender.h

@@ -237,12 +237,16 @@ class LocalAudioSinkAdapter : public AudioTrackSinkInterface,
            /*absolute_capture_timestamp_ms=*/absl::nullopt);
   }
 
+  // AudioSinkInterface implementation.
+  int NumPreferredChannels() const override { return num_preferred_channels_; }
+
   // cricket::AudioSource implementation.
   void SetSink(cricket::AudioSource::Sink* sink) override;
 
   cricket::AudioSource::Sink* sink_;
   // Critical section protecting |sink_|.
   Mutex lock_;
+  int num_preferred_channels_ = -1;
 };
 
 class AudioRtpSender : public DtmfProviderInterface, public RtpSenderBase {