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

Issue 1361633002: Remove the [Un]RegisterVoiceProcessor() API. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 5 years, 3 months ago
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Index: talk/media/webrtc/fakewebrtcvoiceengine.h
diff --git a/talk/media/webrtc/fakewebrtcvoiceengine.h b/talk/media/webrtc/fakewebrtcvoiceengine.h
index bc95839392974a688b33896c3d64c53680bb1d12..9b7b8f7dc06241ad81b4857c504d37cf8e6cdeed 100644
--- a/talk/media/webrtc/fakewebrtcvoiceengine.h
+++ b/talk/media/webrtc/fakewebrtcvoiceengine.h
@@ -34,7 +34,6 @@
#include "talk/media/base/codec.h"
#include "talk/media/base/rtputils.h"
-#include "talk/media/base/voiceprocessor.h"
#include "talk/media/webrtc/fakewebrtccommon.h"
#include "talk/media/webrtc/webrtcvoe.h"
#include "webrtc/base/basictypes.h"
@@ -187,8 +186,7 @@ class FakeAudioProcessing : public webrtc::AudioProcessing {
class FakeWebRtcVoiceEngine
: public webrtc::VoEAudioProcessing,
public webrtc::VoEBase, public webrtc::VoECodec, public webrtc::VoEDtmf,
- public webrtc::VoEHardware,
- public webrtc::VoEExternalMedia, public webrtc::VoENetEqStats,
+ public webrtc::VoEHardware, public webrtc::VoENetEqStats,
public webrtc::VoENetwork, public webrtc::VoERTP_RTCP,
public webrtc::VoEVideoSync, public webrtc::VoEVolumeControl {
public:
@@ -215,7 +213,6 @@ class FakeWebRtcVoiceEngine
opus_dtx(false),
red(false),
nack(false),
- media_processor_registered(false),
rx_agc_enabled(false),
rx_agc_mode(webrtc::kAgcDefault),
cn8_type(13),
@@ -246,7 +243,6 @@ class FakeWebRtcVoiceEngine
bool opus_dtx;
bool red;
bool nack;
- bool media_processor_registered;
bool rx_agc_enabled;
webrtc::AgcModes rx_agc_mode;
webrtc::AgcConfig rx_agc_config;
@@ -294,8 +290,7 @@ class FakeWebRtcVoiceEngine
playout_fail_channel_(-1),
send_fail_channel_(-1),
recording_sample_rate_(-1),
- playout_sample_rate_(-1),
- media_processor_(NULL) {
+ playout_sample_rate_(-1) {
memset(&agc_config_, 0, sizeof(agc_config_));
}
~FakeWebRtcVoiceEngine() {
@@ -307,9 +302,6 @@ class FakeWebRtcVoiceEngine
}
}
- bool IsExternalMediaProcessorRegistered() const {
- return media_processor_ != NULL;
- }
bool IsInited() const { return inited_; }
int GetLastChannel() const { return last_channel_; }
int GetChannelFromLocalSsrc(uint32 local_ssrc) const {
@@ -388,19 +380,6 @@ class FakeWebRtcVoiceEngine
void set_fail_create_channel(bool fail_create_channel) {
fail_create_channel_ = fail_create_channel;
}
- void TriggerProcessPacket(MediaProcessorDirection direction) {
- webrtc::ProcessingTypes pt =
- (direction == cricket::MPD_TX) ?
- webrtc::kRecordingPerChannel : webrtc::kPlaybackAllChannelsMixed;
- if (media_processor_ != NULL) {
- media_processor_->Process(0,
- pt,
- NULL,
- 0,
- 0,
- true);
- }
- }
int AddChannel(const webrtc::Config& config) {
if (fail_create_channel_) {
return -1;
@@ -1101,31 +1080,6 @@ class FakeWebRtcVoiceEngine
return (dtmf_info_.dtmf_event_code == event_code &&
dtmf_info_.dtmf_length_ms == length_ms);
}
- // webrtc::VoEExternalMedia
- WEBRTC_FUNC(RegisterExternalMediaProcessing,
- (int channel, webrtc::ProcessingTypes type,
- webrtc::VoEMediaProcess& processObject)) {
- WEBRTC_CHECK_CHANNEL(channel);
- if (channels_[channel]->media_processor_registered) {
- return -1;
- }
- channels_[channel]->media_processor_registered = true;
- media_processor_ = &processObject;
- return 0;
- }
- WEBRTC_FUNC(DeRegisterExternalMediaProcessing,
- (int channel, webrtc::ProcessingTypes type)) {
- WEBRTC_CHECK_CHANNEL(channel);
- if (!channels_[channel]->media_processor_registered) {
- return -1;
- }
- channels_[channel]->media_processor_registered = false;
- media_processor_ = NULL;
- return 0;
- }
- WEBRTC_STUB(GetAudioFrame, (int channel, int desired_sample_rate_hz,
- webrtc::AudioFrame* frame));
- WEBRTC_STUB(SetExternalMixing, (int channel, bool enable));
int GetNetEqCapacity() const {
auto ch = channels_.find(last_channel_);
ASSERT(ch != channels_.end());
@@ -1199,7 +1153,6 @@ class FakeWebRtcVoiceEngine
int recording_sample_rate_;
int playout_sample_rate_;
DtmfInfo dtmf_info_;
- webrtc::VoEMediaProcess* media_processor_;
FakeAudioProcessing audio_processing_;
};
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