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Remove default receive channel from WVoE; baby step 1.

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Rx AGC config bits copied from https://codereview.webrtc.org/1315903004.

BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1388723002

Cr-Commit-Position: refs/heads/master@{#10233}
This commit is contained in:
solenberg
2015-10-09 01:37:09 -07:00
committed by Commit bot
parent fb6c02d5a1
commit 0b67546d8c
4 changed files with 56 additions and 133 deletions
Download Patch File Download Diff File Expand all files Collapse all files

19
talk/media/base/mediachannel.h
View File

@ -160,7 +160,6 @@ struct AudioOptions {
void SetAll(const AudioOptions& change) {
echo_cancellation.SetFrom(change.echo_cancellation);
auto_gain_control.SetFrom(change.auto_gain_control);
rx_auto_gain_control.SetFrom(change.rx_auto_gain_control);
noise_suppression.SetFrom(change.noise_suppression);
highpass_filter.SetFrom(change.highpass_filter);
stereo_swapping.SetFrom(change.stereo_swapping);
@ -181,10 +180,6 @@ struct AudioOptions {
tx_agc_digital_compression_gain.SetFrom(
change.tx_agc_digital_compression_gain);
tx_agc_limiter.SetFrom(change.tx_agc_limiter);
rx_agc_target_dbov.SetFrom(change.rx_agc_target_dbov);
rx_agc_digital_compression_gain.SetFrom(
change.rx_agc_digital_compression_gain);
rx_agc_limiter.SetFrom(change.rx_agc_limiter);
recording_sample_rate.SetFrom(change.recording_sample_rate);
playout_sample_rate.SetFrom(change.playout_sample_rate);
dscp.SetFrom(change.dscp);
@ -194,7 +189,6 @@ struct AudioOptions {
bool operator==(const AudioOptions& o) const {
return echo_cancellation == o.echo_cancellation &&
auto_gain_control == o.auto_gain_control &&
rx_auto_gain_control == o.rx_auto_gain_control &&
noise_suppression == o.noise_suppression &&
highpass_filter == o.highpass_filter &&
stereo_swapping == o.stereo_swapping &&
@ -213,9 +207,6 @@ struct AudioOptions {
tx_agc_target_dbov == o.tx_agc_target_dbov &&
tx_agc_digital_compression_gain == o.tx_agc_digital_compression_gain &&
tx_agc_limiter == o.tx_agc_limiter &&
rx_agc_target_dbov == o.rx_agc_target_dbov &&
rx_agc_digital_compression_gain == o.rx_agc_digital_compression_gain &&
rx_agc_limiter == o.rx_agc_limiter &&
recording_sample_rate == o.recording_sample_rate &&
playout_sample_rate == o.playout_sample_rate &&
dscp == o.dscp &&
@ -227,7 +218,6 @@ struct AudioOptions {
ost << "AudioOptions {";
ost << ToStringIfSet("aec", echo_cancellation);
ost << ToStringIfSet("agc", auto_gain_control);
ost << ToStringIfSet("rx_agc", rx_auto_gain_control);
ost << ToStringIfSet("ns", noise_suppression);
ost << ToStringIfSet("hf", highpass_filter);
ost << ToStringIfSet("swap", stereo_swapping);
@ -248,10 +238,6 @@ struct AudioOptions {
ost << ToStringIfSet("tx_agc_digital_compression_gain",
tx_agc_digital_compression_gain);
ost << ToStringIfSet("tx_agc_limiter", tx_agc_limiter);
ost << ToStringIfSet("rx_agc_target_dbov", rx_agc_target_dbov);
ost << ToStringIfSet("rx_agc_digital_compression_gain",
rx_agc_digital_compression_gain);
ost << ToStringIfSet("rx_agc_limiter", rx_agc_limiter);
ost << ToStringIfSet("recording_sample_rate", recording_sample_rate);
ost << ToStringIfSet("playout_sample_rate", playout_sample_rate);
ost << ToStringIfSet("dscp", dscp);
@ -265,8 +251,6 @@ struct AudioOptions {
Settable<bool> echo_cancellation;
// Audio processing to adjust the sensitivity of the local mic dynamically.
Settable<bool> auto_gain_control;
// Audio processing to apply gain to the remote audio.
Settable<bool> rx_auto_gain_control;
// Audio processing to filter out background noise.
Settable<bool> noise_suppression;
// Audio processing to remove background noise of lower frequencies.
@ -291,9 +275,6 @@ struct AudioOptions {
Settable<uint16_t> tx_agc_target_dbov;
Settable<uint16_t> tx_agc_digital_compression_gain;
Settable<bool> tx_agc_limiter;
Settable<uint16_t> rx_agc_target_dbov;
Settable<uint16_t> rx_agc_digital_compression_gain;
Settable<bool> rx_agc_limiter;
Settable<uint32_t> recording_sample_rate;
Settable<uint32_t> playout_sample_rate;
// Set DSCP value for packet sent from audio channel.

33
talk/media/webrtc/fakewebrtcvoiceengine.h
View File

@ -208,8 +208,6 @@ class FakeWebRtcVoiceEngine
opus_dtx(false),
red(false),
nack(false),
rx_agc_enabled(false),
rx_agc_mode(webrtc::kAgcDefault),
cn8_type(13),
cn16_type(105),
dtmf_type(106),
@ -224,7 +222,6 @@ class FakeWebRtcVoiceEngine
neteq_capacity(-1),
neteq_fast_accelerate(false) {
memset(&send_codec, 0, sizeof(send_codec));
memset(&rx_agc_config, 0, sizeof(rx_agc_config));
}
bool external_transport;
bool send;
@ -238,9 +235,6 @@ class FakeWebRtcVoiceEngine
bool opus_dtx;
bool red;
bool nack;
bool rx_agc_enabled;
webrtc::AgcModes rx_agc_mode;
webrtc::AgcConfig rx_agc_config;
int cn8_type;
int cn16_type;
int dtmf_type;
@ -1001,27 +995,12 @@ class FakeWebRtcVoiceEngine
WEBRTC_STUB(SetRxNsStatus, (int channel, bool enable, webrtc::NsModes mode));
WEBRTC_STUB(GetRxNsStatus, (int channel, bool& enabled,
webrtc::NsModes& mode));
WEBRTC_FUNC(SetRxAgcStatus, (int channel, bool enable,
webrtc::AgcModes mode)) {
channels_[channel]->rx_agc_enabled = enable;
channels_[channel]->rx_agc_mode = mode;
return 0;
}
WEBRTC_FUNC(GetRxAgcStatus, (int channel, bool& enabled,
webrtc::AgcModes& mode)) {
enabled = channels_[channel]->rx_agc_enabled;
mode = channels_[channel]->rx_agc_mode;
return 0;
}
WEBRTC_FUNC(SetRxAgcConfig, (int channel, webrtc::AgcConfig config)) {
channels_[channel]->rx_agc_config = config;
return 0;
}
WEBRTC_FUNC(GetRxAgcConfig, (int channel, webrtc::AgcConfig& config)) {
config = channels_[channel]->rx_agc_config;
return 0;
}
WEBRTC_STUB(SetRxAgcStatus, (int channel, bool enable,
webrtc::AgcModes mode));
WEBRTC_STUB(GetRxAgcStatus, (int channel, bool& enabled,
webrtc::AgcModes& mode));
WEBRTC_STUB(SetRxAgcConfig, (int channel, webrtc::AgcConfig config));
WEBRTC_STUB(GetRxAgcConfig, (int channel, webrtc::AgcConfig& config));
WEBRTC_STUB(RegisterRxVadObserver, (int, webrtc::VoERxVadCallback&));
WEBRTC_STUB(DeRegisterRxVadObserver, (int channel));

120
talk/media/webrtc/webrtcvoiceengine.cc
View File

@ -148,6 +148,18 @@ const char kAecDumpByAudioOptionFilename[] = "/sdcard/audio.aecdump";
const char kAecDumpByAudioOptionFilename[] = "audio.aecdump";
#endif
bool ValidateStreamParams(const StreamParams& sp) {
if (sp.ssrcs.empty()) {
LOG(LS_ERROR) << "No SSRCs in stream parameters: " << sp.ToString();
return false;
}
if (sp.ssrcs.size() > 1) {
LOG(LS_ERROR) << "Multiple SSRCs in stream parameters: " << sp.ToString();
return false;
}
return true;
}
// Dumps an AudioCodec in RFC 2327-ish format.
std::string ToString(const AudioCodec& codec) {
std::stringstream ss;
@ -221,6 +233,19 @@ bool FindCodec(const std::vector<AudioCodec>& codecs,
return false;
}
bool VerifyUniquePayloadTypes(const std::vector<AudioCodec>& codecs) {
if (codecs.empty()) {
return true;
}
std::vector<int> payload_types;
for (const AudioCodec& codec : codecs) {
payload_types.push_back(codec.id);
}
std::sort(payload_types.begin(), payload_types.end());
auto it = std::unique(payload_types.begin(), payload_types.end());
return it == payload_types.end();
}
bool IsNackEnabled(const AudioCodec& codec) {
return codec.HasFeedbackParam(FeedbackParam(kRtcpFbParamNack,
kParamValueEmpty));
@ -1445,9 +1470,6 @@ bool WebRtcVoiceMediaChannel::SetOptions(const AudioOptions& options) {
// Check if DSCP value is changed from previous.
bool dscp_option_changed = (options_.dscp != options.dscp);
// TODO(xians): Add support to set different options for different send
// streams after we support multiple APMs.
// We retain all of the existing options, and apply the given ones
// on top. This means there is no way to "clear" options such that
// they go back to the engine default.
@ -1461,55 +1483,6 @@ bool WebRtcVoiceMediaChannel::SetOptions(const AudioOptions& options) {
}
}
// Receiver-side auto gain control happens per channel, so set it here from
// options. Note that, like conference mode, setting it on the engine won't
// have the desired effect, since voice channels don't inherit options from
// the media engine when those options are applied per-channel.
bool rx_auto_gain_control;
if (options.rx_auto_gain_control.Get(&rx_auto_gain_control)) {
if (engine()->voe()->processing()->SetRxAgcStatus(
voe_channel(), rx_auto_gain_control,
webrtc::kAgcFixedDigital) == -1) {
LOG_RTCERR1(SetRxAgcStatus, rx_auto_gain_control);
return false;
} else {
LOG(LS_VERBOSE) << "Rx auto gain set to " << rx_auto_gain_control
<< " with mode " << webrtc::kAgcFixedDigital;
}
}
if (options.rx_agc_target_dbov.IsSet() ||
options.rx_agc_digital_compression_gain.IsSet() ||
options.rx_agc_limiter.IsSet()) {
webrtc::AgcConfig config;
// If only some of the options are being overridden, get the current
// settings for the channel and bail if they aren't available.
if (!options.rx_agc_target_dbov.IsSet() ||
!options.rx_agc_digital_compression_gain.IsSet() ||
!options.rx_agc_limiter.IsSet()) {
if (engine()->voe()->processing()->GetRxAgcConfig(
voe_channel(), config) != 0) {
LOG(LS_ERROR) << "Failed to get default rx agc configuration for "
<< "channel " << voe_channel() << ". Since not all rx "
<< "agc options are specified, unable to safely set rx "
<< "agc options.";
return false;
}
}
config.targetLeveldBOv =
options.rx_agc_target_dbov.GetWithDefaultIfUnset(
config.targetLeveldBOv);
config.digitalCompressionGaindB =
options.rx_agc_digital_compression_gain.GetWithDefaultIfUnset(
config.digitalCompressionGaindB);
config.limiterEnable = options.rx_agc_limiter.GetWithDefaultIfUnset(
config.limiterEnable);
if (engine()->voe()->processing()->SetRxAgcConfig(
voe_channel(), config) == -1) {
LOG_RTCERR4(SetRxAgcConfig, voe_channel(), config.targetLeveldBOv,
config.digitalCompressionGaindB, config.limiterEnable);
return false;
}
}
if (dscp_option_changed) {
rtc::DiffServCodePoint dscp = rtc::DSCP_DEFAULT;
if (options_.dscp.GetWithDefaultIfUnset(false))
@ -1518,9 +1491,7 @@ 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;
@ -1528,9 +1499,14 @@ 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:";
LOG(LS_INFO) << "Setting receive voice codecs.";
RTC_DCHECK(thread_checker_.CalledOnValidThread());
if (!VerifyUniquePayloadTypes(codecs)) {
LOG(LS_ERROR) << "Codec payload types overlap.";
return false;
}
std::vector<AudioCodec> new_codecs;
// Find all new codecs. We allow adding new codecs but don't allow changing
@ -2229,17 +2205,18 @@ bool WebRtcVoiceMediaChannel::AddRecvStream(const StreamParams& sp) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
LOG(LS_INFO) << "AddRecvStream: " << sp.ToString();
rtc::CritScope lock(&receive_channels_cs_);
if (!VERIFY(sp.ssrcs.size() == 1))
return false;
uint32_t ssrc = sp.first_ssrc();
if (ssrc == 0) {
LOG(LS_WARNING) << "AddRecvStream with 0 ssrc is not supported.";
if (!ValidateStreamParams(sp)) {
return false;
}
uint32_t ssrc = sp.first_ssrc();
if (ssrc == 0) {
LOG(LS_WARNING) << "AddRecvStream with ssrc==0 is not supported.";
return false;
}
rtc::CritScope lock(&receive_channels_cs_);
if (receive_channels_.find(ssrc) != receive_channels_.end()) {
LOG(LS_ERROR) << "Stream already exists with ssrc " << ssrc;
return false;
@ -2667,16 +2644,19 @@ void WebRtcVoiceMediaChannel::OnRtcpReceived(
return;
}
// If it is a sender report, find the channel that is listening.
// If it is a sender report, find the receive channel that is listening.
bool has_sent_to_default_channel = false;
if (type == kRtcpTypeSR) {
int which_channel =
GetReceiveChannelId(ParseSsrc(packet->data(), packet->size(), true));
if (which_channel != -1) {
uint32_t ssrc = 0;
if (!GetRtcpSsrc(packet->data(), packet->size(), &ssrc)) {
return;
}
int recv_channel_id = GetReceiveChannelId(ssrc);
if (recv_channel_id != -1) {
engine()->voe()->network()->ReceivedRTCPPacket(
which_channel, packet->data(), packet->size());
recv_channel_id, packet->data(), packet->size());
if (IsDefaultChannel(which_channel))
if (IsDefaultChannel(recv_channel_id))
has_sent_to_default_channel = true;
}
}

17
talk/media/webrtc/webrtcvoiceengine_unittest.cc
View File

@ -2219,23 +2219,6 @@ TEST_F(WebRtcVoiceEngineTestFake, TxAgcConfigViaOptions) {
EXPECT_EQ(13, agc_config.targetLeveldBOv);
}
TEST_F(WebRtcVoiceEngineTestFake, RxAgcConfigViaOptions) {
EXPECT_TRUE(SetupEngine());
int channel_num = voe_.GetLastChannel();
send_parameters_.options.rx_agc_target_dbov.Set(6);
send_parameters_.options.rx_agc_digital_compression_gain.Set(0);
send_parameters_.options.rx_agc_limiter.Set(true);
send_parameters_.options.rx_auto_gain_control.Set(true);
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
webrtc::AgcConfig agc_config;
EXPECT_EQ(0, engine_.voe()->processing()->GetRxAgcConfig(
channel_num, agc_config));
EXPECT_EQ(6, agc_config.targetLeveldBOv);
EXPECT_EQ(0, agc_config.digitalCompressionGaindB);
EXPECT_TRUE(agc_config.limiterEnable);
}
TEST_F(WebRtcVoiceEngineTestFake, SampleRatesViaOptions) {
EXPECT_TRUE(SetupEngine());
cricket::AudioOptions options;