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Deprecated SetAudioPacketSize from RTPSender and removed calls to it.

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The packet size was only used to control how often to output DTMF
packets. However, it likely did not work as intended, since that
interval was only set during initialization. No changes to the packet
size, like what AudioEncoder::Num10MsFramesInNextPacket could
indicate, were picked up. The value was instead taken from an entry in
ACMCodecDB.

Since it was not-fully-functional, its exact value didn't seem to
matter and it was getting in the way of making it possible to supply
an external audio encoder factory, I've decided to remove it
altogether. The DTMF code now uses an interval of 50 ms regardless,
which is a value recommended by the RFC.

BUG=webrtc:5806

Review-Url: https://codereview.webrtc.org/2545753002
Cr-Commit-Position: refs/heads/master@{#15380}
This commit is contained in:
ossu 2016-12-02 02:40:02 -08:00 committed by Commit bot
parent e066b302ab
commit 00bceb1eda
8 changed files with 23 additions and 36 deletions
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7
webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
View File

@ -430,10 +430,11 @@ class RtpRtcp : public Module {
// Audio
// **************************************************************************
// Sets audio packet size, used to determine when it's time to send a DTMF
// packet in silence (CNG).
// 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.
virtual int32_t SetAudioPacketSize(uint16_t packet_size_samples) = 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.

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

@ -777,11 +777,9 @@ int32_t ModuleRtpRtcpImpl::SendTelephoneEventOutband(
return rtp_sender_.SendTelephoneEvent(key, time_ms, level);
}
// Set audio packet size, used to determine when it's time to send a DTMF
// packet in silence (CNG).
int32_t ModuleRtpRtcpImpl::SetAudioPacketSize(
const uint16_t packet_size_samples) {
return rtp_sender_.SetAudioPacketSize(packet_size_samples);
return audio_ ? 0 : -1;
}
int32_t ModuleRtpRtcpImpl::SetAudioLevel(

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

@ -260,8 +260,8 @@ class ModuleRtpRtcpImpl : public RtpRtcp, public RTCPReceiver::ModuleRtpRtcp {
// Audio part.
// Set audio packet size, used to determine when it's time to send a DTMF
// packet in silence (CNG).
// 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).

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

@ -1153,7 +1153,7 @@ int32_t RTPSender::SetAudioPacketSize(uint16_t packet_size_samples) {
if (!audio_configured_) {
return -1;
}
return audio_->SetAudioPacketSize(packet_size_samples);
return 0;
}
int32_t RTPSender::SetAudioLevel(uint8_t level_d_bov) {

6
webrtc/modules/rtp_rtcp/source/rtp_sender.h
View File

@ -179,9 +179,9 @@ class RTPSender {
// Send a DTMF tone using RFC 2833 (4733).
int32_t SendTelephoneEvent(uint8_t key, uint16_t time_ms, uint8_t level);
// Set audio packet size, used to determine when it's time to send a DTMF
// packet in silence (CNG).
int32_t SetAudioPacketSize(uint16_t packet_size_samples);
// 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.

24
webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
View File

@ -31,14 +31,6 @@ RTPSenderAudio::RTPSenderAudio(Clock* clock, RTPSender* rtp_sender)
RTPSenderAudio::~RTPSenderAudio() {}
// set audio packet size, used to determine when it's time to send a DTMF packet
// in silence (CNG)
int32_t RTPSenderAudio::SetAudioPacketSize(uint16_t packet_size_samples) {
rtc::CritScope cs(&send_audio_critsect_);
packet_size_samples_ = packet_size_samples;
return 0;
}
int32_t RTPSenderAudio::RegisterAudioPayload(
const char payloadName[RTP_PAYLOAD_NAME_SIZE],
const int8_t payload_type,
@ -132,19 +124,24 @@ bool RTPSenderAudio::SendAudio(FrameType frame_type,
const uint8_t* payload_data,
size_t payload_size,
const RTPFragmentationHeader* fragmentation) {
// From RFC 4733:
// A source has wide latitude as to how often it sends event updates. A
// natural interval is the spacing between non-event audio packets. [...]
// Alternatively, a source MAY decide to use a different spacing for event
// updates, with a value of 50 ms RECOMMENDED.
constexpr int kDtmfIntervalTimeMs = 50;
uint8_t audio_level_dbov = 0;
uint16_t packet_size_samples = 0;
uint32_t dtmf_payload_freq = 0;
{
rtc::CritScope cs(&send_audio_critsect_);
audio_level_dbov = audio_level_dbov_;
packet_size_samples = packet_size_samples_;
dtmf_payload_freq = dtmf_payload_freq_;
}
// Check if we have pending DTMFs to send
if (!dtmf_event_is_on_ && dtmf_queue_.PendingDtmf()) {
if ((clock_->TimeInMilliseconds() - dtmf_time_last_sent_) > 50) {
if ((clock_->TimeInMilliseconds() - dtmf_time_last_sent_) >
kDtmfIntervalTimeMs) {
// New tone to play
dtmf_timestamp_ = rtp_timestamp;
if (dtmf_queue_.NextDtmf(&dtmf_current_event_)) {
@ -163,7 +160,10 @@ bool RTPSenderAudio::SendAudio(FrameType frame_type,
// kEmptyFrame is used to drive the DTMF when in CN mode
// it can be triggered more frequently than we want to send the
// DTMF packets.
if (packet_size_samples > (rtp_timestamp - dtmf_timestamp_last_sent_)) {
const unsigned int dtmf_interval_time_rtp =
dtmf_payload_freq * kDtmfIntervalTimeMs / 1000;
if ((rtp_timestamp - dtmf_timestamp_last_sent_) <
dtmf_interval_time_rtp) {
// not time to send yet
return true;
}

6
webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
View File

@ -42,10 +42,6 @@ class RTPSenderAudio {
size_t payload_size,
const RTPFragmentationHeader* fragmentation);
// set audio packet size, used to determine when it's time to send a DTMF
// packet in silence (CNG)
int32_t SetAudioPacketSize(uint16_t packet_size_samples);
// Store the audio level in dBov for
// header-extension-for-audio-level-indication.
// Valid range is [0,100]. Actual value is negative.
@ -69,8 +65,6 @@ class RTPSenderAudio {
rtc::CriticalSection send_audio_critsect_;
uint16_t packet_size_samples_ GUARDED_BY(send_audio_critsect_) = 160;
// DTMF.
bool dtmf_event_is_on_ = false;
bool dtmf_event_first_packet_sent_ = false;

6
webrtc/voice_engine/channel.cc
View File

@ -1323,12 +1323,6 @@ int32_t Channel::SetSendCodec(const CodecInst& codec) {
}
}
if (_rtpRtcpModule->SetAudioPacketSize(codec.pacsize) != 0) {
WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, _channelId),
"SetSendCodec() failed to set audio packet size");
return -1;
}
return 0;
}