Relanding https://codereview.webrtc.org/1715883002/ in pieces.
- 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}
diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc
index 160a818..24afcbc 100644
--- a/webrtc/audio/audio_send_stream.cc
+++ b/webrtc/audio/audio_send_stream.cc
@@ -125,8 +125,8 @@
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);
diff --git a/webrtc/audio/audio_send_stream.h b/webrtc/audio/audio_send_stream.h
index cf0a19c..d463b3d 100644
--- a/webrtc/audio/audio_send_stream.h
+++ b/webrtc/audio/audio_send_stream.h
@@ -40,8 +40,8 @@
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;
diff --git a/webrtc/audio/audio_send_stream_unittest.cc b/webrtc/audio/audio_send_stream_unittest.cc
index 6788699..c04a3de 100644
--- a/webrtc/audio/audio_send_stream_unittest.cc
+++ b/webrtc/audio/audio_send_stream_unittest.cc
@@ -46,8 +46,8 @@
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()
diff --git a/webrtc/audio_send_stream.h b/webrtc/audio_send_stream.h
index d1af9e0..24c3d77 100644
--- a/webrtc/audio_send_stream.h
+++ b/webrtc/audio_send_stream.h
@@ -90,8 +90,8 @@
};
// 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
diff --git a/webrtc/media/engine/fakewebrtccall.cc b/webrtc/media/engine/fakewebrtccall.cc
index aa94d48..3277e75 100644
--- a/webrtc/media/engine/fakewebrtccall.cc
+++ b/webrtc/media/engine/fakewebrtccall.cc
@@ -39,8 +39,8 @@
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;
diff --git a/webrtc/media/engine/fakewebrtccall.h b/webrtc/media/engine/fakewebrtccall.h
index 89a644a..41a92df 100644
--- a/webrtc/media/engine/fakewebrtccall.h
+++ b/webrtc/media/engine/fakewebrtccall.h
@@ -35,8 +35,8 @@
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 @@
}
// 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_;
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
index f57db27..a496316 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -1178,8 +1178,7 @@
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);
diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
index 634969b..32ab937 100644
--- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
+++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
@@ -322,7 +322,7 @@
(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++) {
diff --git a/webrtc/test/mock_voe_channel_proxy.h b/webrtc/test/mock_voe_channel_proxy.h
index f5c8733..c2211f8 100644
--- a/webrtc/test/mock_voe_channel_proxy.h
+++ b/webrtc/test/mock_voe_channel_proxy.h
@@ -43,8 +43,7 @@
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
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index 11af45e..25ecca1 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -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 @@
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");
diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h
index 75c4fd8..65c34f6 100644
--- a/webrtc/voice_engine/channel.h
+++ b/webrtc/voice_engine/channel.h
@@ -297,10 +297,7 @@
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,
diff --git a/webrtc/voice_engine/channel_proxy.cc b/webrtc/voice_engine/channel_proxy.cc
index da7864f..10c8821 100644
--- a/webrtc/voice_engine/channel_proxy.cc
+++ b/webrtc/voice_engine/channel_proxy.cc
@@ -148,11 +148,9 @@
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) {
diff --git a/webrtc/voice_engine/channel_proxy.h b/webrtc/voice_engine/channel_proxy.h
index 3461cf3..dec726c 100644
--- a/webrtc/voice_engine/channel_proxy.h
+++ b/webrtc/voice_engine/channel_proxy.h
@@ -68,7 +68,7 @@
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);
diff --git a/webrtc/voice_engine/dtmf_inband.h b/webrtc/voice_engine/dtmf_inband.h
index 6000d99..795c5ce 100644
--- a/webrtc/voice_engine/dtmf_inband.h
+++ b/webrtc/voice_engine/dtmf_inband.h
@@ -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: