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Revert r8476 "Set decoder output frequency in AudioDecoder::Decode call"

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This change uncovered issue 4143, evading the Memcheck suppression
since the signature is changed in the Decode function.

A fix for this is in the making; see
https://review.webrtc.org/36309004. This CL will be re-landed once the
fix is in place.

BUG=4143
TBR=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42089004

Cr-Commit-Position: refs/heads/master@{#8488}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8488 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
henrik.lundin@webrtc.org
2015-02-24 21:17:50 +00:00
parent 348072845a
commit 903182bd8e
20 changed files with 184 additions and 201 deletions
Download Patch File Download Diff File Expand all files Collapse all files

64
webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
View File

@ -38,12 +38,8 @@
namespace webrtc {
// PCMu
int AudioDecoderPcmU::Decode(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
DCHECK_EQ(sample_rate_hz, 8000);
int AudioDecoderPcmU::Decode(const uint8_t* encoded, size_t encoded_len,
int16_t* decoded, SpeechType* speech_type) {
int16_t temp_type = 1; // Default is speech.
int16_t ret = WebRtcG711_DecodeU(encoded, static_cast<int16_t>(encoded_len),
decoded, &temp_type);
@ -58,12 +54,8 @@ int AudioDecoderPcmU::PacketDuration(const uint8_t* encoded,
}
// PCMa
int AudioDecoderPcmA::Decode(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
DCHECK_EQ(sample_rate_hz, 8000);
int AudioDecoderPcmA::Decode(const uint8_t* encoded, size_t encoded_len,
int16_t* decoded, SpeechType* speech_type) {
int16_t temp_type = 1; // Default is speech.
int16_t ret = WebRtcG711_DecodeA(encoded, static_cast<int16_t>(encoded_len),
decoded, &temp_type);
@ -81,14 +73,8 @@ int AudioDecoderPcmA::PacketDuration(const uint8_t* encoded,
#ifdef WEBRTC_CODEC_PCM16
AudioDecoderPcm16B::AudioDecoderPcm16B() {}
int AudioDecoderPcm16B::Decode(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
DCHECK(sample_rate_hz == 8000 || sample_rate_hz == 16000 ||
sample_rate_hz == 32000 || sample_rate_hz == 48000)
<< "Unsupported sample rate " << sample_rate_hz;
int AudioDecoderPcm16B::Decode(const uint8_t* encoded, size_t encoded_len,
int16_t* decoded, SpeechType* speech_type) {
int16_t ret =
WebRtcPcm16b_Decode(encoded, static_cast<int16_t>(encoded_len), decoded);
*speech_type = ConvertSpeechType(1);
@ -117,12 +103,8 @@ AudioDecoderIlbc::~AudioDecoderIlbc() {
WebRtcIlbcfix_DecoderFree(dec_state_);
}
int AudioDecoderIlbc::Decode(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
DCHECK_EQ(sample_rate_hz, 8000);
int AudioDecoderIlbc::Decode(const uint8_t* encoded, size_t encoded_len,
int16_t* decoded, SpeechType* speech_type) {
int16_t temp_type = 1; // Default is speech.
int16_t ret = WebRtcIlbcfix_Decode(dec_state_, encoded,
static_cast<int16_t>(encoded_len), decoded,
@ -150,12 +132,8 @@ AudioDecoderG722::~AudioDecoderG722() {
WebRtcG722_FreeDecoder(dec_state_);
}
int AudioDecoderG722::Decode(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
DCHECK_EQ(sample_rate_hz, 16000);
int AudioDecoderG722::Decode(const uint8_t* encoded, size_t encoded_len,
int16_t* decoded, SpeechType* speech_type) {
int16_t temp_type = 1; // Default is speech.
int16_t ret =
WebRtcG722_Decode(dec_state_, encoded, static_cast<int16_t>(encoded_len),
@ -185,12 +163,8 @@ AudioDecoderG722Stereo::~AudioDecoderG722Stereo() {
WebRtcG722_FreeDecoder(dec_state_right_);
}
int AudioDecoderG722Stereo::Decode(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
DCHECK_EQ(sample_rate_hz, 16000);
int AudioDecoderG722Stereo::Decode(const uint8_t* encoded, size_t encoded_len,
int16_t* decoded, SpeechType* speech_type) {
int16_t temp_type = 1; // Default is speech.
// De-interleave the bit-stream into two separate payloads.
uint8_t* encoded_deinterleaved = new uint8_t[encoded_len];
@ -270,12 +244,8 @@ AudioDecoderOpus::~AudioDecoderOpus() {
WebRtcOpus_DecoderFree(dec_state_);
}
int AudioDecoderOpus::Decode(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
DCHECK_EQ(sample_rate_hz, 48000);
int AudioDecoderOpus::Decode(const uint8_t* encoded, size_t encoded_len,
int16_t* decoded, SpeechType* speech_type) {
int16_t temp_type = 1; // Default is speech.
int16_t ret = WebRtcOpus_Decode(dec_state_, encoded,
static_cast<int16_t>(encoded_len), decoded,
@ -287,13 +257,11 @@ int AudioDecoderOpus::Decode(const uint8_t* encoded,
}
int AudioDecoderOpus::DecodeRedundant(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
size_t encoded_len, int16_t* decoded,
SpeechType* speech_type) {
if (!PacketHasFec(encoded, encoded_len)) {
// This packet is a RED packet.
return Decode(encoded, encoded_len, sample_rate_hz, decoded, speech_type);
return Decode(encoded, encoded_len, decoded, speech_type);
}
int16_t temp_type = 1; // Default is speech.