Propagate base minimum delay to audio_receiver_stream
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}
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@ -100,6 +100,10 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
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// Maximum playout delay.
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int SetMaximumPlayoutDelay(int time_ms) override;
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bool SetBaseMinimumPlayoutDelayMs(int delay_ms) override;
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int GetBaseMinimumPlayoutDelayMs() const override;
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absl::optional<uint32_t> PlayoutTimestamp() override;
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int FilteredCurrentDelayMs() const override;
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@ -708,6 +712,15 @@ int AudioCodingModuleImpl::SetMaximumPlayoutDelay(int time_ms) {
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return receiver_.SetMaximumDelay(time_ms);
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}
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bool AudioCodingModuleImpl::SetBaseMinimumPlayoutDelayMs(int delay_ms) {
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// All necessary validation happens on NetEq level.
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return receiver_.SetBaseMinimumDelayMs(delay_ms);
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}
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int AudioCodingModuleImpl::GetBaseMinimumPlayoutDelayMs() const {
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return receiver_.GetBaseMinimumDelayMs();
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}
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// Get 10 milliseconds of raw audio data to play out.
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// Automatic resample to the requested frequency.
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int AudioCodingModuleImpl::PlayoutData10Ms(int desired_freq_hz,
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@ -275,6 +275,15 @@ class AudioCodingModule {
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//
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virtual int SetMaximumPlayoutDelay(int time_ms) = 0;
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// Sets a base minimum for the playout delay. Base minimum delay sets lower
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// bound minimum delay value which is set via SetMinimumPlayoutDelay.
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//
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// Returns true if value was successfully set, false overwise.
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virtual bool SetBaseMinimumPlayoutDelayMs(int delay_ms) = 0;
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// Returns current value of base minimum delay in milliseconds.
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virtual int GetBaseMinimumPlayoutDelayMs() const = 0;
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///////////////////////////////////////////////////////////////////////////
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// int32_t PlayoutTimestamp()
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// The send timestamp of an RTP packet is associated with the decoded
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