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Clean away use of RtpAudioFeedback interface from RTP/RTCP sender code.

Browse Source 
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1803923003

Cr-Commit-Position: refs/heads/master@{#12003}
This commit is contained in:
solenberg 2016-03-15 11:41:53 -07:00 committed by Commit bot
parent 6baec0351a
commit 6021fe2b1e
13 changed files with 10 additions and 51 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

1
webrtc/call/rtc_event_log_unittest.cc
View File

@ -308,7 +308,6 @@ size_t GenerateRtpPacket(uint32_t extensions_bitvector,
RTPSender rtp_sender(false, // bool audio
clock, // Clock* clock
nullptr, // Transport*
nullptr, // RtpAudioFeedback*
nullptr, // PacedSender*
nullptr, // PacketRouter*
nullptr, // SendTimeObserver*

3
webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
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@ -55,8 +55,6 @@ class RtpRtcp : public Module {
* intra_frame_callback - Called when the receiver request a intra frame.
* bandwidth_callback - Called when we receive a changed estimate from
* the receiver of out stream.
* audio_messages - Telephone events. May not be NULL; default
* callback will do nothing.
* remote_bitrate_estimator - Estimates the bandwidth available for a set of
* streams from the same client.
* paced_sender - Spread any bursts of packets into smaller
@ -72,7 +70,6 @@ class RtpRtcp : public Module {
TransportFeedbackObserver* transport_feedback_callback;
RtcpRttStats* rtt_stats;
RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer;
RtpAudioFeedback* audio_messages;
RemoteBitrateEstimator* remote_bitrate_estimator;
RtpPacketSender* paced_sender;
TransportSequenceNumberAllocator* transport_sequence_number_allocator;

2
webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
View File

@ -219,8 +219,6 @@ class MockRtpRtcp : public RtpRtcp {
MOCK_METHOD1(RegisterRtcpStatisticsCallback, void(RtcpStatisticsCallback*));
MOCK_METHOD0(GetRtcpStatisticsCallback, RtcpStatisticsCallback*());
MOCK_METHOD1(SendFeedbackPacket, bool(const rtcp::TransportFeedback& packet));
MOCK_METHOD1(RegisterAudioCallback,
int32_t(RtpAudioFeedback* messagesCallback));
MOCK_METHOD1(SetAudioPacketSize,
int32_t(const uint16_t packetSizeSamples));
MOCK_METHOD3(SendTelephoneEventOutband,

2
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
View File

@ -54,7 +54,6 @@ RtpRtcp::Configuration::Configuration()
transport_feedback_callback(nullptr),
rtt_stats(nullptr),
rtcp_packet_type_counter_observer(nullptr),
audio_messages(NullObjectRtpAudioFeedback()),
remote_bitrate_estimator(nullptr),
paced_sender(nullptr),
transport_sequence_number_allocator(nullptr),
@ -80,7 +79,6 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
: rtp_sender_(configuration.audio,
configuration.clock,
configuration.outgoing_transport,
configuration.audio_messages,
configuration.paced_sender,
configuration.transport_sequence_number_allocator,
configuration.transport_feedback_callback,

3
webrtc/modules/rtp_rtcp/source/rtp_sender.cc
View File

@ -108,7 +108,6 @@ RTPSender::RTPSender(
bool audio,
Clock* clock,
Transport* transport,
RtpAudioFeedback* audio_feedback,
RtpPacketSender* paced_sender,
TransportSequenceNumberAllocator* sequence_number_allocator,
TransportFeedbackObserver* transport_feedback_observer,
@ -125,7 +124,7 @@ RTPSender::RTPSender(
bitrates_(bitrate_callback),
total_bitrate_sent_(clock, bitrates_.total_bitrate_observer()),
audio_configured_(audio),
audio_(audio ? new RTPSenderAudio(clock, this, audio_feedback) : nullptr),
audio_(audio ? new RTPSenderAudio(clock, this) : nullptr),
video_(audio ? nullptr : new RTPSenderVideo(clock, this)),
paced_sender_(paced_sender),
transport_sequence_number_allocator_(sequence_number_allocator),

1
webrtc/modules/rtp_rtcp/source/rtp_sender.h
View File

@ -90,7 +90,6 @@ class RTPSender : public RTPSenderInterface {
RTPSender(bool audio,
Clock* clock,
Transport* transport,
RtpAudioFeedback* audio_feedback,
RtpPacketSender* paced_sender,
TransportSequenceNumberAllocator* sequence_number_allocator,
TransportFeedbackObserver* transport_feedback_callback,

11
webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
View File

@ -21,12 +21,9 @@ namespace webrtc {
static const int kDtmfFrequencyHz = 8000;
RTPSenderAudio::RTPSenderAudio(Clock* clock,
RTPSender* rtpSender,
RtpAudioFeedback* audio_feedback)
RTPSenderAudio::RTPSenderAudio(Clock* clock, RTPSender* rtpSender)
: _clock(clock),
_rtpSender(rtpSender),
_audioFeedback(audio_feedback),
_sendAudioCritsect(CriticalSectionWrapper::CreateCriticalSection()),
_packetSizeSamples(160),
_dtmfEventIsOn(false),
@ -158,7 +155,6 @@ int32_t RTPSenderAudio::SendAudio(FrameType frameType,
// TODO(pwestin) Breakup function in smaller functions.
size_t payloadSize = dataSize;
size_t maxPayloadLength = _rtpSender->MaxPayloadLength();
bool dtmfToneStarted = false;
uint16_t dtmfLengthMS = 0;
uint8_t key = 0;
int red_payload_type;
@ -185,15 +181,10 @@ int32_t RTPSenderAudio::SendAudio(FrameType frameType,
_dtmfEventFirstPacketSent = false;
_dtmfKey = key;
_dtmfLengthSamples = (kDtmfFrequencyHz / 1000) * dtmfLengthMS;
dtmfToneStarted = true;
_dtmfEventIsOn = true;
}
}
}
if (dtmfToneStarted) {
if (_audioFeedback)
_audioFeedback->OnPlayTelephoneEvent(key, dtmfLengthMS, _dtmfLevel);
}
// A source MAY send events and coded audio packets for the same time
// but we don't support it

5
webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
View File

@ -21,9 +21,7 @@
namespace webrtc {
class RTPSenderAudio : public DTMFqueue {
public:
RTPSenderAudio(Clock* clock,
RTPSender* rtpSender,
RtpAudioFeedback* audio_feedback);
RTPSenderAudio(Clock* clock, RTPSender* rtpSender);
virtual ~RTPSenderAudio();
int32_t RegisterAudioPayload(const char payloadName[RTP_PAYLOAD_NAME_SIZE],
@ -73,7 +71,6 @@ class RTPSenderAudio : public DTMFqueue {
private:
Clock* const _clock;
RTPSender* const _rtpSender;
RtpAudioFeedback* const _audioFeedback;
rtc::scoped_ptr<CriticalSectionWrapper> _sendAudioCritsect;

10
webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
View File

@ -141,7 +141,7 @@ class RtpSenderTest : public ::testing::Test {
void SetUp() override { SetUpRtpSender(true); }
void SetUpRtpSender(bool pacer) {
rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport_, nullptr,
rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport_,
pacer ? &mock_paced_sender_ : nullptr,
&seq_num_allocator_, nullptr, nullptr,
nullptr, nullptr, &mock_rtc_event_log_));
@ -954,7 +954,7 @@ TEST_F(RtpSenderTest, SendPadding) {
TEST_F(RtpSenderTest, SendRedundantPayloads) {
MockTransport transport;
rtp_sender_.reset(new RTPSender(
false, &fake_clock_, &transport, nullptr, &mock_paced_sender_, nullptr,
false, &fake_clock_, &transport, &mock_paced_sender_, nullptr,
nullptr, nullptr, nullptr, nullptr, &mock_rtc_event_log_));
rtp_sender_->SetSequenceNumber(kSeqNum);
rtp_sender_->SetRtxPayloadType(kRtxPayload, kPayload);
@ -1096,7 +1096,7 @@ TEST_F(RtpSenderTest, FrameCountCallbacks) {
FrameCounts frame_counts_;
} callback;
rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport_, nullptr,
rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport_,
&mock_paced_sender_, nullptr, nullptr,
nullptr, &callback, nullptr, nullptr));
@ -1152,7 +1152,7 @@ TEST_F(RtpSenderTest, BitrateCallbacks) {
BitrateStatistics total_stats_;
BitrateStatistics retransmit_stats_;
} callback;
rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport_, nullptr,
rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport_,
nullptr, nullptr, nullptr, &callback, nullptr,
nullptr, nullptr));
@ -1205,7 +1205,7 @@ class RtpSenderAudioTest : public RtpSenderTest {
void SetUp() override {
payload_ = kAudioPayload;
rtp_sender_.reset(new RTPSender(true, &fake_clock_, &transport_, nullptr,
rtp_sender_.reset(new RTPSender(true, &fake_clock_, &transport_,
nullptr, nullptr, nullptr, nullptr, nullptr,
nullptr, nullptr));
rtp_sender_->SetSequenceNumber(kSeqNum);

2
webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
View File

@ -106,7 +106,6 @@ class RtpRtcpAudioTest : public ::testing::Test {
configuration.clock = &fake_clock;
configuration.receive_statistics = receive_statistics1_.get();
configuration.outgoing_transport = transport1;
configuration.audio_messages = audioFeedback;
module1 = RtpRtcp::CreateRtpRtcp(configuration);
rtp_receiver1_.reset(RtpReceiver::CreateAudioReceiver(
@ -115,7 +114,6 @@ class RtpRtcpAudioTest : public ::testing::Test {
configuration.receive_statistics = receive_statistics2_.get();
configuration.outgoing_transport = transport2;
configuration.audio_messages = audioFeedback;
module2 = RtpRtcp::CreateRtpRtcp(configuration);
rtp_receiver2_.reset(RtpReceiver::CreateAudioReceiver(

13
webrtc/voice_engine/channel.cc
View File

@ -353,15 +353,6 @@ bool Channel::SendRtcp(const uint8_t* data, size_t len) {
return true;
}
void Channel::OnPlayTelephoneEvent(uint8_t event,
uint16_t lengthMs,
uint8_t volume) {
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::OnPlayTelephoneEvent(event=%u, lengthMs=%u,"
" volume=%u)",
event, lengthMs, volume);
}
void Channel::OnIncomingSSRCChanged(uint32_t ssrc) {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::OnIncomingSSRCChanged(SSRC=%d)", ssrc);
@ -731,7 +722,7 @@ Channel::Channel(int32_t channelId,
ReceiveStatistics::Create(Clock::GetRealTimeClock())),
rtp_receiver_(
RtpReceiver::CreateAudioReceiver(Clock::GetRealTimeClock(),
this,
nullptr,
this,
this,
rtp_payload_registry_.get())),
@ -816,7 +807,6 @@ Channel::Channel(int32_t channelId,
RtpRtcp::Configuration configuration;
configuration.audio = true;
configuration.outgoing_transport = this;
configuration.audio_messages = this;
configuration.receive_statistics = rtp_receive_statistics_.get();
configuration.bandwidth_callback = rtcp_observer_.get();
if (pacing_enabled_) {
@ -2201,7 +2191,6 @@ int Channel::SendTelephoneEventOutband(int event, int duration_ms) {
if (!Sending()) {
return -1;
}
if (_rtpRtcpModule->SendTelephoneEventOutband(
event, duration_ms, kTelephoneEventAttenuationdB) != 0) {
_engineStatisticsPtr->SetLastError(

6
webrtc/voice_engine/channel.h
View File

@ -156,7 +156,6 @@ class Channel
public FileCallback, // receiving notification from file player &
// recorder
public Transport,
public RtpAudioFeedback,
public AudioPacketizationCallback, // receive encoded packets from the
// ACM
public ACMVADCallback, // receive voice activity from the ACM
@ -385,11 +384,6 @@ class Channel
void OnIncomingSSRCChanged(uint32_t ssrc) override;
void OnIncomingCSRCChanged(uint32_t CSRC, bool added) override;
// From RtpAudioFeedback in the RTP/RTCP module
void OnPlayTelephoneEvent(uint8_t event,
uint16_t lengthMs,
uint8_t volume) override;
// From Transport (called by the RTP/RTCP module)
bool SendRtp(const uint8_t* data,
size_t len,

2
webrtc/voice_engine/include/voe_errors.h
View File

@ -33,7 +33,7 @@
#define VE_INVALID_IP_ADDRESS 8019
#define VE_ALREADY_PLAYING 8020
#define VE_NOT_ALL_VERSION_INFO 8021
#define VE_DTMF_OUTOF_RANGE 8022
// 8022 is not used
#define VE_INVALID_CHANNELS 8023
#define VE_SET_PLTYPE_FAILED 8024
// 8025 is not used