Move min transmit bitrate to VideoEncoderConfig.
min_transmit_bitrate_bps needs to be reconfigurable during a call (since
this is currently set only for screensharing through libjingle and can't
be set once and for all for the entire Call.
R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1667
Review URL: https://webrtc-codereview.appspot.com/28779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7518 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/config.cc b/webrtc/config.cc
index e0324b9..3d205f1 100644
--- a/webrtc/config.cc
+++ b/webrtc/config.cc
@@ -50,4 +50,32 @@
ss << '}';
return ss.str();
}
+
+std::string VideoEncoderConfig::ToString() const {
+ std::stringstream ss;
+
+ ss << "{streams: {";
+ for (size_t i = 0; i < streams.size(); ++i) {
+ ss << streams[i].ToString();
+ if (i != streams.size() - 1)
+ ss << "}, {";
+ }
+ ss << '}';
+ ss << ", content_type: ";
+ switch (content_type) {
+ case kRealtimeVideo:
+ ss << "kRealtimeVideo";
+ break;
+ case kScreenshare:
+ ss << "kScreenshare";
+ break;
+ }
+ ss << ", encoder_specific_settings: ";
+ ss << (encoder_specific_settings != NULL ? "(ptr)" : "NULL");
+
+ ss << ", min_transmit_bitrate_bps: " << min_transmit_bitrate_bps;
+ ss << '}';
+ return ss.str();
+}
+
} // namespace webrtc
diff --git a/webrtc/config.h b/webrtc/config.h
index 6f3fb1d..af78d94 100644
--- a/webrtc/config.h
+++ b/webrtc/config.h
@@ -115,11 +115,21 @@
};
VideoEncoderConfig()
- : content_type(kRealtimeVideo), encoder_specific_settings(NULL) {}
+ : content_type(kRealtimeVideo),
+ encoder_specific_settings(NULL),
+ min_transmit_bitrate_bps(0) {}
+
+ std::string ToString() const;
std::vector<VideoStream> streams;
ContentType content_type;
void* encoder_specific_settings;
+
+ // Padding will be used up to this bitrate regardless of the bitrate produced
+ // by the encoder. Padding above what's actually produced by the encoder helps
+ // maintaining a higher bitrate estimate. Padding will however not be sent
+ // unless the estimated bandwidth indicates that the link can handle it.
+ int min_transmit_bitrate_bps;
};
} // namespace webrtc
diff --git a/webrtc/video/call_perf_tests.cc b/webrtc/video/call_perf_tests.cc
index 1087239..9776fb7 100644
--- a/webrtc/video/call_perf_tests.cc
+++ b/webrtc/video/call_perf_tests.cc
@@ -543,9 +543,9 @@
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) OVERRIDE {
if (pad_to_min_bitrate_) {
- send_config->rtp.min_transmit_bitrate_bps = kMinTransmitBitrateBps;
+ encoder_config->min_transmit_bitrate_bps = kMinTransmitBitrateBps;
} else {
- assert(send_config->rtp.min_transmit_bitrate_bps == 0);
+ assert(encoder_config->min_transmit_bitrate_bps == 0);
}
}
diff --git a/webrtc/video/end_to_end_tests.cc b/webrtc/video/end_to_end_tests.cc
index ee36dd5..3d7d3fa 100644
--- a/webrtc/video/end_to_end_tests.cc
+++ b/webrtc/video/end_to_end_tests.cc
@@ -1653,15 +1653,16 @@
encoder_config->streams[i].target_bitrate_bps = 15000;
encoder_config->streams[i].max_bitrate_bps = 20000;
}
- // Significantly higher than max bitrates for all video streams -> forcing
- // padding to trigger redundant padding on all RTX SSRCs.
- send_config->rtp.min_transmit_bitrate_bps = 100000;
send_config->rtp.rtx.payload_type = kSendRtxPayloadType;
send_config->rtp.rtx.pad_with_redundant_payloads = true;
for (size_t i = 0; i < kNumSsrcs; ++i)
send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[i]);
+
+ // Significantly higher than max bitrates for all video streams -> forcing
+ // padding to trigger redundant padding on all RTX SSRCs.
+ encoder_config->min_transmit_bitrate_bps = 100000;
}
virtual void PerformTest() OVERRIDE {
diff --git a/webrtc/video/video_send_stream.cc b/webrtc/video/video_send_stream.cc
index 635c37f..7657250 100644
--- a/webrtc/video/video_send_stream.cc
+++ b/webrtc/video/video_send_stream.cc
@@ -66,8 +66,6 @@
ss << '}';
ss << ", max_packet_size: " << max_packet_size;
- if (min_transmit_bitrate_bps != 0)
- ss << ", min_transmit_bitrate_bps: " << min_transmit_bitrate_bps;
ss << ", extensions: {";
for (size_t i = 0; i < extensions.size(); ++i) {
@@ -137,10 +135,6 @@
assert(config_.rtp.ssrcs.size() > 0);
- assert(config_.rtp.min_transmit_bitrate_bps >= 0);
- rtp_rtcp_->SetMinTransmitBitrate(channel_,
- config_.rtp.min_transmit_bitrate_bps / 1000);
-
for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) {
const std::string& extension = config_.rtp.extensions[i].name;
int id = config_.rtp.extensions[i].id;
@@ -298,6 +292,7 @@
bool VideoSendStream::ReconfigureVideoEncoder(
const VideoEncoderConfig& config) {
+ LOG(LS_INFO) << "(Re)configureVideoEncoder: " << config.ToString();
const std::vector<VideoStream>& streams = config.streams;
assert(!streams.empty());
assert(config_.rtp.ssrcs.size() >= streams.size());
@@ -407,6 +402,10 @@
if (codec_->SetSendCodec(channel_, video_codec) != 0)
return false;
+ assert(config.min_transmit_bitrate_bps >= 0);
+ rtp_rtcp_->SetMinTransmitBitrate(channel_,
+ config.min_transmit_bitrate_bps / 1000);
+
use_default_bitrate_ = false;
return true;
}
diff --git a/webrtc/video/video_send_stream_tests.cc b/webrtc/video/video_send_stream_tests.cc
index 6c2fa39..79b1cd5 100644
--- a/webrtc/video/video_send_stream_tests.cc
+++ b/webrtc/video/video_send_stream_tests.cc
@@ -1075,7 +1075,7 @@
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) OVERRIDE {
- send_config->rtp.min_transmit_bitrate_bps = kMinTransmitBitrateBps;
+ encoder_config->min_transmit_bitrate_bps = kMinTransmitBitrateBps;
}
virtual void PerformTest() OVERRIDE {
diff --git a/webrtc/video_send_stream.h b/webrtc/video_send_stream.h
index dd2bec1..aa5033a 100644
--- a/webrtc/video_send_stream.h
+++ b/webrtc/video_send_stream.h
@@ -60,6 +60,7 @@
struct EncoderSettings {
EncoderSettings() : payload_type(-1), encoder(NULL) {}
+
std::string ToString() const;
std::string payload_name;
@@ -72,9 +73,7 @@
static const size_t kDefaultMaxPacketSize = 1500 - 40; // TCP over IPv4.
struct Rtp {
- Rtp()
- : max_packet_size(kDefaultMaxPacketSize),
- min_transmit_bitrate_bps(0) {}
+ Rtp() : max_packet_size(kDefaultMaxPacketSize) {}
std::string ToString() const;
std::vector<uint32_t> ssrcs;
@@ -82,11 +81,6 @@
// Max RTP packet size delivered to send transport from VideoEngine.
size_t max_packet_size;
- // Padding will be used up to this bitrate regardless of the bitrate
- // produced by the encoder. Padding above what's actually produced by the
- // encoder helps maintaining a higher bitrate estimate.
- int min_transmit_bitrate_bps;
-
// RTP header extensions to use for this send stream.
std::vector<RtpExtension> extensions;
