Clean up abs-send-time for audio.

webrtc/audio/audio_send_stream_unittest.cc

Index: webrtc/audio/audio_send_stream_unittest.cc
diff --git a/webrtc/audio/audio_send_stream_unittest.cc b/webrtc/audio/audio_send_stream_unittest.cc
index a2832ded00cb330d86ffae8805955326cb1033a2..04ac8638a23988852893ef7842e5b966add8e77d 100644
--- a/webrtc/audio/audio_send_stream_unittest.cc
+++ b/webrtc/audio/audio_send_stream_unittest.cc
@@ -35,7 +35,6 @@ const int kChannelId = 1;
 const uint32_t kSsrc = 1234;
 const char* kCName = "foo_name";
 const int kAudioLevelId = 2;
-const int kAbsSendTimeId = 3;
 const int kTransportSequenceNumberId = 4;
 const int kEchoDelayMedian = 254;
 const int kEchoDelayStdDev = -3;
@@ -88,8 +87,6 @@ struct ConfigHelper {
           EXPECT_CALL(*channel_proxy_, SetRTCP_CNAME(StrEq(kCName))).Times(1);
           EXPECT_CALL(*channel_proxy_, SetNACKStatus(true, 10)).Times(1);
           EXPECT_CALL(*channel_proxy_,
-              SetSendAbsoluteSenderTimeStatus(true, kAbsSendTimeId)).Times(1);
-          EXPECT_CALL(*channel_proxy_,
               SetSendAudioLevelIndicationStatus(true, kAudioLevelId)).Times(1);
           EXPECT_CALL(*channel_proxy_, EnableSendTransportSequenceNumber(
                                            kTransportSequenceNumberId))
@@ -119,8 +116,6 @@ struct ConfigHelper {
     stream_config_.rtp.c_name = kCName;
     stream_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId));
-    stream_config_.rtp.extensions.push_back(
-        RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeId));

[the sun, 2016/10/27 16:09:17]

Same here - add audiolevel, or some other extension, to test ToString().

     stream_config_.rtp.extensions.push_back(RtpExtension(
         RtpExtension::kTransportSequenceNumberUri, kTransportSequenceNumberId));
     // Use ISAC as default codec so as to prevent unnecessary |voice_engine_|
@@ -215,8 +210,6 @@ struct ConfigHelper {
 TEST(AudioSendStreamTest, ConfigToString) {
   AudioSendStream::Config config(nullptr);
   config.rtp.ssrc = kSsrc;
-  config.rtp.extensions.push_back(
-      RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeId));
   config.rtp.c_name = kCName;
   config.voe_channel_id = kChannelId;
   config.min_bitrate_kbps = 12;
@@ -230,15 +223,13 @@ TEST(AudioSendStreamTest, ConfigToString) {
   config.send_codec_spec.cng_plfreq = 56;
   config.send_codec_spec.codec_inst = kIsacCodec;
   EXPECT_EQ(
-      "{rtp: {ssrc: 1234, extensions: [{uri: "
-      "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}], "
-      "nack: {rtp_history_ms: 0}, c_name: foo_name}, send_transport: nullptr, "
-      "voe_channel_id: 1, min_bitrate_kbps: 12, max_bitrate_kbps: 34, "
-      "send_codec_spec: {nack_enabled: true, transport_cc_enabled: false, "
-      "enable_codec_fec: true, enable_opus_dtx: false, opus_max_playback_rate: "
-      "32000, cng_payload_type: 42, cng_plfreq: 56, codec_inst: {pltype: "
-      "103, plname: \"isac\", plfreq: 16000, pacsize: 320, channels: 1, rate: "
-      "32000}}}",
+      "{rtp: {ssrc: 1234, extensions: [], nack: {rtp_history_ms: 0}, c_name: "
+      "foo_name}, send_transport: nullptr, voe_channel_id: 1, "
+      "min_bitrate_kbps: 12, max_bitrate_kbps: 34, send_codec_spec: "
+      "{nack_enabled: true, transport_cc_enabled: false, enable_codec_fec: "
+      "true, enable_opus_dtx: false, opus_max_playback_rate: 32000, "
+      "cng_payload_type: 42, cng_plfreq: 56, codec_inst: {pltype: 103, plname: "
+      "\"isac\", plfreq: 16000, pacsize: 320, channels: 1, rate: 32000}}}",
       config.ToString());
 }