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}
diff --git a/api/media_stream_interface.h b/api/media_stream_interface.h
index bd4a2c0..8892ee5 100644
--- a/api/media_stream_interface.h
+++ b/api/media_stream_interface.h
@@ -216,6 +216,11 @@
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() {}
};
diff --git a/media/base/audio_source.h b/media/base/audio_source.h
index 8a87968..51fe0e1 100644
--- a/media/base/audio_source.h
+++ b/media/base/audio_source.h
@@ -36,6 +36,11 @@
// 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() {}
};
diff --git a/media/base/fake_media_engine.h b/media/base/fake_media_engine.h
index 1751dd8..42940bf 100644
--- a/media/base/fake_media_engine.h
+++ b/media/base/fake_media_engine.h
@@ -371,6 +371,7 @@
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:
diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc
index bc382d0..2ed78b4 100644
--- a/media/engine/webrtc_voice_engine.cc
+++ b/media/engine/webrtc_voice_engine.cc
@@ -1114,6 +1114,14 @@
*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 @@
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 @@
// 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 {
diff --git a/pc/rtp_sender.cc b/pc/rtp_sender.cc
index 0da6dfc..5a7e237 100644
--- a/pc/rtp_sender.cc
+++ b/pc/rtp_sender.cc
@@ -405,6 +405,7 @@
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();
}
}
diff --git a/pc/rtp_sender.h b/pc/rtp_sender.h
index c2fe91f..51ae1e9 100644
--- a/pc/rtp_sender.h
+++ b/pc/rtp_sender.h
@@ -237,12 +237,16 @@
/*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 {
