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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

81
modules/audio_coding/acm2/audio_coding_module.cc
View File

@ -74,38 +74,13 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
// Initialize receiver, resets codec database etc.
int InitializeReceiver() override;
// Get current receive frequency.
int ReceiveFrequency() const override;
// Get current playout frequency.
int PlayoutFrequency() const override;
void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs) override;
// Get current received codec.
absl::optional<std::pair<int, SdpAudioFormat>> ReceiveCodec() const override;
// Incoming packet from network parsed and ready for decode.
int IncomingPacket(const uint8_t* incoming_payload,
const size_t payload_length,
const RTPHeader& rtp_info) override;
// Minimum playout delay.
int SetMinimumPlayoutDelay(int time_ms) override;
// Maximum playout delay.
int SetMaximumPlayoutDelay(int time_ms) override;
bool SetBaseMinimumPlayoutDelayMs(int delay_ms) override;
int GetBaseMinimumPlayoutDelayMs() const override;
absl::optional<uint32_t> PlayoutTimestamp() override;
int FilteredCurrentDelayMs() const override;
int TargetDelayMs() const override;
// Get 10 milliseconds of raw audio data to play out, and
// automatic resample to the requested frequency if > 0.
int PlayoutData10Ms(int desired_freq_hz,
@ -605,30 +580,12 @@ int AudioCodingModuleImpl::InitializeReceiverSafe() {
return 0;
}
// Get current receive frequency.
int AudioCodingModuleImpl::ReceiveFrequency() const {
const auto last_packet_sample_rate = receiver_.last_packet_sample_rate_hz();
return last_packet_sample_rate ? *last_packet_sample_rate
: receiver_.last_output_sample_rate_hz();
}
// Get current playout frequency.
int AudioCodingModuleImpl::PlayoutFrequency() const {
return receiver_.last_output_sample_rate_hz();
}
void AudioCodingModuleImpl::SetReceiveCodecs(
const std::map<int, SdpAudioFormat>& codecs) {
rtc::CritScope lock(&acm_crit_sect_);
receiver_.SetCodecs(codecs);
}
absl::optional<std::pair<int, SdpAudioFormat>>
AudioCodingModuleImpl::ReceiveCodec() const {
rtc::CritScope lock(&acm_crit_sect_);
return receiver_.LastDecoder();
}
// Incoming packet from network parsed and ready for decode.
int AudioCodingModuleImpl::IncomingPacket(const uint8_t* incoming_payload,
const size_t payload_length,
@ -639,32 +596,6 @@ int AudioCodingModuleImpl::IncomingPacket(const uint8_t* incoming_payload,
rtc::ArrayView<const uint8_t>(incoming_payload, payload_length));
}
// Minimum playout delay (Used for lip-sync).
int AudioCodingModuleImpl::SetMinimumPlayoutDelay(int time_ms) {
if ((time_ms < 0) || (time_ms > 10000)) {
RTC_LOG(LS_ERROR) << "Delay must be in the range of 0-10000 milliseconds.";
return -1;
}
return receiver_.SetMinimumDelay(time_ms);
}
int AudioCodingModuleImpl::SetMaximumPlayoutDelay(int time_ms) {
if ((time_ms < 0) || (time_ms > 10000)) {
RTC_LOG(LS_ERROR) << "Delay must be in the range of 0-10000 milliseconds.";
return -1;
}
return receiver_.SetMaximumDelay(time_ms);
}
bool AudioCodingModuleImpl::SetBaseMinimumPlayoutDelayMs(int delay_ms) {
// All necessary validation happens on NetEq level.
return receiver_.SetBaseMinimumDelayMs(delay_ms);
}
int AudioCodingModuleImpl::GetBaseMinimumPlayoutDelayMs() const {
return receiver_.GetBaseMinimumDelayMs();
}
// Get 10 milliseconds of raw audio data to play out.
// Automatic resample to the requested frequency.
int AudioCodingModuleImpl::PlayoutData10Ms(int desired_freq_hz,
@ -696,18 +627,6 @@ int AudioCodingModuleImpl::RegisterVADCallback(ACMVADCallback* vad_callback) {
return 0;
}
absl::optional<uint32_t> AudioCodingModuleImpl::PlayoutTimestamp() {
return receiver_.GetPlayoutTimestamp();
}
int AudioCodingModuleImpl::FilteredCurrentDelayMs() const {
return receiver_.FilteredCurrentDelayMs();
}
int AudioCodingModuleImpl::TargetDelayMs() const {
return receiver_.TargetDelayMs();
}
bool AudioCodingModuleImpl::HaveValidEncoder(const char* caller_name) const {
if (!encoder_stack_) {
RTC_LOG(LS_ERROR) << caller_name << " failed: No send codec is registered.";