Implement AudioSendStream::GetStats().
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1414743004
Cr-Commit-Position: refs/heads/master@{#10424}
diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc
index 0d0c072..ccfdca5 100644
--- a/webrtc/audio/audio_send_stream.cc
+++ b/webrtc/audio/audio_send_stream.cc
@@ -12,8 +12,13 @@
#include <string>
+#include "webrtc/audio/conversion.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
+#include "webrtc/voice_engine/include/voe_audio_processing.h"
+#include "webrtc/voice_engine/include/voe_codec.h"
+#include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
+#include "webrtc/voice_engine/include/voe_volume_control.h"
namespace webrtc {
std::string AudioSendStream::Config::Rtp::ToString() const {
@@ -22,8 +27,9 @@
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 << '}';
@@ -42,30 +48,134 @@
}
namespace internal {
-AudioSendStream::AudioSendStream(const webrtc::AudioSendStream::Config& config)
- : config_(config) {
+AudioSendStream::AudioSendStream(const webrtc::AudioSendStream::Config& config,
+ VoiceEngine* voice_engine)
+ : config_(config),
+ voice_engine_(voice_engine),
+ voe_base_(voice_engine) {
LOG(LS_INFO) << "AudioSendStream: " << config_.ToString();
- RTC_DCHECK(config.voe_channel_id != -1);
+ RTC_DCHECK_NE(config.voe_channel_id, -1);
+ RTC_DCHECK(voice_engine_);
}
AudioSendStream::~AudioSendStream() {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
LOG(LS_INFO) << "~AudioSendStream: " << config_.ToString();
}
webrtc::AudioSendStream::Stats AudioSendStream::GetStats() const {
- return webrtc::AudioSendStream::Stats();
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ webrtc::AudioSendStream::Stats stats;
+ stats.local_ssrc = config_.rtp.ssrc;
+ ScopedVoEInterface<VoEAudioProcessing> processing(voice_engine_);
+ ScopedVoEInterface<VoECodec> codec(voice_engine_);
+ ScopedVoEInterface<VoERTP_RTCP> rtp(voice_engine_);
+ ScopedVoEInterface<VoEVolumeControl> volume(voice_engine_);
+ unsigned int ssrc = 0;
+ webrtc::CallStatistics call_stats = {0};
+ if (rtp->GetLocalSSRC(config_.voe_channel_id, ssrc) == -1 ||
+ rtp->GetRTCPStatistics(config_.voe_channel_id, call_stats) == -1) {
+ return stats;
+ }
+
+ stats.bytes_sent = call_stats.bytesSent;
+ stats.packets_sent = call_stats.packetsSent;
+
+ webrtc::CodecInst codec_inst = {0};
+ if (codec->GetSendCodec(config_.voe_channel_id, codec_inst) != -1) {
+ RTC_DCHECK_NE(codec_inst.pltype, -1);
+ stats.codec_name = codec_inst.plname;
+
+ // Get data from the last remote RTCP report.
+ std::vector<webrtc::ReportBlock> blocks;
+ if (rtp->GetRemoteRTCPReportBlocks(config_.voe_channel_id, &blocks) != -1) {
+ for (const webrtc::ReportBlock& block : blocks) {
+ // Lookup report for send ssrc only.
+ if (block.source_SSRC == stats.local_ssrc) {
+ stats.packets_lost = block.cumulative_num_packets_lost;
+ stats.fraction_lost = Q8ToFloat(block.fraction_lost);
+ stats.ext_seqnum = block.extended_highest_sequence_number;
+ // Convert samples to milliseconds.
+ if (codec_inst.plfreq / 1000 > 0) {
+ stats.jitter_ms =
+ block.interarrival_jitter / (codec_inst.plfreq / 1000);
+ }
+ break;
+ }
+ }
+ }
+ }
+
+ // RTT isn't known until a RTCP report is received. Until then, VoiceEngine
+ // returns 0 to indicate an error value.
+ if (call_stats.rttMs > 0) {
+ stats.rtt_ms = call_stats.rttMs;
+ }
+
+ // Local speech level.
+ {
+ unsigned int level = 0;
+ if (volume->GetSpeechInputLevelFullRange(level) != -1) {
+ stats.audio_level = static_cast<int32_t>(level);
+ }
+ }
+
+ // TODO(ajm): Re-enable this metric once we have a reliable implementation.
+ stats.aec_quality_min = -1;
+
+ bool echo_metrics_on = false;
+ if (processing->GetEcMetricsStatus(echo_metrics_on) != -1 &&
+ echo_metrics_on) {
+ // These can also be negative, but in practice -1 is only used to signal
+ // insufficient data, since the resolution is limited to multiples of 4 ms.
+ int median = -1;
+ int std = -1;
+ float dummy = 0.0f;
+ if (processing->GetEcDelayMetrics(median, std, dummy) != -1) {
+ stats.echo_delay_median_ms = median;
+ stats.echo_delay_std_ms = std;
+ }
+
+ // These can take on valid negative values, so use the lowest possible level
+ // as default rather than -1.
+ int erl = -100;
+ int erle = -100;
+ int dummy1 = 0;
+ int dummy2 = 0;
+ if (processing->GetEchoMetrics(erl, erle, dummy1, dummy2) != -1) {
+ stats.echo_return_loss = erl;
+ stats.echo_return_loss_enhancement = erle;
+ }
+ }
+
+ // TODO(solenberg): Collect typing noise warnings here too!
+ // bool typing_noise_detected = typing_noise_detected_;
+
+ return stats;
+}
+
+const webrtc::AudioSendStream::Config& AudioSendStream::config() const {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ return config_;
}
void AudioSendStream::Start() {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
}
void AudioSendStream::Stop() {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
}
void AudioSendStream::SignalNetworkState(NetworkState state) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
}
bool AudioSendStream::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;
}
} // namespace internal