Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(146)

Unified Diff: webrtc/media/engine/fakewebrtcvoiceengine.h

Issue 1741933002: Prevent a voice channel from sending data before a renderer is set. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Modifying copyright header to satisfy presubmit bot. Created 4 years, 9 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
« no previous file with comments | « webrtc/media/engine/fakewebrtccall.h ('k') | webrtc/media/engine/webrtcvoiceengine.h » ('j') | no next file with comments »
Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
Index: webrtc/media/engine/fakewebrtcvoiceengine.h
diff --git a/webrtc/media/engine/fakewebrtcvoiceengine.h b/webrtc/media/engine/fakewebrtcvoiceengine.h
index 941dfef3b22bf8783f60fa84d739e19c78760244..926f2473e610cd4e0f7ae68eb2a005eef97276dc 100644
--- a/webrtc/media/engine/fakewebrtcvoiceengine.h
+++ b/webrtc/media/engine/fakewebrtcvoiceengine.h
@@ -130,7 +130,6 @@ class FakeWebRtcVoiceEngine
struct Channel {
explicit Channel()
: external_transport(false),
- send(false),
playout(false),
volume_scale(1.0),
vad(false),
@@ -151,7 +150,6 @@ class FakeWebRtcVoiceEngine
memset(&send_codec, 0, sizeof(send_codec));
}
bool external_transport;
- bool send;
bool playout;
float volume_scale;
bool vad;
@@ -193,7 +191,6 @@ class FakeWebRtcVoiceEngine
agc_mode_(webrtc::kAgcDefault),
observer_(NULL),
playout_fail_channel_(-1),
- send_fail_channel_(-1),
recording_sample_rate_(-1),
playout_sample_rate_(-1) {
memset(&agc_config_, 0, sizeof(agc_config_));
@@ -213,9 +210,6 @@ class FakeWebRtcVoiceEngine
bool GetPlayout(int channel) {
return channels_[channel]->playout;
}
- bool GetSend(int channel) {
- return channels_[channel]->send;
- }
bool GetVAD(int channel) {
return channels_[channel]->vad;
}
@@ -268,9 +262,6 @@ class FakeWebRtcVoiceEngine
void set_playout_fail_channel(int channel) {
playout_fail_channel_ = channel;
}
- void set_send_fail_channel(int channel) {
- send_fail_channel_ = channel;
- }
void set_fail_create_channel(bool fail_create_channel) {
fail_create_channel_ = fail_create_channel;
}
@@ -347,28 +338,14 @@ class FakeWebRtcVoiceEngine
return -1;
}
}
- WEBRTC_FUNC(StartSend, (int channel)) {
- if (send_fail_channel_ != channel) {
- WEBRTC_CHECK_CHANNEL(channel);
- channels_[channel]->send = true;
- return 0;
- } else {
- // When send_fail_channel_ == channel, fail the StartSend on this
- // channel.
- return -1;
- }
- }
+ WEBRTC_STUB(StartSend, (int channel));
WEBRTC_STUB(StopReceive, (int channel));
WEBRTC_FUNC(StopPlayout, (int channel)) {
WEBRTC_CHECK_CHANNEL(channel);
channels_[channel]->playout = false;
return 0;
}
- WEBRTC_FUNC(StopSend, (int channel)) {
- WEBRTC_CHECK_CHANNEL(channel);
- channels_[channel]->send = false;
- return 0;
- }
+ WEBRTC_STUB(StopSend, (int channel));
WEBRTC_STUB(GetVersion, (char version[1024]));
WEBRTC_STUB(LastError, ());
WEBRTC_FUNC(AssociateSendChannel, (int channel,
@@ -797,7 +774,6 @@ class FakeWebRtcVoiceEngine
webrtc::AgcConfig agc_config_;
webrtc::VoiceEngineObserver* observer_;
int playout_fail_channel_;
- int send_fail_channel_;
int recording_sample_rate_;
int playout_sample_rate_;
FakeAudioProcessing audio_processing_;
« no previous file with comments | « webrtc/media/engine/fakewebrtccall.h ('k') | webrtc/media/engine/webrtcvoiceengine.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698