Bump to WebRTC M120 release

Some API deprecation -- ExperimentalAgc and ExperimentalNs are gone.
We're continuing to carry iSAC even though it's gone upstream, but maybe
we'll want to drop that soon.
This commit is contained in:
Arun Raghavan
2023-12-12 10:42:58 -05:00
parent 9a202fb8c2
commit c6abf6cd3f
479 changed files with 20900 additions and 11996 deletions

View File

@ -26,10 +26,11 @@ rtc_library("audio_frame_operations") {
"../../api/audio:audio_frame_api",
"../../common_audio",
"../../rtc_base:checks",
"../../rtc_base:deprecation",
"../../rtc_base:rtc_base_approved",
"../../rtc_base:logging",
"../../rtc_base:safe_conversions",
"../../system_wrappers:field_trial",
]
absl_deps = [ "//third_party/abseil-cpp/absl/base:core_headers" ]
}
if (rtc_include_tests) {
@ -44,7 +45,9 @@ if (rtc_include_tests) {
":audio_frame_operations",
"../../api/audio:audio_frame_api",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
"../../rtc_base:logging",
"../../rtc_base:macromagic",
"../../rtc_base:stringutils",
"../../test:field_trial",
"../../test:test_support",
"//testing/gtest",

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@ -131,8 +131,8 @@ void AudioFrameOperations::DownmixChannels(const int16_t* src_audio,
return;
}
RTC_NOTREACHED() << "src_channels: " << src_channels
<< ", dst_channels: " << dst_channels;
RTC_DCHECK_NOTREACHED() << "src_channels: " << src_channels
<< ", dst_channels: " << dst_channels;
}
void AudioFrameOperations::DownmixChannels(size_t dst_channels,
@ -149,8 +149,8 @@ void AudioFrameOperations::DownmixChannels(size_t dst_channels,
int err = QuadToStereo(frame);
RTC_DCHECK_EQ(err, 0);
} else {
RTC_NOTREACHED() << "src_channels: " << frame->num_channels_
<< ", dst_channels: " << dst_channels;
RTC_DCHECK_NOTREACHED() << "src_channels: " << frame->num_channels_
<< ", dst_channels: " << dst_channels;
}
}
@ -169,10 +169,10 @@ void AudioFrameOperations::UpmixChannels(size_t target_number_of_channels,
if (!frame->muted()) {
// Up-mixing done in place. Going backwards through the frame ensure nothing
// is irrevocably overwritten.
int16_t* frame_data = frame->mutable_data();
for (int i = frame->samples_per_channel_ - 1; i >= 0; i--) {
for (size_t j = 0; j < target_number_of_channels; ++j) {
frame->mutable_data()[target_number_of_channels * i + j] =
frame->data()[i];
frame_data[target_number_of_channels * i + j] = frame_data[i];
}
}
}
@ -222,14 +222,14 @@ void AudioFrameOperations::Mute(AudioFrame* frame,
size_t end = count;
float start_g = 0.0f;
if (current_frame_muted) {
// Fade out the last |count| samples of frame.
// Fade out the last `count` samples of frame.
RTC_DCHECK(!previous_frame_muted);
start = frame->samples_per_channel_ - count;
end = frame->samples_per_channel_;
start_g = 1.0f;
inc = -inc;
} else {
// Fade in the first |count| samples of frame.
// Fade in the first `count` samples of frame.
RTC_DCHECK(previous_frame_muted);
}

View File

@ -14,8 +14,8 @@
#include <stddef.h>
#include <stdint.h>
#include "absl/base/attributes.h"
#include "api/audio/audio_frame.h"
#include "rtc_base/deprecation.h"
namespace webrtc {
@ -24,38 +24,40 @@ namespace webrtc {
// than a class.
class AudioFrameOperations {
public:
// Add samples in |frame_to_add| with samples in |result_frame|
// putting the results in |results_frame|. The fields
// |vad_activity_| and |speech_type_| of the result frame are
// updated. If |result_frame| is empty (|samples_per_channel_|==0),
// the samples in |frame_to_add| are added to it. The number of
// Add samples in `frame_to_add` with samples in `result_frame`
// putting the results in `results_frame`. The fields
// `vad_activity_` and `speech_type_` of the result frame are
// updated. If `result_frame` is empty (`samples_per_channel_`==0),
// the samples in `frame_to_add` are added to it. The number of
// channels and number of samples per channel must match except when
// |result_frame| is empty.
// `result_frame` is empty.
static void Add(const AudioFrame& frame_to_add, AudioFrame* result_frame);
// |frame.num_channels_| will be updated. This version checks for sufficient
// buffer size and that |num_channels_| is mono. Use UpmixChannels
// `frame.num_channels_` will be updated. This version checks for sufficient
// buffer size and that `num_channels_` is mono. Use UpmixChannels
// instead. TODO(bugs.webrtc.org/8649): remove.
RTC_DEPRECATED static int MonoToStereo(AudioFrame* frame);
ABSL_DEPRECATED("bugs.webrtc.org/8649")
static int MonoToStereo(AudioFrame* frame);
// |frame.num_channels_| will be updated. This version checks that
// |num_channels_| is stereo. Use DownmixChannels
// `frame.num_channels_` will be updated. This version checks that
// `num_channels_` is stereo. Use DownmixChannels
// instead. TODO(bugs.webrtc.org/8649): remove.
RTC_DEPRECATED static int StereoToMono(AudioFrame* frame);
ABSL_DEPRECATED("bugs.webrtc.org/8649")
static int StereoToMono(AudioFrame* frame);
// Downmixes 4 channels |src_audio| to stereo |dst_audio|. This is an in-place
// operation, meaning |src_audio| and |dst_audio| may point to the same
// Downmixes 4 channels `src_audio` to stereo `dst_audio`. This is an in-place
// operation, meaning `src_audio` and `dst_audio` may point to the same
// buffer.
static void QuadToStereo(const int16_t* src_audio,
size_t samples_per_channel,
int16_t* dst_audio);
// |frame.num_channels_| will be updated. This version checks that
// |num_channels_| is 4 channels.
// `frame.num_channels_` will be updated. This version checks that
// `num_channels_` is 4 channels.
static int QuadToStereo(AudioFrame* frame);
// Downmixes |src_channels| |src_audio| to |dst_channels| |dst_audio|.
// This is an in-place operation, meaning |src_audio| and |dst_audio|
// Downmixes `src_channels` `src_audio` to `dst_channels` `dst_audio`.
// This is an in-place operation, meaning `src_audio` and `dst_audio`
// may point to the same buffer. Supported channel combinations are
// Stereo to Mono, Quad to Mono, and Quad to Stereo.
static void DownmixChannels(const int16_t* src_audio,
@ -64,27 +66,27 @@ class AudioFrameOperations {
size_t dst_channels,
int16_t* dst_audio);
// |frame.num_channels_| will be updated. This version checks that
// |num_channels_| and |dst_channels| are valid and performs relevant downmix.
// `frame.num_channels_` will be updated. This version checks that
// `num_channels_` and `dst_channels` are valid and performs relevant downmix.
// Supported channel combinations are N channels to Mono, and Quad to Stereo.
static void DownmixChannels(size_t dst_channels, AudioFrame* frame);
// |frame.num_channels_| will be updated. This version checks that
// |num_channels_| and |dst_channels| are valid and performs relevant
// `frame.num_channels_` will be updated. This version checks that
// `num_channels_` and `dst_channels` are valid and performs relevant
// downmix. Supported channel combinations are Mono to N
// channels. The single channel is replicated.
static void UpmixChannels(size_t target_number_of_channels,
AudioFrame* frame);
// Swap the left and right channels of |frame|. Fails silently if |frame| is
// Swap the left and right channels of `frame`. Fails silently if `frame` is
// not stereo.
static void SwapStereoChannels(AudioFrame* frame);
// Conditionally zero out contents of |frame| for implementing audio mute:
// |previous_frame_muted| && |current_frame_muted| - Zero out whole frame.
// |previous_frame_muted| && !|current_frame_muted| - Fade-in at frame start.
// !|previous_frame_muted| && |current_frame_muted| - Fade-out at frame end.
// !|previous_frame_muted| && !|current_frame_muted| - Leave frame untouched.
// Conditionally zero out contents of `frame` for implementing audio mute:
// `previous_frame_muted` && `current_frame_muted` - Zero out whole frame.
// `previous_frame_muted` && !`current_frame_muted` - Fade-in at frame start.
// !`previous_frame_muted` && `current_frame_muted` - Fade-out at frame end.
// !`previous_frame_muted` && !`current_frame_muted` - Leave frame untouched.
static void Mute(AudioFrame* frame,
bool previous_frame_muted,
bool current_frame_muted);
@ -92,7 +94,7 @@ class AudioFrameOperations {
// Zero out contents of frame.
static void Mute(AudioFrame* frame);
// Halve samples in |frame|.
// Halve samples in `frame`.
static void ApplyHalfGain(AudioFrame* frame);
static int Scale(float left, float right, AudioFrame* frame);