Revert "Add Sender and Receiver interfaces for MediaTransport audio"
This reverts commit 0d8eed6ac77fadf7f9bcf70c671710d60b1ee62d.
Reason for revert: crashes of unit tests.
Original change's description:
> 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}
TBR=solenberg@webrtc.org,nisse@webrtc.org,mellem@webrtc.org,sukhanov@webrtc.org,psla@webrtc.org,sukhanov@google.com
Change-Id: I02e409e1bbe2b2dea8a7b1aa08fa44d4146bda8f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9719
Reviewed-on: https://webrtc-review.googlesource.com/c/123232
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26733}diff --git a/api/media_transport_interface.cc b/api/media_transport_interface.cc
index 0dfec76..de04e19 100644
--- a/api/media_transport_interface.cc
+++ b/api/media_transport_interface.cc
@@ -54,18 +54,6 @@
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 e753ddc..2f5431f 100644
--- a/api/media_transport_interface.h
+++ b/api/media_transport_interface.h
@@ -187,28 +187,10 @@
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 fde732e..c466170 100644
--- a/api/test/loopback_media_transport.cc
+++ b/api/test/loopback_media_transport.cc
@@ -109,7 +109,6 @@
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);
@@ -117,58 +116,6 @@
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) {
@@ -370,10 +317,7 @@
MediaTransportEncodedAudioFrame frame) {
{
rtc::CritScope lock(&sink_lock_);
- const auto it = audio_sinks_.find(channel_id);
- if (it != audio_sinks_.end()) {
- it->second->OnData(frame);
- } else if (audio_sink_) {
+ 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 d2c503b..bcfdb63 100644
--- a/api/test/loopback_media_transport.h
+++ b/api/test/loopback_media_transport.h
@@ -11,7 +11,6 @@
#ifndef API_TEST_LOOPBACK_MEDIA_TRANSPORT_H_
#define API_TEST_LOOPBACK_MEDIA_TRANSPORT_H_
-#include <map>
#include <memory>
#include <utility>
#include <vector>
@@ -86,13 +85,6 @@
~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;
@@ -139,9 +131,6 @@
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);
@@ -155,17 +144,11 @@
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 8fe432d..b827405 100644
--- a/api/test/loopback_media_transport_unittest.cc
+++ b/api/test/loopback_media_transport_unittest.cc
@@ -22,8 +22,6 @@
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 5dae4d3..7285ad4 100644
--- a/api/transport/media/audio_transport.cc
+++ b/api/transport/media/audio_transport.cc
@@ -51,10 +51,4 @@
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 d3afbf3..dcbdcd7 100644
--- a/api/transport/media/audio_transport.h
+++ b/api/transport/media/audio_transport.h
@@ -111,29 +111,9 @@
public:
virtual ~MediaTransportAudioSinkInterface() = default;
- // Called when new encoded audio frame is received, and no receiver is
- // registered. Deprecated.
- virtual void OnData(uint64_t channel_id,
- 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;
+ virtual void OnData(uint64_t channel_id,
+ MediaTransportEncodedAudioFrame frame) = 0;
};
} // namespace webrtc
diff --git a/audio/channel_receive.cc b/audio/channel_receive.cc
index 40dc2c1..0e218ed 100644
--- a/audio/channel_receive.cc
+++ b/audio/channel_receive.cc
@@ -58,14 +58,15 @@
constexpr int kVoiceEngineMaxMinPlayoutDelayMs = 10000;
RTPHeader CreateRTPHeaderForMediaTransportFrame(
- const MediaTransportEncodedAudioFrame& frame) {
+ const MediaTransportEncodedAudioFrame& frame,
+ uint64_t channel_id) {
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();
- // Note: SSRC is no longer used by NetEq, so not set.
+ rtp_header.ssrc = static_cast<uint32_t>(channel_id);
// The rest are initialized by the RTPHeader constructor.
return rtp_header;
@@ -166,12 +167,8 @@
int64_t GetRTT() const;
// MediaTransportAudioSinkInterface override;
- void OnData(MediaTransportEncodedAudioFrame frame) override;
- // TODO(nisse): Deprecated variant. Delete.
- void OnData(uint64_t /* channel_id */,
- MediaTransportEncodedAudioFrame frame) override {
- OnData(std::move(frame));
- }
+ void OnData(uint64_t channel_id,
+ MediaTransportEncodedAudioFrame frame) override;
int32_t OnReceivedPayloadData(const uint8_t* payloadData,
size_t payloadSize,
@@ -296,7 +293,8 @@
}
// MediaTransportAudioSinkInterface override.
-void ChannelReceive::OnData(MediaTransportEncodedAudioFrame frame) {
+void ChannelReceive::OnData(uint64_t channel_id,
+ MediaTransportEncodedAudioFrame frame) {
RTC_CHECK(media_transport_);
if (!Playing()) {
@@ -308,7 +306,7 @@
// Send encoded audio frame to Decoder / NetEq.
if (audio_coding_->IncomingPacket(
frame.encoded_data().data(), frame.encoded_data().size(),
- CreateRTPHeaderForMediaTransportFrame(frame)) != 0) {
+ CreateRTPHeaderForMediaTransportFrame(frame, channel_id)) != 0) {
RTC_DLOG(LS_ERROR) << "ChannelReceive::OnData: unable to "
"push data to the ACM";
}
