| Index: webrtc/media/engine/fakewebrtcvoiceengine.h
|
| diff --git a/webrtc/media/engine/fakewebrtcvoiceengine.h b/webrtc/media/engine/fakewebrtcvoiceengine.h
|
| index 180d06bcaf23b0af6a4722d12de289a79fe31667..13721ea5546c4a531c866628a24db273d6766d3f 100644
|
| --- a/webrtc/media/engine/fakewebrtcvoiceengine.h
|
| +++ b/webrtc/media/engine/fakewebrtcvoiceengine.h
|
| @@ -11,20 +11,11 @@
|
| #ifndef WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_
|
| #define WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_
|
|
|
| -#include <stddef.h>
|
| -
|
| -#include <list>
|
| #include <map>
|
| #include <vector>
|
|
|
| #include "webrtc/base/checks.h"
|
| -#include "webrtc/base/stringutils.h"
|
| -#include "webrtc/base/checks.h"
|
| -#include "webrtc/config.h"
|
| -#include "webrtc/media/base/codec.h"
|
| -#include "webrtc/media/base/rtputils.h"
|
| #include "webrtc/media/engine/webrtcvoe.h"
|
| -#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
|
| #include "webrtc/modules/audio_processing/include/audio_processing.h"
|
|
|
| namespace webrtc {
|
| @@ -35,31 +26,15 @@ class TransmitMixer;
|
|
|
| namespace cricket {
|
|
|
| -static const int kOpusBandwidthNb = 4000;
|
| -static const int kOpusBandwidthMb = 6000;
|
| -static const int kOpusBandwidthWb = 8000;
|
| -static const int kOpusBandwidthSwb = 12000;
|
| -static const int kOpusBandwidthFb = 20000;
|
| -
|
| #define WEBRTC_CHECK_CHANNEL(channel) \
|
| if (channels_.find(channel) == channels_.end()) return -1;
|
|
|
| #define WEBRTC_STUB(method, args) \
|
| int method args override { return 0; }
|
|
|
| -#define WEBRTC_STUB_CONST(method, args) \
|
| - int method args const override { return 0; }
|
| -
|
| -#define WEBRTC_BOOL_STUB(method, args) \
|
| - bool method args override { return true; }
|
| -
|
| -#define WEBRTC_VOID_STUB(method, args) \
|
| - void method args override {}
|
| -
|
| #define WEBRTC_FUNC(method, args) int method args override
|
|
|
| -class FakeWebRtcVoiceEngine
|
| - : public webrtc::VoEBase, public webrtc::VoECodec {
|
| +class FakeWebRtcVoiceEngine : public webrtc::VoEBase {
|
| public:
|
| struct Channel {
|
| std::vector<webrtc::CodecInst> recv_codecs;
|
| @@ -140,66 +115,6 @@ class FakeWebRtcVoiceEngine
|
| WEBRTC_STUB(AssociateSendChannel, (int channel,
|
| int accociate_send_channel));
|
|
|
| - // webrtc::VoECodec
|
| - WEBRTC_STUB(NumOfCodecs, ());
|
| - WEBRTC_STUB(GetCodec, (int index, webrtc::CodecInst& codec));
|
| - WEBRTC_STUB(SetSendCodec, (int channel, const webrtc::CodecInst& codec));
|
| - WEBRTC_STUB(GetSendCodec, (int channel, webrtc::CodecInst& codec));
|
| - WEBRTC_STUB(SetBitRate, (int channel, int bitrate_bps));
|
| - WEBRTC_STUB(GetRecCodec, (int channel, webrtc::CodecInst& codec));
|
| - WEBRTC_FUNC(SetRecPayloadType, (int channel,
|
| - const webrtc::CodecInst& codec)) {
|
| - WEBRTC_CHECK_CHANNEL(channel);
|
| - Channel* ch = channels_[channel];
|
| - // Check if something else already has this slot.
|
| - if (codec.pltype != -1) {
|
| - for (std::vector<webrtc::CodecInst>::iterator it =
|
| - ch->recv_codecs.begin(); it != ch->recv_codecs.end(); ++it) {
|
| - if (it->pltype == codec.pltype &&
|
| - _stricmp(it->plname, codec.plname) != 0) {
|
| - return -1;
|
| - }
|
| - }
|
| - }
|
| - // Otherwise try to find this codec and update its payload type.
|
| - int result = -1; // not found
|
| - for (std::vector<webrtc::CodecInst>::iterator it = ch->recv_codecs.begin();
|
| - it != ch->recv_codecs.end(); ++it) {
|
| - if (strcmp(it->plname, codec.plname) == 0 &&
|
| - it->plfreq == codec.plfreq &&
|
| - it->channels == codec.channels) {
|
| - it->pltype = codec.pltype;
|
| - result = 0;
|
| - }
|
| - }
|
| - return result;
|
| - }
|
| - WEBRTC_STUB(SetSendCNPayloadType, (int channel, int type,
|
| - webrtc::PayloadFrequencies frequency));
|
| - WEBRTC_FUNC(GetRecPayloadType, (int channel, webrtc::CodecInst& codec)) {
|
| - WEBRTC_CHECK_CHANNEL(channel);
|
| - Channel* ch = channels_[channel];
|
| - for (std::vector<webrtc::CodecInst>::iterator it = ch->recv_codecs.begin();
|
| - it != ch->recv_codecs.end(); ++it) {
|
| - if (strcmp(it->plname, codec.plname) == 0 &&
|
| - it->plfreq == codec.plfreq &&
|
| - it->channels == codec.channels &&
|
| - it->pltype != -1) {
|
| - codec.pltype = it->pltype;
|
| - return 0;
|
| - }
|
| - }
|
| - return -1; // not found
|
| - }
|
| - WEBRTC_STUB(SetVADStatus, (int channel, bool enable, webrtc::VadModes mode,
|
| - bool disableDTX));
|
| - WEBRTC_STUB(GetVADStatus, (int channel, bool& enabled,
|
| - webrtc::VadModes& mode, bool& disabledDTX));
|
| - WEBRTC_STUB(SetFECStatus, (int channel, bool enable));
|
| - WEBRTC_STUB(GetFECStatus, (int channel, bool& enable));
|
| - WEBRTC_STUB(SetOpusMaxPlaybackRate, (int channel, int frequency_hz));
|
| - WEBRTC_STUB(SetOpusDtx, (int channel, bool enable_dtx));
|
| -
|
| size_t GetNetEqCapacity() const {
|
| auto ch = channels_.find(last_channel_);
|
| RTC_DCHECK(ch != channels_.end());
|
|
|