Converted a bunch of error checking in Audio[Receive|Send]Stream to RTC_CHECKs instead. They should never fail.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1442483003
Cr-Commit-Position: refs/heads/master@{#10654}
diff --git a/webrtc/audio/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc
index 8caac6f..bdccea2 100644
--- a/webrtc/audio/audio_receive_stream.cc
+++ b/webrtc/audio/audio_receive_stream.cc
@@ -109,13 +109,12 @@
ScopedVoEInterface<VoERTP_RTCP> rtp(voice_engine);
ScopedVoEInterface<VoEVideoSync> sync(voice_engine);
ScopedVoEInterface<VoEVolumeControl> volume(voice_engine);
- unsigned int ssrc = 0;
+
webrtc::CallStatistics call_stats = {0};
+ int error = rtp->GetRTCPStatistics(config_.voe_channel_id, call_stats);
+ RTC_DCHECK_EQ(0, error);
webrtc::CodecInst codec_inst = {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, call_stats) == -1 ||
- codec->GetRecCodec(config_.voe_channel_id, codec_inst) == -1) {
+ if (codec->GetRecCodec(config_.voe_channel_id, codec_inst) == -1) {
return stats;
}
@@ -123,6 +122,7 @@
stats.packets_rcvd = call_stats.packetsReceived;
stats.packets_lost = call_stats.cumulativeLost;
stats.fraction_lost = Q8ToFloat(call_stats.fractionLost);
+ stats.capture_start_ntp_time_ms = call_stats.capture_start_ntp_time_ms_;
if (codec_inst.pltype != -1) {
stats.codec_name = codec_inst.plname;
}
@@ -139,35 +139,33 @@
}
{
unsigned int level = 0;
- if (volume->GetSpeechOutputLevelFullRange(config_.voe_channel_id, level) !=
- -1) {
- stats.audio_level = static_cast<int32_t>(level);
- }
+ error = volume->GetSpeechOutputLevelFullRange(config_.voe_channel_id,
+ level);
+ RTC_DCHECK_EQ(0, error);
+ stats.audio_level = static_cast<int32_t>(level);
}
+ // Get jitter buffer and total delay (alg + jitter + playout) stats.
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);
- }
+ error = neteq->GetNetworkStatistics(config_.voe_channel_id, ns);
+ RTC_DCHECK_EQ(0, error);
+ 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 = call_stats.capture_start_ntp_time_ms_;
+ error = neteq->GetDecodingCallStatistics(config_.voe_channel_id, &ds);
+ RTC_DCHECK_EQ(0, error);
+ 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;
return stats;
}
@@ -205,7 +203,6 @@
// thread. Then this check can be enabled.
// RTC_DCHECK(!thread_checker_.CalledOnValidThread());
RTPHeader header;
-
if (!rtp_header_parser_->Parse(packet, length, &header)) {
return false;
}
diff --git a/webrtc/audio/audio_receive_stream_unittest.cc b/webrtc/audio/audio_receive_stream_unittest.cc
index edd804f..715b52a 100644
--- a/webrtc/audio/audio_receive_stream_unittest.cc
+++ b/webrtc/audio/audio_receive_stream_unittest.cc
@@ -75,8 +75,6 @@
using testing::DoAll;
using testing::SetArgPointee;
using testing::SetArgReferee;
- EXPECT_CALL(voice_engine_, GetRemoteSSRC(kChannelId, _))
- .WillOnce(DoAll(SetArgReferee<1>(0), Return(0)));
EXPECT_CALL(voice_engine_, GetRTCPStatistics(kChannelId, _))
.WillOnce(DoAll(SetArgReferee<1>(kCallStats), Return(0)));
EXPECT_CALL(voice_engine_, GetRecCodec(kChannelId, _))
diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc
index 14112de..a7b98c7 100644
--- a/webrtc/audio/audio_send_stream.cc
+++ b/webrtc/audio/audio_send_stream.cc
@@ -119,17 +119,20 @@
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};
- // TODO(solenberg): Change error code checking to RTC_CHECK_EQ(..., -1), if
- // possible...
- if (rtp->GetLocalSSRC(config_.voe_channel_id, ssrc) == -1 ||
- rtp->GetRTCPStatistics(config_.voe_channel_id, call_stats) == -1) {
- return stats;
- }
+ webrtc::CallStatistics call_stats = {0};
+ int error = rtp->GetRTCPStatistics(config_.voe_channel_id, call_stats);
+ RTC_DCHECK_EQ(0, error);
stats.bytes_sent = call_stats.bytesSent;
stats.packets_sent = call_stats.packetsSent;
+ // 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;
+ }
+ // TODO(solenberg): [was ajm]: Re-enable this metric once we have a reliable
+ // implementation.
+ stats.aec_quality_min = -1;
webrtc::CodecInst codec_inst = {0};
if (codec->GetSendCodec(config_.voe_channel_id, codec_inst) != -1) {
@@ -138,53 +141,45 @@
// 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;
+ error = rtp->GetRemoteRTCPReportBlocks(config_.voe_channel_id, &blocks);
+ RTC_DCHECK_EQ(0, error);
+ 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);
- }
+ error = volume->GetSpeechInputLevelFullRange(level);
+ RTC_DCHECK_EQ(0, error);
+ 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) {
+ error = processing->GetEcMetricsStatus(echo_metrics_on);
+ RTC_DCHECK_EQ(0, error);
+ if (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;
- }
+ error = processing->GetEcDelayMetrics(median, std, dummy);
+ RTC_DCHECK_EQ(0, error);
+ 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.
@@ -192,10 +187,10 @@
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;
- }
+ error = processing->GetEchoMetrics(erl, erle, dummy1, dummy2);
+ RTC_DCHECK_EQ(0, error);
+ stats.echo_return_loss = erl;
+ stats.echo_return_loss_enhancement = erle;
}
internal::AudioState* audio_state =
diff --git a/webrtc/audio/audio_send_stream_unittest.cc b/webrtc/audio/audio_send_stream_unittest.cc
index 1801e9d..27d4029 100644
--- a/webrtc/audio/audio_send_stream_unittest.cc
+++ b/webrtc/audio/audio_send_stream_unittest.cc
@@ -86,8 +86,6 @@
block.fraction_lost = 0;
report_blocks.push_back(block); // Duplicate SSRC, bad fraction_lost.
- EXPECT_CALL(voice_engine_, GetLocalSSRC(kChannelId, _))
- .WillRepeatedly(DoAll(SetArgReferee<1>(0), Return(0)));
EXPECT_CALL(voice_engine_, GetRTCPStatistics(kChannelId, _))
.WillRepeatedly(DoAll(SetArgReferee<1>(kCallStats), Return(0)));
EXPECT_CALL(voice_engine_, GetSendCodec(kChannelId, _))
