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Remove no-op and unused methods from AudioCodingModule

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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