From 83377270dc6044e277ecb7221025438d0dd1939d Mon Sep 17 00:00:00 2001
From: danilchap <danilchap@webrtc.org>
Date: Tue, 25 Jul 2017 07:46:54 -0700
Subject: [PATCH] Remove deprecated RtpRtcp::SetAudioPacketSize was deprecated
 in https://codereview.webrtc.org/2545753002

BUG=webrtc:5806

Review-Url: https://codereview.webrtc.org/2986793002
Cr-Commit-Position: refs/heads/master@{#19134}
---
 webrtc/modules/rtp_rtcp/include/rtp_rtcp.h      | 6 ------
 webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h   | 1 -
 webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc | 5 -----
 webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h  | 4 ----
 webrtc/modules/rtp_rtcp/source/rtp_sender.cc    | 7 -------
 webrtc/modules/rtp_rtcp/source/rtp_sender.h     | 4 ----
 6 files changed, 27 deletions(-)

diff --git a/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h b/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
index 56329115eb..c2a0256432 100644
--- a/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
+++ b/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
@@ -409,12 +409,6 @@ class RtpRtcp : public Module {
   // Audio
   // **************************************************************************
 
-  // This function is deprecated. It was previously used to determine when it
-  // was time to send a DTMF packet in silence (CNG).
-  // Returns -1 on failure else 0.
-  RTC_DEPRECATED virtual int32_t SetAudioPacketSize(
-      uint16_t packet_size_samples) = 0;
-
   // Sends a TelephoneEvent tone using RFC 2833 (4733).
   // Returns -1 on failure else 0.
   virtual int32_t SendTelephoneEventOutband(uint8_t key,
diff --git a/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h b/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
index 40a6c76fd5..54d84d595d 100644
--- a/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
+++ b/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
@@ -174,7 +174,6 @@ class MockRtpRtcp : public RtpRtcp {
   MOCK_METHOD1(RegisterRtcpStatisticsCallback, void(RtcpStatisticsCallback*));
   MOCK_METHOD0(GetRtcpStatisticsCallback, RtcpStatisticsCallback*());
   MOCK_METHOD1(SendFeedbackPacket, bool(const rtcp::TransportFeedback& packet));
-  MOCK_METHOD1(SetAudioPacketSize, int32_t(uint16_t packet_size_samples));
   MOCK_METHOD3(SendTelephoneEventOutband,
                int32_t(uint8_t key, uint16_t time_ms, uint8_t level));
   MOCK_METHOD1(SetSendREDPayloadType, int32_t(int8_t payload_type));
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index 66bbc242b7..c981ad7744 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -772,11 +772,6 @@ int32_t ModuleRtpRtcpImpl::SendTelephoneEventOutband(
   return rtp_sender_->SendTelephoneEvent(key, time_ms, level);
 }
 
-int32_t ModuleRtpRtcpImpl::SetAudioPacketSize(
-    const uint16_t packet_size_samples) {
-  return audio_ ? 0 : -1;
-}
-
 int32_t ModuleRtpRtcpImpl::SetAudioLevel(
     const uint8_t level_d_bov) {
   return rtp_sender_->SetAudioLevel(level_d_bov);
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index 4ff2cf41c1..7a28be536d 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -250,10 +250,6 @@ class ModuleRtpRtcpImpl : public RtpRtcp, public RTCPReceiver::ModuleRtpRtcp {
 
   // Audio part.
 
-  // This function is deprecated. It was previously used to determine when it
-  // was time to send a DTMF packet in silence (CNG).
-  int32_t SetAudioPacketSize(uint16_t packet_size_samples) override;
-
   // Send a TelephoneEvent tone using RFC 2833 (4733).
   int32_t SendTelephoneEventOutband(uint8_t key,
                                     uint16_t time_ms,
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index dc7b20e9db..024e028393 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -1132,13 +1132,6 @@ int32_t RTPSender::SendTelephoneEvent(uint8_t key,
   return audio_->SendTelephoneEvent(key, time_ms, level);
 }
 
-int32_t RTPSender::SetAudioPacketSize(uint16_t packet_size_samples) {
-  if (!audio_configured_) {
-    return -1;
-  }
-  return 0;
-}
-
 int32_t RTPSender::SetAudioLevel(uint8_t level_d_bov) {
   return audio_->SetAudioLevel(level_d_bov);
 }
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.h b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
index d4aca38c90..9ab3e33f8a 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
@@ -176,10 +176,6 @@ class RTPSender {
   // Send a DTMF tone using RFC 2833 (4733).
   int32_t SendTelephoneEvent(uint8_t key, uint16_t time_ms, uint8_t level);
 
-  // This function is deprecated. It was previously used to determine when it
-  // was time to send a DTMF packet in silence (CNG).
-  RTC_DEPRECATED int32_t SetAudioPacketSize(uint16_t packet_size_samples);
-
   // Store the audio level in d_bov for
   // header-extension-for-audio-level-indication.
   int32_t SetAudioLevel(uint8_t level_d_bov);