Delete deprecated methods on AudioCodingModule
Bug: None Change-Id: I05f1ab6cdd6ac52972835af7ea94aacf5f64d673 Reviewed-on: https://webrtc-review.googlesource.com/95482 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24388}
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@ -21,7 +21,6 @@
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#include "common_types.h" // NOLINT(build/include)
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#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
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#include "modules/audio_coding/neteq/include/neteq.h"
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#include "rtc_base/deprecation.h"
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#include "rtc_base/function_view.h"
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#include "system_wrappers/include/clock.h"
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@ -612,20 +611,6 @@ class AudioCodingModule {
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//
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virtual int LeastRequiredDelayMs() const = 0;
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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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// audio of the packet in question. This function returns the timestamp of
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// the latest audio obtained by calling PlayoutData10ms().
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//
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// Input:
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// -timestamp : a reference to a uint32_t to receive the
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// timestamp.
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// Return value:
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// 0 if the output is a correct timestamp.
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// -1 if failed to output the correct timestamp.
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//
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RTC_DEPRECATED virtual int32_t PlayoutTimestamp(uint32_t* timestamp) = 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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@ -674,14 +659,6 @@ class AudioCodingModule {
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AudioFrame* audio_frame,
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bool* muted) = 0;
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/////////////////////////////////////////////////////////////////////////////
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// Same as above, but without the muted parameter. This methods should not be
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// used if enable_fast_accelerate was set to true in NetEq::Config.
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// TODO(henrik.lundin) Remove this method when downstream dependencies are
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// ready.
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virtual int32_t PlayoutData10Ms(int32_t desired_freq_hz,
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AudioFrame* audio_frame) = 0;
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///////////////////////////////////////////////////////////////////////////
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// Codec specific
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//
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