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Move event logging of config into AudioSendStream.

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It was previously logged in Call, but streams are not always created
with the full configuration, which caused header extensions to be
missing from the log.

Bug: webrtc:9885
Change-Id: I86c0424004c4629ebab0f6b155b83fb90e15b131
Reviewed-on: https://webrtc-review.googlesource.com/c/108601
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25483}
This commit is contained in:
Oskar Sundbom
2018-10-30 16:11:02 +01:00
committed by Commit Bot
parent 6bf20549bb
commit 56ef305b67
3 changed files with 42 additions and 16 deletions
Download Patch File Download Diff File Expand all files Collapse all files

1
audio/BUILD.gn
View File

@ -65,6 +65,7 @@ rtc_static_library("audio") {
"../common_audio:common_audio_c",
"../logging:rtc_event_audio",
"../logging:rtc_event_log_api",
"../logging:rtc_stream_config",
"../modules/audio_coding",
"../modules/audio_coding:audio_format_conversion",
"../modules/audio_coding:audio_network_adaptor_config",

39
audio/audio_send_stream.cc
View File

@ -28,6 +28,9 @@
#include "call/rtp_transport_controller_send_interface.h"
#include "common_audio/vad/include/vad.h"
#include "common_types.h" // NOLINT(build/include)
#include "logging/rtc_event_log/events/rtc_event_audio_send_stream_config.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "logging/rtc_event_log/rtc_stream_config.h"
#include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "rtc_base/checks.h"
@ -69,6 +72,39 @@ std::unique_ptr<voe::ChannelSendProxy> CreateChannelAndProxy(
worker_queue, module_process_thread, media_transport, rtcp_rtt_stats,
event_log, frame_encryptor, crypto_options, extmap_allow_mixed));
}
void UpdateEventLogStreamConfig(RtcEventLog* event_log,
const AudioSendStream::Config& config,
const AudioSendStream::Config* old_config) {
using SendCodecSpec = AudioSendStream::Config::SendCodecSpec;
// Only update if any of the things we log have changed.
auto payload_types_equal = [](const absl::optional<SendCodecSpec>& a,
const absl::optional<SendCodecSpec>& b) {
if (a.has_value() && b.has_value()) {
return a->format.name == b->format.name &&
a->payload_type == b->payload_type;
}
return !a.has_value() && !b.has_value();
};
if (old_config && config.rtp.ssrc == old_config->rtp.ssrc &&
config.rtp.extensions == old_config->rtp.extensions &&
payload_types_equal(config.send_codec_spec,
old_config->send_codec_spec)) {
return;
}
auto rtclog_config = absl::make_unique<rtclog::StreamConfig>();
rtclog_config->local_ssrc = config.rtp.ssrc;
rtclog_config->rtp_extensions = config.rtp.extensions;
if (config.send_codec_spec) {
rtclog_config->codecs.emplace_back(config.send_codec_spec->format.name,
config.send_codec_spec->payload_type, 0);
}
event_log->Log(absl::make_unique<RtcEventAudioSendStreamConfig>(
std::move(rtclog_config)));
}
} // namespace
// Helper class to track the actively sending lifetime of this stream.
@ -221,6 +257,9 @@ void AudioSendStream::ConfigureStream(
bool first_time) {
RTC_LOG(LS_INFO) << "AudioSendStream::ConfigureStream: "
<< new_config.ToString();
UpdateEventLogStreamConfig(stream->event_log_, new_config,
first_time ? nullptr : &stream->config_);
const auto& channel_proxy = stream->channel_proxy_;
const auto& old_config = stream->config_;

18
call/call.cc
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@ -30,7 +30,6 @@
#include "call/rtp_stream_receiver_controller.h"
#include "call/rtp_transport_controller_send.h"
#include "logging/rtc_event_log/events/rtc_event_audio_receive_stream_config.h"
#include "logging/rtc_event_log/events/rtc_event_audio_send_stream_config.h"
#include "logging/rtc_event_log/events/rtc_event_rtcp_packet_incoming.h"
#include "logging/rtc_event_log/events/rtc_event_rtp_packet_incoming.h"
#include "logging/rtc_event_log/events/rtc_event_video_receive_stream_config.h"
@ -147,18 +146,6 @@ std::unique_ptr<rtclog::StreamConfig> CreateRtcLogStreamConfig(
return rtclog_config;
}
std::unique_ptr<rtclog::StreamConfig> CreateRtcLogStreamConfig(
const AudioSendStream::Config& config) {
auto rtclog_config = absl::make_unique<rtclog::StreamConfig>();
rtclog_config->local_ssrc = config.rtp.ssrc;
rtclog_config->rtp_extensions = config.rtp.extensions;
if (config.send_codec_spec) {
rtclog_config->codecs.emplace_back(config.send_codec_spec->format.name,
config.send_codec_spec->payload_type, 0);
}
return rtclog_config;
}
} // namespace
namespace internal {
@ -579,9 +566,8 @@ webrtc::AudioSendStream* Call::CreateAudioSendStream(
const webrtc::AudioSendStream::Config& config) {
TRACE_EVENT0("webrtc", "Call::CreateAudioSendStream");
RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
event_log_->Log(absl::make_unique<RtcEventAudioSendStreamConfig>(
CreateRtcLogStreamConfig(config)));
// Stream config is logged in AudioSendStream::ConfigureStream, as it may
// change during the stream's lifetime.
absl::optional<RtpState> suspended_rtp_state;
{
const auto& iter = suspended_audio_send_ssrcs_.find(config.rtp.ssrc);