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Routing BitrateAllocationUpdate to audio codec.

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This will be used in a later CL to use the link capacity field in the
update to control the Opus encoder.

Bug: webrtc:9718
Change-Id: If2ad16a8f4656e8cdf10c33f5fb060ef7ca5caba
Reviewed-on: https://webrtc-review.googlesource.com/c/111510
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25761}
This commit is contained in:
Sebastian Jansson 2018-11-21 19:19:00 +01:00 committed by Commit Bot
parent 3890c1ae6d
commit 254d869c00
5 changed files with 16 additions and 11 deletions
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3
audio/audio_send_stream.cc
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@ -475,8 +475,7 @@ uint32_t AudioSendStream::OnBitrateUpdated(BitrateAllocationUpdate update) {
if (update.target_bitrate > max_bitrate)
update.target_bitrate = max_bitrate;
channel_send_->SetBitrate(update.target_bitrate.bps(),
update.bwe_period.ms());
channel_send_->OnBitrateAllocation(update);
// The amount of audio protection is not exposed by the encoder, hence
// always returning 0.

10
audio/audio_send_stream_unittest.cc
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@ -42,6 +42,7 @@ namespace {
using testing::_;
using testing::Eq;
using testing::Ne;
using testing::Field;
using testing::Invoke;
using testing::Return;
using testing::StrEq;
@ -472,7 +473,9 @@ TEST(AudioSendStreamTest, DoesNotPassHigherBitrateThanMaxBitrate) {
ConfigHelper helper(false, true);
auto send_stream = helper.CreateAudioSendStream();
EXPECT_CALL(*helper.channel_send(),
SetBitrate(helper.config().max_bitrate_bps, _));
OnBitrateAllocation(
Field(&BitrateAllocationUpdate::target_bitrate,
Eq(DataRate::bps(helper.config().max_bitrate_bps)))));
BitrateAllocationUpdate update;
update.target_bitrate = DataRate::bps(helper.config().max_bitrate_bps + 5000);
update.packet_loss_ratio = 0;
@ -484,7 +487,10 @@ TEST(AudioSendStreamTest, DoesNotPassHigherBitrateThanMaxBitrate) {
TEST(AudioSendStreamTest, ProbingIntervalOnBitrateUpdated) {
ConfigHelper helper(false, true);
auto send_stream = helper.CreateAudioSendStream();
EXPECT_CALL(*helper.channel_send(), SetBitrate(_, 5000));
EXPECT_CALL(*helper.channel_send(),
OnBitrateAllocation(Field(&BitrateAllocationUpdate::bwe_period,
Eq(TimeDelta::ms(5000)))));
BitrateAllocationUpdate update;
update.target_bitrate = DataRate::bps(helper.config().max_bitrate_bps + 5000);
update.packet_loss_ratio = 0;

10
audio/channel_send.cc
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@ -111,7 +111,7 @@ class ChannelSend
void StopSend() override;
// Codecs
void SetBitrate(int bitrate_bps, int64_t probing_interval_ms) override;
void OnBitrateAllocation(BitrateAllocationUpdate update) override;
int GetBitrate() const override;
// Network
@ -898,7 +898,7 @@ void ChannelSend::ModifyEncoder(
audio_coding_->ModifyEncoder(modifier);
}
void ChannelSend::SetBitrate(int bitrate_bps, int64_t probing_interval_ms) {
void ChannelSend::OnBitrateAllocation(BitrateAllocationUpdate update) {
// This method can be called on the worker thread, module process thread
// or on a TaskQueue via VideoSendStreamImpl::OnEncoderConfigurationChanged.
// TODO(solenberg): Figure out a good way to check this or enforce calling
@ -909,11 +909,11 @@ void ChannelSend::SetBitrate(int bitrate_bps, int64_t probing_interval_ms) {
audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
if (*encoder) {
(*encoder)->OnReceivedUplinkBandwidth(bitrate_bps, probing_interval_ms);
(*encoder)->OnReceivedUplinkAllocation(update);
}
});
retransmission_rate_limiter_->SetMaxRate(bitrate_bps);
configured_bitrate_bps_ = bitrate_bps;
retransmission_rate_limiter_->SetMaxRate(update.target_bitrate.bps());
configured_bitrate_bps_ = update.target_bitrate.bps();
}
int ChannelSend::GetBitrate() const {

2
audio/channel_send.h
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@ -80,7 +80,7 @@ class ChannelSendInterface {
virtual bool SetSendTelephoneEventPayloadType(int payload_type,
int payload_frequency) = 0;
virtual bool SendTelephoneEventOutband(int event, int duration_ms) = 0;
virtual void SetBitrate(int bitrate_bps, int64_t probing_interval_ms) = 0;
virtual void OnBitrateAllocation(BitrateAllocationUpdate update) = 0;
virtual int GetBitrate() const = 0;
virtual void SetInputMute(bool muted) = 0;

2
audio/mock_voe_channel_proxy.h
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@ -88,7 +88,7 @@ class MockChannelSend : public voe::ChannelSendInterface {
MOCK_METHOD2(SetSendTelephoneEventPayloadType,
bool(int payload_type, int payload_frequency));
MOCK_METHOD2(SendTelephoneEventOutband, bool(int event, int duration_ms));
MOCK_METHOD2(SetBitrate, void(int bitrate_bps, int64_t probing_interval_ms));
MOCK_METHOD1(OnBitrateAllocation, void(BitrateAllocationUpdate update));
MOCK_METHOD1(SetInputMute, void(bool muted));
MOCK_METHOD1(RegisterTransport, void(Transport* transport));
MOCK_METHOD2(ReceivedRTCPPacket, bool(const uint8_t* packet, size_t length));