Adding a send side API for streaming
Review URL: https://webrtc-codereview.appspot.com/1070009 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3457 4adac7df-926f-26a2-2b94-8c16560cd09d
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@ -22,7 +22,7 @@
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#ifndef WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_RTP_RTCP_H_
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#define WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_RTP_RTCP_H_
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#include "common_types.h"
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#include "webrtc/common_types.h"
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namespace webrtc {
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@ -199,6 +199,12 @@ class WEBRTC_DLLEXPORT ViERTP_RTCP {
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const unsigned char payload_typeRED,
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const unsigned char payload_typeFEC) = 0;
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// Enables send side support for delayed video streaming (actual delay will
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// be exhibited on the receiver side).
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// Target delay should be set to zero for real-time mode.
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virtual int EnableSenderStreamingMode(int video_channel,
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int target_delay_ms) = 0;
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// This function enables RTCP key frame requests.
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virtual int SetKeyFrameRequestMethod(
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const int video_channel, const ViEKeyFrameRequestMethod method) = 0;
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@ -687,6 +687,21 @@ void ViEAutoTest::ViERtpRtcpAPITest()
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EXPECT_EQ(0, ViE.rtp_rtcp->SetTransmissionSmoothingStatus(
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tbChannel.videoChannel, false));
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// Streaming Mode.
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EXPECT_EQ(-1, ViE.rtp_rtcp->EnableSenderStreamingMode(
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invalid_channel_id, 0));
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int invalid_delay = -1;
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EXPECT_EQ(-1, ViE.rtp_rtcp->EnableSenderStreamingMode(
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tbChannel.videoChannel, invalid_delay));
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invalid_delay = 15000;
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EXPECT_EQ(-1, ViE.rtp_rtcp->EnableSenderStreamingMode(
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tbChannel.videoChannel, invalid_delay));
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EXPECT_EQ(0, ViE.rtp_rtcp->EnableSenderStreamingMode(
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tbChannel.videoChannel, 5000));
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// Real-time mode.
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EXPECT_EQ(0, ViE.rtp_rtcp->EnableSenderStreamingMode(
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tbChannel.videoChannel, 0));
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//***************************************************************
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// Testing finished. Tear down Video Engine
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//***************************************************************
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@ -35,6 +35,7 @@ namespace webrtc {
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const int kMaxDecodeWaitTimeMs = 50;
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const int kInvalidRtpExtensionId = 0;
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static const int kMaxTargetDelayMs = 10000;
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// Helper class receiving statistics callbacks.
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class ChannelStatsObserver : public StatsObserver {
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@ -102,7 +103,8 @@ ViEChannel::ViEChannel(WebRtc_Word32 channel_id,
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color_enhancement_(false),
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file_recorder_(channel_id),
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mtu_(0),
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sender_(sender) {
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sender_(sender),
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nack_history_size_sender_(kSendSidePacketHistorySize) {
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WEBRTC_TRACE(kTraceMemory, kTraceVideo, ViEId(engine_id, channel_id),
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"ViEChannel::ViEChannel(channel_id: %d, engine_id: %d)",
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channel_id, engine_id);
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@ -151,7 +153,7 @@ WebRtc_Word32 ViEChannel::Init() {
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"%s: RTP::SetRTCPStatus failure", __FUNCTION__);
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}
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if (paced_sender_) {
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if (rtp_rtcp_->SetStorePacketsStatus(true, kSendSidePacketHistorySize) !=
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if (rtp_rtcp_->SetStorePacketsStatus(true, nack_history_size_sender_) !=
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0) {
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WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
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"%s:SetStorePacketsStatus failure", __FUNCTION__);
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@ -295,10 +297,10 @@ WebRtc_Word32 ViEChannel::SetSendCodec(const VideoCodec& video_codec,
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"%s: RTP::SetRTCPStatus failure", __FUNCTION__);
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}
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if (nack_method != kNackOff) {
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rtp_rtcp->SetStorePacketsStatus(true, kSendSidePacketHistorySize);
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rtp_rtcp->SetStorePacketsStatus(true, nack_history_size_sender_);
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rtp_rtcp->SetNACKStatus(nack_method, kMaxPacketAgeToNack);
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} else if (paced_sender_) {
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rtp_rtcp->SetStorePacketsStatus(true, kSendSidePacketHistorySize);
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rtp_rtcp->SetStorePacketsStatus(true, nack_history_size_sender_);
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}
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if (fec_enabled) {
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rtp_rtcp->SetGenericFECStatus(fec_enabled, payload_type_red,
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@ -628,7 +630,7 @@ WebRtc_Word32 ViEChannel::ProcessNACKRequest(const bool enable) {
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}
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WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
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"%s: Using NACK method %d", __FUNCTION__, nackMethod);
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rtp_rtcp_->SetStorePacketsStatus(true, kSendSidePacketHistorySize);
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rtp_rtcp_->SetStorePacketsStatus(true, nack_history_size_sender_);
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vcm_.RegisterPacketRequestCallback(this);
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@ -639,7 +641,7 @@ WebRtc_Word32 ViEChannel::ProcessNACKRequest(const bool enable) {
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it++) {
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RtpRtcp* rtp_rtcp = *it;
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rtp_rtcp->SetNACKStatus(nackMethod, kMaxPacketAgeToNack);
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rtp_rtcp->SetStorePacketsStatus(true, kSendSidePacketHistorySize);
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rtp_rtcp->SetStorePacketsStatus(true, nack_history_size_sender_);
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}
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} else {
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CriticalSectionScoped cs(rtp_rtcp_cs_.get());
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@ -721,6 +723,45 @@ WebRtc_Word32 ViEChannel::SetHybridNACKFECStatus(
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return ProcessFECRequest(enable, payload_typeRED, payload_typeFEC);
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}
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int ViEChannel::EnableSenderStreamingMode(int target_delay_ms) {
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if ((target_delay_ms < 0) || (target_delay_ms > kMaxTargetDelayMs)) {
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WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
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"%s: Target streaming delay out of bounds: %d", __FUNCTION__,
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target_delay_ms);
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return -1;
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}
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if (target_delay_ms == 0) {
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// Real-time mode.
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nack_history_size_sender_ = kSendSidePacketHistorySize;
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vcm_.EnableFrameDropper(true);
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} else {
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// The max size of the nack list should be large enough to accommodate the
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// the number of packets(frames) resulting from the increased delay.
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// Roughly estimating for ~15 packets per frame @ 30fps.
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nack_history_size_sender_ = target_delay_ms * 15 * 30 / 1000;
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// Don't allow a number lower than the default value.
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if (nack_history_size_sender_ < kSendSidePacketHistorySize) {
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nack_history_size_sender_ = kSendSidePacketHistorySize;
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}
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// Disable external VCM frame-dropper. In streaming mode, we are more
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// flexible with rate control constraints.
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vcm_.EnableFrameDropper(false);
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}
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// Setting nack_history_size_.
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// First disabling (forcing free) and then resetting to desired value.
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if (rtp_rtcp_->SetStorePacketsStatus(false, 0) != 0) {
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WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
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"%s:SetStorePacketsStatus failure", __FUNCTION__);
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return -1;
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}
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if (rtp_rtcp_->SetStorePacketsStatus(true, nack_history_size_sender_) != 0) {
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WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
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"%s:SetStorePacketsStatus failure", __FUNCTION__);
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return -1;
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}
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return 0;
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}
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WebRtc_Word32 ViEChannel::SetKeyFrameRequestMethod(
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const KeyFrameRequestMethod method) {
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WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
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@ -116,6 +116,7 @@ class ViEChannel
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WebRtc_Word32 SetHybridNACKFECStatus(const bool enable,
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const unsigned char payload_typeRED,
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const unsigned char payload_typeFEC);
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int EnableSenderStreamingMode(int target_delay_ms);
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WebRtc_Word32 SetKeyFrameRequestMethod(const KeyFrameRequestMethod method);
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bool EnableRemb(bool enable);
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int SetSendTimestampOffsetStatus(bool enable, int id);
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@ -422,6 +423,8 @@ class ViEChannel
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// User set MTU, -1 if not set.
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uint16_t mtu_;
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const bool sender_;
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int nack_history_size_sender_;
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};
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} // namespace webrtc
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@ -553,6 +553,33 @@ int ViERTP_RTCPImpl::SetHybridNACKFECStatus(
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return 0;
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}
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int ViERTP_RTCPImpl::EnableSenderStreamingMode(int video_channel,
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int target_delay_ms) {
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WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
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ViEId(shared_data_->instance_id(), video_channel),
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"%s(channel: %d, target_delay: %d)",
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__FUNCTION__, video_channel, target_delay_ms);
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ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
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ViEChannel* vie_channel = cs.Channel(video_channel);
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if (!vie_channel) {
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WEBRTC_TRACE(kTraceError, kTraceVideo,
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ViEId(shared_data_->instance_id(), video_channel),
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"%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
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shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
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return -1;
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}
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// Update the channel's streaming mode settings.
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if (vie_channel->EnableSenderStreamingMode(target_delay_ms) != 0) {
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WEBRTC_TRACE(kTraceError, kTraceVideo,
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ViEId(shared_data_->instance_id(), video_channel),
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"%s: failed for channel %d", __FUNCTION__, video_channel);
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shared_data_->SetLastError(kViERtpRtcpUnknownError);
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return -1;
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}
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return 0;
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}
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int ViERTP_RTCPImpl::SetKeyFrameRequestMethod(
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const int video_channel,
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const ViEKeyFrameRequestMethod method) {
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@ -64,6 +64,8 @@ class ViERTP_RTCPImpl
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virtual int SetHybridNACKFECStatus(const int video_channel, const bool enable,
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const unsigned char payload_typeRED,
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const unsigned char payload_typeFEC);
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virtual int EnableSenderStreamingMode(int video_channel,
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int target_delay_ms);
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virtual int SetKeyFrameRequestMethod(const int video_channel,
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const ViEKeyFrameRequestMethod method);
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virtual int SetTMMBRStatus(const int video_channel, const bool enable);
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