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Removed old code and default implementations.

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R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6588 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
asapersson@webrtc.org 2014-07-03 07:35:21 +00:00
parent 9c89e932c9
commit dfdaeb92d8
4 changed files with 5 additions and 75 deletions
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41
webrtc/video_engine/include/vie_base.h
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@ -43,29 +43,11 @@ class CpuOveruseObserver {
virtual ~CpuOveruseObserver() {} virtual ~CpuOveruseObserver() {}
}; };
// Limits on standard deviation for under/overuse.
#ifdef WEBRTC_ANDROID
const float kOveruseStdDevMs = 32.0f;
const float kNormalUseStdDevMs = 27.0f;
#elif WEBRTC_LINUX
const float kOveruseStdDevMs = 20.0f;
const float kNormalUseStdDevMs = 14.0f;
#elif WEBRTC_MAC
const float kOveruseStdDevMs = 27.0f;
const float kNormalUseStdDevMs = 21.0f;
#elif WEBRTC_WIN
const float kOveruseStdDevMs = 20.0f;
const float kNormalUseStdDevMs = 14.0f;
#else
const float kOveruseStdDevMs = 30.0f;
const float kNormalUseStdDevMs = 20.0f;
#endif
struct CpuOveruseOptions { struct CpuOveruseOptions {
CpuOveruseOptions() CpuOveruseOptions()
: enable_capture_jitter_method(true), : enable_capture_jitter_method(true),
low_capture_jitter_threshold_ms(kNormalUseStdDevMs), low_capture_jitter_threshold_ms(20.0f),
high_capture_jitter_threshold_ms(kOveruseStdDevMs), high_capture_jitter_threshold_ms(30.0f),
enable_encode_usage_method(false), enable_encode_usage_method(false),
low_encode_usage_threshold_percent(60), low_encode_usage_threshold_percent(60),
high_encode_usage_threshold_percent(90), high_encode_usage_threshold_percent(90),
@ -216,26 +198,11 @@ class WEBRTC_DLLEXPORT ViEBase {
CpuOveruseObserver* observer) = 0; CpuOveruseObserver* observer) = 0;
// Sets options for cpu overuse detector. // Sets options for cpu overuse detector.
// TODO(asapersson): Remove default implementation.
virtual int SetCpuOveruseOptions(int channel, virtual int SetCpuOveruseOptions(int channel,
const CpuOveruseOptions& options) { const CpuOveruseOptions& options) = 0;
return -1;
}
// Gets cpu overuse measures. // Gets cpu overuse measures.
// TODO(asapersson): Remove default implementation. virtual int GetCpuOveruseMetrics(int channel, CpuOveruseMetrics* metrics) = 0;
virtual int GetCpuOveruseMetrics(int channel,
CpuOveruseMetrics* metrics) {
return -1;
}
// TODO(asapersson): Remove this function when libjingle has been updated.
virtual int CpuOveruseMeasures(int channel,
int* capture_jitter_ms,
int* avg_encode_time_ms,
int* encode_usage_percent,
int* capture_queue_delay_ms_per_s) {
return -1;
}
// Specifies the VoiceEngine and VideoEngine channel pair to use for // Specifies the VoiceEngine and VideoEngine channel pair to use for
// audio/video synchronization. // audio/video synchronization.

3
webrtc/video_engine/include/vie_rtp_rtcp.h
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@ -261,8 +261,7 @@ class WEBRTC_DLLEXPORT ViERTP_RTCP {
// Enables/disables RTCP Receiver Reference Time Report Block extension/ // Enables/disables RTCP Receiver Reference Time Report Block extension/
// DLRR Report Block extension (RFC 3611). // DLRR Report Block extension (RFC 3611).
// TODO(asapersson): Remove default implementation. virtual int SetRtcpXrRrtrStatus(int video_channel, bool enable) = 0;
virtual int SetRtcpXrRrtrStatus(int video_channel, bool enable) { return -1; }
// Enables transmission smoothening, i.e. packets belonging to the same frame // Enables transmission smoothening, i.e. packets belonging to the same frame
// will be sent over a longer period of time instead of sending them // will be sent over a longer period of time instead of sending them

31
webrtc/video_engine/vie_base_impl.cc
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@ -121,37 +121,6 @@ int ViEBaseImpl::SetCpuOveruseOptions(int video_channel,
return -1; return -1;
} }
int ViEBaseImpl::CpuOveruseMeasures(int video_channel,
int* capture_jitter_ms,
int* avg_encode_time_ms,
int* encode_usage_percent,
int* capture_queue_delay_ms_per_s) {
ViEChannelManagerScoped cs(*(shared_data_.channel_manager()));
ViEChannel* vie_channel = cs.Channel(video_channel);
if (!vie_channel) {
shared_data_.SetLastError(kViEBaseInvalidChannelId);
return -1;
}
ViEEncoder* vie_encoder = cs.Encoder(video_channel);
assert(vie_encoder);
ViEInputManagerScoped is(*(shared_data_.input_manager()));
ViEFrameProviderBase* provider = is.FrameProvider(vie_encoder);
if (provider) {
ViECapturer* capturer = is.Capture(provider->Id());
if (capturer) {
CpuOveruseMetrics metrics;
capturer->GetCpuOveruseMetrics(&metrics);
*capture_jitter_ms = metrics.capture_jitter_ms;
*avg_encode_time_ms = metrics.avg_encode_time_ms;
*encode_usage_percent = metrics.encode_usage_percent;
*capture_queue_delay_ms_per_s = metrics.capture_queue_delay_ms_per_s;
return 0;
}
}
return -1;
}
int ViEBaseImpl::GetCpuOveruseMetrics(int video_channel, int ViEBaseImpl::GetCpuOveruseMetrics(int video_channel,
CpuOveruseMetrics* metrics) { CpuOveruseMetrics* metrics) {
ViEChannelManagerScoped cs(*(shared_data_.channel_manager())); ViEChannelManagerScoped cs(*(shared_data_.channel_manager()));

5
webrtc/video_engine/vie_base_impl.h
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@ -37,11 +37,6 @@ class ViEBaseImpl
const CpuOveruseOptions& options); const CpuOveruseOptions& options);
virtual int GetCpuOveruseMetrics(int channel, virtual int GetCpuOveruseMetrics(int channel,
CpuOveruseMetrics* metrics); CpuOveruseMetrics* metrics);
virtual int CpuOveruseMeasures(int channel,
int* capture_jitter_ms,
int* avg_encode_time_ms,
int* encode_usage_percent,
int* capture_queue_delay_ms_per_s);
virtual int CreateChannel(int& video_channel); // NOLINT virtual int CreateChannel(int& video_channel); // NOLINT
virtual int CreateChannel(int& video_channel, // NOLINT virtual int CreateChannel(int& video_channel, // NOLINT
const Config* config); const Config* config);