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Try to by pass audio encoder if pre-encoded set #120

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Try to by pass audio encoder if pre-encoded set #120

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07a53ff
Try to by pass audio encoder if pre-encoded set
kuba-- May 24, 2024
96cf86f
refine endiness
kuba-- Jun 1, 2024
d402272
Update audio_transport_impl.cc
kuba-- Jun 2, 2024
f9ae56d
Update audio_encoder_opus.cc
kuba-- Jun 5, 2024
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2 changes: 2 additions & 0 deletions api/audio_codecs/L16/audio_encoder_L16.cc
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Original file line number Diff line number Diff line change
Expand Up @@ -29,6 +29,7 @@ absl::optional<AudioEncoderL16::Config> AudioEncoderL16::SdpToConfig(
}
Config config;
config.sample_rate_hz = format.clockrate_hz;
config.pre_encoded = format.pre_encoded;
config.num_channels = rtc::dchecked_cast<int>(format.num_channels);
auto ptime_iter = format.parameters.find("ptime");
if (ptime_iter != format.parameters.end()) {
Expand Down Expand Up @@ -66,6 +67,7 @@ std::unique_ptr<AudioEncoder> AudioEncoderL16::MakeAudioEncoder(
c.num_channels = config.num_channels;
c.frame_size_ms = config.frame_size_ms;
c.payload_type = payload_type;
c.pre_encoded = config.pre_encoded;
if (!config.IsOk()) {
RTC_DCHECK_NOTREACHED();
return nullptr;
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1 change: 1 addition & 0 deletions api/audio_codecs/L16/audio_encoder_L16.h
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Expand Up @@ -38,6 +38,7 @@ struct RTC_EXPORT AudioEncoderL16 {
int sample_rate_hz = 8000;
int num_channels = 1;
int frame_size_ms = 10;
bool pre_encoded = false;
};
static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
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14 changes: 14 additions & 0 deletions api/audio_codecs/audio_encoder.cc
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Expand Up @@ -110,5 +110,19 @@ ANAStats AudioEncoder::GetANAStats() const {
return ANAStats();
}

size_t AudioEncoder::AppendPreEncodeData(rtc::ArrayView<const int16_t> audio,
rtc::Buffer* encoded) {
const size_t old_size = encoded->size();
for (const int16_t it : audio) {
uint8_t arr[2] = {
static_cast<uint8_t>((it >> 8) & 0x00ff),
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Is endianess an issue here?

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I assumed we will have decoder (and this is how I tested), so the most important was to be consistent with decoder.
Otherwise, this is good question. I think I'll convert it to Little Endian (as more popular architecture). Another alternative is to rely on real architecture (by using C like union implementation) or expose this information as part of API, but It may look too complicated, because who knows what output SDP device will decode it.

static_cast<uint8_t>(it & 0x00ff),
};

encoded->AppendData(arr, 2);
}
return (encoded->size() - old_size);
}

constexpr int AudioEncoder::kMaxNumberOfChannels;
} // namespace webrtc
4 changes: 4 additions & 0 deletions api/audio_codecs/audio_encoder.h
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Expand Up @@ -255,6 +255,10 @@ class AudioEncoder {
virtual EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
rtc::ArrayView<const int16_t> audio,
rtc::Buffer* encoded) = 0;

// The AppendPreEncodeData function adds raw audio data to the end of the encoded buffer.
virtual size_t AppendPreEncodeData(rtc::ArrayView<const int16_t> audio,
rtc::Buffer* encoded);
};
} // namespace webrtc
#endif // API_AUDIO_CODECS_AUDIO_ENCODER_H_
11 changes: 8 additions & 3 deletions api/audio_codecs/audio_format.cc
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Expand Up @@ -22,7 +22,10 @@ SdpAudioFormat::SdpAudioFormat(SdpAudioFormat&&) = default;
SdpAudioFormat::SdpAudioFormat(absl::string_view name,
int clockrate_hz,
size_t num_channels)
: name(name), clockrate_hz(clockrate_hz), num_channels(num_channels) {}
: name(name),
clockrate_hz(clockrate_hz),
num_channels(num_channels),
pre_encoded(false) {}

SdpAudioFormat::SdpAudioFormat(absl::string_view name,
int clockrate_hz,
Expand All @@ -31,7 +34,8 @@ SdpAudioFormat::SdpAudioFormat(absl::string_view name,
: name(name),
clockrate_hz(clockrate_hz),
num_channels(num_channels),
parameters(param) {}
parameters(param),
pre_encoded(false) {}

SdpAudioFormat::SdpAudioFormat(absl::string_view name,
int clockrate_hz,
Expand All @@ -40,7 +44,8 @@ SdpAudioFormat::SdpAudioFormat(absl::string_view name,
: name(name),
clockrate_hz(clockrate_hz),
num_channels(num_channels),
parameters(std::move(param)) {}
parameters(std::move(param)),
pre_encoded(false) {}

bool SdpAudioFormat::Matches(const SdpAudioFormat& o) const {
return absl::EqualsIgnoreCase(name, o.name) &&
Expand Down
1 change: 1 addition & 0 deletions api/audio_codecs/audio_format.h
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Expand Up @@ -56,6 +56,7 @@ struct RTC_EXPORT SdpAudioFormat {
int clockrate_hz;
size_t num_channels;
Parameters parameters;
bool pre_encoded;
};

// Information about how an audio format is treated by the codec implementation.
Expand Down
3 changes: 3 additions & 0 deletions api/audio_codecs/g711/audio_encoder_g711.cc
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Expand Up @@ -31,6 +31,7 @@ absl::optional<AudioEncoderG711::Config> AudioEncoderG711::SdpToConfig(
config.type = is_pcmu ? Config::Type::kPcmU : Config::Type::kPcmA;
config.num_channels = rtc::dchecked_cast<int>(format.num_channels);
config.frame_size_ms = 20;
config.pre_encoded = format.pre_encoded;
auto ptime_iter = format.parameters.find("ptime");
if (ptime_iter != format.parameters.end()) {
const auto ptime = rtc::StringToNumber<int>(ptime_iter->second);
Expand Down Expand Up @@ -75,13 +76,15 @@ std::unique_ptr<AudioEncoder> AudioEncoderG711::MakeAudioEncoder(
AudioEncoderPcmU::Config impl_config;
impl_config.num_channels = config.num_channels;
impl_config.frame_size_ms = config.frame_size_ms;
impl_config.pre_encoded = config.pre_encoded;
impl_config.payload_type = payload_type;
return std::make_unique<AudioEncoderPcmU>(impl_config);
}
case Config::Type::kPcmA: {
AudioEncoderPcmA::Config impl_config;
impl_config.num_channels = config.num_channels;
impl_config.frame_size_ms = config.frame_size_ms;
impl_config.pre_encoded = config.pre_encoded;
impl_config.payload_type = payload_type;
return std::make_unique<AudioEncoderPcmA>(impl_config);
}
Expand Down
1 change: 1 addition & 0 deletions api/audio_codecs/g711/audio_encoder_g711.h
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Expand Up @@ -37,6 +37,7 @@ struct RTC_EXPORT AudioEncoderG711 {
Type type = Type::kPcmU;
int num_channels = 1;
int frame_size_ms = 20;
bool pre_encoded = false;
};
static absl::optional<AudioEncoderG711::Config> SdpToConfig(
const SdpAudioFormat& audio_format);
Expand Down
1 change: 1 addition & 0 deletions api/audio_codecs/g722/audio_encoder_g722.cc
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Expand Up @@ -30,6 +30,7 @@ absl::optional<AudioEncoderG722Config> AudioEncoderG722::SdpToConfig(

AudioEncoderG722Config config;
config.num_channels = rtc::checked_cast<int>(format.num_channels);
config.pre_encoded = format.pre_encoded;
auto ptime_iter = format.parameters.find("ptime");
if (ptime_iter != format.parameters.end()) {
auto ptime = rtc::StringToNumber<int>(ptime_iter->second);
Expand Down
1 change: 1 addition & 0 deletions api/audio_codecs/g722/audio_encoder_g722_config.h
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Expand Up @@ -22,6 +22,7 @@ struct AudioEncoderG722Config {
}
int frame_size_ms = 20;
int num_channels = 1;
bool pre_encoded = false;
};

} // namespace webrtc
Expand Down
1 change: 1 addition & 0 deletions api/audio_codecs/ilbc/audio_encoder_ilbc.cc
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Expand Up @@ -45,6 +45,7 @@ absl::optional<AudioEncoderIlbcConfig> AudioEncoderIlbc::SdpToConfig(
}

AudioEncoderIlbcConfig config;
config.pre_encoded = format.pre_encoded;
auto ptime_iter = format.parameters.find("ptime");
if (ptime_iter != format.parameters.end()) {
auto ptime = rtc::StringToNumber<int>(ptime_iter->second);
Expand Down
1 change: 1 addition & 0 deletions api/audio_codecs/ilbc/audio_encoder_ilbc_config.h
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Expand Up @@ -21,6 +21,7 @@ struct AudioEncoderIlbcConfig {
int frame_size_ms = 30; // Valid values are 20, 30, 40, and 60 ms.
// Note that frame size 40 ms produces encodings with two 20 ms frames in
// them, and frame size 60 ms consists of two 30 ms frames.
bool pre_encoded = false;
};

} // namespace webrtc
Expand Down
3 changes: 2 additions & 1 deletion api/audio_codecs/opus/audio_encoder_opus_config.cc
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Expand Up @@ -44,7 +44,8 @@ AudioEncoderOpusConfig::AudioEncoderOpusConfig()
complexity_threshold_window_bps(1500),
dtx_enabled(false),
uplink_bandwidth_update_interval_ms(200),
payload_type(-1) {}
payload_type(-1),
pre_encoded(false) {}
AudioEncoderOpusConfig::AudioEncoderOpusConfig(const AudioEncoderOpusConfig&) =
default;
AudioEncoderOpusConfig::~AudioEncoderOpusConfig() = default;
Expand Down
2 changes: 2 additions & 0 deletions api/audio_codecs/opus/audio_encoder_opus_config.h
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Expand Up @@ -67,6 +67,8 @@ struct RTC_EXPORT AudioEncoderOpusConfig {
// NOTE: This member isn't necessary, and will soon go away. See
// https://bugs.chromium.org/p/webrtc/issues/detail?id=7847
int payload_type;

bool pre_encoded;
};

} // namespace webrtc
Expand Down
5 changes: 4 additions & 1 deletion api/audio_options.cc
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Expand Up @@ -56,6 +56,7 @@ void AudioOptions::SetAll(const AudioOptions& change) {
SetFrom(&audio_network_adaptor, change.audio_network_adaptor);
SetFrom(&audio_network_adaptor_config, change.audio_network_adaptor_config);
SetFrom(&init_recording_on_send, change.init_recording_on_send);
SetFrom(&pre_encoded, change.pre_encoded);
}

bool AudioOptions::operator==(const AudioOptions& o) const {
Expand All @@ -75,7 +76,8 @@ bool AudioOptions::operator==(const AudioOptions& o) const {
combined_audio_video_bwe == o.combined_audio_video_bwe &&
audio_network_adaptor == o.audio_network_adaptor &&
audio_network_adaptor_config == o.audio_network_adaptor_config &&
init_recording_on_send == o.init_recording_on_send;
init_recording_on_send == o.init_recording_on_send &&
pre_encoded == o.pre_encoded;
}

std::string AudioOptions::ToString() const {
Expand All @@ -100,6 +102,7 @@ std::string AudioOptions::ToString() const {
ToStringIfSet(&result, "combined_audio_video_bwe", combined_audio_video_bwe);
ToStringIfSet(&result, "audio_network_adaptor", audio_network_adaptor);
ToStringIfSet(&result, "init_recording_on_send", init_recording_on_send);
ToStringIfSet(&result, "pre_encoded", pre_encoded);
result << "}";
return result.str();
}
Expand Down
2 changes: 2 additions & 0 deletions api/audio_options.h
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Expand Up @@ -73,6 +73,8 @@ struct RTC_EXPORT AudioOptions {
// true.
// TODO(webrtc:13566): Remove this option. See issue for details.
absl::optional<bool> init_recording_on_send;
// Audio is already pre-encoded, so we can pass encoding.
absl::optional<bool> pre_encoded;
};

} // namespace cricket
Expand Down
13 changes: 0 additions & 13 deletions audio/audio_transport_impl.cc
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Expand Up @@ -210,17 +210,6 @@ int32_t AudioTransportImpl::NeedMorePlayData(const size_t nSamples,
int64_t* elapsed_time_ms,
int64_t* ntp_time_ms) {
TRACE_EVENT0("webrtc", "AudioTransportImpl::SendProcessedData");
RTC_DCHECK_EQ(sizeof(int16_t) * nChannels, nBytesPerSample);
RTC_DCHECK_GE(nChannels, 1);
RTC_DCHECK_LE(nChannels, 2);
RTC_DCHECK_GE(
samplesPerSec,
static_cast<uint32_t>(AudioProcessing::NativeRate::kSampleRate8kHz));

// 100 = 1 second / data duration (10 ms).
RTC_DCHECK_EQ(nSamples * 100, samplesPerSec);
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Do these checks prevent using pre-encoded data? Since it's on the receive end, seems like this should be unaffected? In particular, dropping the 10ms rule seems like a pretty big deal since it's assumed fairly throughout the system that play data would be given in 10ms chunks.

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I'll double check the samples one (it should be fine). For sure if you link rust-sdk examples with debug version of webrtc it will crash in this place, so I commented this block for testing purpose.

#
# Fatal error in: ../../../../_source/macos_arm64/webrtc/src/audio/audio_transport_impl.cc, line 213
# last system error: 0
# Check failed: sizeof(int16_t) * nChannels == nBytesPerSample (4 vs. 2)

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I also worried about OggReader where we have:

const OGG_PAGE_DURATION: Duration = Duration::from_millis(20);

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I think we can change that to be 10. since webrtc's internal clock for audio is 10ms

RTC_DCHECK_LE(nBytesPerSample * nSamples * nChannels,
AudioFrame::kMaxDataSizeBytes);

mixer_->Mix(nChannels, &mixed_frame_);
*elapsed_time_ms = mixed_frame_.elapsed_time_ms_;
Expand All @@ -229,12 +218,10 @@ int32_t AudioTransportImpl::NeedMorePlayData(const size_t nSamples,
if (audio_processing_) {
const auto error =
ProcessReverseAudioFrame(audio_processing_, &mixed_frame_);
RTC_DCHECK_EQ(error, AudioProcessing::kNoError);
}

nSamplesOut = Resample(mixed_frame_, samplesPerSec, &render_resampler_,
static_cast<int16_t*>(audioSamples));
RTC_DCHECK_EQ(nSamplesOut, nChannels * nSamples);
return 0;
}

Expand Down
2 changes: 2 additions & 0 deletions media/engine/webrtc_voice_engine.cc
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Expand Up @@ -394,6 +394,7 @@ void WebRtcVoiceEngine::Init() {
options.audio_jitter_buffer_max_packets = 200;
options.audio_jitter_buffer_fast_accelerate = false;
options.audio_jitter_buffer_min_delay_ms = 0;
options.pre_encoded = false;
ApplyOptions(options);
}
initialized_ = true;
Expand Down Expand Up @@ -1650,6 +1651,7 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
IsCodec(voice_codec, kRedCodecName))) {
webrtc::SdpAudioFormat format(voice_codec.name, voice_codec.clockrate,
voice_codec.channels, voice_codec.params);
format.pre_encoded = options_.pre_encoded.value_or(false);

voice_codec_info = engine()->encoder_factory_->QueryAudioEncoder(format);
if (!voice_codec_info) {
Expand Down
20 changes: 12 additions & 8 deletions modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
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Expand Up @@ -29,7 +29,8 @@ AudioEncoderPcm::AudioEncoderPcm(const Config& config, int sample_rate_hz)
static_cast<size_t>(config.frame_size_ms / 10)),
full_frame_samples_(config.num_channels * config.frame_size_ms *
sample_rate_hz / 1000),
first_timestamp_in_buffer_(0) {
first_timestamp_in_buffer_(0),
pre_encoded_(config.pre_encoded) {
RTC_CHECK_GT(sample_rate_hz, 0) << "Sample rate must be larger than 0 Hz";
RTC_CHECK_EQ(config.frame_size_ms % 10, 0)
<< "Frame size must be an integer multiple of 10 ms.";
Expand Down Expand Up @@ -74,13 +75,16 @@ AudioEncoder::EncodedInfo AudioEncoderPcm::EncodeImpl(
EncodedInfo info;
info.encoded_timestamp = first_timestamp_in_buffer_;
info.payload_type = payload_type_;
info.encoded_bytes = encoded->AppendData(
full_frame_samples_ * BytesPerSample(),
[&](rtc::ArrayView<uint8_t> encoded) {
return EncodeCall(&speech_buffer_[0], full_frame_samples_,
encoded.data());
});
speech_buffer_.clear();
if (pre_encoded_) {
info.encoded_bytes = AppendPreEncodeData(audio, encoded);
} else {
info.encoded_bytes = encoded->AppendData(
full_frame_samples_ * BytesPerSample(),
[&](rtc::ArrayView<uint8_t> encoded) {
return EncodeCall(&speech_buffer_[0], full_frame_samples_,
encoded.data());
});
} speech_buffer_.clear();
info.encoder_type = GetCodecType();
return info;
}
Expand Down
4 changes: 3 additions & 1 deletion modules/audio_coding/codecs/g711/audio_encoder_pcm.h
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Expand Up @@ -29,10 +29,11 @@ class AudioEncoderPcm : public AudioEncoder {
int frame_size_ms;
size_t num_channels;
int payload_type;
bool pre_encoded;

protected:
explicit Config(int pt)
: frame_size_ms(20), num_channels(1), payload_type(pt) {}
: frame_size_ms(20), num_channels(1), payload_type(pt), pre_encoded(false) {}
};

~AudioEncoderPcm() override;
Expand Down Expand Up @@ -67,6 +68,7 @@ class AudioEncoderPcm : public AudioEncoder {
const int sample_rate_hz_;
const size_t num_channels_;
const int payload_type_;
bool pre_encoded_;
const size_t num_10ms_frames_per_packet_;
const size_t full_frame_samples_;
std::vector<int16_t> speech_buffer_;
Expand Down
69 changes: 38 additions & 31 deletions modules/audio_coding/codecs/g722/audio_encoder_g722.cc
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Expand Up @@ -33,7 +33,8 @@ AudioEncoderG722Impl::AudioEncoderG722Impl(const AudioEncoderG722Config& config,
num_10ms_frames_buffered_(0),
first_timestamp_in_buffer_(0),
encoders_(new EncoderState[num_channels_]),
interleave_buffer_(2 * num_channels_) {
interleave_buffer_(2 * num_channels_),
pre_encoded_(config.pre_encoded) {
RTC_CHECK(config.IsOk());
const size_t samples_per_channel =
kSampleRateHz / 100 * num_10ms_frames_per_packet_;
Expand Down Expand Up @@ -103,38 +104,44 @@ AudioEncoder::EncodedInfo AudioEncoderG722Impl::EncodeImpl(
return EncodedInfo();
}

// Encode each channel separately.
RTC_CHECK_EQ(num_10ms_frames_buffered_, num_10ms_frames_per_packet_);
num_10ms_frames_buffered_ = 0;
const size_t samples_per_channel = SamplesPerChannel();
for (size_t i = 0; i < num_channels_; ++i) {
const size_t bytes_encoded = WebRtcG722_Encode(
encoders_[i].encoder, encoders_[i].speech_buffer.get(),
samples_per_channel, encoders_[i].encoded_buffer.data());
RTC_CHECK_EQ(bytes_encoded, samples_per_channel / 2);
}

const size_t bytes_to_encode = samples_per_channel / 2 * num_channels_;
EncodedInfo info;
info.encoded_bytes = encoded->AppendData(
bytes_to_encode, [&](rtc::ArrayView<uint8_t> encoded) {
// Interleave the encoded bytes of the different channels. Each separate
// channel and the interleaved stream encodes two samples per byte, most
// significant half first.
for (size_t i = 0; i < samples_per_channel / 2; ++i) {
for (size_t j = 0; j < num_channels_; ++j) {
uint8_t two_samples = encoders_[j].encoded_buffer.data()[i];
interleave_buffer_.data()[j] = two_samples >> 4;
interleave_buffer_.data()[num_channels_ + j] = two_samples & 0xf;
if (pre_encoded_) {
info.encoded_bytes = AppendPreEncodeData(audio, encoded);
} else {
// Encode each channel separately.
RTC_CHECK_EQ(num_10ms_frames_buffered_, num_10ms_frames_per_packet_);
num_10ms_frames_buffered_ = 0;
const size_t samples_per_channel = SamplesPerChannel();
for (size_t i = 0; i < num_channels_; ++i) {
const size_t bytes_encoded = WebRtcG722_Encode(
encoders_[i].encoder, encoders_[i].speech_buffer.get(),
samples_per_channel, encoders_[i].encoded_buffer.data());
RTC_CHECK_EQ(bytes_encoded, samples_per_channel / 2);
}

const size_t bytes_to_encode = samples_per_channel / 2 * num_channels_;

info.encoded_bytes = encoded->AppendData(
bytes_to_encode, [&](rtc::ArrayView<uint8_t> encoded) {
// Interleave the encoded bytes of the different channels. Each separate
// channel and the interleaved stream encodes two samples per byte, most
// significant half first.
for (size_t i = 0; i < samples_per_channel / 2; ++i) {
for (size_t j = 0; j < num_channels_; ++j) {
uint8_t two_samples = encoders_[j].encoded_buffer.data()[i];
interleave_buffer_.data()[j] = two_samples >> 4;
interleave_buffer_.data()[num_channels_ + j] = two_samples & 0xf;
}
for (size_t j = 0; j < num_channels_; ++j)
encoded[i * num_channels_ + j] =
interleave_buffer_.data()[2 * j] << 4 |
interleave_buffer_.data()[2 * j + 1];
}
for (size_t j = 0; j < num_channels_; ++j)
encoded[i * num_channels_ + j] =
interleave_buffer_.data()[2 * j] << 4 |
interleave_buffer_.data()[2 * j + 1];
}

return bytes_to_encode;
});

return bytes_to_encode;
});
}

info.encoded_timestamp = first_timestamp_in_buffer_;
info.payload_type = payload_type_;
info.encoder_type = CodecType::kG722;
Expand Down
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