Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(477)

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

Issue 1934513002: Remove usage of VoENetwork from VoEWrapper and FakeWebRtcVoiceEngine. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 8 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 | « no previous file | webrtc/media/engine/webrtcvoe.h » ('j') | webrtc/media/engine/webrtcvoiceengine.cc » ('J')
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 4aa6ea3d4380de25d3b127f750ddf3e1894a6b2d..8c9d14d168766b6906e95e8e792304b171211513 100644
--- a/webrtc/media/engine/fakewebrtcvoiceengine.h
+++ b/webrtc/media/engine/fakewebrtcvoiceengine.h
@@ -121,15 +121,13 @@ class FakeAudioProcessing : public webrtc::AudioProcessing {
class FakeWebRtcVoiceEngine
: public webrtc::VoEAudioProcessing,
public webrtc::VoEBase, public webrtc::VoECodec,
- public webrtc::VoEHardware,
- public webrtc::VoENetwork, public webrtc::VoERTP_RTCP,
+ public webrtc::VoEHardware, public webrtc::VoERTP_RTCP,
public webrtc::VoEVolumeControl {
public:
struct Channel {
Channel() {
memset(&send_codec, 0, sizeof(send_codec));
}
- bool external_transport = false;
bool playout = false;
float volume_scale = 1.0f;
bool vad = false;
@@ -146,8 +144,6 @@ class FakeWebRtcVoiceEngine
int associate_send_channel = -1;
std::vector<webrtc::CodecInst> recv_codecs;
webrtc::CodecInst send_codec;
- webrtc::PacketTime last_rtp_packet_time;
- std::list<std::string> packets;
int neteq_capacity = -1;
bool neteq_fast_accelerate = false;
};
@@ -191,10 +187,6 @@ class FakeWebRtcVoiceEngine
int GetNACKMaxPackets(int channel) {
return channels_[channel]->nack_max_packets;
}
- const webrtc::PacketTime& GetLastRtpPacketTime(int channel) {
- RTC_DCHECK(channels_.find(channel) != channels_.end());
- return channels_[channel]->last_rtp_packet_time;
- }
int GetSendCNPayloadType(int channel, bool wideband) {
return (wideband) ?
channels_[channel]->cn16_type :
@@ -455,40 +447,6 @@ class FakeWebRtcVoiceEngine
WEBRTC_STUB(EnableBuiltInNS, (bool enable));
virtual bool BuiltInNSIsAvailable() const { return false; }
- // webrtc::VoENetwork
- WEBRTC_FUNC(RegisterExternalTransport, (int channel,
- webrtc::Transport& transport)) {
- WEBRTC_CHECK_CHANNEL(channel);
- channels_[channel]->external_transport = true;
- return 0;
- }
- WEBRTC_FUNC(DeRegisterExternalTransport, (int channel)) {
- WEBRTC_CHECK_CHANNEL(channel);
- channels_[channel]->external_transport = false;
- return 0;
- }
- WEBRTC_FUNC(ReceivedRTPPacket, (int channel, const void* data,
- size_t length)) {
- WEBRTC_CHECK_CHANNEL(channel);
- if (!channels_[channel]->external_transport) return -1;
- channels_[channel]->packets.push_back(
- std::string(static_cast<const char*>(data), length));
- return 0;
- }
- WEBRTC_FUNC(ReceivedRTPPacket, (int channel, const void* data,
- size_t length,
- const webrtc::PacketTime& packet_time)) {
- WEBRTC_CHECK_CHANNEL(channel);
- if (ReceivedRTPPacket(channel, data, length) == -1) {
- return -1;
- }
- channels_[channel]->last_rtp_packet_time = packet_time;
- return 0;
- }
-
- WEBRTC_STUB(ReceivedRTCPPacket, (int channel, const void* data,
- size_t length));
-
// webrtc::VoERTP_RTCP
WEBRTC_FUNC(SetLocalSSRC, (int channel, unsigned int ssrc)) {
WEBRTC_CHECK_CHANNEL(channel);
« no previous file with comments | « no previous file | webrtc/media/engine/webrtcvoe.h » ('j') | webrtc/media/engine/webrtcvoiceengine.cc » ('J')

Powered by Google App Engine
This is Rietveld 408576698