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Remove VoEFile from VoeWrapper and the remaining places in libjingle where it was being used.

Browse Source 
BUG=webrtc:4690
R=pbos@webrtc.org

Review URL: https://codereview.webrtc.org/1360773002 .

Cr-Commit-Position: refs/heads/master@{#10026}
This commit is contained in:
Fredrik Solenberg
2015-09-23 12:05:37 +02:00
parent f706c8ae91
commit 09677342ae
5 changed files with 2 additions and 76 deletions
Download Patch File Download Diff File Expand all files Collapse all files

62
talk/media/webrtc/fakewebrtcvoiceengine.h
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@ -187,7 +187,7 @@ class FakeAudioProcessing : public webrtc::AudioProcessing {
class FakeWebRtcVoiceEngine
: public webrtc::VoEAudioProcessing,
public webrtc::VoEBase, public webrtc::VoECodec, public webrtc::VoEDtmf,
public webrtc::VoEFile, public webrtc::VoEHardware,
public webrtc::VoEHardware,
public webrtc::VoEExternalMedia, public webrtc::VoENetEqStats,
public webrtc::VoENetwork, public webrtc::VoERTP_RTCP,
public webrtc::VoEVideoSync, public webrtc::VoEVolumeControl {
@ -209,7 +209,6 @@ class FakeWebRtcVoiceEngine
volume_scale(1.0),
volume_pan_left(1.0),
volume_pan_right(1.0),
file(false),
vad(false),
codec_fec(false),
max_encoding_bandwidth(0),
@ -241,7 +240,6 @@ class FakeWebRtcVoiceEngine
float volume_scale;
float volume_pan_left;
float volume_pan_right;
bool file;
bool vad;
bool codec_fec;
int max_encoding_bandwidth;
@ -723,64 +721,6 @@ class FakeWebRtcVoiceEngine
return 0;
}
// webrtc::VoEFile
WEBRTC_FUNC(StartPlayingFileLocally, (int channel, const char* fileNameUTF8,
bool loop, webrtc::FileFormats format,
float volumeScaling, int startPointMs,
int stopPointMs)) {
WEBRTC_CHECK_CHANNEL(channel);
channels_[channel]->file = true;
return 0;
}
WEBRTC_FUNC(StartPlayingFileLocally, (int channel, webrtc::InStream* stream,
webrtc::FileFormats format,
float volumeScaling, int startPointMs,
int stopPointMs)) {
WEBRTC_CHECK_CHANNEL(channel);
channels_[channel]->file = true;
return 0;
}
WEBRTC_FUNC(StopPlayingFileLocally, (int channel)) {
WEBRTC_CHECK_CHANNEL(channel);
channels_[channel]->file = false;
return 0;
}
WEBRTC_FUNC(IsPlayingFileLocally, (int channel)) {
WEBRTC_CHECK_CHANNEL(channel);
return (channels_[channel]->file) ? 1 : 0;
}
WEBRTC_STUB(StartPlayingFileAsMicrophone, (int channel,
const char* fileNameUTF8,
bool loop,
bool mixWithMicrophone,
webrtc::FileFormats format,
float volumeScaling));
WEBRTC_STUB(StartPlayingFileAsMicrophone, (int channel,
webrtc::InStream* stream,
bool mixWithMicrophone,
webrtc::FileFormats format,
float volumeScaling));
WEBRTC_STUB(StopPlayingFileAsMicrophone, (int channel));
WEBRTC_STUB(IsPlayingFileAsMicrophone, (int channel));
WEBRTC_STUB(StartRecordingPlayout, (int channel, const char* fileNameUTF8,
webrtc::CodecInst* compression,
int maxSizeBytes));
WEBRTC_STUB(StartRecordingPlayout, (int channel, webrtc::OutStream* stream,
webrtc::CodecInst* compression));
WEBRTC_STUB(StopRecordingPlayout, (int channel));
WEBRTC_FUNC(StartRecordingMicrophone, (const char* fileNameUTF8,
webrtc::CodecInst* compression,
int maxSizeBytes)) {
return 0;
}
WEBRTC_FUNC(StartRecordingMicrophone, (webrtc::OutStream* stream,
webrtc::CodecInst* compression)) {
return 0;
}
WEBRTC_FUNC(StopRecordingMicrophone, ()) {
return 0;
}
// webrtc::VoEHardware
WEBRTC_FUNC(GetNumOfRecordingDevices, (int& num)) {
return GetNumDevices(num);

6
talk/media/webrtc/webrtcvoe.h
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@ -95,7 +95,7 @@ class VoEWrapper {
public:
VoEWrapper()
: engine_(webrtc::VoiceEngine::Create()), processing_(engine_),
base_(engine_), codec_(engine_), dtmf_(engine_), file_(engine_),
base_(engine_), codec_(engine_), dtmf_(engine_),
hw_(engine_), media_(engine_), neteq_(engine_), network_(engine_),
rtp_(engine_), sync_(engine_), volume_(engine_) {
}
@ -103,7 +103,6 @@ class VoEWrapper {
webrtc::VoEBase* base,
webrtc::VoECodec* codec,
webrtc::VoEDtmf* dtmf,
webrtc::VoEFile* file,
webrtc::VoEHardware* hw,
webrtc::VoEExternalMedia* media,
webrtc::VoENetEqStats* neteq,
@ -116,7 +115,6 @@ class VoEWrapper {
base_(base),
codec_(codec),
dtmf_(dtmf),
file_(file),
hw_(hw),
media_(media),
neteq_(neteq),
@ -131,7 +129,6 @@ class VoEWrapper {
webrtc::VoEBase* base() const { return base_.get(); }
webrtc::VoECodec* codec() const { return codec_.get(); }
webrtc::VoEDtmf* dtmf() const { return dtmf_.get(); }
webrtc::VoEFile* file() const { return file_.get(); }
webrtc::VoEHardware* hw() const { return hw_.get(); }
webrtc::VoEExternalMedia* media() const { return media_.get(); }
webrtc::VoENetEqStats* neteq() const { return neteq_.get(); }
@ -147,7 +144,6 @@ class VoEWrapper {
scoped_voe_ptr<webrtc::VoEBase> base_;
scoped_voe_ptr<webrtc::VoECodec> codec_;
scoped_voe_ptr<webrtc::VoEDtmf> dtmf_;
scoped_voe_ptr<webrtc::VoEFile> file_;
scoped_voe_ptr<webrtc::VoEHardware> hw_;
scoped_voe_ptr<webrtc::VoEExternalMedia> media_;
scoped_voe_ptr<webrtc::VoENetEqStats> neteq_;

5
talk/media/webrtc/webrtcvoiceengine.cc
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@ -2308,11 +2308,6 @@ bool WebRtcVoiceMediaChannel::ChangeSend(int channel, SendFlags send) {
LOG_RTCERR1(StartSend, channel);
return false;
}
if (engine()->voe()->file() &&
engine()->voe()->file()->StopPlayingFileAsMicrophone(channel) == -1) {
LOG_RTCERR1(StopPlayingFileAsMicrophone, channel);
return false;
}
} else { // SEND_NOTHING
RTC_DCHECK(send == SEND_NOTHING);
if (engine()->voe()->base()->StopSend(channel) == -1) {

4
talk/media/webrtc/webrtcvoiceengine.h
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@ -47,10 +47,6 @@
#include "webrtc/common.h"
#include "webrtc/config.h"
namespace webrtc {
class VideoEngine;
}
namespace cricket {
class AudioDeviceModule;

1
talk/media/webrtc/webrtcvoiceengine_unittest.cc
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@ -68,7 +68,6 @@ class FakeVoEWrapper : public cricket::VoEWrapper {
engine, // base
engine, // codec
engine, // dtmf
engine, // file
engine, // hw
engine, // media
engine, // neteq