From 1b48a48075e46ae79d6860f10cd74c4c1bdc50b0 Mon Sep 17 00:00:00 2001
From: Eloi BAIL <eloi.bail@savoirfairelinux.com>
Date: Wed, 21 Jan 2015 15:01:52 -0500
Subject: [PATCH] daemon: handle libav opus native decoder
Since libavcodec56 libav native opus decoder is used instead of libopus
decoder.
This decoder only ouput float planar audio data (AV_SAMPLE_FMT_FLTP)
This commit detect the output format and convert it to signed 16bits
in case of float planar.
Refs #64245
Change-Id: Idb8240a1158a0add446d7df3b4a1ed51089e2781
---
daemon/src/media/audio/audiobuffer.cpp | 20 ++++++++++++++++++++
daemon/src/media/audio/audiobuffer.h | 6 ++++++
daemon/src/media/media_decoder.cpp | 9 +++++++--
3 files changed, 33 insertions(+), 2 deletions(-)
diff --git a/daemon/src/media/audio/audiobuffer.cpp b/daemon/src/media/audio/audiobuffer.cpp
index 40d11e1087..5310536e9e 100644
--- a/daemon/src/media/audio/audiobuffer.cpp
+++ b/daemon/src/media/audio/audiobuffer.cpp
@@ -214,6 +214,26 @@ void AudioBuffer::deinterleave(const std::vector<ring::AudioSample>& in, AudioFo
deinterleave(in.data(), in.size()/format.nb_channels, format.nb_channels);
}
+void AudioBuffer::convertFloatPlanarToSigned16(uint8_t** extended_data, size_t frame_num, unsigned nb_channels)
+{
+ if (extended_data == nullptr)
+ return;
+
+ // Resize buffer
+ setChannelNum(nb_channels);
+ resize(frame_num);
+
+ for (unsigned j = 0, c = channels(); j < c; j++){
+ float* inputChannel = (float*)extended_data[j];
+ for (unsigned i=0, f=frames(); i < f; i++){
+ float inputChannelVal = *inputChannel++;
+ // avoid saturation: limit val between -1 and 1
+ inputChannelVal = std::max(-1.0f, std::min(inputChannelVal, 1.0f));
+ samples_[j][i] = (int16_t) (inputChannelVal * 32768.0f);
+ }
+ }
+}
+
size_t AudioBuffer::mix(const AudioBuffer& other, bool up /* = true */)
{
const bool upmix = up && (other.samples_.size() < samples_.size());
diff --git a/daemon/src/media/audio/audiobuffer.h b/daemon/src/media/audio/audiobuffer.h
index 3d22eb9e47..fffd5c2c7e 100644
--- a/daemon/src/media/audio/audiobuffer.h
+++ b/daemon/src/media/audio/audiobuffer.h
@@ -288,6 +288,12 @@ class AudioBuffer {
*/
void deinterleave(const std::vector<ring::AudioSample>& in, AudioFormat format);
+ /**
+ * convert float planar data to signed 16
+ */
+ void convertFloatPlanarToSigned16(uint8_t** extended_data, size_t frame_num, unsigned nb_channels = 1);
+
+
/**
* In-place gain transformation.
*
diff --git a/daemon/src/media/media_decoder.cpp b/daemon/src/media/media_decoder.cpp
index 4a14fef788..2b8a517da7 100644
--- a/daemon/src/media/media_decoder.cpp
+++ b/daemon/src/media/media_decoder.cpp
@@ -432,8 +432,13 @@ void MediaDecoder::writeToRingBuffer(AVFrame* decoded_frame,
ring::AudioBuffer out(decoded_frame->nb_samples, decoderFormat);
- out.deinterleave(reinterpret_cast<const ring::AudioSample*>(decoded_frame->data[0]),
- decoded_frame->nb_samples, decoderCtx_->channels);
+ if ( decoderCtx_->sample_fmt == AV_SAMPLE_FMT_FLTP ) {
+ out.convertFloatPlanarToSigned16(decoded_frame->extended_data,
+ decoded_frame->nb_samples, decoderCtx_->channels);
+ } else if ( decoderCtx_->sample_fmt == AV_SAMPLE_FMT_S16 ) {
+ out.deinterleave(reinterpret_cast<const ring::AudioSample*>(decoded_frame->data[0]),
+ decoded_frame->nb_samples, decoderCtx_->channels);
+ }
if ((unsigned)decoded_frame->sample_rate != outFormat.sample_rate) {
if (!resampler_) {
RING_DBG("Creating audio resampler");
--
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