commit abbe84172148f54448b5288f979fd9c320ffb0f3
author Jonas Olsson <jonasolsson@webrtc.org> Tue Apr 03 13:40:05 2018 +0200
committer Commit Bot <commit-bot@chromium.org> Tue Apr 03 12:51:00 2018 +0000
tree 57a21592e846aa06771259fcb7c73ac5ef93a5e4
parent fe9d8178df890744978f75e050f5c87ad7697785

This CL removes all usages of our custom ostream << overloads.

This prepares us for removing them altogether.

Bug: webrtc:8982
Change-Id: I66002cc8d4bf0e07925766d568d2498422f0f38e
Reviewed-on: https://webrtc-review.googlesource.com/64142
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Henrik Grunell <henrikg@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22707}

modules/audio_coding/BUILD.gn

diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index ea69666..1fb13e5 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -148,6 +148,7 @@
            ":audio_coding_module_typedefs",
            ":neteq",
            ":rent_a_codec",
+           "../../rtc_base:audio_format_to_string",
            "../../rtc_base:rtc_base_approved",
            "../../api:optional",
            "../../logging:rtc_event_log_api",
@@ -1087,6 +1088,7 @@
     "../../api/audio_codecs:audio_codecs_api",
     "../../common_audio",
     "../../common_audio:common_audio_c",
+    "../../rtc_base:audio_format_to_string",
     "../../rtc_base:checks",
     "../../rtc_base:gtest_prod",
     "../../rtc_base:rtc_base_approved",

modules/audio_coding/acm2/acm_receiver.cc

diff --git a/modules/audio_coding/acm2/acm_receiver.cc b/modules/audio_coding/acm2/acm_receiver.cc
index 0d5dcae..4d0209b 100644
--- a/modules/audio_coding/acm2/acm_receiver.cc
+++ b/modules/audio_coding/acm2/acm_receiver.cc
@@ -26,6 +26,7 @@
 #include "rtc_base/format_macros.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/strings/audio_format_to_string.h"
 #include "system_wrappers/include/clock.h"
 
 namespace webrtc {
@@ -260,7 +261,8 @@
       neteq_->RegisterPayloadType(rtp_payload_type, audio_format);
   if (!success) {
     RTC_LOG(LERROR) << "AcmReceiver::AddCodec failed for payload type "
-                    << rtp_payload_type << ", decoder format " << audio_format;
+                    << rtp_payload_type << ", decoder format "
+                    << rtc::ToString(audio_format);
   }
   return success;
 }

modules/audio_coding/neteq/decoder_database.cc

diff --git a/modules/audio_coding/neteq/decoder_database.cc b/modules/audio_coding/neteq/decoder_database.cc
index 7244c8d..40b0dc7 100644
--- a/modules/audio_coding/neteq/decoder_database.cc
+++ b/modules/audio_coding/neteq/decoder_database.cc
@@ -15,6 +15,7 @@
 #include "api/audio_codecs/audio_decoder.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
+#include "rtc_base/strings/audio_format_to_string.h"
 
 namespace webrtc {
 
@@ -95,7 +96,7 @@
     RTC_DCHECK(factory_);
     decoder_ = factory_->MakeAudioDecoder(audio_format_, codec_pair_id_);
   }
-  RTC_DCHECK(decoder_) << "Failed to create: " << audio_format_;
+  RTC_DCHECK(decoder_) << "Failed to create: " << rtc::ToString(audio_format_);
   return decoder_.get();
 }
 

modules/audio_coding/neteq/neteq_impl.cc

diff --git a/modules/audio_coding/neteq/neteq_impl.cc b/modules/audio_coding/neteq/neteq_impl.cc
index 78015d9..4e780c3 100644
--- a/modules/audio_coding/neteq/neteq_impl.cc
+++ b/modules/audio_coding/neteq/neteq_impl.cc
@@ -46,6 +46,7 @@
 #include "rtc_base/logging.h"
 #include "rtc_base/numerics/safe_conversions.h"
 #include "rtc_base/sanitizer.h"
+#include "rtc_base/strings/audio_format_to_string.h"
 #include "rtc_base/system/fallthrough.h"
 #include "rtc_base/trace_event.h"
 #include "system_wrappers/include/field_trial.h"
@@ -262,7 +263,8 @@
 bool NetEqImpl::RegisterPayloadType(int rtp_payload_type,
                                     const SdpAudioFormat& audio_format) {
   RTC_LOG(LS_VERBOSE) << "NetEqImpl::RegisterPayloadType: payload type "
-                      << rtp_payload_type << ", codec " << audio_format;
+                      << rtp_payload_type << ", codec "
+                      << rtc::ToString(audio_format);
   rtc::CritScope lock(&crit_sect_);
   return decoder_database_->RegisterPayload(rtp_payload_type, audio_format) ==
          DecoderDatabase::kOK;