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AudioReceiveStream: Clean up ConfigureStream.

Browse source 
Increases readability of configuration setup by separating
first time configuration setup and reconfiguration.

Bug: None
Change-Id: Ifee29c31bce505a971ab5993cb862a72db94725a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220700
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34159}
This commit is contained in:
Markus Handell 2021-05-29 13:21:28 +02:00 committed by WebRTC LUCI CQ
parent c81afe365d
commit acd16af8c6
2 changed files with 40 additions and 52 deletions
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88
audio/audio_receive_stream.cc
View file

@ -18,6 +18,7 @@
#include "api/audio_codecs/audio_format.h"
#include "api/call/audio_sink.h"
#include "api/rtp_parameters.h"
#include "api/sequence_checker.h"
#include "audio/audio_send_stream.h"
#include "audio/audio_state.h"
#include "audio/channel_receive.h"
@ -118,7 +119,8 @@ AudioReceiveStream::AudioReceiveStream(
const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
webrtc::RtcEventLog* event_log,
std::unique_ptr<voe::ChannelReceiveInterface> channel_receive)
: audio_state_(audio_state),
: config_(config),
audio_state_(audio_state),
source_tracker_(clock),
channel_receive_(std::move(channel_receive)) {
RTC_LOG(LS_INFO) << "AudioReceiveStream: " << config.rtp.remote_ssrc;
@ -140,7 +142,15 @@ AudioReceiveStream::AudioReceiveStream(
// Register with transport.
rtp_stream_receiver_ = receiver_controller->CreateReceiver(
config.rtp.remote_ssrc, channel_receive_.get());
ConfigureStream(this, config, true);
// Complete configuration.
// TODO(solenberg): Config NACK history window (which is a packet count),
// using the actual packet size for the configured codec.
channel_receive_->SetNACKStatus(config.rtp.nack.rtp_history_ms != 0,
config.rtp.nack.rtp_history_ms / 20);
channel_receive_->SetReceiveCodecs(config.decoder_map);
channel_receive_->SetDepacketizerToDecoderFrameTransformer(
config.frame_transformer);
}
AudioReceiveStream::~AudioReceiveStream() {
@ -154,7 +164,34 @@ AudioReceiveStream::~AudioReceiveStream() {
void AudioReceiveStream::Reconfigure(
const webrtc::AudioReceiveStream::Config& config) {
RTC_DCHECK(worker_thread_checker_.IsCurrent());
ConfigureStream(this, config, false);
// Configuration parameters which cannot be changed.
RTC_DCHECK(config_.rtp.remote_ssrc == config.rtp.remote_ssrc);
RTC_DCHECK(config_.rtcp_send_transport == config.rtcp_send_transport);
// Decoder factory cannot be changed because it is configured at
// voe::Channel construction time.
RTC_DCHECK(config_.decoder_factory == config.decoder_factory);
// SSRC can't be changed mid-stream.
RTC_DCHECK_EQ(config_.rtp.local_ssrc, config.rtp.local_ssrc);
RTC_DCHECK_EQ(config_.rtp.remote_ssrc, config.rtp.remote_ssrc);
// TODO(solenberg): Config NACK history window (which is a packet count),
// using the actual packet size for the configured codec.
if (config_.rtp.nack.rtp_history_ms != config.rtp.nack.rtp_history_ms) {
channel_receive_->SetNACKStatus(config.rtp.nack.rtp_history_ms != 0,
config.rtp.nack.rtp_history_ms / 20);
}
if (config_.decoder_map != config.decoder_map) {
channel_receive_->SetReceiveCodecs(config.decoder_map);
}
if (config_.frame_transformer != config.frame_transformer) {
channel_receive_->SetDepacketizerToDecoderFrameTransformer(
config.frame_transformer);
}
config_ = config;
}
void AudioReceiveStream::Start() {
@ -384,50 +421,5 @@ internal::AudioState* AudioReceiveStream::audio_state() const {
RTC_DCHECK(audio_state);
return audio_state;
}
void AudioReceiveStream::ConfigureStream(AudioReceiveStream* stream,
const Config& new_config,
bool first_time) {
RTC_LOG(LS_INFO) << "AudioReceiveStream::ConfigureStream: "
<< new_config.ToString();
RTC_DCHECK(stream);
const auto& channel_receive = stream->channel_receive_;
const auto& old_config = stream->config_;
// Configuration parameters which cannot be changed.
RTC_DCHECK(first_time ||
old_config.rtp.remote_ssrc == new_config.rtp.remote_ssrc);
RTC_DCHECK(first_time ||
old_config.rtcp_send_transport == new_config.rtcp_send_transport);
// Decoder factory cannot be changed because it is configured at
// voe::Channel construction time.
RTC_DCHECK(first_time ||
old_config.decoder_factory == new_config.decoder_factory);
if (!first_time) {
// SSRC can't be changed mid-stream.
RTC_DCHECK_EQ(old_config.rtp.local_ssrc, new_config.rtp.local_ssrc);
RTC_DCHECK_EQ(old_config.rtp.remote_ssrc, new_config.rtp.remote_ssrc);
}
// TODO(solenberg): Config NACK history window (which is a packet count),
// using the actual packet size for the configured codec.
if (first_time || old_config.rtp.nack.rtp_history_ms !=
new_config.rtp.nack.rtp_history_ms) {
channel_receive->SetNACKStatus(new_config.rtp.nack.rtp_history_ms != 0,
new_config.rtp.nack.rtp_history_ms / 20);
}
if (first_time || old_config.decoder_map != new_config.decoder_map) {
channel_receive->SetReceiveCodecs(new_config.decoder_map);
}
if (first_time ||
old_config.frame_transformer != new_config.frame_transformer) {
channel_receive->SetDepacketizerToDecoderFrameTransformer(
new_config.frame_transformer);
}
stream->config_ = new_config;
}
} // namespace internal
} // namespace webrtc

4
audio/audio_receive_stream.h
View file

@ -102,10 +102,6 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream,
const AudioSendStream* GetAssociatedSendStreamForTesting() const;
private:
static void ConfigureStream(AudioReceiveStream* stream,
const Config& new_config,
bool first_time);
AudioState* audio_state() const;
SequenceChecker worker_thread_checker_;