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Propagate base minimum delay to audio_receiver_stream

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Bug: webrtc:10287
Change-Id: Id7914976ef5b7eb708802119932b554d9ce4879e
Reviewed-on: https://webrtc-review.googlesource.com/c/121563
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26563}
This commit is contained in:
Ruslan Burakov
2019-02-06 09:45:56 +01:00
committed by Commit Bot
parent 9ce800d6d1
commit 3b50f9f9ce
9 changed files with 71 additions and 0 deletions
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13
modules/audio_coding/acm2/audio_coding_module.cc
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@ -100,6 +100,10 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
// 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;
@ -708,6 +712,15 @@ int AudioCodingModuleImpl::SetMaximumPlayoutDelay(int time_ms) {
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,