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Delete a chain of methods in ViE, VoE and ACM

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The end goal is to remove AcmReceiver::SetInitialDelay. This change is
in preparation for that goal. It turns out that
AcmReceiver::SetInitialDelay was only invoked through the following
call chain, where each method in the chain is never referenced from
anywhere else (except from tests in some cases):

ViEChannel::SetReceiverBufferingMode
-> ViESyncModule::SetTargetBufferingDelay
-> VoEVideoSync::SetInitialPlayoutDelay
-> Channel::SetInitialPlayoutDelay
-> AudioCodingModule::SetInitialPlayoutDelay
-> AcmReceiver::SetInitialDelay

The start of the chain, ViEChannel::SetReceiverBufferingMode was never
referenced.

This change deletes all the methods above except
AcmReceiver::SetInitialDelay itself, which will be handled in a
follow-up change.

BUG=webrtc:3520

Review URL: https://codereview.webrtc.org/1421013006

Cr-Commit-Position: refs/heads/master@{#10471}
This commit is contained in:
henrik.lundin
2015-11-01 11:43:30 -08:00
committed by Commit bot
parent e502bbe138
commit 74f0f3551e
18 changed files with 1 additions and 341 deletions
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11
webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
View File

@ -765,17 +765,6 @@ int AudioCodingModuleImpl::UnregisterReceiveCodec(uint8_t payload_type) {
return receiver_.RemoveCodec(payload_type);
}
int AudioCodingModuleImpl::SetInitialPlayoutDelay(int delay_ms) {
{
CriticalSectionScoped lock(acm_crit_sect_.get());
// Initialize receiver, if it is not initialized. Otherwise, initial delay
// is reset upon initialization of the receiver.
if (!receiver_initialized_)
InitializeReceiverSafe();
}
return receiver_.SetInitialDelay(delay_ms);
}
int AudioCodingModuleImpl::EnableNack(size_t max_nack_list_size) {
return receiver_.EnableNack(max_nack_list_size);
}

4
webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
View File

@ -150,10 +150,6 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
// Smallest latency NetEq will maintain.
int LeastRequiredDelayMs() const override;
// Impose an initial delay on playout. ACM plays silence until |delay_ms|
// audio is accumulated in NetEq buffer, then starts decoding payloads.
int SetInitialPlayoutDelay(int delay_ms) override;
// Get playout timestamp.
int PlayoutTimestamp(uint32_t* timestamp) override;

24
webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
View File

@ -249,30 +249,6 @@ TEST_F(AudioCodingModuleTestOldApi, DISABLED_ON_ANDROID(InitializedToZero)) {
EXPECT_EQ(0, stats.decoded_plc_cng);
}
// Apply an initial playout delay. Calls to AudioCodingModule::PlayoutData10ms()
// should result in generating silence, check the associated field.
TEST_F(AudioCodingModuleTestOldApi,
DISABLED_ON_ANDROID(SilenceGeneratorCalled)) {
RegisterCodec();
AudioDecodingCallStats stats;
const int kInitialDelay = 100;
acm_->SetInitialPlayoutDelay(kInitialDelay);
int num_calls = 0;
for (int time_ms = 0; time_ms < kInitialDelay;
time_ms += kFrameSizeMs, ++num_calls) {
InsertPacketAndPullAudio();
}
acm_->GetDecodingCallStatistics(&stats);
EXPECT_EQ(0, stats.calls_to_neteq);
EXPECT_EQ(num_calls, stats.calls_to_silence_generator);
EXPECT_EQ(0, stats.decoded_normal);
EXPECT_EQ(0, stats.decoded_cng);
EXPECT_EQ(0, stats.decoded_plc);
EXPECT_EQ(0, stats.decoded_plc_cng);
}
// Insert some packets and pull audio. Check statistics are valid. Then,
// simulate packet loss and check if PLC and PLC-to-CNG statistics are
// correctly updated.