Add Sender and Receiver interfaces for MediaTransport audio
Implement in LoopbackMediaTransport.
Bug: webrtc:9719
Change-Id: I429ac3f78d99b8ea4f9ac85b9a3600b215b61a55
Reviewed-on: https://webrtc-review.googlesource.com/c/121957
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26731}diff --git a/api/media_transport_interface.cc b/api/media_transport_interface.cc
index de04e19..0dfec76 100644
--- a/api/media_transport_interface.cc
+++ b/api/media_transport_interface.cc
@@ -54,6 +54,18 @@
MediaTransportInterface::MediaTransportInterface() = default;
MediaTransportInterface::~MediaTransportInterface() = default;
+std::unique_ptr<MediaTransportAudioSender>
+MediaTransportInterface::CreateAudioSender(uint64_t channel_id) {
+ return nullptr;
+}
+
+std::unique_ptr<MediaTransportAudioReceiver>
+MediaTransportInterface::CreateAudioReceiver(
+ uint64_t channel_id,
+ MediaTransportAudioSinkInterface* sink) {
+ return nullptr;
+}
+
void MediaTransportInterface::SetKeyFrameRequestCallback(
MediaTransportKeyFrameRequestCallback* callback) {}
diff --git a/api/media_transport_interface.h b/api/media_transport_interface.h
index 2f5431f..e753ddc 100644
--- a/api/media_transport_interface.h
+++ b/api/media_transport_interface.h
@@ -187,10 +187,28 @@
MediaTransportInterface();
virtual ~MediaTransportInterface();
+ // Creates an object representing the send end-point of a audio stream using
+ // this transport.
+ // TODO(bugs.webrtc.org/9719): Make pure virtual after downstream
+ // implementations are updated.
+ virtual std::unique_ptr<MediaTransportAudioSender> CreateAudioSender(
+ uint64_t channel_id);
+
+ // Creates an object representing the receive end-point of a audio stream
+ // using this transport.
+ // TODO(bugs.webrtc.org/9719): Make pure virtual after downstream
+ // implementations are updated.
+ virtual std::unique_ptr<MediaTransportAudioReceiver> CreateAudioReceiver(
+ uint64_t channel_id,
+ // TODO(nisse): Add Rtt observer, or route that via Call to the receive
+ // stream instead?
+ MediaTransportAudioSinkInterface* sink);
+
// Start asynchronous send of audio frame. The status returned by this method
// only pertains to the synchronous operations (e.g.
// serialization/packetization), not to the asynchronous operation.
-
+ // TODO(nisse): Deprecated, should be deleted when implementations are updated
+ // to use CreateAudioSender.
virtual RTCError SendAudioFrame(uint64_t channel_id,
MediaTransportEncodedAudioFrame frame) = 0;
diff --git a/api/test/loopback_media_transport.cc b/api/test/loopback_media_transport.cc
index c466170..fde732e 100644
--- a/api/test/loopback_media_transport.cc
+++ b/api/test/loopback_media_transport.cc
@@ -109,6 +109,7 @@
MediaTransportPair::LoopbackMediaTransport::~LoopbackMediaTransport() {
rtc::CritScope lock(&sink_lock_);
+ RTC_CHECK(audio_sinks_.empty());
RTC_CHECK(audio_sink_ == nullptr);
RTC_CHECK(video_sink_ == nullptr);
RTC_CHECK(data_sink_ == nullptr);
@@ -116,6 +117,58 @@
RTC_CHECK(rtt_observers_.empty());
}
+class MediaTransportPair::LoopbackMediaTransport::AudioSender
+ : public MediaTransportAudioSender {
+ public:
+ AudioSender(LoopbackMediaTransport* transport, uint64_t channel_id)
+ : transport_(transport), channel_id_(channel_id) {}
+ void SendAudioFrame(MediaTransportEncodedAudioFrame frame) override {
+ transport_->SendAudioFrame(channel_id_, std::move(frame));
+ }
+
+ private:
+ LoopbackMediaTransport* transport_;
+ uint64_t channel_id_;
+};
+
+class MediaTransportPair::LoopbackMediaTransport::AudioReceiver
+ : public MediaTransportAudioReceiver {
+ public:
+ AudioReceiver(LoopbackMediaTransport* transport, uint64_t channel_id)
+ : transport_(transport), channel_id_(channel_id) {}
+ ~AudioReceiver() override {
+ transport_->UnregisterAudioReceiver(channel_id_);
+ }
+
+ private:
+ LoopbackMediaTransport* transport_;
+ uint64_t channel_id_;
+};
+
+std::unique_ptr<MediaTransportAudioSender>
+MediaTransportPair::LoopbackMediaTransport::CreateAudioSender(
+ uint64_t channel_id) {
+ return absl::make_unique<AudioSender>(this, channel_id);
+}
+
+std::unique_ptr<MediaTransportAudioReceiver>
+MediaTransportPair::LoopbackMediaTransport::CreateAudioReceiver(
+ uint64_t channel_id,
+ MediaTransportAudioSinkInterface* sink) {
+ rtc::CritScope cs(&sink_lock_);
+ auto res = audio_sinks_.emplace(channel_id, sink);
+ RTC_DCHECK(res.second);
+ return absl::make_unique<AudioReceiver>(this, channel_id);
+}
+
+void MediaTransportPair::LoopbackMediaTransport::UnregisterAudioReceiver(
+ uint64_t channel_id) {
+ rtc::CritScope cs(&sink_lock_);
+ auto it = audio_sinks_.find(channel_id);
+ RTC_DCHECK(it != audio_sinks_.end());
+ audio_sinks_.erase(it);
+}
+
RTCError MediaTransportPair::LoopbackMediaTransport::SendAudioFrame(
uint64_t channel_id,
MediaTransportEncodedAudioFrame frame) {
@@ -317,7 +370,10 @@
MediaTransportEncodedAudioFrame frame) {
{
rtc::CritScope lock(&sink_lock_);
- if (audio_sink_) {
+ const auto it = audio_sinks_.find(channel_id);
+ if (it != audio_sinks_.end()) {
+ it->second->OnData(frame);
+ } else if (audio_sink_) {
audio_sink_->OnData(channel_id, frame);
}
}
diff --git a/api/test/loopback_media_transport.h b/api/test/loopback_media_transport.h
index bcfdb63..d2c503b 100644
--- a/api/test/loopback_media_transport.h
+++ b/api/test/loopback_media_transport.h
@@ -11,6 +11,7 @@
#ifndef API_TEST_LOOPBACK_MEDIA_TRANSPORT_H_
#define API_TEST_LOOPBACK_MEDIA_TRANSPORT_H_
+#include <map>
#include <memory>
#include <utility>
#include <vector>
@@ -85,6 +86,13 @@
~LoopbackMediaTransport() override;
+ std::unique_ptr<MediaTransportAudioSender> CreateAudioSender(
+ uint64_t channel_id) override;
+
+ std::unique_ptr<MediaTransportAudioReceiver> CreateAudioReceiver(
+ uint64_t channel_id,
+ MediaTransportAudioSinkInterface* sink) override;
+
RTCError SendAudioFrame(uint64_t channel_id,
MediaTransportEncodedAudioFrame frame) override;
@@ -131,6 +139,9 @@
const MediaTransportAllocatedBitrateLimits& limits) override;
private:
+ class AudioReceiver;
+ class AudioSender;
+
void OnData(uint64_t channel_id, MediaTransportEncodedAudioFrame frame);
void OnData(uint64_t channel_id, MediaTransportEncodedVideoFrame frame);
@@ -144,11 +155,17 @@
void OnRemoteCloseChannel(int channel_id);
void OnStateChanged() RTC_RUN_ON(thread_);
+ void UnregisterAudioReceiver(uint64_t channel_id);
rtc::Thread* const thread_;
rtc::CriticalSection sink_lock_;
rtc::CriticalSection stats_lock_;
+ std::map<uint64_t, MediaTransportAudioSinkInterface*> audio_sinks_
+ RTC_GUARDED_BY(sink_lock_);
+
+ // TODO(bugs.webrtc.org/9719): Delete when everything is converted to
+ // CreateAudioReceiver.
MediaTransportAudioSinkInterface* audio_sink_ RTC_GUARDED_BY(sink_lock_) =
nullptr;
MediaTransportVideoSinkInterface* video_sink_ RTC_GUARDED_BY(sink_lock_) =
diff --git a/api/test/loopback_media_transport_unittest.cc b/api/test/loopback_media_transport_unittest.cc
index b827405..8fe432d 100644
--- a/api/test/loopback_media_transport_unittest.cc
+++ b/api/test/loopback_media_transport_unittest.cc
@@ -22,6 +22,8 @@
class MockMediaTransportAudioSinkInterface
: public MediaTransportAudioSinkInterface {
public:
+ MOCK_METHOD1(OnData, void(MediaTransportEncodedAudioFrame));
+ // TODO(nisse): Deprecated version, delete.
MOCK_METHOD2(OnData, void(uint64_t, MediaTransportEncodedAudioFrame));
};
diff --git a/api/transport/media/audio_transport.cc b/api/transport/media/audio_transport.cc
index 7285ad4..5dae4d3 100644
--- a/api/transport/media/audio_transport.cc
+++ b/api/transport/media/audio_transport.cc
@@ -51,4 +51,10 @@
MediaTransportEncodedAudioFrame::MediaTransportEncodedAudioFrame(
MediaTransportEncodedAudioFrame&&) = default;
+void MediaTransportAudioSinkInterface::OnData(
+ uint64_t channel_id,
+ MediaTransportEncodedAudioFrame frame) {
+ OnData(frame);
+}
+
} // namespace webrtc
diff --git a/api/transport/media/audio_transport.h b/api/transport/media/audio_transport.h
index dcbdcd7..d3afbf3 100644
--- a/api/transport/media/audio_transport.h
+++ b/api/transport/media/audio_transport.h
@@ -111,9 +111,29 @@
public:
virtual ~MediaTransportAudioSinkInterface() = default;
- // Called when new encoded audio frame is received.
+ // Called when new encoded audio frame is received, and no receiver is
+ // registered. Deprecated.
virtual void OnData(uint64_t channel_id,
- MediaTransportEncodedAudioFrame frame) = 0;
+ MediaTransportEncodedAudioFrame frame);
+
+ // Called when new encoded audio frame is received.
+ // TODO(bugs.webrtc.org/9719): Make pure virtual after downstream
+ // implementations are updated.
+ virtual void OnData(MediaTransportEncodedAudioFrame frame) {}
+};
+
+class MediaTransportAudioSender {
+ public:
+ virtual ~MediaTransportAudioSender() = default;
+
+ virtual void SendAudioFrame(MediaTransportEncodedAudioFrame frame) = 0;
+};
+
+// Similar to RtpStreamReceiverInterface, only owns the association with the
+// demuxer.
+class MediaTransportAudioReceiver {
+ public:
+ virtual ~MediaTransportAudioReceiver() = default;
};
} // namespace webrtc
diff --git a/audio/channel_receive.cc b/audio/channel_receive.cc
index 0e218ed..40dc2c1 100644
--- a/audio/channel_receive.cc
+++ b/audio/channel_receive.cc
@@ -58,15 +58,14 @@
constexpr int kVoiceEngineMaxMinPlayoutDelayMs = 10000;
RTPHeader CreateRTPHeaderForMediaTransportFrame(
- const MediaTransportEncodedAudioFrame& frame,
- uint64_t channel_id) {
+ const MediaTransportEncodedAudioFrame& frame) {
webrtc::RTPHeader rtp_header;
rtp_header.payloadType = frame.payload_type();
rtp_header.payload_type_frequency = frame.sampling_rate_hz();
rtp_header.timestamp = frame.starting_sample_index();
rtp_header.sequenceNumber = frame.sequence_number();
- rtp_header.ssrc = static_cast<uint32_t>(channel_id);
+ // Note: SSRC is no longer used by NetEq, so not set.
// The rest are initialized by the RTPHeader constructor.
return rtp_header;
@@ -167,8 +166,12 @@
int64_t GetRTT() const;
// MediaTransportAudioSinkInterface override;
- void OnData(uint64_t channel_id,
- MediaTransportEncodedAudioFrame frame) override;
+ void OnData(MediaTransportEncodedAudioFrame frame) override;
+ // TODO(nisse): Deprecated variant. Delete.
+ void OnData(uint64_t /* channel_id */,
+ MediaTransportEncodedAudioFrame frame) override {
+ OnData(std::move(frame));
+ }
int32_t OnReceivedPayloadData(const uint8_t* payloadData,
size_t payloadSize,
@@ -293,8 +296,7 @@
}
// MediaTransportAudioSinkInterface override.
-void ChannelReceive::OnData(uint64_t channel_id,
- MediaTransportEncodedAudioFrame frame) {
+void ChannelReceive::OnData(MediaTransportEncodedAudioFrame frame) {
RTC_CHECK(media_transport_);
if (!Playing()) {
@@ -306,7 +308,7 @@
// Send encoded audio frame to Decoder / NetEq.
if (audio_coding_->IncomingPacket(
frame.encoded_data().data(), frame.encoded_data().size(),
- CreateRTPHeaderForMediaTransportFrame(frame, channel_id)) != 0) {
+ CreateRTPHeaderForMediaTransportFrame(frame)) != 0) {
RTC_DLOG(LS_ERROR) << "ChannelReceive::OnData: unable to "
"push data to the ACM";
}
