diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.cc b/webrtc/modules/audio_coding/codecs/audio_encoder.cc index 8974cf18bc..c0c20bec4d 100644 --- a/webrtc/modules/audio_coding/codecs/audio_encoder.cc +++ b/webrtc/modules/audio_coding/codecs/audio_encoder.cc @@ -52,8 +52,4 @@ void AudioEncoder::SetProjectedPacketLossRate(double fraction) {} void AudioEncoder::SetTargetBitrate(int target_bps) {} -void AudioEncoder::SetMaxBitrate(int max_bps) {} - -void AudioEncoder::SetMaxPayloadSize(int max_payload_size_bytes) {} - } // namespace webrtc diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.h b/webrtc/modules/audio_coding/codecs/audio_encoder.h index c053b7fdef..cda9d86f2e 100644 --- a/webrtc/modules/audio_coding/codecs/audio_encoder.h +++ b/webrtc/modules/audio_coding/codecs/audio_encoder.h @@ -138,16 +138,6 @@ class AudioEncoder { // encoder is free to adjust or disregard the given bitrate (the default // implementation does the latter). virtual void SetTargetBitrate(int target_bps); - - // Sets the maximum bitrate which must not be exceeded for any packet. The - // encoder is free to adjust or disregard this value (the default - // implementation does the latter). - virtual void SetMaxBitrate(int max_bps); - - // Sets an upper limit on the size of packet payloads produced by the - // encoder. The encoder is free to adjust or disregard this value (the - // default implementation does the latter). - virtual void SetMaxPayloadSize(int max_payload_size_bytes); }; } // namespace webrtc #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_ diff --git a/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc b/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc index ab3bd770e6..2fe58c9ba5 100644 --- a/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc +++ b/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc @@ -199,14 +199,6 @@ void AudioEncoderCng::SetTargetBitrate(int bits_per_second) { speech_encoder_->SetTargetBitrate(bits_per_second); } -void AudioEncoderCng::SetMaxBitrate(int max_bps) { - speech_encoder_->SetMaxBitrate(max_bps); -} - -void AudioEncoderCng::SetMaxPayloadSize(int max_payload_size_bytes) { - speech_encoder_->SetMaxPayloadSize(max_payload_size_bytes); -} - AudioEncoder::EncodedInfo AudioEncoderCng::EncodePassive( size_t frames_to_encode, size_t max_encoded_bytes, diff --git a/webrtc/modules/audio_coding/codecs/cng/include/audio_encoder_cng.h b/webrtc/modules/audio_coding/codecs/cng/include/audio_encoder_cng.h index fd2aa129c8..b025bc2e44 100644 --- a/webrtc/modules/audio_coding/codecs/cng/include/audio_encoder_cng.h +++ b/webrtc/modules/audio_coding/codecs/cng/include/audio_encoder_cng.h @@ -67,8 +67,6 @@ class AudioEncoderCng final : public AudioEncoder { void SetMaxPlaybackRate(int frequency_hz) override; void SetProjectedPacketLossRate(double fraction) override; void SetTargetBitrate(int target_bps) override; - void SetMaxBitrate(int max_bps) override; - void SetMaxPayloadSize(int max_payload_size_bytes) override; private: EncodedInfo EncodePassive(size_t frames_to_encode, diff --git a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h index 5484395ad8..686b45a742 100644 --- a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h +++ b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h @@ -67,8 +67,6 @@ class AudioEncoderIsacT final : public AudioEncoder { size_t max_encoded_bytes, uint8_t* encoded) override; void Reset() override; - void SetMaxPayloadSize(int max_payload_size_bytes) override; - void SetMaxBitrate(int max_rate_bps) override; private: // This value is taken from STREAM_SIZE_MAX_60 for iSAC float (60 ms) and diff --git a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h index ad09c3f90d..3cc635c612 100644 --- a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h +++ b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h @@ -157,20 +157,6 @@ void AudioEncoderIsacT::Reset() { RecreateEncoderInstance(config_); } -template -void AudioEncoderIsacT::SetMaxPayloadSize(int max_payload_size_bytes) { - auto conf = config_; - conf.max_payload_size_bytes = max_payload_size_bytes; - RecreateEncoderInstance(conf); -} - -template -void AudioEncoderIsacT::SetMaxBitrate(int max_rate_bps) { - auto conf = config_; - conf.max_bit_rate = max_rate_bps; - RecreateEncoderInstance(conf); -} - template void AudioEncoderIsacT::RecreateEncoderInstance(const Config& config) { CHECK(config.IsOk()); diff --git a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc index 46febf7662..c8ae53fe29 100644 --- a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc +++ b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc @@ -121,12 +121,4 @@ void AudioEncoderCopyRed::SetTargetBitrate(int bits_per_second) { speech_encoder_->SetTargetBitrate(bits_per_second); } -void AudioEncoderCopyRed::SetMaxBitrate(int max_bps) { - speech_encoder_->SetMaxBitrate(max_bps); -} - -void AudioEncoderCopyRed::SetMaxPayloadSize(int max_payload_size_bytes) { - speech_encoder_->SetMaxPayloadSize(max_payload_size_bytes); -} - } // namespace webrtc diff --git a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h index d0fcd41ed8..1d35f95877 100644 --- a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h +++ b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h @@ -54,8 +54,6 @@ class AudioEncoderCopyRed final : public AudioEncoder { void SetMaxPlaybackRate(int frequency_hz) override; void SetProjectedPacketLossRate(double fraction) override; void SetTargetBitrate(int target_bps) override; - void SetMaxBitrate(int max_bps) override; - void SetMaxPayloadSize(int max_payload_size_bytes) override; private: AudioEncoder* speech_encoder_; diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc index 571a509b0e..cb07cd6eb7 100644 --- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc +++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc @@ -739,30 +739,6 @@ int AudioCodingModuleImpl::IncomingPayload(const uint8_t* incoming_payload, 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_.get()); - - if (!HaveValidEncoder("SetISACMaxRate")) { - return -1; - } - - codec_manager_.CurrentEncoder()->SetMaxBitrate(max_bit_per_sec); - return 0; -} - -// TODO(henrik.lundin): Remove? Only used in tests. Deprecated in VoiceEngine. -int AudioCodingModuleImpl::SetISACMaxPayloadSize(int max_size_bytes) { - CriticalSectionScoped lock(acm_crit_sect_.get()); - - if (!HaveValidEncoder("SetISACMaxPayloadSize")) { - return -1; - } - - codec_manager_.CurrentEncoder()->SetMaxPayloadSize(max_size_bytes); - return 0; -} - int AudioCodingModuleImpl::SetOpusApplication(OpusApplicationMode application) { CriticalSectionScoped lock(acm_crit_sect_.get()); if (!HaveValidEncoder("SetOpusApplication")) { diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h index 568bf92d3f..837cd11004 100644 --- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h +++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h @@ -185,10 +185,6 @@ class AudioCodingModuleImpl final : public AudioCodingModule { int GetNetworkStatistics(NetworkStatistics* statistics) override; - int SetISACMaxRate(int max_bit_per_sec) override; - - int SetISACMaxPayloadSize(int max_size_bytes) override; - int SetOpusApplication(OpusApplicationMode application) override; // If current send codec is Opus, informs it about the maximum playback rate diff --git a/webrtc/modules/audio_coding/main/interface/audio_coding_module.h b/webrtc/modules/audio_coding/main/interface/audio_coding_module.h index 085dd619ce..0d3d5da818 100644 --- a/webrtc/modules/audio_coding/main/interface/audio_coding_module.h +++ b/webrtc/modules/audio_coding/main/interface/audio_coding_module.h @@ -722,40 +722,6 @@ class AudioCodingModule { // Codec specific // - /////////////////////////////////////////////////////////////////////////// - // int32_t SetISACMaxRate() - // Set the maximum instantaneous rate of iSAC. For a payload of B bits - // with a frame-size of T sec the instantaneous rate is B/T bits per - // second. Therefore, (B/T < |max_rate_bps|) and - // (B < |max_payload_len_bytes| * 8) are always satisfied for iSAC payloads, - // c.f SetISACMaxPayloadSize(). - // - // Input: - // -max_rate_bps : maximum instantaneous bit-rate given in bits/sec. - // - // Return value: - // -1 if failed to set the maximum rate. - // 0 if the maximum rate is set successfully. - // - virtual int SetISACMaxRate(int max_rate_bps) = 0; - - /////////////////////////////////////////////////////////////////////////// - // int32_t SetISACMaxPayloadSize() - // Set the maximum payload size of iSAC packets. No iSAC payload, - // regardless of its frame-size, may exceed the given limit. For - // an iSAC payload of size B bits and frame-size T seconds we have; - // (B < |max_payload_len_bytes| * 8) and (B/T < |max_rate_bps|), c.f. - // SetISACMaxRate(). - // - // Input: - // -max_payload_len_bytes : maximum payload size in bytes. - // - // Return value: - // -1 if failed to set the maximum payload-size. - // 0 if the given length is set successfully. - // - virtual int SetISACMaxPayloadSize(int max_payload_len_bytes) = 0; - /////////////////////////////////////////////////////////////////////////// // int SetOpusApplication() // Sets the intended application if current send codec is Opus. Opus uses this diff --git a/webrtc/modules/audio_coding/main/test/iSACTest.cc b/webrtc/modules/audio_coding/main/test/iSACTest.cc index cc41e3bc1b..bd796d1ce7 100644 --- a/webrtc/modules/audio_coding/main/test/iSACTest.cc +++ b/webrtc/modules/audio_coding/main/test/iSACTest.cc @@ -35,8 +35,6 @@ namespace webrtc { void SetISACConfigDefault(ACMTestISACConfig& isacConfig) { isacConfig.currentRateBitPerSec = 0; isacConfig.currentFrameSizeMsec = 0; - isacConfig.maxRateBitPerSec = 0; - isacConfig.maxPayloadSizeByte = 0; isacConfig.encodingMode = -1; isacConfig.initRateBitPerSec = 0; isacConfig.initFrameSizeInMsec = 0; @@ -67,15 +65,6 @@ int16_t SetISAConfig(ACMTestISACConfig& isacConfig, AudioCodingModule* acm, } } - if (isacConfig.maxRateBitPerSec > 0) { - // Set max rate. - EXPECT_EQ(0, acm->SetISACMaxRate(isacConfig.maxRateBitPerSec)); - } - if (isacConfig.maxPayloadSizeByte > 0) { - // Set max payload size. - EXPECT_EQ(0, acm->SetISACMaxPayloadSize(isacConfig.maxPayloadSizeByte)); - } - return 0; } @@ -193,39 +182,6 @@ void ISACTest::Perform() { testNr++; EncodeDecode(testNr, wbISACConfig, swbISACConfig); - int user_input; - if ((_testMode == 0) || (_testMode == 1)) { - swbISACConfig.maxPayloadSizeByte = static_cast(200); - wbISACConfig.maxPayloadSizeByte = static_cast(200); - } else { - printf("Enter the max payload-size for side A: "); - CHECK_ERROR(scanf("%d", &user_input)); - swbISACConfig.maxPayloadSizeByte = (uint16_t) user_input; - printf("Enter the max payload-size for side B: "); - CHECK_ERROR(scanf("%d", &user_input)); - wbISACConfig.maxPayloadSizeByte = (uint16_t) user_input; - } - testNr++; - EncodeDecode(testNr, wbISACConfig, swbISACConfig); - - SetISACConfigDefault(wbISACConfig); - SetISACConfigDefault(swbISACConfig); - - if ((_testMode == 0) || (_testMode == 1)) { - swbISACConfig.maxRateBitPerSec = static_cast(48000); - wbISACConfig.maxRateBitPerSec = static_cast(48000); - } else { - printf("Enter the max rate for side A: "); - CHECK_ERROR(scanf("%d", &user_input)); - swbISACConfig.maxRateBitPerSec = (uint32_t) user_input; - printf("Enter the max rate for side B: "); - CHECK_ERROR(scanf("%d", &user_input)); - wbISACConfig.maxRateBitPerSec = (uint32_t) user_input; - } - - testNr++; - EncodeDecode(testNr, wbISACConfig, swbISACConfig); - testNr++; if (_testMode == 0) { SwitchingSamplingRate(testNr, 4); diff --git a/webrtc/modules/audio_coding/main/test/iSACTest.h b/webrtc/modules/audio_coding/main/test/iSACTest.h index f4223f7512..8f892d907b 100644 --- a/webrtc/modules/audio_coding/main/test/iSACTest.h +++ b/webrtc/modules/audio_coding/main/test/iSACTest.h @@ -29,8 +29,6 @@ namespace webrtc { struct ACMTestISACConfig { int32_t currentRateBitPerSec; int16_t currentFrameSizeMsec; - uint32_t maxRateBitPerSec; - int16_t maxPayloadSizeByte; int16_t encodingMode; uint32_t initRateBitPerSec; int16_t initFrameSizeInMsec;