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Avoid recreating the audio stream when a frame decryptor is set.

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This is to be consistent with how things work on the video side but
also much less drastic than the current implementation. Aim is to
remove RecreateAudioReceiveStream(), which would improve efficiency
as well as allow for specific handling of the cases that currently
trigger recreation.

Bug: webrtc:11993
Change-Id: Ia81a5e66d44e41ea4eb2bff800e0b1583821c96a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221860
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34282}
This commit is contained in:
Tommi
2021-06-14 08:11:10 +02:00
committed by WebRTC LUCI CQ
parent e5f1a3992e
commit e097282f11
10 changed files with 44 additions and 3 deletions
Download Patch File Download Diff File Expand all files Collapse all files

1
audio/BUILD.gn
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@ -152,6 +152,7 @@ if (rtc_include_tests) {
"../api/audio_codecs:audio_codecs_api",
"../api/audio_codecs/opus:audio_decoder_opus",
"../api/audio_codecs/opus:audio_encoder_opus",
"../api/crypto:frame_decryptor_interface",
"../api/rtc_event_log",
"../api/task_queue:default_task_queue_factory",
"../api/units:time_delta",

8
audio/audio_receive_stream.cc
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@ -246,6 +246,14 @@ void AudioReceiveStream::SetUseTransportCcAndNackHistory(bool use_transport_cc,
}
}
void AudioReceiveStream::SetFrameDecryptor(
rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor) {
// TODO(bugs.webrtc.org/11993): This is called via WebRtcAudioReceiveStream,
// expect to be called on the network thread.
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
channel_receive_->SetFrameDecryptor(std::move(frame_decryptor));
}
webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats(
bool get_and_clear_legacy_stats) const {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);

2
audio/audio_receive_stream.h
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@ -91,6 +91,8 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream,
void SetDecoderMap(std::map<int, SdpAudioFormat> decoder_map) override;
void SetUseTransportCcAndNackHistory(bool use_transport_cc,
int history_ms) override;
void SetFrameDecryptor(rtc::scoped_refptr<webrtc::FrameDecryptorInterface>
frame_decryptor) override;
webrtc::AudioReceiveStream::Stats GetStats(
bool get_and_clear_legacy_stats) const override;

16
audio/channel_receive.cc
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@ -177,6 +177,9 @@ class ChannelReceive : public ChannelReceiveInterface {
rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
override;
void SetFrameDecryptor(rtc::scoped_refptr<webrtc::FrameDecryptorInterface>
frame_decryptor) override;
private:
void ReceivePacket(const uint8_t* packet,
size_t packet_length,
@ -275,10 +278,12 @@ class ChannelReceive : public ChannelReceiveInterface {
SequenceChecker construction_thread_;
// E2EE Audio Frame Decryption
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor_;
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor_
RTC_GUARDED_BY(worker_thread_checker_);
webrtc::CryptoOptions crypto_options_;
webrtc::AbsoluteCaptureTimeInterpolator absolute_capture_time_interpolator_;
webrtc::AbsoluteCaptureTimeInterpolator absolute_capture_time_interpolator_
RTC_GUARDED_BY(worker_thread_checker_);
webrtc::CaptureClockOffsetUpdater capture_clock_offset_updater_;
@ -889,6 +894,13 @@ void ChannelReceive::SetDepacketizerToDecoderFrameTransformer(
InitFrameTransformerDelegate(std::move(frame_transformer));
}
void ChannelReceive::SetFrameDecryptor(
rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor) {
// TODO(bugs.webrtc.org/11993): Expect to be called on the network thread.
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
frame_decryptor_ = std::move(frame_decryptor);
}
NetworkStatistics ChannelReceive::GetNetworkStatistics(
bool get_and_clear_legacy_stats) const {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);

3
audio/channel_receive.h
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@ -159,6 +159,9 @@ class ChannelReceiveInterface : public RtpPacketSinkInterface {
virtual void SetDepacketizerToDecoderFrameTransformer(
rtc::scoped_refptr<webrtc::FrameTransformerInterface>
frame_transformer) = 0;
virtual void SetFrameDecryptor(
rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor) = 0;
};
std::unique_ptr<ChannelReceiveInterface> CreateChannelReceive(

6
audio/mock_voe_channel_proxy.h
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@ -17,6 +17,7 @@
#include <utility>
#include <vector>
#include "api/crypto/frame_decryptor_interface.h"
#include "api/test/mock_frame_encryptor.h"
#include "audio/channel_receive.h"
#include "audio/channel_send.h"
@ -98,6 +99,11 @@ class MockChannelReceive : public voe::ChannelReceiveInterface {
SetDepacketizerToDecoderFrameTransformer,
(rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer),
(override));
MOCK_METHOD(
void,
SetFrameDecryptor,
(rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor),
(override));
};
class MockChannelSend : public voe::ChannelSendInterface {

2
call/audio_receive_stream.h
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@ -177,6 +177,8 @@ class AudioReceiveStream {
virtual void SetDecoderMap(std::map<int, SdpAudioFormat> decoder_map) = 0;
virtual void SetUseTransportCcAndNackHistory(bool use_transport_cc,
int history_ms) = 0;
virtual void SetFrameDecryptor(
rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor) = 0;
// Starts stream activity.
// When a stream is active, it can receive, process and deliver packets.

5
media/engine/fake_webrtc_call.cc
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@ -113,6 +113,11 @@ void FakeAudioReceiveStream::SetUseTransportCcAndNackHistory(
config_.rtp.nack.rtp_history_ms = history_ms;
}
void FakeAudioReceiveStream::SetFrameDecryptor(
rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor) {
config_.frame_decryptor = std::move(frame_decryptor);
}
webrtc::AudioReceiveStream::Stats FakeAudioReceiveStream::GetStats(
bool get_and_clear_legacy_stats) const {
return stats_;

2
media/engine/fake_webrtc_call.h
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@ -112,6 +112,8 @@ class FakeAudioReceiveStream final : public webrtc::AudioReceiveStream {
std::map<int, webrtc::SdpAudioFormat> decoder_map) override;
void SetUseTransportCcAndNackHistory(bool use_transport_cc,
int history_ms) override;
void SetFrameDecryptor(rtc::scoped_refptr<webrtc::FrameDecryptorInterface>
frame_decryptor) override;
webrtc::AudioReceiveStream::Stats GetStats(
bool get_and_clear_legacy_stats) const override;

2
media/engine/webrtc_voice_engine.cc
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@ -1218,7 +1218,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
config_.frame_decryptor = frame_decryptor;
RecreateAudioReceiveStream();
stream_->SetFrameDecryptor(std::move(frame_decryptor));
}
void SetLocalSsrc(uint32_t local_ssrc) {