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Remove default receive channel from WVoE; baby step 2.

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Rename voe_channel_ to default_send_channel_id_.

BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1388733002

Cr-Commit-Position: refs/heads/master@{#10261}
This commit is contained in:
solenberg 2015-10-13 03:06:58 -07:00 committed by Commit bot
parent b8fd39caa2
commit 8fb30c328b
3 changed files with 64 additions and 58 deletions
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94
talk/media/webrtc/webrtcvoiceengine.cc
View File

@ -1390,7 +1390,7 @@ WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel(WebRtcVoiceEngine* engine,
const AudioOptions& options,
webrtc::Call* call)
: engine_(engine),
voe_channel_(engine->CreateMediaVoiceChannel()),
default_send_channel_id_(engine->CreateMediaVoiceChannel()),
send_bitrate_setting_(false),
send_bitrate_bps_(0),
options_(),
@ -1406,16 +1406,16 @@ WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel(WebRtcVoiceEngine* engine,
RTC_DCHECK(thread_checker_.CalledOnValidThread());
engine->RegisterChannel(this);
LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel "
<< voe_channel();
<< default_send_channel_id();
RTC_DCHECK(nullptr != call);
ConfigureSendChannel(voe_channel());
ConfigureSendChannel(default_send_channel_id());
SetOptions(options);
}
WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel() {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel "
<< voe_channel();
<< default_send_channel_id();
// Remove any remaining send streams, the default channel will be deleted
// later.
@ -1433,7 +1433,7 @@ WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel() {
RTC_DCHECK(receive_streams_.empty());
// Delete the default channel.
DeleteChannel(voe_channel());
DeleteChannel(default_send_channel_id());
}
bool WebRtcVoiceMediaChannel::SetSendParameters(
@ -1485,7 +1485,9 @@ bool WebRtcVoiceMediaChannel::SetOptions(const AudioOptions& options) {
LOG(LS_WARNING) << "Failed to set DSCP settings for audio channel";
}
}
RecreateAudioReceiveStreams();
LOG(LS_INFO) << "Set voice channel options. Current options: "
<< options_.ToString();
return true;
@ -1493,9 +1495,10 @@ bool WebRtcVoiceMediaChannel::SetOptions(const AudioOptions& options) {
bool WebRtcVoiceMediaChannel::SetRecvCodecs(
const std::vector<AudioCodec>& codecs) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
// Set the payload types to be used for incoming media.
LOG(LS_INFO) << "Setting receive voice codecs.";
RTC_DCHECK(thread_checker_.CalledOnValidThread());
if (!VerifyUniquePayloadTypes(codecs)) {
LOG(LS_ERROR) << "Codec payload types overlap.";
@ -1830,7 +1833,8 @@ bool WebRtcVoiceMediaChannel::SetRecvRtpHeaderExtensions(
// The default channel may or may not be in |receive_channels_|. Set the rtp
// header extensions for default channel regardless.
if (!SetChannelRecvRtpHeaderExtensions(voe_channel(), extensions)) {
if (!SetChannelRecvRtpHeaderExtensions(default_send_channel_id(),
extensions)) {
return false;
}
@ -1899,7 +1903,8 @@ bool WebRtcVoiceMediaChannel::SetSendRtpHeaderExtensions(
// The default channel may or may not be in |send_channels_|. Set the rtp
// header extensions for default channel regardless.
if (!SetChannelSendRtpHeaderExtensions(voe_channel(), extensions)) {
if (!SetChannelSendRtpHeaderExtensions(default_send_channel_id(),
extensions)) {
return false;
}
@ -1959,7 +1964,7 @@ bool WebRtcVoiceMediaChannel::ChangePlayout(bool playout) {
bool result = true;
if (receive_channels_.empty()) {
// Only toggle the default channel if we don't have any other channels.
result = SetPlayout(voe_channel(), playout);
result = SetPlayout(default_send_channel_id(), playout);
}
for (const auto& ch : receive_channels_) {
if (!SetPlayout(ch.second->channel(), playout)) {
@ -2103,7 +2108,7 @@ bool WebRtcVoiceMediaChannel::AddSendStream(const StreamParams& sp) {
}
}
if (default_channel_is_available) {
channel = voe_channel();
channel = default_send_channel_id();
} else {
// Create a new channel for sending audio data.
channel = engine()->CreateMediaVoiceChannel();
@ -2226,11 +2231,12 @@ bool WebRtcVoiceMediaChannel::AddRecvStream(const StreamParams& sp) {
LOG(LS_INFO) << "Recv stream " << ssrc << " reuse default channel";
default_receive_ssrc_ = ssrc;
WebRtcVoiceChannelRenderer* channel_renderer =
new WebRtcVoiceChannelRenderer(voe_channel(), audio_transport);
new WebRtcVoiceChannelRenderer(default_send_channel_id(),
audio_transport);
receive_channels_.insert(std::make_pair(ssrc, channel_renderer));
receive_stream_params_[ssrc] = sp;
AddAudioReceiveStream(ssrc);
return SetPlayout(voe_channel(), playout_);
return SetPlayout(default_send_channel_id(), playout_);
}
// Create a new channel for receiving audio data.
@ -2239,7 +2245,6 @@ bool WebRtcVoiceMediaChannel::AddRecvStream(const StreamParams& sp) {
LOG_RTCERR0(CreateChannel);
return false;
}
if (!ConfigureRecvChannel(channel)) {
DeleteChannel(channel);
return false;
@ -2259,17 +2264,18 @@ bool WebRtcVoiceMediaChannel::AddRecvStream(const StreamParams& sp) {
bool WebRtcVoiceMediaChannel::ConfigureRecvChannel(int channel) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
// Configure to use external transport, like our default channel.
// Configure to use external transport.
if (engine()->voe()->network()->RegisterExternalTransport(
channel, *this) == -1) {
LOG_RTCERR2(SetExternalTransport, channel, this);
return false;
}
// Use the same SSRC as our default channel (so the RTCP reports are correct).
// Use the same SSRC as our default send channel, so the RTCP reports are
// correct.
unsigned int send_ssrc = 0;
webrtc::VoERTP_RTCP* rtp = engine()->voe()->rtp();
if (rtp->GetLocalSSRC(voe_channel(), send_ssrc) == -1) {
if (rtp->GetLocalSSRC(default_send_channel_id(), send_ssrc) == -1) {
LOG_RTCERR1(GetSendSSRC, channel);
return false;
}
@ -2278,12 +2284,13 @@ bool WebRtcVoiceMediaChannel::ConfigureRecvChannel(int channel) {
return false;
}
// Associate receive channel to default channel (so the receive channel can
// obtain RTT from the send channel)
engine()->voe()->base()->AssociateSendChannel(channel, voe_channel());
// Associate receive channel to default send channel (so the receive channel
// can obtain RTT from the send channel).
engine()->voe()->base()->AssociateSendChannel(channel,
default_send_channel_id());
LOG(LS_INFO) << "VoiceEngine channel #"
<< channel << " is associated with channel #"
<< voe_channel() << ".";
<< default_send_channel_id() << ".";
// Use the same recv payload types as our default channel.
ResetRecvCodecs(channel);
@ -2294,7 +2301,7 @@ bool WebRtcVoiceMediaChannel::ConfigureRecvChannel(int channel) {
voe_codec.pltype = codec.id;
voe_codec.rate = 0; // Needed to make GetRecPayloadType work for ISAC
if (engine()->voe()->codec()->GetRecPayloadType(
voe_channel(), voe_codec) != -1) {
default_send_channel_id(), voe_codec) != -1) {
if (engine()->voe()->codec()->SetRecPayloadType(
channel, voe_codec) == -1) {
LOG_RTCERR2(SetRecPayloadType, channel, ToString(voe_codec));
@ -2306,8 +2313,8 @@ bool WebRtcVoiceMediaChannel::ConfigureRecvChannel(int channel) {
}
if (InConferenceMode()) {
// To be in par with the video, voe_channel() is not used for receiving in
// a conference call.
// To be in par with the video, default_send_channel_id() is not used for
// receiving in a conference call.
if (receive_channels_.empty() && default_receive_ssrc_ == 0 && playout_) {
// This is the first stream in a multi user meeting. We can now
// disable playback of the default stream. This since the default
@ -2316,9 +2323,10 @@ bool WebRtcVoiceMediaChannel::ConfigureRecvChannel(int channel) {
// the default channel will be mixed in with the other streams
// throughout the whole meeting, which might be disturbing.
LOG(LS_INFO) << "Disabling playback on the default voice channel";
SetPlayout(voe_channel(), false);
SetPlayout(default_send_channel_id(), false);
}
}
SetNack(channel, nack_enabled_);
// Set RTP header extension for the new channel.
@ -2355,7 +2363,7 @@ bool WebRtcVoiceMediaChannel::RemoveRecvStream(uint32_t ssrc) {
RTC_DCHECK(IsDefaultChannel(channel));
// Recycle the default channel is for recv stream.
if (playout_)
SetPlayout(voe_channel(), false);
SetPlayout(default_send_channel_id(), false);
default_receive_ssrc_ = 0;
return true;
@ -2383,7 +2391,7 @@ bool WebRtcVoiceMediaChannel::RemoveRecvStream(uint32_t ssrc) {
}
if (enable_default_channel_playout && playout_) {
LOG(LS_INFO) << "Enabling playback on the default voice channel";
SetPlayout(voe_channel(), true);
SetPlayout(default_send_channel_id(), true);
}
return true;
@ -2429,10 +2437,9 @@ bool WebRtcVoiceMediaChannel::GetActiveStreams(
int WebRtcVoiceMediaChannel::GetOutputLevel() {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
// return the highest output level of all streams
int highest = GetOutputLevel(voe_channel());
int highest = GetOutputLevel(default_send_channel_id());
for (const auto& ch : receive_channels_) {
int level = GetOutputLevel(ch.second->channel());
highest = std::max(level, highest);
highest = std::max(GetOutputLevel(ch.second->channel()), highest);
}
return highest;
}
@ -2471,7 +2478,7 @@ bool WebRtcVoiceMediaChannel::SetOutputVolume(uint32_t ssrc, double volume) {
// Default channel is not in receive_channels_ if it is not being used for
// playout.
if (default_receive_ssrc_ == 0)
channels.push_back(voe_channel());
channels.push_back(default_send_channel_id());
for (const auto& ch : receive_channels_) {
channels.push_back(ch.second->channel());
}
@ -2520,7 +2527,7 @@ bool WebRtcVoiceMediaChannel::InsertDtmf(uint32_t ssrc,
}
}
if (default_channel_is_inuse) {
channel = voe_channel();
channel = default_send_channel_id();
} else if (!send_channels_.empty()) {
channel = send_channels_.begin()->second->channel();
}
@ -2569,7 +2576,7 @@ void WebRtcVoiceMediaChannel::OnPacketReceived(
int which_channel =
GetReceiveChannelId(ParseSsrc(packet->data(), packet->size(), false));
if (which_channel == -1) {
which_channel = voe_channel();
which_channel = default_send_channel_id();
}
// Pass it off to the decoder.
@ -2631,7 +2638,8 @@ void WebRtcVoiceMediaChannel::OnRtcpReceived(
}
bool WebRtcVoiceMediaChannel::MuteStream(uint32_t ssrc, bool muted) {
int channel = (ssrc == 0) ? voe_channel() : GetSendChannelId(ssrc);
int channel =
(ssrc == 0) ? default_send_channel_id() : GetSendChannelId(ssrc);
if (channel == -1) {
LOG(LS_WARNING) << "The specified ssrc " << ssrc << " is not in use.";
return false;
@ -2825,7 +2833,7 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
channels.push_back(ch.second->channel());
}
if (channels.empty()) {
channels.push_back(voe_channel());
channels.push_back(default_send_channel_id());
}
// Get the SSRC and stats for each receiver, based on our own calculations.
@ -2923,17 +2931,18 @@ int WebRtcVoiceMediaChannel::GetOutputLevel(int channel) {
int WebRtcVoiceMediaChannel::GetReceiveChannelId(uint32_t ssrc) const {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
ChannelMap::const_iterator it = receive_channels_.find(ssrc);
if (it != receive_channels_.end())
if (it != receive_channels_.end()) {
return it->second->channel();
return (ssrc == default_receive_ssrc_) ? voe_channel() : -1;
}
return (ssrc == default_receive_ssrc_) ? default_send_channel_id() : -1;
}
int WebRtcVoiceMediaChannel::GetSendChannelId(uint32_t ssrc) const {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
ChannelMap::const_iterator it = send_channels_.find(ssrc);
if (it != send_channels_.end())
if (it != send_channels_.end()) {
return it->second->channel();
}
return -1;
}
@ -2988,8 +2997,8 @@ bool WebRtcVoiceMediaChannel::EnableRtcp(int channel) {
}
// TODO(juberti): Enable VQMon and RTCP XR reports, once we know what
// what we want to do with them.
// engine()->voe().EnableVQMon(voe_channel(), true);
// engine()->voe().EnableRTCP_XR(voe_channel(), true);
// engine()->voe().EnableVQMon(default_send_channel_id(), true);
// engine()->voe().EnableRTCP_XR(default_send_channel_id(), true);
return true;
}
@ -3136,8 +3145,9 @@ bool WebRtcVoiceMediaChannel::SetRecvCodecsInternal(
// TODO(xians): Figure out how we use the default channel in conference
// mode.
if (engine()->voe()->codec()->SetRecPayloadType(
voe_channel(), voe_codec) == -1) {
LOG_RTCERR2(SetRecPayloadType, voe_channel(), ToString(voe_codec));
default_send_channel_id(), voe_codec) == -1) {
LOG_RTCERR2(SetRecPayloadType, default_send_channel_id(),
ToString(voe_codec));
return false;
}
}

14
talk/media/webrtc/webrtcvoiceengine.h
View File

@ -180,8 +180,8 @@ class WebRtcVoiceMediaChannel : public VoiceMediaChannel,
webrtc::Call* call);
~WebRtcVoiceMediaChannel() override;
int voe_channel() const { return voe_channel_; }
bool valid() const { return voe_channel_ != -1; }
int default_send_channel_id() const { return default_send_channel_id_; }
bool valid() const { return default_send_channel_id_ != -1; }
const AudioOptions& options() const { return options_; }
bool SetSendParameters(const AudioSendParameters& params) override;
@ -286,7 +286,7 @@ class WebRtcVoiceMediaChannel : public VoiceMediaChannel,
return options_.conference_mode.GetWithDefaultIfUnset(false);
}
bool IsDefaultChannel(int channel_id) const {
return channel_id == voe_channel();
return channel_id == default_send_channel_id_;
}
bool SetSendCodecs(int channel, const std::vector<AudioCodec>& codecs);
bool SetSendBitrateInternal(int bps);
@ -308,7 +308,7 @@ class WebRtcVoiceMediaChannel : public VoiceMediaChannel,
rtc::ThreadChecker thread_checker_;
WebRtcVoiceEngine* const engine_;
const int voe_channel_;
const int default_send_channel_id_;
std::vector<AudioCodec> recv_codecs_;
std::vector<AudioCodec> send_codecs_;
rtc::scoped_ptr<webrtc::CodecInst> send_codec_;
@ -325,12 +325,12 @@ class WebRtcVoiceMediaChannel : public VoiceMediaChannel,
webrtc::Call* const call_;
// send_channels_ contains the channels which are being used for sending.
// When the default channel (voe_channel) is used for sending, it is
// contained in send_channels_, otherwise not.
// When the default channel (default_send_channel_id) is used for sending, it
// is contained in send_channels_, otherwise not.
ChannelMap send_channels_;
std::vector<RtpHeaderExtension> send_extensions_;
uint32_t default_receive_ssrc_;
// Note the default channel (voe_channel()) can reside in both
// Note the default channel (default_send_channel_id()) can reside in both
// receive_channels_ and send_channels_ in non-conference mode and in that
// case it will only be there if a non-zero default_receive_ssrc_ is set.
ChannelMap receive_channels_; // for multiple sources

14
talk/media/webrtc/webrtcvoiceengine_unittest.cc
View File

@ -2856,12 +2856,12 @@ TEST_F(WebRtcVoiceEngineTestFake, TestGetReceiveChannelIdIn1To1Calls) {
cricket::WebRtcVoiceMediaChannel* media_channel =
static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_);
// Test that GetChannelNum returns the default channel if the SSRC is unknown.
EXPECT_EQ(media_channel->voe_channel(),
EXPECT_EQ(media_channel->default_send_channel_id(),
media_channel->GetReceiveChannelId(0));
cricket::StreamParams stream;
stream.ssrcs.push_back(kSsrc2);
EXPECT_TRUE(channel_->AddRecvStream(stream));
EXPECT_EQ(media_channel->voe_channel(),
EXPECT_EQ(media_channel->default_send_channel_id(),
media_channel->GetReceiveChannelId(kSsrc2));
}
@ -2876,7 +2876,7 @@ TEST_F(WebRtcVoiceEngineTestFake, TestGetChannelNumInConferenceCalls) {
EXPECT_TRUE(channel_->AddRecvStream(stream));
cricket::WebRtcVoiceMediaChannel* media_channel =
static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_);
EXPECT_LT(media_channel->voe_channel(),
EXPECT_LT(media_channel->default_send_channel_id(),
media_channel->GetReceiveChannelId(kSsrc2));
}
@ -3081,9 +3081,7 @@ TEST_F(WebRtcVoiceEngineTestFake, AssociateChannelUnset1On1) {
// does not send RTCP SR.
TEST_F(WebRtcVoiceEngineTestFake, AssociateDefaultChannelOnSecondRecvChannel) {
EXPECT_TRUE(SetupEngine());
cricket::WebRtcVoiceMediaChannel* media_channel =
static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_);
int default_channel = media_channel->voe_channel();
int default_channel = voe_.GetLastChannel();
EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(1)));
int recv_ch_1 = voe_.GetLastChannel();
EXPECT_EQ(recv_ch_1, default_channel);
@ -3103,9 +3101,7 @@ TEST_F(WebRtcVoiceEngineTestFake, AssociateDefaultChannelOnConference) {
EXPECT_TRUE(SetupEngine());
send_parameters_.options = options_conference_;
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
cricket::WebRtcVoiceMediaChannel* media_channel =
static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_);
int default_channel = media_channel->voe_channel();
int default_channel = voe_.GetLastChannel();
EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(1)));
int recv_ch = voe_.GetLastChannel();
EXPECT_NE(recv_ch, default_channel);