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Relanding https://codereview.webrtc.org/1715883002/ in pieces.

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- Use better types in AudioSendStream::SendTelephoneEvent() and related methods.

BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1782053002

Cr-Commit-Position: refs/heads/master@{#11953}
This commit is contained in:
solenberg 2016-03-11 03:06:41 -08:00 committed by Commit bot
parent 4bf0c71774
commit 8842c3e41b
14 changed files with 35 additions and 37 deletions
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4
webrtc/audio/audio_send_stream.cc
View File

@ -125,8 +125,8 @@ bool AudioSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
return false;
}
bool AudioSendStream::SendTelephoneEvent(int payload_type, uint8_t event,
uint32_t duration_ms) {
bool AudioSendStream::SendTelephoneEvent(int payload_type, int event,
int duration_ms) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
return channel_proxy_->SetSendTelephoneEventPayloadType(payload_type) &&
channel_proxy_->SendTelephoneEventOutband(event, duration_ms);

4
webrtc/audio/audio_send_stream.h
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@ -40,8 +40,8 @@ class AudioSendStream final : public webrtc::AudioSendStream {
bool DeliverRtcp(const uint8_t* packet, size_t length) override;
// webrtc::AudioSendStream implementation.
bool SendTelephoneEvent(int payload_type, uint8_t event,
uint32_t duration_ms) override;
bool SendTelephoneEvent(int payload_type, int event,
int duration_ms) override;
webrtc::AudioSendStream::Stats GetStats() const override;
const webrtc::AudioSendStream::Config& config() const;

4
webrtc/audio/audio_send_stream_unittest.cc
View File

@ -46,8 +46,8 @@ const CallStatistics kCallStats = {
const CodecInst kCodecInst = {-121, "codec_name_send", 48000, -231, 0, -671};
const ReportBlock kReportBlock = {456, 780, 123, 567, 890, 132, 143, 13354};
const int kTelephoneEventPayloadType = 123;
const uint8_t kTelephoneEventCode = 45;
const uint32_t kTelephoneEventDuration = 6789;
const int kTelephoneEventCode = 45;
const int kTelephoneEventDuration = 6789;
struct ConfigHelper {
ConfigHelper()

4
webrtc/audio_send_stream.h
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@ -90,8 +90,8 @@ class AudioSendStream : public SendStream {
};
// TODO(solenberg): Make payload_type a config property instead.
virtual bool SendTelephoneEvent(int payload_type, uint8_t event,
uint32_t duration_ms) = 0;
virtual bool SendTelephoneEvent(int payload_type, int event,
int duration_ms) = 0;
virtual Stats GetStats() const = 0;
};
} // namespace webrtc

4
webrtc/media/engine/fakewebrtccall.cc
View File

@ -39,8 +39,8 @@ FakeAudioSendStream::TelephoneEvent
return latest_telephone_event_;
}
bool FakeAudioSendStream::SendTelephoneEvent(int payload_type, uint8_t event,
uint32_t duration_ms) {
bool FakeAudioSendStream::SendTelephoneEvent(int payload_type, int event,
int duration_ms) {
latest_telephone_event_.payload_type = payload_type;
latest_telephone_event_.event_code = event;
latest_telephone_event_.duration_ms = duration_ms;

8
webrtc/media/engine/fakewebrtccall.h
View File

@ -35,8 +35,8 @@ class FakeAudioSendStream final : public webrtc::AudioSendStream {
public:
struct TelephoneEvent {
int payload_type = -1;
uint8_t event_code = 0;
uint32_t duration_ms = 0;
int event_code = 0;
int duration_ms = 0;
};
explicit FakeAudioSendStream(const webrtc::AudioSendStream::Config& config);
@ -56,8 +56,8 @@ class FakeAudioSendStream final : public webrtc::AudioSendStream {
}
// webrtc::AudioSendStream implementation.
bool SendTelephoneEvent(int payload_type, uint8_t event,
uint32_t duration_ms) override;
bool SendTelephoneEvent(int payload_type, int event,
int duration_ms) override;
webrtc::AudioSendStream::Stats GetStats() const override;
TelephoneEvent latest_telephone_event_;

3
webrtc/media/engine/webrtcvoiceengine.cc
View File

@ -1178,8 +1178,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
RTC_CHECK(stream_);
}
bool SendTelephoneEvent(int payload_type, uint8_t event,
uint32_t duration_ms) {
bool SendTelephoneEvent(int payload_type, int event, int duration_ms) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_DCHECK(stream_);
return stream_->SendTelephoneEvent(payload_type, event, duration_ms);

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

@ -322,7 +322,7 @@ TEST_F(RtpRtcpAudioTest, DTMF) {
(voice_codec.rate < 0) ? 0 : voice_codec.rate));
// Start DTMF test.
uint32_t timeStamp = 160;
int timeStamp = 160;
// Send a DTMF tone using RFC 2833 (4733).
for (int i = 0; i < 16; i++) {

3
webrtc/test/mock_voe_channel_proxy.h
View File

@ -43,8 +43,7 @@ class MockVoEChannelProxy : public voe::ChannelProxy {
MOCK_CONST_METHOD0(GetSpeechOutputLevelFullRange, int32_t());
MOCK_CONST_METHOD0(GetDelayEstimate, uint32_t());
MOCK_METHOD1(SetSendTelephoneEventPayloadType, bool(int payload_type));
MOCK_METHOD2(SendTelephoneEventOutband, bool(uint8_t event,
uint32_t duration_ms));
MOCK_METHOD2(SendTelephoneEventOutband, bool(int event, int duration_ms));
};
} // namespace test
} // namespace webrtc

20
webrtc/voice_engine/channel.cc
View File

@ -47,6 +47,8 @@
namespace webrtc {
namespace voe {
const int kTelephoneEventAttenuationdB = 10;
class TransportFeedbackProxy : public TransportFeedbackObserver {
public:
TransportFeedbackProxy() : feedback_observer_(nullptr) {
@ -2212,21 +2214,21 @@ int Channel::GetChannelOutputVolumeScaling(float& scaling) const {
return 0;
}
int Channel::SendTelephoneEventOutband(unsigned char eventCode,
int lengthMs,
int attenuationDb,
bool playDtmfEvent) {
int Channel::SendTelephoneEventOutband(int event, int duration_ms) {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::SendTelephoneEventOutband(..., playDtmfEvent=%d)",
playDtmfEvent);
"Channel::SendTelephoneEventOutband(...)");
RTC_DCHECK_LE(0, event);
RTC_DCHECK_GE(255, event);
RTC_DCHECK_LE(0, duration_ms);
RTC_DCHECK_GE(65535, duration_ms);
if (!Sending()) {
return -1;
}
_playOutbandDtmfEvent = playDtmfEvent;
_playOutbandDtmfEvent = false;
if (_rtpRtcpModule->SendTelephoneEventOutband(eventCode, lengthMs,
attenuationDb) != 0) {
if (_rtpRtcpModule->SendTelephoneEventOutband(
event, duration_ms, kTelephoneEventAttenuationdB) != 0) {
_engineStatisticsPtr->SetLastError(
VE_SEND_DTMF_FAILED, kTraceWarning,
"SendTelephoneEventOutband() failed to send event");

5
webrtc/voice_engine/channel.h
View File

@ -297,10 +297,7 @@ class Channel
int GetRtpRtcp(RtpRtcp** rtpRtcpModule, RtpReceiver** rtp_receiver) const;
// VoEDtmf
int SendTelephoneEventOutband(unsigned char eventCode,
int lengthMs,
int attenuationDb,
bool playDtmfEvent);
int SendTelephoneEventOutband(int event, int duration_ms);
int SendTelephoneEventInband(unsigned char eventCode,
int lengthMs,
int attenuationDb,

6
webrtc/voice_engine/channel_proxy.cc
View File

@ -148,11 +148,9 @@ bool ChannelProxy::SetSendTelephoneEventPayloadType(int payload_type) {
return channel()->SetSendTelephoneEventPayloadType(payload_type) == 0;
}
bool ChannelProxy::SendTelephoneEventOutband(uint8_t event,
uint32_t duration_ms) {
bool ChannelProxy::SendTelephoneEventOutband(int event, int duration_ms) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
return
channel()->SendTelephoneEventOutband(event, duration_ms, 10, false) == 0;
return channel()->SendTelephoneEventOutband(event, duration_ms) == 0;
}
void ChannelProxy::SetSink(std::unique_ptr<AudioSinkInterface> sink) {

2
webrtc/voice_engine/channel_proxy.h
View File

@ -68,7 +68,7 @@ class ChannelProxy {
virtual uint32_t GetDelayEstimate() const;
virtual bool SetSendTelephoneEventPayloadType(int payload_type);
virtual bool SendTelephoneEventOutband(uint8_t event, uint32_t duration_ms);
virtual bool SendTelephoneEventOutband(int event, int duration_ms);
virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink);

3
webrtc/voice_engine/dtmf_inband.h
View File

@ -17,6 +17,9 @@
namespace webrtc {
// TODO(solenberg): Used as a DTMF tone generator in voe::OutputMixer. Pull out
// the one in NetEq and use that instead? We don't need several
// implemenations of this.
class DtmfInband
{
public: