Revert of Implement AudioReceiveStream::GetStats(). (patchset #19 id:360001 of https://codereview.webrtc.org/1390753002/ )
Reason for revert:
webrtc_perf_tests started failing on Win32 Release, Mac32 Release and Linux64 Release (all running large tests). These were not caught by try bots.
Original issue's description:
> Implement AudioReceiveStream::GetStats().
>
> R=tommi@webrtc.org
> TBR=hta@webrtc.org
> BUG=webrtc:4690
>
> Committed: https://chromium.googlesource.com/external/webrtc/+/a457752f4afc496ed7f4d6b584b08d8635f18cc0
TBR=tommi@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1411083006
Cr-Commit-Position: refs/heads/master@{#10340}
diff --git a/talk/media/webrtc/fakewebrtccall.cc b/talk/media/webrtc/fakewebrtccall.cc
index 399ab13..a0386b0 100644
--- a/talk/media/webrtc/fakewebrtccall.cc
+++ b/talk/media/webrtc/fakewebrtccall.cc
@@ -40,16 +40,15 @@
RTC_DCHECK(config.voe_channel_id != -1);
}
+webrtc::AudioReceiveStream::Stats FakeAudioReceiveStream::GetStats() const {
+ return webrtc::AudioReceiveStream::Stats();
+}
+
const webrtc::AudioReceiveStream::Config&
FakeAudioReceiveStream::GetConfig() const {
return config_;
}
-void FakeAudioReceiveStream::SetStats(
- const webrtc::AudioReceiveStream::Stats& stats) {
- stats_ = stats;
-}
-
void FakeAudioReceiveStream::IncrementReceivedPackets() {
received_packets_++;
}
diff --git a/talk/media/webrtc/fakewebrtccall.h b/talk/media/webrtc/fakewebrtccall.h
index 0ec854f..fb271f2 100644
--- a/talk/media/webrtc/fakewebrtccall.h
+++ b/talk/media/webrtc/fakewebrtccall.h
@@ -42,8 +42,11 @@
explicit FakeAudioReceiveStream(
const webrtc::AudioReceiveStream::Config& config);
+ // webrtc::AudioReceiveStream implementation.
+ webrtc::AudioReceiveStream::Stats GetStats() const override;
+
const webrtc::AudioReceiveStream::Config& GetConfig() const;
- void SetStats(const webrtc::AudioReceiveStream::Stats& stats);
+
int received_packets() const { return received_packets_; }
void IncrementReceivedPackets();
@@ -61,13 +64,7 @@
return true;
}
- // webrtc::AudioReceiveStream implementation.
- webrtc::AudioReceiveStream::Stats GetStats() const override {
- return stats_;
- }
-
webrtc::AudioReceiveStream::Config config_;
- webrtc::AudioReceiveStream::Stats stats_;
int received_packets_;
};
diff --git a/talk/media/webrtc/fakewebrtcvoiceengine.h b/talk/media/webrtc/fakewebrtcvoiceengine.h
index 9b91327..1167b6b 100644
--- a/talk/media/webrtc/fakewebrtcvoiceengine.h
+++ b/talk/media/webrtc/fakewebrtcvoiceengine.h
@@ -65,6 +65,25 @@
static const int kOpusBandwidthSwb = 12000;
static const int kOpusBandwidthFb = 20000;
+static const webrtc::NetworkStatistics kNetStats = {
+ 1, // uint16_t currentBufferSize;
+ 2, // uint16_t preferredBufferSize;
+ true, // bool jitterPeaksFound;
+ 1234, // uint16_t currentPacketLossRate;
+ 567, // uint16_t currentDiscardRate;
+ 8901, // uint16_t currentExpandRate;
+ 234, // uint16_t currentSpeechExpandRate;
+ 5678, // uint16_t currentPreemptiveRate;
+ 9012, // uint16_t currentAccelerateRate;
+ 3456, // uint16_t currentSecondaryDecodedRate;
+ 7890, // int32_t clockDriftPPM;
+ 54, // meanWaitingTimeMs;
+ 32, // int medianWaitingTimeMs;
+ 1, // int minWaitingTimeMs;
+ 98, // int maxWaitingTimeMs;
+ 7654, // int addedSamples;
+}; // These random but non-trivial numbers are used for testing.
+
#define WEBRTC_CHECK_CHANNEL(channel) \
if (channels_.find(channel) == channels_.end()) return -1;
@@ -162,9 +181,9 @@
class FakeWebRtcVoiceEngine
: public webrtc::VoEAudioProcessing,
public webrtc::VoEBase, public webrtc::VoECodec, public webrtc::VoEDtmf,
- public webrtc::VoEHardware,
+ public webrtc::VoEHardware, public webrtc::VoENetEqStats,
public webrtc::VoENetwork, public webrtc::VoERTP_RTCP,
- public webrtc::VoEVolumeControl {
+ public webrtc::VoEVideoSync, public webrtc::VoEVolumeControl {
public:
struct DtmfInfo {
DtmfInfo()
@@ -508,7 +527,26 @@
return 0;
}
WEBRTC_STUB(SetBitRate, (int channel, int bitrate_bps));
- WEBRTC_STUB(GetRecCodec, (int channel, webrtc::CodecInst& codec));
+ WEBRTC_FUNC(GetRecCodec, (int channel, webrtc::CodecInst& codec)) {
+ WEBRTC_CHECK_CHANNEL(channel);
+ const Channel* c = channels_[channel];
+ for (std::list<std::string>::const_iterator it_packet = c->packets.begin();
+ it_packet != c->packets.end(); ++it_packet) {
+ int pltype;
+ if (!GetRtpPayloadType(it_packet->data(), it_packet->length(), &pltype)) {
+ continue;
+ }
+ for (std::vector<webrtc::CodecInst>::const_iterator it_codec =
+ c->recv_codecs.begin(); it_codec != c->recv_codecs.end();
+ ++it_codec) {
+ if (it_codec->pltype == pltype) {
+ codec = *it_codec;
+ return 0;
+ }
+ }
+ }
+ return -1;
+ }
WEBRTC_FUNC(SetRecPayloadType, (int channel,
const webrtc::CodecInst& codec)) {
WEBRTC_CHECK_CHANNEL(channel);
@@ -687,6 +725,20 @@
WEBRTC_STUB(EnableBuiltInNS, (bool enable));
virtual bool BuiltInNSIsAvailable() const { return false; }
+ // webrtc::VoENetEqStats
+ WEBRTC_FUNC(GetNetworkStatistics, (int channel,
+ webrtc::NetworkStatistics& ns)) {
+ WEBRTC_CHECK_CHANNEL(channel);
+ memcpy(&ns, &kNetStats, sizeof(webrtc::NetworkStatistics));
+ return 0;
+ }
+
+ WEBRTC_FUNC_CONST(GetDecodingCallStatistics, (int channel,
+ webrtc::AudioDecodingCallStats*)) {
+ WEBRTC_CHECK_CHANNEL(channel);
+ return 0;
+ }
+
// webrtc::VoENetwork
WEBRTC_FUNC(RegisterExternalTransport, (int channel,
webrtc::Transport& transport)) {
@@ -835,6 +887,18 @@
return 0;
}
+ // webrtc::VoEVideoSync
+ WEBRTC_STUB(GetPlayoutBufferSize, (int& bufferMs));
+ WEBRTC_STUB(GetPlayoutTimestamp, (int channel, unsigned int& timestamp));
+ WEBRTC_STUB(GetRtpRtcp, (int, webrtc::RtpRtcp**, webrtc::RtpReceiver**));
+ WEBRTC_STUB(SetInitTimestamp, (int channel, unsigned int timestamp));
+ WEBRTC_STUB(SetInitSequenceNumber, (int channel, short sequenceNumber));
+ WEBRTC_STUB(SetMinimumPlayoutDelay, (int channel, int delayMs));
+ WEBRTC_STUB(SetInitialPlayoutDelay, (int channel, int delay_ms));
+ WEBRTC_STUB(GetDelayEstimate, (int channel, int* jitter_buffer_delay_ms,
+ int* playout_buffer_delay_ms));
+ WEBRTC_STUB_CONST(GetLeastRequiredDelayMs, (int channel));
+
// webrtc::VoEVolumeControl
WEBRTC_STUB(SetSpeakerVolume, (unsigned int));
WEBRTC_STUB(GetSpeakerVolume, (unsigned int&));
diff --git a/talk/media/webrtc/webrtcvoe.h b/talk/media/webrtc/webrtcvoe.h
index db6a64a..844831f 100644
--- a/talk/media/webrtc/webrtcvoe.h
+++ b/talk/media/webrtc/webrtcvoe.h
@@ -38,9 +38,13 @@
#include "webrtc/voice_engine/include/voe_codec.h"
#include "webrtc/voice_engine/include/voe_dtmf.h"
#include "webrtc/voice_engine/include/voe_errors.h"
+#include "webrtc/voice_engine/include/voe_external_media.h"
+#include "webrtc/voice_engine/include/voe_file.h"
#include "webrtc/voice_engine/include/voe_hardware.h"
+#include "webrtc/voice_engine/include/voe_neteq_stats.h"
#include "webrtc/voice_engine/include/voe_network.h"
#include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
+#include "webrtc/voice_engine/include/voe_video_sync.h"
#include "webrtc/voice_engine/include/voe_volume_control.h"
namespace cricket {
@@ -92,16 +96,18 @@
VoEWrapper()
: engine_(webrtc::VoiceEngine::Create()), processing_(engine_),
base_(engine_), codec_(engine_), dtmf_(engine_),
- hw_(engine_), network_(engine_),
- rtp_(engine_), volume_(engine_) {
+ hw_(engine_), neteq_(engine_), network_(engine_),
+ rtp_(engine_), sync_(engine_), volume_(engine_) {
}
VoEWrapper(webrtc::VoEAudioProcessing* processing,
webrtc::VoEBase* base,
webrtc::VoECodec* codec,
webrtc::VoEDtmf* dtmf,
webrtc::VoEHardware* hw,
+ webrtc::VoENetEqStats* neteq,
webrtc::VoENetwork* network,
webrtc::VoERTP_RTCP* rtp,
+ webrtc::VoEVideoSync* sync,
webrtc::VoEVolumeControl* volume)
: engine_(NULL),
processing_(processing),
@@ -109,8 +115,10 @@
codec_(codec),
dtmf_(dtmf),
hw_(hw),
+ neteq_(neteq),
network_(network),
rtp_(rtp),
+ sync_(sync),
volume_(volume) {
}
~VoEWrapper() {}
@@ -120,8 +128,10 @@
webrtc::VoECodec* codec() const { return codec_.get(); }
webrtc::VoEDtmf* dtmf() const { return dtmf_.get(); }
webrtc::VoEHardware* hw() const { return hw_.get(); }
+ webrtc::VoENetEqStats* neteq() const { return neteq_.get(); }
webrtc::VoENetwork* network() const { return network_.get(); }
webrtc::VoERTP_RTCP* rtp() const { return rtp_.get(); }
+ webrtc::VoEVideoSync* sync() const { return sync_.get(); }
webrtc::VoEVolumeControl* volume() const { return volume_.get(); }
int error() { return base_->LastError(); }
@@ -132,8 +142,10 @@
scoped_voe_ptr<webrtc::VoECodec> codec_;
scoped_voe_ptr<webrtc::VoEDtmf> dtmf_;
scoped_voe_ptr<webrtc::VoEHardware> hw_;
+ scoped_voe_ptr<webrtc::VoENetEqStats> neteq_;
scoped_voe_ptr<webrtc::VoENetwork> network_;
scoped_voe_ptr<webrtc::VoERTP_RTCP> rtp_;
+ scoped_voe_ptr<webrtc::VoEVideoSync> sync_;
scoped_voe_ptr<webrtc::VoEVolumeControl> volume_;
};
diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc
index d880e4b..2a3df2d 100644
--- a/talk/media/webrtc/webrtcvoiceengine.cc
+++ b/talk/media/webrtc/webrtcvoiceengine.cc
@@ -2694,6 +2694,11 @@
}
}
+ webrtc::CallStatistics cs;
+ unsigned int ssrc;
+ webrtc::CodecInst codec;
+ unsigned int level;
+
for (const auto& ch : send_channels_) {
const int channel = ch.second->channel();
@@ -2701,8 +2706,6 @@
// remote side told us it got from its RTCP report.
VoiceSenderInfo sinfo;
- webrtc::CallStatistics cs = {0};
- unsigned int ssrc = 0;
if (engine()->voe()->rtp()->GetRTCPStatistics(channel, cs) == -1 ||
engine()->voe()->rtp()->GetLocalSSRC(channel, ssrc) == -1) {
continue;
@@ -2723,7 +2726,6 @@
sinfo.packets_lost = -1;
sinfo.ext_seqnum = -1;
std::vector<webrtc::ReportBlock> receive_blocks;
- webrtc::CodecInst codec = {0};
if (engine()->voe()->rtp()->GetRemoteRTCPReportBlocks(
channel, &receive_blocks) != -1 &&
engine()->voe()->codec()->GetSendCodec(channel, codec) != -1) {
@@ -2744,7 +2746,6 @@
}
// Local speech level.
- unsigned int level = 0;
sinfo.audio_level = (engine()->voe()->volume()->
GetSpeechInputLevelFullRange(level) != -1) ? level : -1;
@@ -2765,36 +2766,76 @@
}
// Get the SSRC and stats for each receiver.
- info->receivers.clear();
- for (const auto& stream : receive_streams_) {
- webrtc::AudioReceiveStream::Stats stats = stream.second->GetStats();
- VoiceReceiverInfo rinfo;
- rinfo.add_ssrc(stats.remote_ssrc);
- rinfo.bytes_rcvd = stats.bytes_rcvd;
- rinfo.packets_rcvd = stats.packets_rcvd;
- rinfo.packets_lost = stats.packets_lost;
- rinfo.fraction_lost = stats.fraction_lost;
- rinfo.codec_name = stats.codec_name;
- rinfo.ext_seqnum = stats.ext_seqnum;
- rinfo.jitter_ms = stats.jitter_ms;
- rinfo.jitter_buffer_ms = stats.jitter_buffer_ms;
- rinfo.jitter_buffer_preferred_ms = stats.jitter_buffer_preferred_ms;
- rinfo.delay_estimate_ms = stats.delay_estimate_ms;
- rinfo.audio_level = stats.audio_level;
- rinfo.expand_rate = stats.expand_rate;
- rinfo.speech_expand_rate = stats.speech_expand_rate;
- rinfo.secondary_decoded_rate = stats.secondary_decoded_rate;
- rinfo.accelerate_rate = stats.accelerate_rate;
- rinfo.preemptive_expand_rate = stats.preemptive_expand_rate;
- rinfo.decoding_calls_to_silence_generator =
- stats.decoding_calls_to_silence_generator;
- rinfo.decoding_calls_to_neteq = stats.decoding_calls_to_neteq;
- rinfo.decoding_normal = stats.decoding_normal;
- rinfo.decoding_plc = stats.decoding_plc;
- rinfo.decoding_cng = stats.decoding_cng;
- rinfo.decoding_plc_cng = stats.decoding_plc_cng;
- rinfo.capture_start_ntp_time_ms = stats.capture_start_ntp_time_ms;
- info->receivers.push_back(rinfo);
+ for (const auto& ch : receive_channels_) {
+ int ch_id = ch.second->channel();
+ memset(&cs, 0, sizeof(cs));
+ if (engine()->voe()->rtp()->GetRemoteSSRC(ch_id, ssrc) != -1 &&
+ engine()->voe()->rtp()->GetRTCPStatistics(ch_id, cs) != -1 &&
+ engine()->voe()->codec()->GetRecCodec(ch_id, codec) != -1) {
+ VoiceReceiverInfo rinfo;
+ rinfo.add_ssrc(ssrc);
+ rinfo.bytes_rcvd = cs.bytesReceived;
+ rinfo.packets_rcvd = cs.packetsReceived;
+ // The next four fields are from the most recently sent RTCP report.
+ // Convert Q8 to floating point.
+ rinfo.fraction_lost = static_cast<float>(cs.fractionLost) / (1 << 8);
+ rinfo.packets_lost = cs.cumulativeLost;
+ rinfo.ext_seqnum = cs.extendedMax;
+ rinfo.capture_start_ntp_time_ms = cs.capture_start_ntp_time_ms_;
+ if (codec.pltype != -1) {
+ rinfo.codec_name = codec.plname;
+ }
+ // Convert samples to milliseconds.
+ if (codec.plfreq / 1000 > 0) {
+ rinfo.jitter_ms = cs.jitterSamples / (codec.plfreq / 1000);
+ }
+
+ // Get jitter buffer and total delay (alg + jitter + playout) stats.
+ webrtc::NetworkStatistics ns;
+ if (engine()->voe()->neteq() &&
+ engine()->voe()->neteq()->GetNetworkStatistics(
+ ch_id, ns) != -1) {
+ rinfo.jitter_buffer_ms = ns.currentBufferSize;
+ rinfo.jitter_buffer_preferred_ms = ns.preferredBufferSize;
+ rinfo.expand_rate =
+ static_cast<float>(ns.currentExpandRate) / (1 << 14);
+ rinfo.speech_expand_rate =
+ static_cast<float>(ns.currentSpeechExpandRate) / (1 << 14);
+ rinfo.secondary_decoded_rate =
+ static_cast<float>(ns.currentSecondaryDecodedRate) / (1 << 14);
+ rinfo.accelerate_rate =
+ static_cast<float>(ns.currentAccelerateRate) / (1 << 14);
+ rinfo.preemptive_expand_rate =
+ static_cast<float>(ns.currentPreemptiveRate) / (1 << 14);
+ }
+
+ webrtc::AudioDecodingCallStats ds;
+ if (engine()->voe()->neteq() &&
+ engine()->voe()->neteq()->GetDecodingCallStatistics(
+ ch_id, &ds) != -1) {
+ rinfo.decoding_calls_to_silence_generator =
+ ds.calls_to_silence_generator;
+ rinfo.decoding_calls_to_neteq = ds.calls_to_neteq;
+ rinfo.decoding_normal = ds.decoded_normal;
+ rinfo.decoding_plc = ds.decoded_plc;
+ rinfo.decoding_cng = ds.decoded_cng;
+ rinfo.decoding_plc_cng = ds.decoded_plc_cng;
+ }
+
+ if (engine()->voe()->sync()) {
+ int jitter_buffer_delay_ms = 0;
+ int playout_buffer_delay_ms = 0;
+ engine()->voe()->sync()->GetDelayEstimate(
+ ch_id, &jitter_buffer_delay_ms, &playout_buffer_delay_ms);
+ rinfo.delay_estimate_ms = jitter_buffer_delay_ms +
+ playout_buffer_delay_ms;
+ }
+
+ // Get speech level.
+ rinfo.audio_level = (engine()->voe()->volume()->
+ GetSpeechOutputLevelFullRange(ch_id, level) != -1) ? level : -1;
+ info->receivers.push_back(rinfo);
+ }
}
return true;
diff --git a/talk/media/webrtc/webrtcvoiceengine_unittest.cc b/talk/media/webrtc/webrtcvoiceengine_unittest.cc
index 491af19..b0fc2bb 100644
--- a/talk/media/webrtc/webrtcvoiceengine_unittest.cc
+++ b/talk/media/webrtc/webrtcvoiceengine_unittest.cc
@@ -38,27 +38,27 @@
#include "webrtc/p2p/base/faketransportcontroller.h"
#include "talk/session/media/channel.h"
+// Tests for the WebRtcVoiceEngine/VoiceChannel code.
+
using cricket::kRtpAudioLevelHeaderExtension;
using cricket::kRtpAbsoluteSenderTimeHeaderExtension;
-namespace {
-
-const cricket::AudioCodec kPcmuCodec(0, "PCMU", 8000, 64000, 1, 0);
-const cricket::AudioCodec kIsacCodec(103, "ISAC", 16000, 32000, 1, 0);
-const cricket::AudioCodec kOpusCodec(111, "opus", 48000, 64000, 2, 0);
-const cricket::AudioCodec kG722CodecVoE(9, "G722", 16000, 64000, 1, 0);
-const cricket::AudioCodec kG722CodecSdp(9, "G722", 8000, 64000, 1, 0);
-const cricket::AudioCodec kRedCodec(117, "red", 8000, 0, 1, 0);
-const cricket::AudioCodec kCn8000Codec(13, "CN", 8000, 0, 1, 0);
-const cricket::AudioCodec kCn16000Codec(105, "CN", 16000, 0, 1, 0);
-const cricket::AudioCodec kTelephoneEventCodec(106, "telephone-event", 8000, 0,
- 1, 0);
-const cricket::AudioCodec* const kAudioCodecs[] = {
+static const cricket::AudioCodec kPcmuCodec(0, "PCMU", 8000, 64000, 1, 0);
+static const cricket::AudioCodec kIsacCodec(103, "ISAC", 16000, 32000, 1, 0);
+static const cricket::AudioCodec kOpusCodec(111, "opus", 48000, 64000, 2, 0);
+static const cricket::AudioCodec kG722CodecVoE(9, "G722", 16000, 64000, 1, 0);
+static const cricket::AudioCodec kG722CodecSdp(9, "G722", 8000, 64000, 1, 0);
+static const cricket::AudioCodec kRedCodec(117, "red", 8000, 0, 1, 0);
+static const cricket::AudioCodec kCn8000Codec(13, "CN", 8000, 0, 1, 0);
+static const cricket::AudioCodec kCn16000Codec(105, "CN", 16000, 0, 1, 0);
+static const cricket::AudioCodec
+ kTelephoneEventCodec(106, "telephone-event", 8000, 0, 1, 0);
+static const cricket::AudioCodec* const kAudioCodecs[] = {
&kPcmuCodec, &kIsacCodec, &kOpusCodec, &kG722CodecVoE, &kRedCodec,
&kCn8000Codec, &kCn16000Codec, &kTelephoneEventCodec,
};
-const uint32_t kSsrc1 = 0x99;
-const uint32_t kSsrc2 = 0x98;
+static uint32_t kSsrc1 = 0x99;
+static uint32_t kSsrc2 = 0x98;
class FakeVoEWrapper : public cricket::VoEWrapper {
public:
@@ -68,8 +68,10 @@
engine, // codec
engine, // dtmf
engine, // hw
+ engine, // neteq
engine, // network
engine, // rtp
+ engine, // sync
engine) { // volume
}
};
@@ -84,7 +86,6 @@
int SetTraceCallback(webrtc::TraceCallback* callback) override { return 0; }
unsigned int filter_;
};
-} // namespace
class WebRtcVoiceEngineTestFake : public testing::Test {
public:
@@ -292,71 +293,6 @@
EXPECT_EQ(-1, voe_.GetReceiveRtpExtensionId(new_channel_num, ext));
}
- const webrtc::AudioReceiveStream::Stats& GetAudioReceiveStreamStats() const {
- static webrtc::AudioReceiveStream::Stats stats;
- if (stats.remote_ssrc == 0) {
- stats.remote_ssrc = 123;
- stats.bytes_rcvd = 456;
- stats.packets_rcvd = 768;
- stats.packets_lost = 101;
- stats.fraction_lost = 23.45f;
- stats.codec_name = "codec_name";
- stats.ext_seqnum = 678;
- stats.jitter_ms = 901;
- stats.jitter_buffer_ms = 234;
- stats.jitter_buffer_preferred_ms = 567;
- stats.delay_estimate_ms = 890;
- stats.audio_level = 1234;
- stats.expand_rate = 5.67f;
- stats.speech_expand_rate = 8.90f;
- stats.secondary_decoded_rate = 1.23f;
- stats.accelerate_rate = 4.56f;
- stats.preemptive_expand_rate = 7.89f;
- stats.decoding_calls_to_silence_generator = 012;
- stats.decoding_calls_to_neteq = 345;
- stats.decoding_normal = 67890;
- stats.decoding_plc = 1234;
- stats.decoding_cng = 5678;
- stats.decoding_plc_cng = 9012;
- stats.capture_start_ntp_time_ms = 3456;
- }
- return stats;
- }
- void SetAudioReceiveStreamStats() {
- for (auto* s : call_.GetAudioReceiveStreams()) {
- s->SetStats(GetAudioReceiveStreamStats());
- }
- }
- void VerifyVoiceReceiverInfo(const cricket::VoiceReceiverInfo& info) {
- const auto& kStats = GetAudioReceiveStreamStats();
- EXPECT_EQ(info.local_stats.front().ssrc, kStats.remote_ssrc);
- EXPECT_EQ(info.bytes_rcvd, kStats.bytes_rcvd);
- EXPECT_EQ(info.packets_rcvd, kStats.packets_rcvd);
- EXPECT_EQ(info.packets_lost, kStats.packets_lost);
- EXPECT_EQ(info.fraction_lost, kStats.fraction_lost);
- EXPECT_EQ(info.codec_name, kStats.codec_name);
- EXPECT_EQ(info.ext_seqnum, kStats.ext_seqnum);
- EXPECT_EQ(info.jitter_ms, kStats.jitter_ms);
- EXPECT_EQ(info.jitter_buffer_ms, kStats.jitter_buffer_ms);
- EXPECT_EQ(info.jitter_buffer_preferred_ms,
- kStats.jitter_buffer_preferred_ms);
- EXPECT_EQ(info.delay_estimate_ms, kStats.delay_estimate_ms);
- EXPECT_EQ(info.audio_level, kStats.audio_level);
- EXPECT_EQ(info.expand_rate, kStats.expand_rate);
- EXPECT_EQ(info.speech_expand_rate, kStats.speech_expand_rate);
- EXPECT_EQ(info.secondary_decoded_rate, kStats.secondary_decoded_rate);
- EXPECT_EQ(info.accelerate_rate, kStats.accelerate_rate);
- EXPECT_EQ(info.preemptive_expand_rate, kStats.preemptive_expand_rate);
- EXPECT_EQ(info.decoding_calls_to_silence_generator,
- kStats.decoding_calls_to_silence_generator);
- EXPECT_EQ(info.decoding_calls_to_neteq, kStats.decoding_calls_to_neteq);
- EXPECT_EQ(info.decoding_normal, kStats.decoding_normal);
- EXPECT_EQ(info.decoding_plc, kStats.decoding_plc);
- EXPECT_EQ(info.decoding_cng, kStats.decoding_cng);
- EXPECT_EQ(info.decoding_plc_cng, kStats.decoding_plc_cng);
- EXPECT_EQ(info.capture_start_ntp_time_ms, kStats.capture_start_ntp_time_ms);
- }
-
protected:
cricket::FakeCall call_;
cricket::FakeWebRtcVoiceEngine voe_;
@@ -2072,23 +2008,38 @@
EXPECT_EQ(cricket::kIntStatValue, info.senders[i].jitter_ms);
EXPECT_EQ(kPcmuCodec.name, info.senders[i].codec_name);
}
+ EXPECT_EQ(0u, info.receivers.size());
- // We have added one receive stream. We should see empty stats.
- EXPECT_EQ(info.receivers.size(), 1u);
- EXPECT_EQ(info.receivers[0].local_stats.front().ssrc, 0);
-
- // Remove the kSsrc2 stream. No receiver stats.
- EXPECT_TRUE(channel_->RemoveRecvStream(kSsrc2));
+ // Registered stream's remote SSRC is kSsrc2. Send a packet with SSRC=1.
+ // We should drop the packet and no stats should be available.
+ DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
EXPECT_EQ(true, channel_->GetStats(&info));
EXPECT_EQ(0u, info.receivers.size());
- // Deliver a new packet - a default receive stream should be created and we
- // should see stats again.
+ // Remove the kSsrc2 stream and deliver a new packet - a default receive
+ // stream should be created and we should see stats.
+ EXPECT_TRUE(channel_->RemoveRecvStream(kSsrc2));
DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
- SetAudioReceiveStreamStats();
EXPECT_EQ(true, channel_->GetStats(&info));
EXPECT_EQ(1u, info.receivers.size());
- VerifyVoiceReceiverInfo(info.receivers[0]);
+
+ EXPECT_EQ(cricket::kIntStatValue, info.receivers[0].bytes_rcvd);
+ EXPECT_EQ(cricket::kIntStatValue, info.receivers[0].packets_rcvd);
+ EXPECT_EQ(cricket::kIntStatValue, info.receivers[0].packets_lost);
+ EXPECT_EQ(cricket::kIntStatValue, info.receivers[0].ext_seqnum);
+ EXPECT_EQ(kPcmuCodec.name, info.receivers[0].codec_name);
+ EXPECT_EQ(static_cast<float>(cricket::kNetStats.currentExpandRate) /
+ (1 << 14), info.receivers[0].expand_rate);
+ EXPECT_EQ(static_cast<float>(cricket::kNetStats.currentSpeechExpandRate) /
+ (1 << 14), info.receivers[0].speech_expand_rate);
+ EXPECT_EQ(static_cast<float>(cricket::kNetStats.currentSecondaryDecodedRate) /
+ (1 << 14), info.receivers[0].secondary_decoded_rate);
+ EXPECT_EQ(
+ static_cast<float>(cricket::kNetStats.currentAccelerateRate) / (1 << 14),
+ info.receivers[0].accelerate_rate);
+ EXPECT_EQ(
+ static_cast<float>(cricket::kNetStats.currentPreemptiveRate) / (1 << 14),
+ info.receivers[0].preemptive_expand_rate);
}
// Test that we can add and remove receive streams, and do proper send/playout.
@@ -2349,22 +2300,33 @@
// EXPECT_EQ(cricket::kIntStatValue, info.senders[0].echo_return_loss);
// EXPECT_EQ(cricket::kIntStatValue,
// info.senders[0].echo_return_loss_enhancement);
- // We have added one receive stream. We should see empty stats.
- EXPECT_EQ(info.receivers.size(), 1u);
- EXPECT_EQ(info.receivers[0].local_stats.front().ssrc, 0);
+ EXPECT_EQ(0u, info.receivers.size());
- // Remove the kSsrc2 stream. No receiver stats.
- EXPECT_TRUE(channel_->RemoveRecvStream(kSsrc2));
+ // Registered stream's remote SSRC is kSsrc2. Send a packet with SSRC=1.
+ // We should drop the packet and no stats should be available.
+ DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
EXPECT_EQ(true, channel_->GetStats(&info));
EXPECT_EQ(0u, info.receivers.size());
- // Deliver a new packet - a default receive stream should be created and we
- // should see stats again.
+ // Remove the kSsrc2 stream and deliver a new packet - a default receive
+ // stream should be created and we should see stats.
+ EXPECT_TRUE(channel_->RemoveRecvStream(kSsrc2));
DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
- SetAudioReceiveStreamStats();
EXPECT_EQ(true, channel_->GetStats(&info));
EXPECT_EQ(1u, info.receivers.size());
- VerifyVoiceReceiverInfo(info.receivers[0]);
+
+ EXPECT_EQ(cricket::kIntStatValue, info.receivers[0].bytes_rcvd);
+ EXPECT_EQ(cricket::kIntStatValue, info.receivers[0].packets_rcvd);
+ EXPECT_EQ(cricket::kIntStatValue, info.receivers[0].packets_lost);
+ EXPECT_EQ(cricket::kIntStatValue, info.receivers[0].ext_seqnum);
+ EXPECT_EQ(kPcmuCodec.name, info.receivers[0].codec_name);
+ EXPECT_EQ(static_cast<float>(cricket::kNetStats.currentExpandRate) /
+ (1 << 14), info.receivers[0].expand_rate);
+ EXPECT_EQ(static_cast<float>(cricket::kNetStats.currentSpeechExpandRate) /
+ (1 << 14), info.receivers[0].speech_expand_rate);
+ EXPECT_EQ(static_cast<float>(cricket::kNetStats.currentSecondaryDecodedRate) /
+ (1 << 14), info.receivers[0].secondary_decoded_rate);
+ // TODO(sriniv): Add testing for more receiver fields.
}
// Test that we can set the outgoing SSRC properly with multiple streams.
diff --git a/webrtc/audio/BUILD.gn b/webrtc/audio/BUILD.gn
index d5061db..c6f4b6b 100644
--- a/webrtc/audio/BUILD.gn
+++ b/webrtc/audio/BUILD.gn
@@ -14,8 +14,6 @@
"audio_receive_stream.h",
"audio_send_stream.cc",
"audio_send_stream.h",
- "conversion.h",
- "scoped_voe_interface.h",
]
configs += [ "..:common_config" ]
diff --git a/webrtc/audio/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc
index 0fd96d0..c725e37 100644
--- a/webrtc/audio/audio_receive_stream.cc
+++ b/webrtc/audio/audio_receive_stream.cc
@@ -12,17 +12,10 @@
#include <string>
-#include "webrtc/audio/conversion.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "webrtc/system_wrappers/interface/tick_util.h"
-#include "webrtc/voice_engine/include/voe_base.h"
-#include "webrtc/voice_engine/include/voe_codec.h"
-#include "webrtc/voice_engine/include/voe_neteq_stats.h"
-#include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
-#include "webrtc/voice_engine/include/voe_video_sync.h"
-#include "webrtc/voice_engine/include/voe_volume_control.h"
namespace webrtc {
std::string AudioReceiveStream::Config::Rtp::ToString() const {
@@ -31,9 +24,8 @@
ss << ", extensions: [";
for (size_t i = 0; i < extensions.size(); ++i) {
ss << extensions[i].ToString();
- if (i != extensions.size() - 1) {
+ if (i != extensions.size() - 1)
ss << ", ";
- }
}
ss << ']';
ss << '}';
@@ -44,9 +36,8 @@
std::stringstream ss;
ss << "{rtp: " << rtp.ToString();
ss << ", voe_channel_id: " << voe_channel_id;
- if (!sync_group.empty()) {
+ if (!sync_group.empty())
ss << ", sync_group: " << sync_group;
- }
ss << '}';
return ss.str();
}
@@ -54,18 +45,13 @@
namespace internal {
AudioReceiveStream::AudioReceiveStream(
RemoteBitrateEstimator* remote_bitrate_estimator,
- const webrtc::AudioReceiveStream::Config& config,
- VoiceEngine* voice_engine)
+ const webrtc::AudioReceiveStream::Config& config)
: remote_bitrate_estimator_(remote_bitrate_estimator),
config_(config),
- voice_engine_(voice_engine),
- voe_base_(voice_engine),
rtp_header_parser_(RtpHeaderParser::Create()) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
LOG(LS_INFO) << "AudioReceiveStream: " << config_.ToString();
RTC_DCHECK(config.voe_channel_id != -1);
RTC_DCHECK(remote_bitrate_estimator_ != nullptr);
- RTC_DCHECK(voice_engine_ != nullptr);
RTC_DCHECK(rtp_header_parser_ != nullptr);
for (const auto& ext : config.rtp.extensions) {
// One-byte-extension local identifiers are in the range 1-14 inclusive.
@@ -87,117 +73,33 @@
}
AudioReceiveStream::~AudioReceiveStream() {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
LOG(LS_INFO) << "~AudioReceiveStream: " << config_.ToString();
}
webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
- webrtc::AudioReceiveStream::Stats stats;
- stats.remote_ssrc = config_.rtp.remote_ssrc;
- ScopedVoEInterface<VoECodec> codec(voice_engine_);
- ScopedVoEInterface<VoENetEqStats> neteq(voice_engine_);
- ScopedVoEInterface<VoERTP_RTCP> rtp(voice_engine_);
- ScopedVoEInterface<VoEVideoSync> sync(voice_engine_);
- ScopedVoEInterface<VoEVolumeControl> volume(voice_engine_);
- unsigned int ssrc = 0;
- webrtc::CallStatistics cs = {0};
- webrtc::CodecInst ci = {0};
- // Only collect stats if we have seen some traffic with the SSRC.
- if (rtp->GetRemoteSSRC(config_.voe_channel_id, ssrc) == -1 ||
- rtp->GetRTCPStatistics(config_.voe_channel_id, cs) == -1 ||
- codec->GetRecCodec(config_.voe_channel_id, ci) == -1) {
- return stats;
- }
-
- stats.bytes_rcvd = cs.bytesReceived;
- stats.packets_rcvd = cs.packetsReceived;
- stats.packets_lost = cs.cumulativeLost;
- stats.fraction_lost = static_cast<float>(cs.fractionLost) / (1 << 8);
- if (ci.pltype != -1) {
- stats.codec_name = ci.plname;
- }
-
- stats.ext_seqnum = cs.extendedMax;
- if (ci.plfreq / 1000 > 0) {
- stats.jitter_ms = cs.jitterSamples / (ci.plfreq / 1000);
- }
- {
- int jitter_buffer_delay_ms = 0;
- int playout_buffer_delay_ms = 0;
- sync->GetDelayEstimate(config_.voe_channel_id, &jitter_buffer_delay_ms,
- &playout_buffer_delay_ms);
- stats.delay_estimate_ms =
- jitter_buffer_delay_ms + playout_buffer_delay_ms;
- }
- {
- unsigned int level = 0;
- if (volume->GetSpeechOutputLevelFullRange(config_.voe_channel_id, level)
- != -1) {
- stats.audio_level = static_cast<int32_t>(level);
- }
- }
-
- webrtc::NetworkStatistics ns = {0};
- if (neteq->GetNetworkStatistics(config_.voe_channel_id, ns) != -1) {
- // Get jitter buffer and total delay (alg + jitter + playout) stats.
- stats.jitter_buffer_ms = ns.currentBufferSize;
- stats.jitter_buffer_preferred_ms = ns.preferredBufferSize;
- stats.expand_rate = Q14ToFloat(ns.currentExpandRate);
- stats.speech_expand_rate = Q14ToFloat(ns.currentSpeechExpandRate);
- stats.secondary_decoded_rate = Q14ToFloat(ns.currentSecondaryDecodedRate);
- stats.accelerate_rate = Q14ToFloat(ns.currentAccelerateRate);
- stats.preemptive_expand_rate = Q14ToFloat(ns.currentPreemptiveRate);
- }
-
- webrtc::AudioDecodingCallStats ds;
- if (neteq->GetDecodingCallStatistics(config_.voe_channel_id, &ds) != -1) {
- stats.decoding_calls_to_silence_generator =
- ds.calls_to_silence_generator;
- stats.decoding_calls_to_neteq = ds.calls_to_neteq;
- stats.decoding_normal = ds.decoded_normal;
- stats.decoding_plc = ds.decoded_plc;
- stats.decoding_cng = ds.decoded_cng;
- stats.decoding_plc_cng = ds.decoded_plc_cng;
- }
-
- stats.capture_start_ntp_time_ms = cs.capture_start_ntp_time_ms_;
-
- return stats;
+ return webrtc::AudioReceiveStream::Stats();
}
const webrtc::AudioReceiveStream::Config& AudioReceiveStream::config() const {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
return config_;
}
void AudioReceiveStream::Start() {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
}
void AudioReceiveStream::Stop() {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
}
void AudioReceiveStream::SignalNetworkState(NetworkState state) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
}
bool AudioReceiveStream::DeliverRtcp(const uint8_t* packet, size_t length) {
- // TODO(solenberg): Tests call this function on a network thread, libjingle
- // calls on the worker thread. We should move towards always using a network
- // thread. Then this check can be enabled.
- // RTC_DCHECK(!thread_checker_.CalledOnValidThread());
return false;
}
bool AudioReceiveStream::DeliverRtp(const uint8_t* packet,
size_t length,
const PacketTime& packet_time) {
- // TODO(solenberg): Tests call this function on a network thread, libjingle
- // calls on the worker thread. We should move towards always using a network
- // thread. Then this check can be enabled.
- // RTC_DCHECK(!thread_checker_.CalledOnValidThread());
RTPHeader header;
if (!rtp_header_parser_->Parse(packet, length, &header)) {
diff --git a/webrtc/audio/audio_receive_stream.h b/webrtc/audio/audio_receive_stream.h
index 5c77653..1e52724 100644
--- a/webrtc/audio/audio_receive_stream.h
+++ b/webrtc/audio/audio_receive_stream.h
@@ -12,23 +12,18 @@
#define WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_
#include "webrtc/audio_receive_stream.h"
-#include "webrtc/audio/scoped_voe_interface.h"
-#include "webrtc/base/thread_checker.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
-#include "webrtc/voice_engine/include/voe_base.h"
namespace webrtc {
class RemoteBitrateEstimator;
-class VoiceEngine;
namespace internal {
class AudioReceiveStream : public webrtc::AudioReceiveStream {
public:
AudioReceiveStream(RemoteBitrateEstimator* remote_bitrate_estimator,
- const webrtc::AudioReceiveStream::Config& config,
- VoiceEngine* voice_engine);
+ const webrtc::AudioReceiveStream::Config& config);
~AudioReceiveStream() override;
// webrtc::ReceiveStream implementation.
@@ -46,12 +41,8 @@
const webrtc::AudioReceiveStream::Config& config() const;
private:
- rtc::ThreadChecker thread_checker_;
RemoteBitrateEstimator* const remote_bitrate_estimator_;
const webrtc::AudioReceiveStream::Config config_;
- VoiceEngine* voice_engine_;
- // We hold one interface pointer to the VoE to make sure it is kept alive.
- ScopedVoEInterface<VoEBase> voe_base_;
rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_;
};
} // namespace internal
diff --git a/webrtc/audio/audio_receive_stream_unittest.cc b/webrtc/audio/audio_receive_stream_unittest.cc
index 4a1c8c6..d6cce69 100644
--- a/webrtc/audio/audio_receive_stream_unittest.cc
+++ b/webrtc/audio/audio_receive_stream_unittest.cc
@@ -11,14 +11,10 @@
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/audio/audio_receive_stream.h"
-#include "webrtc/audio/conversion.h"
#include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_estimator.h"
#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
-#include "webrtc/test/fake_voice_engine.h"
-namespace {
-
-using webrtc::ByteWriter;
+namespace webrtc {
const size_t kAbsoluteSendTimeLength = 4;
@@ -49,28 +45,23 @@
ByteWriter<uint16_t>::WriteBigEndian(header + 2, 0x1234); // Sequence number.
ByteWriter<uint32_t>::WriteBigEndian(header + 4, 0x5678); // Timestamp.
ByteWriter<uint32_t>::WriteBigEndian(header + 8, 0x4321); // SSRC.
- int32_t rtp_header_length = webrtc::kRtpHeaderSize;
+ int32_t rtp_header_length = kRtpHeaderSize;
BuildAbsoluteSendTimeExtension(header + rtp_header_length, extension_id,
abs_send_time);
rtp_header_length += kAbsoluteSendTimeLength;
return rtp_header_length;
}
-} // namespace
-
-namespace webrtc {
-namespace test {
TEST(AudioReceiveStreamTest, AudioPacketUpdatesBweWithTimestamp) {
MockRemoteBitrateEstimator rbe;
- FakeVoiceEngine fve;
AudioReceiveStream::Config config;
config.combined_audio_video_bwe = true;
- config.voe_channel_id = fve.kReceiveChannelId;
+ config.voe_channel_id = 1;
const int kAbsSendTimeId = 3;
config.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
- internal::AudioReceiveStream recv_stream(&rbe, config, &fve);
+ internal::AudioReceiveStream recv_stream(&rbe, config);
uint8_t rtp_packet[30];
const int kAbsSendTimeValue = 1234;
CreateRtpHeaderWithAbsSendTime(rtp_packet, kAbsSendTimeId, kAbsSendTimeValue);
@@ -83,57 +74,4 @@
EXPECT_TRUE(
recv_stream.DeliverRtp(rtp_packet, sizeof(rtp_packet), packet_time));
}
-
-TEST(AudioReceiveStreamTest, GetStats) {
- const uint32_t kSsrc1 = 667;
-
- MockRemoteBitrateEstimator rbe;
- FakeVoiceEngine fve;
- AudioReceiveStream::Config config;
- config.rtp.remote_ssrc = kSsrc1;
- config.voe_channel_id = fve.kReceiveChannelId;
- internal::AudioReceiveStream recv_stream(&rbe, config, &fve);
-
- AudioReceiveStream::Stats stats = recv_stream.GetStats();
- const CallStatistics& call_stats = fve.GetRecvCallStats();
- const CodecInst& codec_inst = fve.GetRecvRecCodecInst();
- const NetworkStatistics& net_stats = fve.GetRecvNetworkStats();
- const AudioDecodingCallStats& decode_stats =
- fve.GetRecvAudioDecodingCallStats();
- EXPECT_EQ(kSsrc1, stats.remote_ssrc);
- EXPECT_EQ(static_cast<int64_t>(call_stats.bytesReceived), stats.bytes_rcvd);
- EXPECT_EQ(static_cast<uint32_t>(call_stats.packetsReceived),
- stats.packets_rcvd);
- EXPECT_EQ(call_stats.cumulativeLost, stats.packets_lost);
- EXPECT_EQ(static_cast<float>(call_stats.fractionLost) / 256,
- stats.fraction_lost);
- EXPECT_EQ(std::string(codec_inst.plname), stats.codec_name);
- EXPECT_EQ(call_stats.extendedMax, stats.ext_seqnum);
- EXPECT_EQ(call_stats.jitterSamples / (codec_inst.plfreq / 1000),
- stats.jitter_ms);
- EXPECT_EQ(net_stats.currentBufferSize, stats.jitter_buffer_ms);
- EXPECT_EQ(net_stats.preferredBufferSize, stats.jitter_buffer_preferred_ms);
- EXPECT_EQ(static_cast<uint32_t>(fve.kRecvJitterBufferDelay +
- fve.kRecvPlayoutBufferDelay), stats.delay_estimate_ms);
- EXPECT_EQ(static_cast<int32_t>(fve.kRecvSpeechOutputLevel),
- stats.audio_level);
- EXPECT_EQ(Q14ToFloat(net_stats.currentExpandRate), stats.expand_rate);
- EXPECT_EQ(Q14ToFloat(net_stats.currentSpeechExpandRate),
- stats.speech_expand_rate);
- EXPECT_EQ(Q14ToFloat(net_stats.currentSecondaryDecodedRate),
- stats.secondary_decoded_rate);
- EXPECT_EQ(Q14ToFloat(net_stats.currentAccelerateRate), stats.accelerate_rate);
- EXPECT_EQ(Q14ToFloat(net_stats.currentPreemptiveRate),
- stats.preemptive_expand_rate);
- EXPECT_EQ(decode_stats.calls_to_silence_generator,
- stats.decoding_calls_to_silence_generator);
- EXPECT_EQ(decode_stats.calls_to_neteq, stats.decoding_calls_to_neteq);
- EXPECT_EQ(decode_stats.decoded_normal, stats.decoding_normal);
- EXPECT_EQ(decode_stats.decoded_plc, stats.decoding_plc);
- EXPECT_EQ(decode_stats.decoded_cng, stats.decoding_cng);
- EXPECT_EQ(decode_stats.decoded_plc_cng, stats.decoding_plc_cng);
- EXPECT_EQ(call_stats.capture_start_ntp_time_ms_,
- stats.capture_start_ntp_time_ms);
-}
-} // namespace test
} // namespace webrtc
diff --git a/webrtc/audio/conversion.h b/webrtc/audio/conversion.h
deleted file mode 100644
index 4c0b7aa..0000000
--- a/webrtc/audio/conversion.h
+++ /dev/null
@@ -1,21 +0,0 @@
-/*
- * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_AUDIO_CONVERSION_H_
-#define WEBRTC_AUDIO_CONVERSION_H_
-
-namespace webrtc {
-
-inline float Q14ToFloat(uint16_t v) {
- return static_cast<float>(v) / (1 << 14);
-}
-} // namespace webrtc
-
-#endif // WEBRTC_AUDIO_CONVERSION_H_
diff --git a/webrtc/audio/scoped_voe_interface.h b/webrtc/audio/scoped_voe_interface.h
deleted file mode 100644
index 5a88fc9..0000000
--- a/webrtc/audio/scoped_voe_interface.h
+++ /dev/null
@@ -1,43 +0,0 @@
-/*
- * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_AUDIO_SCOPED_VOE_INTERFACE_H_
-#define WEBRTC_AUDIO_SCOPED_VOE_INTERFACE_H_
-
-#include "webrtc/base/checks.h"
-
-namespace webrtc {
-
-class VoiceEngine;
-
-namespace internal {
-
-template<class T> class ScopedVoEInterface {
- public:
- explicit ScopedVoEInterface(webrtc::VoiceEngine* e)
- : ptr_(T::GetInterface(e)) {
- RTC_DCHECK(ptr_);
- }
- ~ScopedVoEInterface() {
- if (ptr_) {
- ptr_->Release();
- }
- }
- T* operator->() {
- RTC_DCHECK(ptr_);
- return ptr_;
- }
- private:
- T* ptr_;
-};
-} // namespace internal
-} // namespace webrtc
-
-#endif // WEBRTC_AUDIO_SCOPED_VOE_INTERFACE_H_
diff --git a/webrtc/audio/webrtc_audio.gypi b/webrtc/audio/webrtc_audio.gypi
index b9d45db..40ccff6 100644
--- a/webrtc/audio/webrtc_audio.gypi
+++ b/webrtc/audio/webrtc_audio.gypi
@@ -18,8 +18,6 @@
'audio/audio_receive_stream.h',
'audio/audio_send_stream.cc',
'audio/audio_send_stream.h',
- 'audio/conversion.h',
- 'audio/scoped_voe_interface.h',
],
},
}
diff --git a/webrtc/audio_receive_stream.h b/webrtc/audio_receive_stream.h
index 3e5a518..70d6480 100644
--- a/webrtc/audio_receive_stream.h
+++ b/webrtc/audio_receive_stream.h
@@ -26,32 +26,7 @@
class AudioReceiveStream : public ReceiveStream {
public:
- struct Stats {
- uint32_t remote_ssrc = 0;
- int64_t bytes_rcvd = 0;
- uint32_t packets_rcvd = 0;
- uint32_t packets_lost = 0;
- float fraction_lost = 0.0f;
- std::string codec_name;
- uint32_t ext_seqnum = 0;
- uint32_t jitter_ms = 0;
- uint32_t jitter_buffer_ms = 0;
- uint32_t jitter_buffer_preferred_ms = 0;
- uint32_t delay_estimate_ms = 0;
- int32_t audio_level = -1;
- float expand_rate = 0.0f;
- float speech_expand_rate = 0.0f;
- float secondary_decoded_rate = 0.0f;
- float accelerate_rate = 0.0f;
- float preemptive_expand_rate = 0.0f;
- int32_t decoding_calls_to_silence_generator = 0;
- int32_t decoding_calls_to_neteq = 0;
- int32_t decoding_normal = 0;
- int32_t decoding_plc = 0;
- int32_t decoding_cng = 0;
- int32_t decoding_plc_cng = 0;
- int64_t capture_start_ntp_time_ms = 0;
- };
+ struct Stats {};
struct Config {
std::string ToString() const;
diff --git a/webrtc/call/bitrate_estimator_tests.cc b/webrtc/call/bitrate_estimator_tests.cc
index 08e36c8..f7044ae 100644
--- a/webrtc/call/bitrate_estimator_tests.cc
+++ b/webrtc/call/bitrate_estimator_tests.cc
@@ -25,7 +25,6 @@
#include "webrtc/test/encoder_settings.h"
#include "webrtc/test/fake_decoder.h"
#include "webrtc/test/fake_encoder.h"
-#include "webrtc/test/fake_voice_engine.h"
#include "webrtc/test/frame_generator_capturer.h"
namespace webrtc {
@@ -131,10 +130,8 @@
}
virtual void SetUp() {
- Call::Config config;
- config.voice_engine = &fake_voice_engine_;
- receiver_call_.reset(Call::Create(config));
- sender_call_.reset(Call::Create(config));
+ receiver_call_.reset(Call::Create(Call::Config()));
+ sender_call_.reset(Call::Create(Call::Config()));
send_transport_.SetReceiver(receiver_call_->Receiver());
receive_transport_.SetReceiver(sender_call_->Receiver());
@@ -268,7 +265,6 @@
test::FakeDecoder fake_decoder_;
};
- test::FakeVoiceEngine fake_voice_engine_;
TraceObserver receiver_trace_;
test::DirectTransport send_transport_;
test::DirectTransport receive_transport_;
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index a69be98..9a036c9 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -123,8 +123,7 @@
VideoSendStream::RtpStateMap suspended_video_send_ssrcs_;
- RtcEventLog* event_log_ = nullptr;
- VoECodec* voe_codec_ = nullptr;
+ RtcEventLog* event_log_;
RTC_DISALLOW_COPY_AND_ASSIGN(Call);
};
@@ -143,7 +142,8 @@
config_(config),
network_enabled_(true),
receive_crit_(RWLockWrapper::CreateRWLock()),
- send_crit_(RWLockWrapper::CreateRWLock()) {
+ send_crit_(RWLockWrapper::CreateRWLock()),
+ event_log_(nullptr) {
RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
RTC_DCHECK_GE(config.bitrate_config.min_bitrate_bps, 0);
RTC_DCHECK_GE(config.bitrate_config.start_bitrate_bps,
@@ -153,11 +153,11 @@
config.bitrate_config.start_bitrate_bps);
}
if (config.voice_engine) {
- // Keep a reference to VoECodec, so we're sure the VoiceEngine lives for the
- // duration of the call.
- voe_codec_ = VoECodec::GetInterface(config.voice_engine);
- if (voe_codec_)
- event_log_ = voe_codec_->GetEventLog();
+ VoECodec* voe_codec = VoECodec::GetInterface(config.voice_engine);
+ if (voe_codec) {
+ event_log_ = voe_codec->GetEventLog();
+ voe_codec->Release();
+ }
}
Trace::CreateTrace();
@@ -179,9 +179,6 @@
module_process_thread_->Stop();
Trace::ReturnTrace();
-
- if (voe_codec_)
- voe_codec_->Release();
}
PacketReceiver* Call::Receiver() {
@@ -232,8 +229,7 @@
TRACE_EVENT0("webrtc", "Call::CreateAudioReceiveStream");
RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
AudioReceiveStream* receive_stream = new AudioReceiveStream(
- channel_group_->GetRemoteBitrateEstimator(false), config,
- config_.voice_engine);
+ channel_group_->GetRemoteBitrateEstimator(false), config);
{
WriteLockScoped write_lock(*receive_crit_);
RTC_DCHECK(audio_receive_ssrcs_.find(config.rtp.remote_ssrc) ==
diff --git a/webrtc/call/call_unittest.cc b/webrtc/call/call_unittest.cc
index 9819b53..9adecc3 100644
--- a/webrtc/call/call_unittest.cc
+++ b/webrtc/call/call_unittest.cc
@@ -13,21 +13,19 @@
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/call.h"
-#include "webrtc/test/fake_voice_engine.h"
namespace {
struct CallHelper {
- CallHelper() : voice_engine_(new webrtc::test::FakeVoiceEngine()) {
+ CallHelper() {
webrtc::Call::Config config;
- config.voice_engine = voice_engine_.get();
+ // TODO(solenberg): Fill in with VoiceEngine* etc.
call_.reset(webrtc::Call::Create(config));
}
webrtc::Call* operator->() { return call_.get(); }
private:
- rtc::scoped_ptr<webrtc::test::FakeVoiceEngine> voice_engine_;
rtc::scoped_ptr<webrtc::Call> call_;
};
} // namespace
diff --git a/webrtc/test/fake_voice_engine.h b/webrtc/test/fake_voice_engine.h
deleted file mode 100644
index 72f6b27..0000000
--- a/webrtc/test/fake_voice_engine.h
+++ /dev/null
@@ -1,421 +0,0 @@
-/*
- * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_AUDIO_FAKE_VOICE_ENGINE_H_
-#define WEBRTC_AUDIO_FAKE_VOICE_ENGINE_H_
-
-#include <vector>
-
-#include "testing/gtest/include/gtest/gtest.h"
-
-#include "webrtc/voice_engine/voice_engine_impl.h"
-
-namespace webrtc {
-namespace test {
-
-// NOTE: This class inherits from VoiceEngineImpl so that its clients will be
-// able to get the various interfaces as usual, via T::GetInterface().
-class FakeVoiceEngine final : public VoiceEngineImpl {
- public:
- const int kSendChannelId = 1;
- const int kReceiveChannelId = 2;
-
- const int kRecvJitterBufferDelay = -7;
- const int kRecvPlayoutBufferDelay = 302;
- const unsigned int kRecvSpeechOutputLevel = 99;
-
- FakeVoiceEngine() : VoiceEngineImpl(new Config(), true) {
- // Increase ref count so this object isn't automatically deleted whenever
- // interfaces are Release():d.
- ++_ref_count;
- }
- ~FakeVoiceEngine() override {
- // Decrease ref count before base class d-tor is called; otherwise it will
- // trigger an assertion.
- --_ref_count;
- }
-
- const CallStatistics& GetRecvCallStats() const {
- static const CallStatistics kStats = {
- 345, 678, 901, 234, -1, 0, 0, 567, 890, 123
- };
- return kStats;
- }
-
- const CodecInst& GetRecvRecCodecInst() const {
- static const CodecInst kStats = {
- 123, "codec_name", 96000, -1, -1, -1
- };
- return kStats;
- }
-
- const NetworkStatistics& GetRecvNetworkStats() const {
- static const NetworkStatistics kStats = {
- 123, 456, false, 0, 0, 789, 12, 345, 678, 901, -1, -1, -1, -1, -1, 0
- };
- return kStats;
- }
-
- const AudioDecodingCallStats& GetRecvAudioDecodingCallStats() const {
- static AudioDecodingCallStats stats;
- if (stats.calls_to_silence_generator == 0) {
- stats.calls_to_silence_generator = 234;
- stats.calls_to_neteq = 567;
- stats.decoded_normal = 890;
- stats.decoded_plc = 123;
- stats.decoded_cng = 456;
- stats.decoded_plc_cng = 789;
- }
- return stats;
- }
-
- // VoEBase
- int RegisterVoiceEngineObserver(VoiceEngineObserver& observer) override {
- return -1;
- }
- int DeRegisterVoiceEngineObserver() override { return -1; }
- int Init(AudioDeviceModule* external_adm = NULL,
- AudioProcessing* audioproc = NULL) override { return -1; }
- AudioProcessing* audio_processing() override { return nullptr; }
- int Terminate() override { return -1; }
- int CreateChannel() override { return -1; }
- int CreateChannel(const Config& config) override { return -1; }
- int DeleteChannel(int channel) override { return -1; }
- int StartReceive(int channel) override { return -1; }
- int StopReceive(int channel) override { return -1; }
- int StartPlayout(int channel) override { return -1; }
- int StopPlayout(int channel) override { return -1; }
- int StartSend(int channel) override { return -1; }
- int StopSend(int channel) override { return -1; }
- int GetVersion(char version[1024]) override { return -1; }
- int LastError() override { return -1; }
- AudioTransport* audio_transport() { return nullptr; }
- int AssociateSendChannel(int channel, int accociate_send_channel) override {
- return -1;
- }
-
- // VoECodec
- int NumOfCodecs() override { return -1; }
- int GetCodec(int index, CodecInst& codec) override { return -1; }
- int SetSendCodec(int channel, const CodecInst& codec) override { return -1; }
- int GetSendCodec(int channel, CodecInst& codec) override { return -1; }
- int SetBitRate(int channel, int bitrate_bps) override { return -1; }
- int GetRecCodec(int channel, CodecInst& codec) override {
- EXPECT_EQ(channel, kReceiveChannelId);
- codec = GetRecvRecCodecInst();
- return 0;
- }
- int SetRecPayloadType(int channel, const CodecInst& codec) override {
- return -1;
- }
- int GetRecPayloadType(int channel, CodecInst& codec) override { return -1; }
- int SetSendCNPayloadType(int channel, int type,
- PayloadFrequencies frequency = kFreq16000Hz) override { return -1; }
- int SetVADStatus(int channel,
- bool enable,
- VadModes mode = kVadConventional,
- bool disableDTX = false) override { return -1; }
- int GetVADStatus(int channel,
- bool& enabled,
- VadModes& mode,
- bool& disabledDTX) override { return -1; }
- int SetOpusMaxPlaybackRate(int channel, int frequency_hz) override {
- return -1;
- }
- int SetOpusDtx(int channel, bool enable_dtx) override { return -1; }
- RtcEventLog* GetEventLog() override { return nullptr; }
-
- // VoEDtmf
- int SendTelephoneEvent(int channel,
- int eventCode,
- bool outOfBand = true,
- int lengthMs = 160,
- int attenuationDb = 10) override { return -1; }
- int SetSendTelephoneEventPayloadType(int channel,
- unsigned char type) override {
- return -1;
- }
- int GetSendTelephoneEventPayloadType(int channel,
- unsigned char& type) override {
- return -1;
- }
- int SetDtmfFeedbackStatus(bool enable,
- bool directFeedback = false) override { return -1; }
- int GetDtmfFeedbackStatus(bool& enabled, bool& directFeedback) override {
- return -1;
- }
- int PlayDtmfTone(int eventCode,
- int lengthMs = 200,
- int attenuationDb = 10) override { return -1; }
-
- // VoEExternalMedia
- int RegisterExternalMediaProcessing(
- int channel,
- ProcessingTypes type,
- VoEMediaProcess& processObject) override { return -1; }
- int DeRegisterExternalMediaProcessing(int channel,
- ProcessingTypes type) override {
- return -1;
- }
- int GetAudioFrame(int channel,
- int desired_sample_rate_hz,
- AudioFrame* frame) override { return -1; }
- int SetExternalMixing(int channel, bool enable) override { return -1; }
-
- // VoEFile
- int StartPlayingFileLocally(
- int channel,
- const char fileNameUTF8[1024],
- bool loop = false,
- FileFormats format = kFileFormatPcm16kHzFile,
- float volumeScaling = 1.0,
- int startPointMs = 0,
- int stopPointMs = 0) override { return -1; }
- int StartPlayingFileLocally(
- int channel,
- InStream* stream,
- FileFormats format = kFileFormatPcm16kHzFile,
- float volumeScaling = 1.0,
- int startPointMs = 0,
- int stopPointMs = 0) override { return -1; }
- int StopPlayingFileLocally(int channel) override { return -1; }
- int IsPlayingFileLocally(int channel) override { return -1; }
- int StartPlayingFileAsMicrophone(
- int channel,
- const char fileNameUTF8[1024],
- bool loop = false,
- bool mixWithMicrophone = false,
- FileFormats format = kFileFormatPcm16kHzFile,
- float volumeScaling = 1.0) override { return -1; }
- int StartPlayingFileAsMicrophone(
- int channel,
- InStream* stream,
- bool mixWithMicrophone = false,
- FileFormats format = kFileFormatPcm16kHzFile,
- float volumeScaling = 1.0) override { return -1; }
- int StopPlayingFileAsMicrophone(int channel) override { return -1; }
- int IsPlayingFileAsMicrophone(int channel) override { return -1; }
- int StartRecordingPlayout(int channel,
- const char* fileNameUTF8,
- CodecInst* compression = NULL,
- int maxSizeBytes = -1) override { return -1; }
- int StopRecordingPlayout(int channel) override { return -1; }
- int StartRecordingPlayout(int channel,
- OutStream* stream,
- CodecInst* compression = NULL) override {
- return -1;
- }
- int StartRecordingMicrophone(const char* fileNameUTF8,
- CodecInst* compression = NULL,
- int maxSizeBytes = -1) override { return -1; }
- int StartRecordingMicrophone(OutStream* stream,
- CodecInst* compression = NULL) override {
- return -1;
- }
- int StopRecordingMicrophone() override { return -1; }
-
- // VoEHardware
- int GetNumOfRecordingDevices(int& devices) override { return -1; }
-
- // Gets the number of audio devices available for playout.
- int GetNumOfPlayoutDevices(int& devices) override { return -1; }
-
- // Gets the name of a specific recording device given by an |index|.
- // On Windows Vista/7, it also retrieves an additional unique ID
- // (GUID) for the recording device.
- int GetRecordingDeviceName(int index,
- char strNameUTF8[128],
- char strGuidUTF8[128]) override { return -1; }
-
- // Gets the name of a specific playout device given by an |index|.
- // On Windows Vista/7, it also retrieves an additional unique ID
- // (GUID) for the playout device.
- int GetPlayoutDeviceName(int index,
- char strNameUTF8[128],
- char strGuidUTF8[128]) override { return -1; }
-
- // Sets the audio device used for recording.
- int SetRecordingDevice(
- int index,
- StereoChannel recordingChannel = kStereoBoth) override { return -1; }
-
- // Sets the audio device used for playout.
- int SetPlayoutDevice(int index) override { return -1; }
-
- // Sets the type of audio device layer to use.
- int SetAudioDeviceLayer(AudioLayers audioLayer) override { return -1; }
-
- // Gets the currently used (active) audio device layer.
- int GetAudioDeviceLayer(AudioLayers& audioLayer) override { return -1; }
-
- // Native sample rate controls (samples/sec)
- int SetRecordingSampleRate(unsigned int samples_per_sec) override {
- return -1;
- }
- int RecordingSampleRate(unsigned int* samples_per_sec) const override {
- return -1;
- }
- int SetPlayoutSampleRate(unsigned int samples_per_sec) override {
- return -1;
- }
- int PlayoutSampleRate(unsigned int* samples_per_sec) const override {
- return -1;
- }
-
- // Queries and controls platform audio effects on Android devices.
- bool BuiltInAECIsAvailable() const override { return false; }
- int EnableBuiltInAEC(bool enable) override { return -1; }
- bool BuiltInAGCIsAvailable() const override { return false; }
- int EnableBuiltInAGC(bool enable) override { return -1; }
- bool BuiltInNSIsAvailable() const override { return false; }
- int EnableBuiltInNS(bool enable) override { return -1; }
-
- // VoENetwork
- int RegisterExternalTransport(int channel, Transport& transport) override {
- return -1;
- }
- int DeRegisterExternalTransport(int channel) override { return -1; }
- int ReceivedRTPPacket(int channel,
- const void* data,
- size_t length) override { return -1; }
- int ReceivedRTPPacket(int channel,
- const void* data,
- size_t length,
- const PacketTime& packet_time) override { return -1; }
- int ReceivedRTCPPacket(int channel,
- const void* data,
- size_t length) { return -1; }
-
- // VoENetEqStats
- int GetNetworkStatistics(int channel, NetworkStatistics& stats) override {
- EXPECT_EQ(channel, kReceiveChannelId);
- stats = GetRecvNetworkStats();
- return 0;
- }
- int GetDecodingCallStatistics(int channel,
- AudioDecodingCallStats* stats) const override {
- EXPECT_EQ(channel, kReceiveChannelId);
- EXPECT_NE(nullptr, stats);
- *stats = GetRecvAudioDecodingCallStats();
- return 0;
- }
-
- // VoERTP_RTCP
- int SetLocalSSRC(int channel, unsigned int ssrc) override { return -1; }
- int GetLocalSSRC(int channel, unsigned int& ssrc) override { return -1; }
- int GetRemoteSSRC(int channel, unsigned int& ssrc) override {
- EXPECT_EQ(channel, kReceiveChannelId);
- ssrc = 0;
- return 0;
- }
- int SetSendAudioLevelIndicationStatus(int channel,
- bool enable,
- unsigned char id = 1) override {
- return -1;
- }
- int SetSendAbsoluteSenderTimeStatus(int channel,
- bool enable,
- unsigned char id) override { return -1; }
- int SetReceiveAbsoluteSenderTimeStatus(int channel,
- bool enable,
- unsigned char id) override {
- return -1;
- }
- int SetRTCPStatus(int channel, bool enable) override { return -1; }
- int GetRTCPStatus(int channel, bool& enabled) override { return -1; }
- int SetRTCP_CNAME(int channel, const char cName[256]) override { return -1; }
- int GetRTCP_CNAME(int channel, char cName[256]) { return -1; }
- int GetRemoteRTCP_CNAME(int channel, char cName[256]) override { return -1; }
- int GetRemoteRTCPData(int channel,
- unsigned int& NTPHigh,
- unsigned int& NTPLow,
- unsigned int& timestamp,
- unsigned int& playoutTimestamp,
- unsigned int* jitter = NULL,
- unsigned short* fractionLost = NULL) override {
- return -1;
- }
- int GetRTPStatistics(int channel,
- unsigned int& averageJitterMs,
- unsigned int& maxJitterMs,
- unsigned int& discardedPackets) override { return -1; }
- int GetRTCPStatistics(int channel, CallStatistics& stats) override {
- EXPECT_EQ(channel, kReceiveChannelId);
- stats = GetRecvCallStats();
- return 0;
- }
- int GetRemoteRTCPReportBlocks(
- int channel,
- std::vector<ReportBlock>* receive_blocks) override { return -1; }
- int SetNACKStatus(int channel, bool enable, int maxNoPackets) override {
- return -1;
- }
-
- // VoEVideoSync
- int GetPlayoutBufferSize(int& buffer_ms) override { return -1; }
- int SetMinimumPlayoutDelay(int channel, int delay_ms) override { return -1; }
- int SetInitialPlayoutDelay(int channel, int delay_ms) override { return -1; }
- int GetDelayEstimate(int channel,
- int* jitter_buffer_delay_ms,
- int* playout_buffer_delay_ms) override {
- EXPECT_EQ(channel, kReceiveChannelId);
- *jitter_buffer_delay_ms = kRecvJitterBufferDelay;
- *playout_buffer_delay_ms = kRecvPlayoutBufferDelay;
- return 0;
- }
- int GetLeastRequiredDelayMs(int channel) const override { return -1; }
- int SetInitTimestamp(int channel, unsigned int timestamp) override {
- return -1;
- }
- int SetInitSequenceNumber(int channel, short sequenceNumber) override {
- return -1;
- }
- int GetPlayoutTimestamp(int channel, unsigned int& timestamp) override {
- return -1;
- }
- int GetRtpRtcp(int channel,
- RtpRtcp** rtpRtcpModule,
- RtpReceiver** rtp_receiver) override { return -1; }
-
- // VoEVolumeControl
- int SetSpeakerVolume(unsigned int volume) override { return -1; }
- int GetSpeakerVolume(unsigned int& volume) override { return -1; }
- int SetMicVolume(unsigned int volume) override { return -1; }
- int GetMicVolume(unsigned int& volume) override { return -1; }
- int SetInputMute(int channel, bool enable) override { return -1; }
- int GetInputMute(int channel, bool& enabled) override { return -1; }
- int GetSpeechInputLevel(unsigned int& level) override { return -1; }
- int GetSpeechOutputLevel(int channel, unsigned int& level) override {
- return -1;
- }
- int GetSpeechInputLevelFullRange(unsigned int& level) override { return -1; }
- int GetSpeechOutputLevelFullRange(int channel,
- unsigned int& level) override {
- EXPECT_EQ(channel, kReceiveChannelId);
- level = kRecvSpeechOutputLevel;
- return 0;
- }
- int SetChannelOutputVolumeScaling(int channel, float scaling) override {
- return -1;
- }
- int GetChannelOutputVolumeScaling(int channel, float& scaling) override {
- return -1;
- }
- int SetOutputVolumePan(int channel, float left, float right) override {
- return -1;
- }
- int GetOutputVolumePan(int channel, float& left, float& right) override {
- return -1;
- }
-};
-} // namespace test
-} // namespace webrtc
-
-#endif // WEBRTC_AUDIO_FAKE_VOICE_ENGINE_H_
diff --git a/webrtc/test/webrtc_test_common.gyp b/webrtc/test/webrtc_test_common.gyp
index 5076900..f8d3365 100644
--- a/webrtc/test/webrtc_test_common.gyp
+++ b/webrtc/test/webrtc_test_common.gyp
@@ -30,7 +30,6 @@
'fake_encoder.h',
'fake_network_pipe.cc',
'fake_network_pipe.h',
- 'fake_voice_engine.h',
'frame_generator_capturer.cc',
'frame_generator_capturer.h',
'layer_filtering_transport.cc',
diff --git a/webrtc/voice_engine/voice_engine_impl.h b/webrtc/voice_engine/voice_engine_impl.h
index a21aabd..07f29c3 100644
--- a/webrtc/voice_engine/voice_engine_impl.h
+++ b/webrtc/voice_engine/voice_engine_impl.h
@@ -128,9 +128,7 @@
// This implements the Release() method for all the inherited interfaces.
int Release() override;
- // This is *protected* so that FakeVoiceEngine can inherit from the class and
- // manipulate the reference count. See: fake_voice_engine.h.
- protected:
+ private:
Atomic32 _ref_count;
rtc::scoped_ptr<const Config> own_config_;
};
