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Delete expired and unused field trial WebRTC-Audio-OpusPlcUsePrevDecodedSamples

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Bug: b/143582588, webrtc:42221607
Change-Id: I49f477ab785801c8ef7143ab8b8654dd7379dfbd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359560
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42783}
This commit is contained in:
Danil Chapovalov 2024-08-14 14:36:31 +02:00 committed by WebRTC LUCI CQ
parent eb26634e6a
commit 759f8d80f0
4 changed files with 1 additions and 54 deletions
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5
experiments/field_trials.py
View File

@ -475,9 +475,6 @@ POLICY_EXEMPT_FIELD_TRIALS: FrozenSet[FieldTrial] = frozenset([
FieldTrial('WebRTC-Audio-OpusBitrateMultipliers',
42221139,
date(2024, 4, 1)),
FieldTrial('WebRTC-Audio-OpusPlcUsePrevDecodedSamples',
143582588,
date(2024, 4, 1)),
FieldTrial('WebRTC-Audio-Red-For-Opus',
42221750,
date(2024, 4, 1)),
@ -878,7 +875,7 @@ POLICY_EXEMPT_FIELD_TRIALS: FrozenSet[FieldTrial] = frozenset([
]) # yapf: disable
POLICY_EXEMPT_FIELD_TRIALS_DIGEST: str = \
'95050aa4a628b16d1220b3674b543c035c24cb6c'
'e750429cb042c02921a227fa18291f2cea23ea3b'
REGISTERED_FIELD_TRIALS: FrozenSet[FieldTrial] = ACTIVE_FIELD_TRIALS.union(
POLICY_EXEMPT_FIELD_TRIALS)

1
modules/audio_coding/BUILD.gn
View File

@ -545,7 +545,6 @@ rtc_library("webrtc_opus_wrapper") {
"../../api:array_view",
"../../rtc_base:checks",
"../../rtc_base:ignore_wundef",
"../../system_wrappers:field_trial",
]
if (rtc_build_opus) {

2
modules/audio_coding/codecs/opus/opus_inst.h
View File

@ -31,8 +31,6 @@ struct WebRtcOpusEncInst {
struct WebRtcOpusDecInst {
OpusDecoder* decoder;
OpusMSDecoder* multistream_decoder;
int prev_decoded_samples;
bool plc_use_prev_decoded_samples;
size_t channels;
int in_dtx_mode;
int sample_rate_hz;

47
modules/audio_coding/codecs/opus/opus_interface.cc
View File

@ -15,7 +15,6 @@
#include "api/array_view.h"
#include "rtc_base/checks.h"
#include "system_wrappers/include/field_trial.h"
enum {
#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
@ -35,9 +34,6 @@ enum {
kWebRtcOpusPlcFrameSizeMs = 10,
};
constexpr char kPlcUsePrevDecodedSamplesFieldTrial[] =
"WebRTC-Audio-OpusPlcUsePrevDecodedSamples";
static int FrameSizePerChannel(int frame_size_ms, int sample_rate_hz) {
RTC_DCHECK_GT(frame_size_ms, 0);
RTC_DCHECK_EQ(frame_size_ms % 10, 0);
@ -51,11 +47,6 @@ static int MaxFrameSizePerChannel(int sample_rate_hz) {
return FrameSizePerChannel(kWebRtcOpusMaxDecodeFrameSizeMs, sample_rate_hz);
}
// Default sample count per channel.
static int DefaultFrameSizePerChannel(int sample_rate_hz) {
return FrameSizePerChannel(20, sample_rate_hz);
}
int16_t WebRtcOpus_EncoderCreate(OpusEncInst** inst,
size_t channels,
int32_t application,
@ -394,12 +385,6 @@ int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst,
// Creation of memory all ok.
state->channels = channels;
state->sample_rate_hz = sample_rate_hz;
state->plc_use_prev_decoded_samples =
webrtc::field_trial::IsEnabled(kPlcUsePrevDecodedSamplesFieldTrial);
if (state->plc_use_prev_decoded_samples) {
state->prev_decoded_samples =
DefaultFrameSizePerChannel(state->sample_rate_hz);
}
state->in_dtx_mode = 0;
*inst = state;
return 0;
@ -438,12 +423,6 @@ int16_t WebRtcOpus_MultistreamDecoderCreate(
// Creation of memory all ok.
state->channels = channels;
state->sample_rate_hz = 48000;
state->plc_use_prev_decoded_samples =
webrtc::field_trial::IsEnabled(kPlcUsePrevDecodedSamplesFieldTrial);
if (state->plc_use_prev_decoded_samples) {
state->prev_decoded_samples =
DefaultFrameSizePerChannel(state->sample_rate_hz);
}
state->in_dtx_mode = 0;
*inst = state;
return 0;
@ -542,17 +521,6 @@ static int DecodePlc(OpusDecInst* inst, int16_t* decoded) {
int plc_samples =
FrameSizePerChannel(kWebRtcOpusPlcFrameSizeMs, inst->sample_rate_hz);
if (inst->plc_use_prev_decoded_samples) {
/* The number of samples we ask for is `number_of_lost_frames` times
* `prev_decoded_samples_`. Limit the number of samples to maximum
* `MaxFrameSizePerChannel()`. */
plc_samples = inst->prev_decoded_samples;
const int max_samples_per_channel =
MaxFrameSizePerChannel(inst->sample_rate_hz);
plc_samples = plc_samples <= max_samples_per_channel
? plc_samples
: max_samples_per_channel;
}
decoded_samples =
DecodeNative(inst, NULL, 0, plc_samples, decoded, &audio_type, 0);
if (decoded_samples < 0) {
@ -581,11 +549,6 @@ int WebRtcOpus_Decode(OpusDecInst* inst,
return -1;
}
if (inst->plc_use_prev_decoded_samples) {
/* Update decoded sample memory, to be used by the PLC in case of losses. */
inst->prev_decoded_samples = decoded_samples;
}
return decoded_samples;
}
@ -639,16 +602,6 @@ int WebRtcOpus_DurationEst(OpusDecInst* inst,
}
int WebRtcOpus_PlcDuration(OpusDecInst* inst) {
if (inst->plc_use_prev_decoded_samples) {
/* The number of samples we ask for is `number_of_lost_frames` times
* `prev_decoded_samples_`. Limit the number of samples to maximum
* `MaxFrameSizePerChannel()`. */
const int plc_samples = inst->prev_decoded_samples;
const int max_samples_per_channel =
MaxFrameSizePerChannel(inst->sample_rate_hz);
return plc_samples <= max_samples_per_channel ? plc_samples
: max_samples_per_channel;
}
return FrameSizePerChannel(kWebRtcOpusPlcFrameSizeMs, inst->sample_rate_hz);
}