zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Remove no-op and unused methods from AudioCodingModule

Browse Source 
This CL removes the following no-op and/or unused methods from
AudioCodingModule and AudioCodingModuleImpl:

ConfigISACBandwidthEstimator
DecoderEstimatedBandwidth
IsInternalDTXReplacedWithWebRtc
REDPayloadISAC
ReplaceInternalDTXWithWebRtc
ResetDecoder
ResetEncoder
SendBitrate
SetReceivedEstimatedBandwidth

R=henrik.lundin@webrtc.org

Review URL: https://codereview.webrtc.org/1308283003 .

Cr-Commit-Position: refs/heads/master@{#9773}
This commit is contained in:
Karl Wiberg
2015-08-25 09:37:04 +02:00
parent 7ef9d9104d
commit dd00f113a9
8 changed files with 9 additions and 334 deletions
Download Patch File Download Diff File Expand all files Collapse all files

138
webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
View File

@ -237,15 +237,6 @@ int32_t AudioCodingModuleImpl::Encode(const InputData& input_data) {
// Sender
//
// TODO(henrik.lundin): Remove this method; only used in tests.
int AudioCodingModuleImpl::ResetEncoder() {
CriticalSectionScoped lock(acm_crit_sect_);
if (!HaveValidEncoder("ResetEncoder")) {
return -1;
}
return 0;
}
// Can be called multiple times for Codec, CNG, RED.
int AudioCodingModuleImpl::RegisterSendCodec(const CodecInst& send_codec) {
CriticalSectionScoped lock(acm_crit_sect_);
@ -279,31 +270,6 @@ int AudioCodingModuleImpl::SendFrequency() const {
return codec_manager_.CurrentEncoder()->SampleRateHz();
}
// Get encode bitrate.
// Adaptive rate codecs return their current encode target rate, while other
// codecs return there longterm avarage or their fixed rate.
// TODO(henrik.lundin): Remove; not used.
int AudioCodingModuleImpl::SendBitrate() const {
FATAL() << "Deprecated";
// This return statement is required to workaround a bug in VS2013 Update 4
// when turning on the whole program optimizations. Without hit the linker
// will hang because it doesn't seem to find an exit path for this function.
// This is likely a bug in link.exe and would probably be fixed in VS2015.
return -1;
// CriticalSectionScoped lock(acm_crit_sect_);
//
// if (!codec_manager_.current_encoder()) {
// WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_,
// "SendBitrate Failed, no codec is registered");
// return -1;
// }
//
// WebRtcACMCodecParams encoder_param;
// codec_manager_.current_encoder()->EncoderParams(&encoder_param);
//
// return encoder_param.codec_inst.rate;
}
void AudioCodingModuleImpl::SetBitRate(int bitrate_bps) {
CriticalSectionScoped lock(acm_crit_sect_);
if (codec_manager_.CurrentEncoder()) {
@ -311,16 +277,6 @@ void AudioCodingModuleImpl::SetBitRate(int bitrate_bps) {
}
}
// Set available bandwidth, inform the encoder about the estimated bandwidth
// received from the remote party.
// TODO(henrik.lundin): Remove; not used.
int AudioCodingModuleImpl::SetReceivedEstimatedBandwidth(int bw) {
CriticalSectionScoped lock(acm_crit_sect_);
FATAL() << "Dead code?";
return -1;
// return codecs_[current_send_codec_idx_]->SetEstimatedBandwidth(bw);
}
// Register a transport callback which will be called to deliver
// the encoded buffers.
int AudioCodingModuleImpl::RegisterTransportCallback(
@ -608,15 +564,6 @@ int AudioCodingModuleImpl::InitializeReceiverSafe() {
return 0;
}
// TODO(turajs): If NetEq opens an API for reseting the state of decoders then
// implement this method. Otherwise it should be removed. I might be that by
// removing and registering a decoder we can achieve the effect of resetting.
// Reset the decoder state.
// TODO(henrik.lundin): Remove; only used in one test, and does nothing.
int AudioCodingModuleImpl::ResetDecoder() {
return 0;
}
// Get current receive frequency.
int AudioCodingModuleImpl::ReceiveFrequency() const {
WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_,
@ -725,22 +672,6 @@ int AudioCodingModuleImpl::SetMaximumPlayoutDelay(int time_ms) {
return receiver_.SetMaximumDelay(time_ms);
}
// Estimate the Bandwidth based on the incoming stream, needed for one way
// audio where the RTCP send the BW estimate.
// This is also done in the RTP module.
int AudioCodingModuleImpl::DecoderEstimatedBandwidth() const {
// We can estimate far-end to near-end bandwidth if the iSAC are sent. Check
// if the last received packets were iSAC packet then retrieve the bandwidth.
int last_audio_codec_id = receiver_.last_audio_codec_id();
if (last_audio_codec_id >= 0 &&
STR_CASE_CMP("ISAC", ACMCodecDB::database_[last_audio_codec_id].plname)) {
CriticalSectionScoped lock(acm_crit_sect_);
FATAL() << "Dead code?";
// return codecs_[last_audio_codec_id]->GetEstimatedBandwidth();
}
return -1;
}
// Set playout mode for: voice, fax, streaming or off.
int AudioCodingModuleImpl::SetPlayoutMode(AudioPlayoutMode mode) {
receiver_.SetPlayoutMode(mode);
@ -813,38 +744,6 @@ int AudioCodingModuleImpl::IncomingPayload(const uint8_t* incoming_payload,
return 0;
}
int AudioCodingModuleImpl::ReplaceInternalDTXWithWebRtc(bool use_webrtc_dtx) {
CriticalSectionScoped lock(acm_crit_sect_);
if (!HaveValidEncoder("ReplaceInternalDTXWithWebRtc")) {
WEBRTC_TRACE(
webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
"Cannot replace codec internal DTX when no send codec is registered.");
return -1;
}
FATAL() << "Dead code?";
// int res = codecs_[current_send_codec_idx_]->ReplaceInternalDTX(
// use_webrtc_dtx);
// Check if VAD is turned on, or if there is any error.
// if (res == 1) {
// vad_enabled_ = true;
// } else if (res < 0) {
// WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
// "Failed to set ReplaceInternalDTXWithWebRtc(%d)",
// use_webrtc_dtx);
// return res;
// }
return 0;
}
int AudioCodingModuleImpl::IsInternalDTXReplacedWithWebRtc(
bool* uses_webrtc_dtx) {
*uses_webrtc_dtx = true;
return 0;
}
// TODO(henrik.lundin): Remove? Only used in tests. Deprecated in VoiceEngine.
int AudioCodingModuleImpl::SetISACMaxRate(int max_bit_per_sec) {
CriticalSectionScoped lock(acm_crit_sect_);
@ -869,23 +768,6 @@ int AudioCodingModuleImpl::SetISACMaxPayloadSize(int max_size_bytes) {
return 0;
}
// TODO(henrik.lundin): Remove? Only used in tests.
int AudioCodingModuleImpl::ConfigISACBandwidthEstimator(
int frame_size_ms,
int rate_bit_per_sec,
bool enforce_frame_size) {
CriticalSectionScoped lock(acm_crit_sect_);
if (!HaveValidEncoder("ConfigISACBandwidthEstimator")) {
return -1;
}
FATAL() << "Dead code?";
return -1;
// return codecs_[current_send_codec_idx_]->ConfigISACBandwidthEstimator(
// frame_size_ms, rate_bit_per_sec, enforce_frame_size);
}
int AudioCodingModuleImpl::SetOpusApplication(OpusApplicationMode application) {
CriticalSectionScoped lock(acm_crit_sect_);
if (!HaveValidEncoder("SetOpusApplication")) {
@ -950,26 +832,6 @@ int AudioCodingModuleImpl::UnregisterReceiveCodec(uint8_t payload_type) {
return receiver_.RemoveCodec(payload_type);
}
// TODO(turajs): correct the type of |length_bytes| when it is corrected in
// GenericCodec.
int AudioCodingModuleImpl::REDPayloadISAC(int isac_rate,
int isac_bw_estimate,
uint8_t* payload,
int16_t* length_bytes) {
CriticalSectionScoped lock(acm_crit_sect_);
if (!HaveValidEncoder("EncodeData")) {
return -1;
}
FATAL() << "Dead code?";
return -1;
// int status;
// status = codecs_[current_send_codec_idx_]->REDPayloadISAC(isac_rate,
// isac_bw_estimate,
// payload,
// length_bytes);
// return status;
}
int AudioCodingModuleImpl::SetInitialPlayoutDelay(int delay_ms) {
{
CriticalSectionScoped lock(acm_crit_sect_);

38
webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
View File

@ -31,7 +31,7 @@ namespace acm2 {
class ACMDTMFDetection;
class AudioCodingModuleImpl : public AudioCodingModule {
class AudioCodingModuleImpl final : public AudioCodingModule {
public:
friend webrtc::AudioCodingImpl;
@ -42,9 +42,6 @@ class AudioCodingModuleImpl : public AudioCodingModule {
// Sender
//
// Reset send codec.
int ResetEncoder() override;
// Can be called multiple times for Codec, CNG, RED.
int RegisterSendCodec(const CodecInst& send_codec) override;
@ -57,20 +54,11 @@ class AudioCodingModuleImpl : public AudioCodingModule {
// Get current send frequency.
int SendFrequency() const override;
// Get encode bit-rate.
// Adaptive rate codecs return their current encode target rate, while other
// codecs return there long-term average or their fixed rate.
int SendBitrate() const override;
// Sets the bitrate to the specified value in bits/sec. In case the codec does
// not support the requested value it will choose an appropriate value
// instead.
void SetBitRate(int bitrate_bps) override;
// Set available bandwidth, inform the encoder about the
// estimated bandwidth received from the remote party.
int SetReceivedEstimatedBandwidth(int bw) override;
// Register a transport callback which will be
// called to deliver the encoded buffers.
int RegisterTransportCallback(AudioPacketizationCallback* transport) override;
@ -124,9 +112,6 @@ class AudioCodingModuleImpl : public AudioCodingModule {
// Initialize receiver, resets codec database etc.
int InitializeReceiver() override;
// Reset the decoder state.
int ResetDecoder() override;
// Get current receive frequency.
int ReceiveFrequency() const override;
@ -180,11 +165,6 @@ class AudioCodingModuleImpl : public AudioCodingModule {
// Get Dtmf playout status.
bool DtmfPlayoutStatus() const override;
// Estimate the Bandwidth based on the incoming stream, needed
// for one way audio where the RTCP send the BW estimate.
// This is also done in the RTP module .
int DecoderEstimatedBandwidth() const override;
// Set playout mode voice, fax.
int SetPlayoutMode(AudioPlayoutMode mode) override;
@ -204,26 +184,10 @@ class AudioCodingModuleImpl : public AudioCodingModule {
int GetNetworkStatistics(NetworkStatistics* statistics) override;
// GET RED payload for iSAC. The method id called when 'this' ACM is
// the default ACM.
// TODO(henrik.lundin) Not used. Remove?
int REDPayloadISAC(int isac_rate,
int isac_bw_estimate,
uint8_t* payload,
int16_t* length_bytes);
int ReplaceInternalDTXWithWebRtc(bool use_webrtc_dtx) override;
int IsInternalDTXReplacedWithWebRtc(bool* uses_webrtc_dtx) override;
int SetISACMaxRate(int max_bit_per_sec) override;
int SetISACMaxPayloadSize(int max_size_bytes) override;
int ConfigISACBandwidthEstimator(int frame_size_ms,
int rate_bit_per_sec,
bool enforce_frame_size = false) override;
int SetOpusApplication(OpusApplicationMode application) override;
// If current send codec is Opus, informs it about the maximum playback rate

125
webrtc/modules/audio_coding/main/interface/audio_coding_module.h
View File

@ -193,17 +193,6 @@ class AudioCodingModule {
// Sender
//
///////////////////////////////////////////////////////////////////////////
// int32_t ResetEncoder()
// This API resets the states of encoder. All the encoder settings, such as
// send-codec or VAD/DTX, will be preserved.
//
// Return value:
// -1 if failed to initialize,
// 0 if succeeded.
//
virtual int32_t ResetEncoder() = 0;
///////////////////////////////////////////////////////////////////////////
// int32_t RegisterSendCodec()
// Registers a codec, specified by |send_codec|, as sending codec.
@ -261,39 +250,11 @@ class AudioCodingModule {
//
virtual int32_t SendFrequency() const = 0;
///////////////////////////////////////////////////////////////////////////
// int32_t Bitrate()
// Get encoding bit-rate in bits per second.
//
// Return value:
// positive; encoding rate in bits/sec,
// -1 if an error is happened.
//
virtual int32_t SendBitrate() const = 0;
///////////////////////////////////////////////////////////////////////////
// Sets the bitrate to the specified value in bits/sec. If the value is not
// supported by the codec, it will choose another appropriate value.
virtual void SetBitRate(int bitrate_bps) = 0;
///////////////////////////////////////////////////////////////////////////
// int32_t SetReceivedEstimatedBandwidth()
// Set available bandwidth [bits/sec] of the up-link channel.
// This information is used for traffic shaping, and is currently only
// supported if iSAC is the send codec.
//
// Input:
// -bw : bandwidth in bits/sec estimated for
// up-link.
// Return value
// -1 if error occurred in setting the bandwidth,
// 0 bandwidth is set successfully.
//
// TODO(henrik.lundin) Unused. Remove?
virtual int32_t SetReceivedEstimatedBandwidth(
const int32_t bw) = 0;
///////////////////////////////////////////////////////////////////////////
// int32_t RegisterTransportCallback()
// Register a transport callback which will be called to deliver
// the encoded buffers whenever Process() is called and a
@ -465,39 +426,6 @@ class AudioCodingModule {
virtual int32_t VAD(bool* dtx_enabled, bool* vad_enabled,
ACMVADMode* vad_mode) const = 0;
///////////////////////////////////////////////////////////////////////////
// int32_t ReplaceInternalDTXWithWebRtc()
// Used to replace codec internal DTX scheme with WebRtc.
//
// Input:
// -use_webrtc_dtx : if false (default) the codec built-in DTX/VAD
// scheme is used, otherwise the internal DTX is
// replaced with WebRtc DTX/VAD.
//
// Return value:
// -1 if failed to replace codec internal DTX with WebRtc,
// 0 if succeeded.
//
virtual int32_t ReplaceInternalDTXWithWebRtc(
const bool use_webrtc_dtx = false) = 0;
///////////////////////////////////////////////////////////////////////////
// int32_t IsInternalDTXReplacedWithWebRtc()
// Get status if the codec internal DTX is replaced with WebRtc DTX.
// This should always be true if codec does not have an internal DTX.
//
// Output:
// -uses_webrtc_dtx : is set to true if the codec internal DTX is
// replaced with WebRtc DTX/VAD, otherwise it is set
// to false.
//
// Return value:
// -1 if failed to determine if codec internal DTX is replaced with WebRtc,
// 0 if succeeded.
//
virtual int32_t IsInternalDTXReplacedWithWebRtc(
bool* uses_webrtc_dtx) = 0;
///////////////////////////////////////////////////////////////////////////
// int32_t RegisterVADCallback()
// Call this method to register a callback function which is called
@ -533,17 +461,6 @@ class AudioCodingModule {
//
virtual int32_t InitializeReceiver() = 0;
///////////////////////////////////////////////////////////////////////////
// int32_t ResetDecoder()
// This API resets the states of decoders. ACM will not lose any
// decoder-related settings, such as registered codecs.
//
// Return value:
// -1 if failed to initialize,
// 0 if succeeded.
//
virtual int32_t ResetDecoder() = 0;
///////////////////////////////////////////////////////////////////////////
// int32_t ReceiveFrequency()
// Get sampling frequency of the last received payload.
@ -738,19 +655,6 @@ class AudioCodingModule {
// TODO(tlegrand): Change function to return the timestamp.
virtual int32_t PlayoutTimestamp(uint32_t* timestamp) = 0;
///////////////////////////////////////////////////////////////////////////
// int32_t DecoderEstimatedBandwidth()
// Get the estimate of the Bandwidth, in bits/second, based on the incoming
// stream. This API is useful in one-way communication scenarios, where
// the bandwidth information is sent in an out-of-band fashion.
// Currently only supported if iSAC is registered as a receiver.
//
// Return value:
// >0 bandwidth in bits/second.
// -1 if failed to get a bandwidth estimate.
//
virtual int32_t DecoderEstimatedBandwidth() const = 0;
///////////////////////////////////////////////////////////////////////////
// int32_t SetPlayoutMode()
// Call this API to set the playout mode. Playout mode could be optimized
@ -849,35 +753,6 @@ class AudioCodingModule {
//
virtual int SetISACMaxPayloadSize(int max_payload_len_bytes) = 0;
///////////////////////////////////////////////////////////////////////////
// int32_t ConfigISACBandwidthEstimator()
// Call this function to configure the bandwidth estimator of ISAC.
// During the adaptation of bit-rate, iSAC automatically adjusts the
// frame-size (either 30 or 60 ms) to save on RTP header. The initial
// frame-size can be specified by the first argument. The configuration also
// regards the initial estimate of bandwidths. The estimator starts from
// this point and converges to the actual bottleneck. This is given by the
// second parameter. Furthermore, it is also possible to control the
// adaptation of frame-size. This is specified by the last parameter.
//
// Input:
// -init_frame_size_ms : initial frame-size in milliseconds. For iSAC-wb
// 30 ms and 60 ms (default) are acceptable values,
// and for iSAC-swb 30 ms is the only acceptable
// value. Zero indicates default value.
// -init_rate_bps : initial estimate of the bandwidth. Values
// between 10000 and 58000 are acceptable.
// -enforce_srame_size : if true, the frame-size will not be adapted.
//
// Return value:
// -1 if failed to configure the bandwidth estimator,
// 0 if the configuration was successfully applied.
//
virtual int32_t ConfigISACBandwidthEstimator(
int init_frame_size_ms,
int init_rate_bps,
bool enforce_frame_size = false) = 0;
///////////////////////////////////////////////////////////////////////////
// int SetOpusApplication()
// Sets the intended application if current send codec is Opus. Opus uses this

1
webrtc/modules/audio_coding/main/test/APITest.cc
View File

@ -1129,7 +1129,6 @@ void APITest::ChangeCodec(char side) {
myChannel = _channel_B2A;
}
myACM->ResetEncoder();
Wait(100);
// Register the next codec

16
webrtc/modules/audio_coding/main/test/TestVADDTX.cc
View File

@ -137,7 +137,6 @@ void TestVadDtx::Run(std::string in_filename, int frequency, int channels,
TestWebRtcVadDtx::TestWebRtcVadDtx()
: vad_enabled_(false),
dtx_enabled_(false),
use_webrtc_dtx_(false),
output_file_num_(0) {
}
@ -191,7 +190,7 @@ void TestWebRtcVadDtx::RunTestCases() {
// Set the expectation and run the test.
void TestWebRtcVadDtx::Test(bool new_outfile) {
int expects[] = {-1, 1, use_webrtc_dtx_, 0, 0};
int expects[] = {-1, 1, dtx_enabled_, 0, 0};
if (new_outfile) {
output_file_num_++;
}
@ -219,17 +218,10 @@ void TestWebRtcVadDtx::SetVAD(bool enable_dtx, bool enable_vad,
EXPECT_EQ(dtx_enabled_ , enable_dtx); // DTX should be set as expected.
bool replaced = false;
acm_send_->IsInternalDTXReplacedWithWebRtc(&replaced);
use_webrtc_dtx_ = dtx_enabled_ && replaced;
if (use_webrtc_dtx_) {
if (dtx_enabled_) {
EXPECT_TRUE(vad_enabled_); // WebRTC DTX cannot run without WebRTC VAD.
}
if (!dtx_enabled_ || !use_webrtc_dtx_) {
// Using no DTX or codec Internal DTX should not affect setting of VAD.
} else {
// Using no DTX should not affect setting of VAD.
EXPECT_EQ(enable_vad, vad_enabled_);
}
}

1
webrtc/modules/audio_coding/main/test/TestVADDTX.h
View File

@ -88,7 +88,6 @@ class TestWebRtcVadDtx final : public TestVadDtx {
bool vad_enabled_;
bool dtx_enabled_;
bool use_webrtc_dtx_;
int output_file_num_;
};

15
webrtc/modules/audio_coding/main/test/TwoWayCommunication.cc
View File

@ -279,8 +279,8 @@ void TwoWayCommunication::Perform() {
// In the following loop we tests that the code can handle misuse of the APIs.
// In the middle of a session with data flowing between two sides, called A
// and B, APIs will be called, like ResetEncoder(), and the code should
// continue to run, and be able to recover.
// and B, APIs will be called, and the code should continue to run, and be
// able to recover.
while (!_inFileA.EndOfFile() && !_inFileB.EndOfFile()) {
msecPassed += 10;
EXPECT_GT(_inFileA.Read10MsData(audioFrame), 0);
@ -305,21 +305,14 @@ void TwoWayCommunication::Perform() {
msecPassed = 0;
secPassed++;
}
// Call RestEncoder for ACM on side A, and InitializeSender for ACM on
// side B.
if (((secPassed % 5) == 4) && (msecPassed == 0)) {
EXPECT_EQ(0, _acmA->ResetEncoder());
}
// Re-register send codec on side B.
if (((secPassed % 5) == 4) && (msecPassed >= 990)) {
EXPECT_EQ(0, _acmB->RegisterSendCodec(codecInst_B));
EXPECT_EQ(0, _acmB->SendCodec(&dummy));
}
// Reset decoder on side B, and initialize receiver on side A.
if (((secPassed % 7) == 6) && (msecPassed == 0)) {
EXPECT_EQ(0, _acmB->ResetDecoder());
// Initialize receiver on side A.
if (((secPassed % 7) == 6) && (msecPassed == 0))
EXPECT_EQ(0, _acmA->InitializeReceiver());
}
// Re-register codec on side A.
if (((secPassed % 7) == 6) && (msecPassed >= 990)) {
EXPECT_EQ(0, _acmA->RegisterReceiveCodec(codecInst_B));

9
webrtc/modules/audio_coding/main/test/iSACTest.cc
View File

@ -75,13 +75,6 @@ int16_t SetISAConfig(ACMTestISACConfig& isacConfig, AudioCodingModule* acm,
// Set max payload size.
EXPECT_EQ(0, acm->SetISACMaxPayloadSize(isacConfig.maxPayloadSizeByte));
}
if ((isacConfig.initFrameSizeInMsec != 0)
|| (isacConfig.initRateBitPerSec != 0)) {
EXPECT_EQ(0, acm->ConfigISACBandwidthEstimator(
static_cast<uint8_t>(isacConfig.initFrameSizeInMsec),
static_cast<uint16_t>(isacConfig.initRateBitPerSec),
isacConfig.enforceFrameSize));
}
return 0;
}
@ -215,8 +208,6 @@ void ISACTest::Perform() {
testNr++;
EncodeDecode(testNr, wbISACConfig, swbISACConfig);
_acmA->ResetEncoder();
_acmB->ResetEncoder();
SetISACConfigDefault(wbISACConfig);
SetISACConfigDefault(swbISACConfig);