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Remove deprecated RtpRtcp::SetAudioPacketSize

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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}
This commit is contained in:
danilchap 2017-07-25 07:46:54 -07:00 committed by Commit Bot
parent e264a9ee9c
commit 83377270dc
6 changed files with 0 additions and 27 deletions
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6
webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
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@ -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,

1
webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
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@ -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));

5
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
View File

@ -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);

4
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
View File

@ -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,

7
webrtc/modules/rtp_rtcp/source/rtp_sender.cc
View File

@ -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);
}

4
webrtc/modules/rtp_rtcp/source/rtp_sender.h
View File

@ -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);