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Unified Diff: webrtc/audio/audio_receive_stream.h

Issue 2709723003: Initial implementation of RtpTransportControllerReceive and related interfaces.
Patch Set: Fix audio. Created 3 years, 8 months ago
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Index: webrtc/audio/audio_receive_stream.h
diff --git a/webrtc/audio/audio_receive_stream.h b/webrtc/audio/audio_receive_stream.h
index adac88387613a43a7f6220567a75c8e881fe313c..f53a138298ce517ca25e7cbb201be8436777ecd4 100644
--- a/webrtc/audio/audio_receive_stream.h
+++ b/webrtc/audio/audio_receive_stream.h
@@ -18,7 +18,10 @@
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/thread_checker.h"
#include "webrtc/call/audio_receive_stream.h"
+#include "webrtc/call/rtp_transport_controller_receive.h"
#include "webrtc/call/syncable.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
the sun 2017/04/18 09:59:00 We only need forward decl here.
nisse-webrtc 2017/04/19 08:35:43 Done.
namespace webrtc {
class PacketRouter;
@@ -32,7 +35,8 @@ class ChannelProxy;
namespace internal {
class AudioSendStream;
-class AudioReceiveStream final : public webrtc::AudioReceiveStream,
+class AudioReceiveStream final : public webrtc::RtpPacketReceiverInterface,
+ public webrtc::AudioReceiveStream,
public AudioMixer::Source,
public Syncable {
public:
@@ -50,8 +54,8 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream,
void SetSink(std::unique_ptr<AudioSinkInterface> sink) override;
void SetGain(float gain) override;
- // TODO(nisse): Intended to be part of an RtpPacketReceiver interface.
- void OnRtpPacket(const RtpPacketReceived& packet);
+ // webrtc::RtpPacketReceiverInterface implementation
+ bool OnRtpPacketReceive(RtpPacketReceived* packet) override;
// AudioMixer::Source
AudioFrameInfo GetAudioFrameWithInfo(int sample_rate_hz,
@@ -78,6 +82,8 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream,
rtc::ThreadChecker worker_thread_checker_;
rtc::ThreadChecker module_process_thread_checker_;
const webrtc::AudioReceiveStream::Config config_;
+ RtpHeaderExtensionMap rtp_header_extensions_;
+
rtc::scoped_refptr<webrtc::AudioState> audio_state_;
std::unique_ptr<voe::ChannelProxy> channel_proxy_;
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