diff options
author | solenberg <solenberg@webrtc.org> | 2015-11-16 09:48:04 -0800 |
---|---|---|
committer | Commit bot <commit-bot@chromium.org> | 2015-11-16 17:48:12 +0000 |
commit | 8b85de2ba1a8885b70bf9fe8beadc54c5c405335 (patch) | |
tree | 70dbb306823e9746031ed9e857b3a7b1a115fe92 /webrtc/audio/audio_send_stream.cc | |
parent | 9a7c838ec419fcef29e40e6089acc90406181644 (diff) | |
download | webrtc-8b85de2ba1a8885b70bf9fe8beadc54c5c405335.tar.gz |
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}
Diffstat (limited to 'webrtc/audio/audio_send_stream.cc')
-rw-r--r-- | webrtc/audio/audio_send_stream.cc | 81 |
1 files changed, 38 insertions, 43 deletions
diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc index 14112deb9a..a7b98c7670 100644 --- a/webrtc/audio/audio_send_stream.cc +++ b/webrtc/audio/audio_send_stream.cc @@ -119,17 +119,20 @@ webrtc::AudioSendStream::Stats AudioSendStream::GetStats() const { 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 @@ webrtc::AudioSendStream::Stats AudioSendStream::GetStats() const { // 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 @@ webrtc::AudioSendStream::Stats AudioSendStream::GetStats() const { 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 = |