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Delete some unused AudioCodingModule methods

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Methods deleted:

  ReceiveFrequency, PlayoutFrequency, ReceiveCodec,
  SetMinimumPlayoutDelay, SetMaximumPlayoutDelay,
  SetBaseMinimumPlayoutDelayMs, GetBaseMinimumPlayoutDelayMs,
  PlayoutTimestamp, FilteredCurrentDelayMs, TargetDelayMs.

Became unused with cl
https://webrtc-review.googlesource.com/c/src/+/111504

Bug: None
Change-Id: Ie50e8e86a622661c3daa9db83a2e66489dcd2d98
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148071
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28918}
This commit is contained in:
Niels Möller
2019-08-13 15:54:15 +02:00
committed by Commit Bot
parent 728a0ee459
commit 5ceb4ac5ed
4 changed files with 12 additions and 188 deletions
Download Patch File Download Diff File Expand all files Collapse all files

25
modules/audio_coding/test/target_delay_unittest.cc
View File

@ -12,6 +12,7 @@
#include "api/audio/audio_frame.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "modules/audio_coding/acm2/acm_receiver.h"
#include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/include/module_common_types.h"
@ -23,19 +24,16 @@ namespace webrtc {
class TargetDelayTest : public ::testing::Test {
protected:
TargetDelayTest()
: acm_(AudioCodingModule::Create(
AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))) {}
: receiver_(
AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory())) {}
~TargetDelayTest() {}
void SetUp() {
EXPECT_TRUE(acm_.get() != NULL);
ASSERT_EQ(0, acm_->InitializeReceiver());
constexpr int pltype = 108;
std::map<int, SdpAudioFormat> receive_codecs = {
{pltype, {"L16", kSampleRateHz, 1}}};
acm_->SetReceiveCodecs(receive_codecs);
receiver_.SetCodecs(receive_codecs);
rtp_header_.payloadType = pltype;
rtp_header_.timestamp = 0;
@ -99,8 +97,9 @@ class TargetDelayTest : public ::testing::Test {
void Push() {
rtp_header_.timestamp += kFrameSizeSamples;
rtp_header_.sequenceNumber++;
ASSERT_EQ(
0, acm_->IncomingPacket(payload_, kFrameSizeSamples * 2, rtp_header_));
ASSERT_EQ(0, receiver_.InsertPacket(rtp_header_,
rtc::ArrayView<const uint8_t>(
payload_, kFrameSizeSamples * 2)));
}
// Pull audio equivalent to the amount of audio in one RTP packet.
@ -108,7 +107,7 @@ class TargetDelayTest : public ::testing::Test {
AudioFrame frame;
bool muted;
for (int k = 0; k < kNum10msPerFrame; ++k) { // Pull one frame.
ASSERT_EQ(0, acm_->PlayoutData10Ms(-1, &frame, &muted));
ASSERT_EQ(0, receiver_.GetAudio(-1, &frame, &muted));
ASSERT_FALSE(muted);
// Had to use ASSERT_TRUE, ASSERT_EQ generated error.
ASSERT_TRUE(kSampleRateHz == frame.sample_rate_hz_);
@ -135,20 +134,20 @@ class TargetDelayTest : public ::testing::Test {
}
int SetMinimumDelay(int delay_ms) {
return acm_->SetMinimumPlayoutDelay(delay_ms);
return receiver_.SetMinimumDelay(delay_ms);
}
int SetMaximumDelay(int delay_ms) {
return acm_->SetMaximumPlayoutDelay(delay_ms);
return receiver_.SetMaximumDelay(delay_ms);
}
int GetCurrentOptimalDelayMs() {
NetworkStatistics stats;
acm_->GetNetworkStatistics(&stats);
receiver_.GetNetworkStatistics(&stats);
return stats.preferredBufferSize;
}
std::unique_ptr<AudioCodingModule> acm_;
acm2::AcmReceiver receiver_;
RTPHeader rtp_header_;
uint8_t payload_[kPayloadLenBytes];
};