Expose RtpCodecParameters to VoiceMediaInfo stats.
Payload type -> RtpCodecParameters maps added for sender and receiver.
This is a follow-up to https://codereview.webrtc.org/2484193002/ which
did the same thing for VideoMediaInfo. This information will be used to
produce RTCCodecStats[1].
Voice[Sender/Receiver]Info is updated with current codec payload type
for every stream which can be used to look up the codec in
VoiceMediaInfo.
[1] https://w3c.github.io/webrtc-stats/#codec-dict*
BUG=chromium:659117
Review-Url: https://codereview.webrtc.org/2503383002
Cr-Commit-Position: refs/heads/master@{#15144}
diff --git a/webrtc/api/call/audio_receive_stream.h b/webrtc/api/call/audio_receive_stream.h
index ec80a9e..2fd6760 100644
--- a/webrtc/api/call/audio_receive_stream.h
+++ b/webrtc/api/call/audio_receive_stream.h
@@ -16,6 +16,7 @@
#include <string>
#include <vector>
+#include "webrtc/base/optional.h"
#include "webrtc/base/scoped_ref_ptr.h"
#include "webrtc/modules/audio_coding/codecs/audio_decoder_factory.h"
#include "webrtc/common_types.h"
@@ -40,6 +41,7 @@
uint32_t packets_lost = 0;
float fraction_lost = 0.0f;
std::string codec_name;
+ rtc::Optional<int> codec_payload_type;
uint32_t ext_seqnum = 0;
uint32_t jitter_ms = 0;
uint32_t jitter_buffer_ms = 0;
diff --git a/webrtc/api/call/audio_send_stream.cc b/webrtc/api/call/audio_send_stream.cc
index 3ce35e7..b619007 100644
--- a/webrtc/api/call/audio_send_stream.cc
+++ b/webrtc/api/call/audio_send_stream.cc
@@ -30,6 +30,7 @@
namespace webrtc {
AudioSendStream::Stats::Stats() = default;
+AudioSendStream::Stats::~Stats() = default;
AudioSendStream::Config::Config(Transport* send_transport)
: send_transport(send_transport) {}
diff --git a/webrtc/api/call/audio_send_stream.h b/webrtc/api/call/audio_send_stream.h
index aab625e..b4ba615 100644
--- a/webrtc/api/call/audio_send_stream.h
+++ b/webrtc/api/call/audio_send_stream.h
@@ -15,6 +15,7 @@
#include <string>
#include <vector>
+#include "webrtc/base/optional.h"
#include "webrtc/config.h"
#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
#include "webrtc/transport.h"
@@ -31,6 +32,7 @@
public:
struct Stats {
Stats();
+ ~Stats();
// TODO(solenberg): Harmonize naming and defaults with receive stream stats.
uint32_t local_ssrc = 0;
@@ -39,6 +41,7 @@
int32_t packets_lost = -1;
float fraction_lost = -1.0f;
std::string codec_name;
+ rtc::Optional<int> codec_payload_type;
int32_t ext_seqnum = -1;
int32_t jitter_ms = -1;
int64_t rtt_ms = -1;
diff --git a/webrtc/audio/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc
index 0d75307..5a8efc2 100644
--- a/webrtc/audio/audio_receive_stream.cc
+++ b/webrtc/audio/audio_receive_stream.cc
@@ -184,6 +184,7 @@
stats.capture_start_ntp_time_ms = call_stats.capture_start_ntp_time_ms_;
if (codec_inst.pltype != -1) {
stats.codec_name = codec_inst.plname;
+ stats.codec_payload_type = rtc::Optional<int>(codec_inst.pltype);
}
stats.ext_seqnum = call_stats.extendedMax;
if (codec_inst.plfreq / 1000 > 0) {
diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc
index 8c03dba..b031762 100644
--- a/webrtc/audio/audio_send_stream.cc
+++ b/webrtc/audio/audio_send_stream.cc
@@ -170,6 +170,7 @@
if (codec->GetSendCodec(config_.voe_channel_id, codec_inst) != -1) {
RTC_DCHECK_NE(codec_inst.pltype, -1);
stats.codec_name = codec_inst.plname;
+ stats.codec_payload_type = rtc::Optional<int>(codec_inst.pltype);
// Get data from the last remote RTCP report.
for (const auto& block : channel_proxy_->GetRemoteRTCPReportBlocks()) {
diff --git a/webrtc/media/base/mediachannel.h b/webrtc/media/base/mediachannel.h
index 80b9d4d..d4d339e 100644
--- a/webrtc/media/base/mediachannel.h
+++ b/webrtc/media/base/mediachannel.h
@@ -553,6 +553,7 @@
float fraction_lost;
int64_t rtt_ms;
std::string codec_name;
+ rtc::Optional<int> codec_payload_type;
std::vector<SsrcSenderInfo> local_stats;
std::vector<SsrcReceiverInfo> remote_stats;
};
@@ -598,6 +599,7 @@
int packets_lost;
float fraction_lost;
std::string codec_name;
+ rtc::Optional<int> codec_payload_type;
std::vector<SsrcReceiverInfo> local_stats;
std::vector<SsrcSenderInfo> remote_stats;
};
@@ -698,9 +700,6 @@
std::vector<SsrcGroup> ssrc_groups;
// TODO(hbos): Move this to |VideoMediaInfo::send_codecs|?
std::string encoder_implementation_name;
- // TODO(hbos): Move this to |MediaSenderInfo| when supported by
- // |VoiceSenderInfo| as well (which also extends that class).
- rtc::Optional<uint32_t> codec_payload_type;
int packets_cached;
int firs_rcvd;
int plis_rcvd;
@@ -746,9 +745,6 @@
std::vector<SsrcGroup> ssrc_groups;
// TODO(hbos): Move this to |VideoMediaInfo::receive_codecs|?
std::string decoder_implementation_name;
- // TODO(hbos): Move this to |MediaReceiverInfo| when supported by
- // |VoiceReceiverInfo| as well (which also extends that class).
- rtc::Optional<uint32_t> codec_payload_type;
int packets_concealed;
int firs_sent;
int plis_sent;
@@ -831,9 +827,13 @@
void Clear() {
senders.clear();
receivers.clear();
+ send_codecs.clear();
+ receive_codecs.clear();
}
std::vector<VoiceSenderInfo> senders;
std::vector<VoiceReceiverInfo> receivers;
+ RtpCodecParametersMap send_codecs;
+ RtpCodecParametersMap receive_codecs;
};
struct VideoMediaInfo {
diff --git a/webrtc/media/engine/webrtcvideoengine2.cc b/webrtc/media/engine/webrtcvideoengine2.cc
index c43dff8..5238348 100644
--- a/webrtc/media/engine/webrtcvideoengine2.cc
+++ b/webrtc/media/engine/webrtcvideoengine2.cc
@@ -2025,8 +2025,8 @@
if (parameters_.codec_settings) {
info.codec_name = parameters_.codec_settings->codec.name;
- info.codec_payload_type = rtc::Optional<uint32_t>(
- static_cast<uint32_t>(parameters_.codec_settings->codec.id));
+ info.codec_payload_type = rtc::Optional<int>(
+ parameters_.codec_settings->codec.id);
}
if (stream_ == NULL)
@@ -2436,8 +2436,8 @@
webrtc::VideoReceiveStream::Stats stats = stream_->GetStats();
info.decoder_implementation_name = stats.decoder_implementation_name;
if (stats.current_payload_type != -1) {
- info.codec_payload_type = rtc::Optional<uint32_t>(
- static_cast<uint32_t>(stats.current_payload_type));
+ info.codec_payload_type = rtc::Optional<int>(
+ stats.current_payload_type);
}
info.bytes_rcvd = stats.rtp_stats.transmitted.payload_bytes +
stats.rtp_stats.transmitted.header_bytes +
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
index 937078c..57921ac 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -2526,7 +2526,7 @@
RTC_DCHECK(info);
// Get SSRC and stats for each sender.
- RTC_DCHECK(info->senders.size() == 0);
+ RTC_DCHECK_EQ(info->senders.size(), 0U);
for (const auto& stream : send_streams_) {
webrtc::AudioSendStream::Stats stats = stream.second->GetStats();
VoiceSenderInfo sinfo;
@@ -2536,6 +2536,7 @@
sinfo.packets_lost = stats.packets_lost;
sinfo.fraction_lost = stats.fraction_lost;
sinfo.codec_name = stats.codec_name;
+ sinfo.codec_payload_type = stats.codec_payload_type;
sinfo.ext_seqnum = stats.ext_seqnum;
sinfo.jitter_ms = stats.jitter_ms;
sinfo.rtt_ms = stats.rtt_ms;
@@ -2551,7 +2552,7 @@
}
// Get SSRC and stats for each receiver.
- RTC_DCHECK(info->receivers.size() == 0);
+ RTC_DCHECK_EQ(info->receivers.size(), 0U);
for (const auto& stream : recv_streams_) {
webrtc::AudioReceiveStream::Stats stats = stream.second->GetStats();
VoiceReceiverInfo rinfo;
@@ -2561,6 +2562,7 @@
rinfo.packets_lost = stats.packets_lost;
rinfo.fraction_lost = stats.fraction_lost;
rinfo.codec_name = stats.codec_name;
+ rinfo.codec_payload_type = stats.codec_payload_type;
rinfo.ext_seqnum = stats.ext_seqnum;
rinfo.jitter_ms = stats.jitter_ms;
rinfo.jitter_buffer_ms = stats.jitter_buffer_ms;
@@ -2584,6 +2586,18 @@
info->receivers.push_back(rinfo);
}
+ // Get codec info
+ for (const AudioCodec& codec : send_codecs_) {
+ webrtc::RtpCodecParameters codec_params = codec.ToCodecParameters();
+ info->send_codecs.insert(
+ std::make_pair(codec_params.payload_type, std::move(codec_params)));
+ }
+ for (const AudioCodec& codec : recv_codecs_) {
+ webrtc::RtpCodecParameters codec_params = codec.ToCodecParameters();
+ info->receive_codecs.insert(
+ std::make_pair(codec_params.payload_type, std::move(codec_params)));
+ }
+
return true;
}
diff --git a/webrtc/media/engine/webrtcvoiceengine_unittest.cc b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
index 3b4670e..e4e4cf1 100644
--- a/webrtc/media/engine/webrtcvoiceengine_unittest.cc
+++ b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
@@ -446,6 +446,7 @@
stats.packets_lost = 9012;
stats.fraction_lost = 34.56f;
stats.codec_name = "codec_name_send";
+ stats.codec_payload_type = rtc::Optional<int>(42);
stats.ext_seqnum = 789;
stats.jitter_ms = 12;
stats.rtt_ms = 345;
@@ -473,6 +474,7 @@
EXPECT_EQ(info.packets_lost, stats.packets_lost);
EXPECT_EQ(info.fraction_lost, stats.fraction_lost);
EXPECT_EQ(info.codec_name, stats.codec_name);
+ EXPECT_EQ(info.codec_payload_type, stats.codec_payload_type);
EXPECT_EQ(info.ext_seqnum, stats.ext_seqnum);
EXPECT_EQ(info.jitter_ms, stats.jitter_ms);
EXPECT_EQ(info.rtt_ms, stats.rtt_ms);
@@ -496,6 +498,7 @@
stats.packets_lost = 101;
stats.fraction_lost = 23.45f;
stats.codec_name = "codec_name_recv";
+ stats.codec_payload_type = rtc::Optional<int>(42);
stats.ext_seqnum = 678;
stats.jitter_ms = 901;
stats.jitter_buffer_ms = 234;
@@ -530,6 +533,7 @@
EXPECT_EQ(info.packets_lost, stats.packets_lost);
EXPECT_EQ(info.fraction_lost, stats.fraction_lost);
EXPECT_EQ(info.codec_name, stats.codec_name);
+ EXPECT_EQ(info.codec_payload_type, stats.codec_payload_type);
EXPECT_EQ(info.ext_seqnum, stats.ext_seqnum);
EXPECT_EQ(info.jitter_ms, stats.jitter_ms);
EXPECT_EQ(info.jitter_buffer_ms, stats.jitter_buffer_ms);
@@ -552,6 +556,20 @@
EXPECT_EQ(info.decoding_muted_output, stats.decoding_muted_output);
EXPECT_EQ(info.capture_start_ntp_time_ms, stats.capture_start_ntp_time_ms);
}
+ void VerifyVoiceSendRecvCodecs(const cricket::VoiceMediaInfo& info) const {
+ EXPECT_EQ(send_parameters_.codecs.size(), info.send_codecs.size());
+ for (const cricket::AudioCodec& codec : send_parameters_.codecs) {
+ ASSERT_EQ(info.send_codecs.count(codec.id), 1U);
+ EXPECT_EQ(info.send_codecs.find(codec.id)->second,
+ codec.ToCodecParameters());
+ }
+ EXPECT_EQ(recv_parameters_.codecs.size(), info.receive_codecs.size());
+ for (const cricket::AudioCodec& codec : recv_parameters_.codecs) {
+ ASSERT_EQ(info.receive_codecs.count(codec.id), 1U);
+ EXPECT_EQ(info.receive_codecs.find(codec.id)->second,
+ codec.ToCodecParameters());
+ }
+ }
protected:
StrictMock<webrtc::test::MockAudioDeviceModule> adm_;
@@ -2275,6 +2293,7 @@
for (const auto& sender : info.senders) {
VerifyVoiceSenderInfo(sender, false);
}
+ VerifyVoiceSendRecvCodecs(info);
// We have added one receive stream. We should see empty stats.
EXPECT_EQ(info.receivers.size(), 1u);
@@ -2300,6 +2319,7 @@
EXPECT_EQ(static_cast<size_t>(arraysize(kSsrcs4)), info.senders.size());
EXPECT_EQ(1u, info.receivers.size());
VerifyVoiceReceiverInfo(info.receivers[0]);
+ VerifyVoiceSendRecvCodecs(info);
}
}
@@ -2483,6 +2503,7 @@
SetSend(true);
EXPECT_EQ(true, channel_->GetStats(&info));
VerifyVoiceSenderInfo(info.senders[0], true);
+ VerifyVoiceSendRecvCodecs(info);
}
// Remove the kSsrc2 stream. No receiver stats.
@@ -2504,6 +2525,7 @@
EXPECT_EQ(1u, info.senders.size());
EXPECT_EQ(1u, info.receivers.size());
VerifyVoiceReceiverInfo(info.receivers[0]);
+ VerifyVoiceSendRecvCodecs(info);
}
}
