Add ToString() to VideoSendStream::Config.

Adds ToString() to subsequent parts as well as a common.gyp to define
ToString() methods for config.h. VideoStream is also moved to config.h.

BUG=3171
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11329004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6170 4adac7df-926f-26a2-2b94-8c16560cd09d

7 files changed, 207 insertions, 27 deletions

diff --git a/common.gyp b/common.gyp
new file mode 100644
index 00000000..b6b6354a
--- /dev/null
+++ b/common.gyp
@@ -0,0 +1,20 @@
+# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+{
+  'includes': ['build/common.gypi'],
+  'targets': [
+    {
+      'target_name': 'webrtc_common',
+      'type': 'static_library',
+      'sources': [
+        'config.h',
+        'config.cc',
+      ],
+    },
+  ],
+}
diff --git a/common_types.h b/common_types.h
index 2d93102d..6892a83f 100644
--- a/common_types.h
+++ b/common_types.h
@@ -13,6 +13,8 @@
 
 #include <stddef.h>
 #include <string.h>
+
+#include <string>
 #include <vector>
 
 #include "webrtc/typedefs.h"
@@ -781,30 +783,6 @@ struct RTPHeader {
   RTPHeaderExtension extension;
 };
 
-struct VideoStream {
-  VideoStream()
-      : width(0),
-        height(0),
-        max_framerate(-1),
-        min_bitrate_bps(-1),
-        target_bitrate_bps(-1),
-        max_bitrate_bps(-1),
-        max_qp(-1) {}
-
-  size_t width;
-  size_t height;
-  int max_framerate;
-
-  int min_bitrate_bps;
-  int target_bitrate_bps;
-  int max_bitrate_bps;
-
-  int max_qp;
-
-  // Bitrate thresholds for enabling additional temporal layers.
-  std::vector<int> temporal_layers;
-};
-
 }  // namespace webrtc
 
 #endif  // WEBRTC_COMMON_TYPES_H_
diff --git a/config.cc b/config.cc
new file mode 100644
index 00000000..e0324b9e
--- /dev/null
+++ b/config.cc
@@ -0,0 +1,53 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "webrtc/config.h"
+
+#include <sstream>
+#include <string>
+
+namespace webrtc {
+std::string FecConfig::ToString() const {
+  std::stringstream ss;
+  ss << "{ulpfec_payload_type: " << ulpfec_payload_type;
+  ss << ", red_payload_type: " << red_payload_type;
+  ss << '}';
+  return ss.str();
+}
+
+std::string RtpExtension::ToString() const {
+  std::stringstream ss;
+  ss << "{name: " << name;
+  ss << ", id: " << id;
+  ss << '}';
+  return ss.str();
+}
+
+std::string VideoStream::ToString() const {
+  std::stringstream ss;
+  ss << "{width: " << width;
+  ss << ", height: " << height;
+  ss << ", max_framerate: " << max_framerate;
+  ss << ", min_bitrate_bps:" << min_bitrate_bps;
+  ss << ", target_bitrate_bps:" << target_bitrate_bps;
+  ss << ", max_bitrate_bps:" << max_bitrate_bps;
+  ss << ", max_qp: " << max_qp;
+
+  ss << ", temporal_layers: {";
+  for (size_t i = 0; i < temporal_layers.size(); ++i) {
+    ss << temporal_layers[i];
+    if (i != temporal_layers.size() - 1)
+      ss << "}, {";
+  }
+  ss << '}';
+
+  ss << '}';
+  return ss.str();
+}
+}  // namespace webrtc
diff --git a/config.h b/config.h
index 105d9a54..7717bbad 100644
--- a/config.h
+++ b/config.h
@@ -57,6 +57,7 @@ struct NackConfig {
 // payload types to '-1' to disable.
 struct FecConfig {
   FecConfig() : ulpfec_payload_type(-1), red_payload_type(-1) {}
+  std::string ToString() const;
   // Payload type used for ULPFEC packets.
   int ulpfec_payload_type;
 
@@ -66,13 +67,40 @@ struct FecConfig {
 
 // RTP header extension to use for the video stream, see RFC 5285.
 struct RtpExtension {
-  static const char* kTOffset;
-  static const char* kAbsSendTime;
   RtpExtension(const char* name, int id) : name(name), id(id) {}
+  std::string ToString() const;
   // TODO(mflodman) Add API to query supported extensions.
+  static const char* kTOffset;
+  static const char* kAbsSendTime;
   std::string name;
   int id;
 };
+
+struct VideoStream {
+  VideoStream()
+      : width(0),
+        height(0),
+        max_framerate(-1),
+        min_bitrate_bps(-1),
+        target_bitrate_bps(-1),
+        max_bitrate_bps(-1),
+        max_qp(-1) {}
+  std::string ToString() const;
+
+  size_t width;
+  size_t height;
+  int max_framerate;
+
+  int min_bitrate_bps;
+  int target_bitrate_bps;
+  int max_bitrate_bps;
+
+  int max_qp;
+
+  // Bitrate thresholds for enabling additional temporal layers.
+  std::vector<int> temporal_layers;
+};
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_CONFIG_H_
diff --git a/video/video_send_stream.cc b/video/video_send_stream.cc
index e6e683a1..c1de2745 100644
--- a/video/video_send_stream.cc
+++ b/video/video_send_stream.cc
@@ -10,6 +10,7 @@
 
 #include "webrtc/video/video_send_stream.h"
 
+#include <sstream>
 #include <string>
 #include <vector>
 
@@ -25,6 +26,98 @@
 #include "webrtc/video_send_stream.h"
 
 namespace webrtc {
+std::string
+VideoSendStream::VideoSendStream::Config::EncoderSettings::ToString() const {
+  std::stringstream ss;
+  ss << "{payload_name: " << payload_name;
+  ss << ", payload_type: " << payload_type;
+  if (encoder != NULL)
+    ss << ", (encoder)";
+  if (encoder_settings != NULL)
+    ss << ", (encoder_settings)";
+
+  ss << ", streams: {";
+  for (size_t i = 0; i < streams.size(); ++i) {
+    ss << streams[i].ToString();
+    if (i != streams.size() - 1)
+      ss << "}, {";
+  }
+  ss << '}';
+
+  ss << '}';
+  return ss.str();
+}
+
+std::string VideoSendStream::VideoSendStream::Config::Rtp::Rtx::ToString()
+    const {
+  std::stringstream ss;
+  ss << "{ssrcs: {";
+  for (size_t i = 0; i < ssrcs.size(); ++i) {
+    ss << ssrcs[i];
+    if (i != ssrcs.size() - 1)
+      ss << "}, {";
+  }
+  ss << '}';
+
+  ss << ", payload_type: " << payload_type;
+  ss << '}';
+  return ss.str();
+}
+
+std::string VideoSendStream::VideoSendStream::Config::Rtp::ToString() const {
+  std::stringstream ss;
+  ss << "{ssrcs: {";
+  for (size_t i = 0; i < ssrcs.size(); ++i) {
+    ss << ssrcs[i];
+    if (i != ssrcs.size() - 1)
+      ss << "}, {";
+  }
+  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) {
+    ss << extensions[i].ToString();
+    if (i != extensions.size() - 1)
+      ss << "}, {";
+  }
+  ss << '}';
+
+  if (nack.rtp_history_ms != 0)
+    ss << ", nack.rtp_history_ms: " << nack.rtp_history_ms;
+  if (fec.ulpfec_payload_type != -1 || fec.red_payload_type != -1)
+    ss << ", fec: " << fec.ToString();
+  if (rtx.payload_type != 0 || !rtx.ssrcs.empty())
+    ss << ", rtx: " << rtx.ToString();
+  if (c_name != "")
+    ss << ", c_name: " << c_name;
+  ss << '}';
+  return ss.str();
+}
+
+std::string VideoSendStream::VideoSendStream::Config::ToString() const {
+  std::stringstream ss;
+  ss << "{encoder_settings: " << encoder_settings.ToString();
+  ss << ", rtp: " << rtp.ToString();
+  if (pre_encode_callback != NULL)
+    ss << ", (pre_encode_callback)";
+  if (post_encode_callback != NULL)
+    ss << ", (post_encode_callback)";
+  if (local_renderer != NULL) {
+    ss << ", (local_renderer, render_delay_ms: " << render_delay_ms << ")";
+  }
+  if (target_delay_ms > 0)
+    ss << ", target_delay_ms: " << target_delay_ms;
+  if (pacing)
+    ss << ", pacing: on";
+  if (suspend_below_min_bitrate)
+    ss << ", suspend_below_min_bitrate: on";
+  ss << '}';
+  return ss.str();
+}
 namespace internal {
 
 VideoSendStream::VideoSendStream(newapi::Transport* transport,
diff --git a/video_send_stream.h b/video_send_stream.h
index aa027b0e..1a94121d 100644
--- a/video_send_stream.h
+++ b/video_send_stream.h
@@ -64,9 +64,13 @@ class VideoSendStream {
           target_delay_ms(0),
           pacing(false),
           suspend_below_min_bitrate(false) {}
+    std::string ToString() const;
+
     struct EncoderSettings {
       EncoderSettings()
           : payload_type(-1), encoder(NULL), encoder_settings(NULL) {}
+      std::string ToString() const;
+
       std::string payload_name;
       int payload_type;
 
@@ -87,6 +91,7 @@ class VideoSendStream {
       Rtp()
           : max_packet_size(kDefaultMaxPacketSize),
             min_transmit_bitrate_bps(0) {}
+      std::string ToString() const;
 
       std::vector<uint32_t> ssrcs;
 
@@ -111,6 +116,7 @@ class VideoSendStream {
       // details.
       struct Rtx {
         Rtx() : payload_type(0) {}
+        std::string ToString() const;
         // SSRCs to use for the RTX streams.
         std::vector<uint32_t> ssrcs;
 
@@ -136,7 +142,7 @@ class VideoSendStream {
 
     // Expected delay needed by the renderer, i.e. the frame will be delivered
     // this many milliseconds, if possible, earlier than expected render time.
-    // Only valid if |renderer| is set.
+    // Only valid if |local_renderer| is set.
     int render_delay_ms;
 
     // Target delay in milliseconds. A positive value indicates this stream is
diff --git a/webrtc.gyp b/webrtc.gyp
index f376c064..d50552d9 100644
--- a/webrtc.gyp
+++ b/webrtc.gyp
@@ -20,6 +20,7 @@
   'variables': {
     'webrtc_all_dependencies': [
       'base/base.gyp:*',
+      'common.gyp:*',
       'common_audio/common_audio.gyp:*',
       'common_video/common_video.gyp:*',
       'modules/modules.gyp:*',
@@ -75,6 +76,7 @@
         '<@(webrtc_video_sources)',
       ],
       'dependencies': [
+        'common.gyp:*',
         '<@(webrtc_video_dependencies)',
       ],
     },