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Bug Fix: WebRTC Unity Plugin Audio One Way

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When audio_only is on for the webrtc unity plugin, there is a bug that
the audio from hologram cannot be heard at the remote side.

Actually we found the audio is transmitted to the remote side, but the
remote side wants video data also to playout everything. So without
video data, the remote side will drop all the audio data.

Thus, on the hologram (using webrtc unity plugin) side, we should not
hook up a dummy camera, but instead we should use media constraint to
request the remote side to send video data.

This CL fixes the bug.

Bug: webrtc:8555
Change-Id: I21ddda65185b645088aa4ac15f47b3f8ffad1873
Reviewed-on: https://webrtc-review.googlesource.com/24680
Commit-Queue: Qiang Chen <qiangchen@chromium.org>
Reviewed-by: George Zhou <gyzhou@chromium.org>
Cr-Commit-Position: refs/heads/master@{#21094}
This commit is contained in:
Qiang Chen
2017-12-05 11:11:21 -08:00
committed by Commit Bot
parent 1b36cb09c4
commit 51e2046dbc
3 changed files with 26 additions and 16 deletions
Download Patch File Download Diff File Expand all files Collapse all files

2
examples/BUILD.gn
View File

@ -657,6 +657,8 @@ if (is_win || is_android) {
deps = [
"../api:libjingle_peerconnection_test_api",
"../api:video_frame_api",
"../api/audio_codecs:builtin_audio_decoder_factory",
"../api/audio_codecs:builtin_audio_encoder_factory",
"../media:rtc_media",
"../media:rtc_media_base",
"../modules/video_capture:video_capture_module",

36
examples/unityplugin/simple_peer_connection.cc
View File

@ -12,6 +12,8 @@
#include <utility>
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/test/fakeconstraints.h"
#include "api/videosourceproxy.h"
#include "media/engine/webrtcvideocapturerfactory.h"
@ -91,7 +93,8 @@ bool SimplePeerConnection::InitializePeerConnection(const char** turn_urls,
g_peer_connection_factory = webrtc::CreatePeerConnectionFactory(
g_worker_thread.get(), g_worker_thread.get(), g_signaling_thread.get(),
nullptr, nullptr, nullptr);
nullptr, webrtc::CreateBuiltinAudioEncoderFactory(),
webrtc::CreateBuiltinAudioDecoderFactory(), nullptr, nullptr);
}
if (!g_peer_connection_factory.get()) {
DeletePeerConnection();
@ -99,19 +102,19 @@ bool SimplePeerConnection::InitializePeerConnection(const char** turn_urls,
}
g_peer_count++;
if (!CreatePeerConnection(turn_urls, no_of_urls, username, credential,
is_receiver)) {
if (!CreatePeerConnection(turn_urls, no_of_urls, username, credential)) {
DeletePeerConnection();
return false;
}
mandatory_receive_ = is_receiver;
return peer_connection_.get() != nullptr;
}
bool SimplePeerConnection::CreatePeerConnection(const char** turn_urls,
const int no_of_urls,
const char* username,
const char* credential,
bool is_receiver) {
const char* credential) {
RTC_DCHECK(g_peer_connection_factory.get() != nullptr);
RTC_DCHECK(peer_connection_.get() == nullptr);
@ -148,11 +151,6 @@ bool SimplePeerConnection::CreatePeerConnection(const char** turn_urls,
webrtc::FakeConstraints constraints;
constraints.SetAllowDtlsSctpDataChannels();
if (is_receiver) {
constraints.SetMandatoryReceiveAudio(true);
constraints.SetMandatoryReceiveVideo(true);
}
peer_connection_ = g_peer_connection_factory->CreatePeerConnection(
config_, &constraints, nullptr, nullptr, this);
@ -192,7 +190,12 @@ bool SimplePeerConnection::CreateOffer() {
if (!peer_connection_.get())
return false;
peer_connection_->CreateOffer(this, nullptr);
webrtc::FakeConstraints constraints;
if (mandatory_receive_) {
constraints.SetMandatoryReceiveAudio(true);
constraints.SetMandatoryReceiveVideo(true);
}
peer_connection_->CreateOffer(this, &constraints);
return true;
}
@ -200,7 +203,12 @@ bool SimplePeerConnection::CreateAnswer() {
if (!peer_connection_.get())
return false;
peer_connection_->CreateAnswer(this, nullptr);
webrtc::FakeConstraints constraints;
if (mandatory_receive_) {
constraints.SetMandatoryReceiveAudio(true);
constraints.SetMandatoryReceiveVideo(true);
}
peer_connection_->CreateAnswer(this, &constraints);
return true;
}
@ -421,8 +429,8 @@ void SimplePeerConnection::AddStreams(bool audio_only) {
RTC_DCHECK(texture_helper != nullptr)
<< "Cannot get the Surface Texture Helper.";
rtc::scoped_refptr<AndroidVideoTrackSource> source(
new rtc::RefCountedObject<AndroidVideoTrackSource>(
rtc::scoped_refptr<webrtc::jni::AndroidVideoTrackSource> source(
new rtc::RefCountedObject<webrtc::jni::AndroidVideoTrackSource>(
g_signaling_thread.get(), env, texture_helper, false));
rtc::scoped_refptr<webrtc::VideoTrackSourceProxy> proxy_source =
webrtc::VideoTrackSourceProxy::Create(g_signaling_thread.get(),

4
examples/unityplugin/simple_peer_connection.h
View File

@ -64,8 +64,7 @@ class SimplePeerConnection : public webrtc::PeerConnectionObserver,
bool CreatePeerConnection(const char** turn_urls,
const int no_of_urls,
const char* username,
const char* credential,
bool is_receiver);
const char* credential);
void CloseDataChannel();
std::unique_ptr<cricket::VideoCapturer> OpenVideoCaptureDevice();
void SetAudioControl();
@ -127,6 +126,7 @@ class SimplePeerConnection : public webrtc::PeerConnectionObserver,
bool is_mute_audio_ = false;
bool is_record_audio_ = false;
bool mandatory_receive_ = false;
// disallow copy-and-assign
SimplePeerConnection(const SimplePeerConnection&) = delete;