mollyim/webrtc
1
0
Fork 0
mirror of https://github.com/mollyim/webrtc.git synced 2025-05-12 21:30:45 +01:00
Code Issues Projects Releases Packages Wiki Activity Actions

Make muted param in GetAudio optional.

Browse source 
It is not necessary for the caller to use it and the mute info can be
found on AudioFrame.muted().

Bug: None
Change-Id: I458f1f2e8489c1d8f8a9078b21f889b2540bdab9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349940
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42236}
This commit is contained in:
Jakob Ivarsson 2024-05-06 16:46:48 +02:00 committed by WebRTC LUCI CQ
parent 86cd7a35dc
commit 1e5f88c5be
6 changed files with 14 additions and 25 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

2
api/neteq/neteq.h
View file

@ -210,7 +210,7 @@ class NetEq {
// Returns kOK on success, or kFail in case of an error.
virtual int GetAudio(
AudioFrame* audio_frame,
bool* muted,
bool* muted = nullptr,
int* current_sample_rate_hz = nullptr,
absl::optional<Operation> action_override = absl::nullopt) = 0;

17
audio/channel_receive.cc
View file

@ -391,9 +391,7 @@ AudioMixer::Source::AudioFrameInfo ChannelReceive::GetAudioFrameWithInfo(
event_log_->Log(std::make_unique<RtcEventAudioPlayout>(remote_ssrc_));
// Get 10ms raw PCM data from the ACM (mixer limits output frequency)
bool muted;
if (acm_receiver_.GetAudio(audio_frame->sample_rate_hz_, audio_frame,
&muted) == -1) {
if (acm_receiver_.GetAudio(audio_frame->sample_rate_hz_, audio_frame) == -1) {
RTC_DLOG(LS_ERROR)
<< "ChannelReceive::GetAudioFrame() PlayoutData10Ms() failed!";
// In all likelihood, the audio in this frame is garbage. We return an
@ -406,13 +404,6 @@ AudioMixer::Source::AudioFrameInfo ChannelReceive::GetAudioFrameWithInfo(
return AudioMixer::Source::AudioFrameInfo::kError;
}
if (muted) {
// TODO(henrik.lundin): We should be able to do better than this. But we
// will have to go through all the cases below where the audio samples may
// be used, and handle the muted case in some way.
AudioFrameOperations::Mute(audio_frame);
}
{
// Pass the audio buffers to an optional sink callback, before applying
// scaling/panning, as that applies to the mix operation.
@ -509,9 +500,9 @@ AudioMixer::Source::AudioFrameInfo ChannelReceive::GetAudioFrameWithInfo(
}
TRACE_EVENT_END2("webrtc", "ChannelReceive::GetAudioFrameWithInfo", "gain",
output_gain, "muted", muted);
return muted ? AudioMixer::Source::AudioFrameInfo::kMuted
: AudioMixer::Source::AudioFrameInfo::kNormal;
output_gain, "muted", audio_frame->muted());
return audio_frame->muted() ? AudioMixer::Source::AudioFrameInfo::kMuted
: AudioMixer::Source::AudioFrameInfo::kNormal;
}
int ChannelReceive::PreferredSampleRate() const {

4
modules/audio_coding/acm2/acm_receiver.cc
View file

@ -151,8 +151,6 @@ int AcmReceiver::InsertPacket(const RTPHeader& rtp_header,
int AcmReceiver::GetAudio(int desired_freq_hz,
AudioFrame* audio_frame,
bool* muted) {
RTC_DCHECK(muted);
int current_sample_rate_hz = 0;
if (neteq_->GetAudio(audio_frame, muted, &current_sample_rate_hz) !=
NetEq::kOK) {
@ -212,7 +210,7 @@ int AcmReceiver::GetAudio(int desired_freq_hz,
sizeof(int16_t) * audio_frame->samples_per_channel_ *
audio_frame->num_channels_);
call_stats_.DecodedByNetEq(audio_frame->speech_type_, *muted);
call_stats_.DecodedByNetEq(audio_frame->speech_type_, audio_frame->muted());
return 0;
}

4
modules/audio_coding/acm2/acm_receiver.h
View file

@ -94,7 +94,9 @@ class AcmReceiver {
// Return value : 0 if OK.
// -1 if NetEq returned an error.
//
int GetAudio(int desired_freq_hz, AudioFrame* audio_frame, bool* muted);
int GetAudio(int desired_freq_hz,
AudioFrame* audio_frame,
bool* muted = nullptr);
// Replace the current set of decoders with the specified set.
void SetCodecs(const std::map<int, SdpAudioFormat>& codecs);

9
modules/audio_coding/neteq/neteq_impl.cc
View file

@ -218,13 +218,15 @@ int NetEqImpl::GetAudio(AudioFrame* audio_frame,
absl::optional<Operation> action_override) {
TRACE_EVENT0("webrtc", "NetEqImpl::GetAudio");
MutexLock lock(&mutex_);
if (GetAudioInternal(audio_frame, muted, action_override) != 0) {
if (GetAudioInternal(audio_frame, action_override) != 0) {
return kFail;
}
RTC_DCHECK_EQ(
audio_frame->sample_rate_hz_,
rtc::dchecked_cast<int>(audio_frame->samples_per_channel_ * 100));
RTC_DCHECK_EQ(*muted, audio_frame->muted());
if (muted != nullptr) {
*muted = audio_frame->muted();
}
audio_frame->speech_type_ = ToSpeechType(LastOutputType());
last_output_sample_rate_hz_ = audio_frame->sample_rate_hz_;
RTC_DCHECK(last_output_sample_rate_hz_ == 8000 ||
@ -747,13 +749,11 @@ bool NetEqImpl::MaybeChangePayloadType(uint8_t payload_type) {
}
int NetEqImpl::GetAudioInternal(AudioFrame* audio_frame,
bool* muted,
absl::optional<Operation> action_override) {
PacketList packet_list;
DtmfEvent dtmf_event;
Operation operation;
bool play_dtmf;
*muted = false;
last_decoded_packet_infos_.clear();
tick_timer_->Increment();
stats_->IncreaseCounter(output_size_samples_, fs_hz_);
@ -786,7 +786,6 @@ int NetEqImpl::GetAudioInternal(AudioFrame* audio_frame,
audio_frame->num_channels_ = sync_buffer_->Channels();
stats_->ExpandedNoiseSamples(output_size_samples_, false);
controller_->NotifyMutedState();
*muted = true;
return 0;
}
int return_value = GetDecision(&operation, &packet_list, &dtmf_event,

3
modules/audio_coding/neteq/neteq_impl.h
View file

@ -133,7 +133,7 @@ class NetEqImpl : public webrtc::NetEq {
int GetAudio(
AudioFrame* audio_frame,
bool* muted,
bool* muted = nullptr,
int* current_sample_rate_hz = nullptr,
absl::optional<Operation> action_override = absl::nullopt) override;
@ -216,7 +216,6 @@ class NetEqImpl : public webrtc::NetEq {
// Delivers 10 ms of audio data. The data is written to `audio_frame`.
// Returns 0 on success, otherwise an error code.
int GetAudioInternal(AudioFrame* audio_frame,
bool* muted,
absl::optional<Operation> action_override)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);